From 7285b1ee0364a9d3cda1824cc7f4799faa42b1b0 Mon Sep 17 00:00:00 2001 From: Pieter De Gendt Date: Fri, 29 Nov 2019 14:57:35 +0100 Subject: [PATCH 001/186] Jack output add soxr resampling --- audio_jack.c | 162 +++++++++++++++++++++++++++++++++++++++++++-------- common.h | 3 + 2 files changed, 140 insertions(+), 25 deletions(-) diff --git a/audio_jack.c b/audio_jack.c index 0aff5025..a8600de9 100644 --- a/audio_jack.c +++ b/audio_jack.c @@ -29,10 +29,20 @@ #include #include -// Two-channel, 16bit audio: -static const int bytes_per_frame = 4; +#ifdef CONFIG_SOXR +#include +#endif + +#define NPORTS 2 + +typedef jack_default_audio_sample_t sample_t; + +#define jack_sample_size sizeof(sample_t) + +// Two-channel, 32bit audio: +static const int bytes_per_frame = NPORTS * jack_sample_size; // Four seconds buffer -- should be plenty -#define buffer_size (44100 * 4 * bytes_per_frame) +#define buffer_size (48000u * 4u * bytes_per_frame) static pthread_mutex_t buffer_mutex = PTHREAD_MUTEX_INITIALIZER; static pthread_mutex_t client_mutex = PTHREAD_MUTEX_INITIALIZER; @@ -62,7 +72,6 @@ audio_output audio_jack = {.name = "jack", // This also affects deinterlacing. // So make it exactly the number of incoming audio channels! -#define NPORTS 2 static jack_port_t *port[NPORTS]; static const char *port_name[NPORTS] = {"out_L", "out_R"}; @@ -76,7 +85,36 @@ static int flush_please = 0; static jack_latency_range_t latest_latency_range[NPORTS]; static int64_t time_of_latest_transfer; -static inline jack_default_audio_sample_t sample_conv(short sample) { +#ifdef CONFIG_SOXR +typedef struct soxr_recipe { + int recipe; + const char *name; +} soxr_recipe_t; + +static soxr_recipe_t soxr_quality_table[] = { + { SOXR_VHQ, "very high" }, + { SOXR_HQ, "high" }, + { SOXR_MQ, "medium" }, + { SOXR_LQ, "low" }, + { SOXR_QQ, "quick" }, + { -1, NULL } +}; + +static int parse_soxr_recipe_name(const char *name) { + for (soxr_recipe_t *s = soxr_quality_table; s->name != NULL; ++s) { + if (!strcmp(s->name, name)) { + return s->recipe; + } + } + return -1; +} + +static soxr_t soxr = NULL; +static soxr_quality_spec_t quality_spec; +static soxr_io_spec_t io_spec; +#endif + +static inline sample_t sample_conv(short sample) { // It sounds correct, but I don't understand it. // Zero int needs to be zero float. Check. // Plus 32767 int is 1.0. Check. @@ -86,17 +124,17 @@ static inline jack_default_audio_sample_t sample_conv(short sample) { return ((sample < 0) ? (-1.0 * sample / SHRT_MIN) : (1.0 * sample / SHRT_MAX)); } -static void deinterleave_and_convert(const char *interleaved_input_buffer, - jack_default_audio_sample_t *jack_output_buffer[], - jack_nframes_t offset, jack_nframes_t nframes) { +static void deinterleave(const char *interleaved_input_buffer, + sample_t *jack_output_buffer[], + jack_nframes_t offset, jack_nframes_t nframes) { jack_nframes_t f; // We're dealing with 16bit audio here: - short *ifp = (short *)interleaved_input_buffer; + sample_t *ifp = (sample_t *)interleaved_input_buffer; // Zero-copy, we're working directly on the target and destination buffers, // so deal with an offset for the second part of the input ringbuffer for (f = offset; f < (nframes + offset); f++) { for (int i = 0; i < NPORTS; i++) { - jack_output_buffer[i][f] = sample_conv(*ifp++); + jack_output_buffer[i][f] = *ifp++; } } } @@ -107,7 +145,7 @@ static void deinterleave_and_convert(const char *interleaved_input_buffer, // output, no file access, no mutexes... // The JACK ringbuffer we use to get the data in here is explicitly lock-free. static int process(jack_nframes_t nframes, __attribute__((unused)) void *arg) { - jack_default_audio_sample_t *buffer[NPORTS]; + sample_t *buffer[NPORTS]; // Expect an array of two elements because of possible ringbuffer wrap-around: jack_ringbuffer_data_t v[2] = {0}; jack_nframes_t i, thisbuf; @@ -115,7 +153,7 @@ static int process(jack_nframes_t nframes, __attribute__((unused)) void *arg) { int frames_required = 0; for (i = 0; i < NPORTS; i++) { - buffer[i] = (jack_default_audio_sample_t *)jack_port_get_buffer(port[i], nframes); + buffer[i] = (sample_t *)jack_port_get_buffer(port[i], nframes); } if (flush_please) { // We just move the read pointer ahead without doing anything with the data. @@ -131,7 +169,7 @@ static int process(jack_nframes_t nframes, __attribute__((unused)) void *arg) { } else { frames_required = thisbuf; } - deinterleave_and_convert(v[i].buf, buffer, frames_written, frames_required); + deinterleave(v[i].buf, buffer, frames_written, frames_required); frames_written += frames_required; nframes -= frames_required; } @@ -183,6 +221,9 @@ int jack_init(__attribute__((unused)) int argc, __attribute__((unused)) char **a // Do the "general" audio options. Note, these options are in the "general" stanza! parse_general_audio_options(); +#ifdef CONFIG_SOXR + config.jack_soxr_resample_recipe = -1; // don't resample by default +#endif // Now the options specific to the backend, from the "jack" stanza: if (config.cfg != NULL) { @@ -193,6 +234,12 @@ int jack_init(__attribute__((unused)) int argc, __attribute__((unused)) char **a if (config_lookup_string(config.cfg, "jack.autoconnect_pattern", &str)) { config.jack_autoconnect_pattern = (char *)str; } +#ifdef CONFIG_SOXR + if (config_lookup_string(config.cfg, "jack.soxr_resample_recipe", &str)) { + debug(1, "SOXR quality %s", str); + config.jack_soxr_resample_recipe = parse_soxr_recipe_name(str); + } +#endif } if (config.jack_client_name == NULL) config.jack_client_name = strdup("shairport-sync"); @@ -213,6 +260,12 @@ int jack_init(__attribute__((unused)) int argc, __attribute__((unused)) char **a die("Could not start JACK server. JackStatus is %x", status); } sample_rate = jack_get_sample_rate(client); +#ifdef CONFIG_SOXR + if (config.jack_soxr_resample_recipe >= SOXR_QQ) { + quality_spec = soxr_quality_spec(config.jack_soxr_resample_recipe, 0); + io_spec = soxr_io_spec(SOXR_INT16_I, SOXR_FLOAT32_I); + } else +#endif if (sample_rate != 44100) { die("The JACK server is running at the wrong sample rate (%d) for Shairport Sync." " Must be 44100 Hz.", @@ -287,14 +340,38 @@ void jack_deinit() { warn("Error closing jack client"); pthread_mutex_unlock(&client_mutex); jack_ringbuffer_free(jackbuf); +#ifdef CONFIG_SOXR + if (soxr) { + soxr_delete(soxr); + soxr = NULL; + } +#endif } -void jack_start(__attribute__((unused)) int i_sample_rate, +void jack_start(int i_sample_rate, __attribute__((unused)) int i_sample_format) { // Nothing to do, JACK client has already been set up at jack_init(). // Also, we have no say over the sample rate or sample format of JACK, - // We convert the 16bit samples to float, and die if the sample rate is != 44k1. - // FIXME: later, resampling would be nice. Fold into soxr if possible. + // We convert the 16bit samples to float, and die if the sample rate is != 44k1 without soxr. +#ifdef CONFIG_SOXR + if (config.jack_soxr_resample_recipe >= SOXR_QQ) { + // we might improve a bit with soxr_clear if the sample_rate doesn't change + if (soxr) { + soxr_delete(soxr); + } + soxr_error_t e = NULL; + soxr = soxr_create(i_sample_rate, + sample_rate, + NPORTS, + &e, + &io_spec, + &quality_spec, + NULL); + if (!soxr) { + die("Unable to create soxr resampler for JACK: %s", e); + } + } +#endif } void jack_flush() { @@ -319,7 +396,7 @@ int jack_delay(long *the_delay) { debug(2, "audio_occupancy_now is %d.", audio_occupancy_now); pthread_mutex_unlock(&buffer_mutex); - int64_t frames_processed_since_latest_latency_check = (delta * 44100) >> 32; + int64_t frames_processed_since_latest_latency_check = (delta * sample_rate) >> 32; // debug(1,"delta: %" PRId64 " frames.",frames_processed_since_latest_latency_check); // jack_latency is set by the graph() callback, it's the average of the maximum // latencies of all our output ports. Adjust this constant baseline delay according @@ -330,18 +407,53 @@ int jack_delay(long *the_delay) { } int play(void *buf, int samples) { - // debug(1,"jack_play of %d samples.",samples); - // copy the samples into the queue - size_t bytes_to_transfer, bytes_transferred; - bytes_to_transfer = samples * bytes_per_frame; + jack_ringbuffer_data_t v[2] = {0}; + size_t i, j, c; + jack_nframes_t thisbuf; // It's ok to lock here since we're not in the realtime callback: pthread_mutex_lock(&buffer_mutex); - bytes_transferred = jack_ringbuffer_write(jackbuf, buf, bytes_to_transfer); + jack_ringbuffer_get_write_vector(jackbuf, v); + short *in = (short *)buf; + sample_t *out; + for (i = 0; i < 2; ++i) { + thisbuf = v[i].len / (jack_sample_size * NPORTS); // #samples per channel + out = (sample_t *)v[i].buf; +#ifdef CONFIG_SOXR + if (soxr) { + size_t i_done, o_done; + soxr_error_t e; + while (samples > 0 && thisbuf > 0) { + e = soxr_process(soxr, + (soxr_in_t)in, + samples, + &i_done, + (soxr_out_t)out, + thisbuf, + &o_done); + if (e) + die("Error during soxr process: %s", e); + + samples -= i_done; + thisbuf -= o_done; + jack_ringbuffer_write_advance(jackbuf, o_done * jack_sample_size * NPORTS); + } + } else { +#endif + j = 0; + for (j = 0; j < thisbuf && samples > 0; ++j) { + for (c = 0; c < NPORTS; ++c) + out[j * NPORTS + c] = sample_conv(*in++); + --samples; + } + jack_ringbuffer_write_advance(jackbuf, j * jack_sample_size * NPORTS); +#ifdef CONFIG_SOXR + } +#endif + } time_of_latest_transfer = get_absolute_time_in_fp(); pthread_mutex_unlock(&buffer_mutex); - if (bytes_transferred < bytes_to_transfer) { - warn("JACK ringbuffer overrun. Only wrote %d of %d bytes.", bytes_transferred, - bytes_to_transfer); + if (samples) { + warn("JACK ringbuffer overrun. Dropped %d samples.", samples); } return 0; } diff --git a/common.h b/common.h index 34cd10ba..45fc9683 100644 --- a/common.h +++ b/common.h @@ -270,6 +270,9 @@ typedef struct { #ifdef CONFIG_JACK char *jack_client_name; char *jack_autoconnect_pattern; +#ifdef CONFIG_SOXR + int jack_soxr_resample_recipe; +#endif #endif } shairport_cfg; From 311909a2dd0665a5058937f3d9d87f691f97ae60 Mon Sep 17 00:00:00 2001 From: Pieter De Gendt Date: Tue, 3 Dec 2019 12:14:52 +0100 Subject: [PATCH 002/186] Fix input buffer while resampling --- audio_jack.c | 1 + 1 file changed, 1 insertion(+) diff --git a/audio_jack.c b/audio_jack.c index a8600de9..499687a3 100644 --- a/audio_jack.c +++ b/audio_jack.c @@ -433,6 +433,7 @@ int play(void *buf, int samples) { if (e) die("Error during soxr process: %s", e); + in += i_done * NPORTS; // advance our input buffer samples -= i_done; thisbuf -= o_done; jack_ringbuffer_write_advance(jackbuf, o_done * jack_sample_size * NPORTS); From ed9f8d174c450ee477023f9def2539a4f1f13a86 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 18 Dec 2019 10:40:41 +0000 Subject: [PATCH 003/186] Update README.md --- README.md | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/README.md b/README.md index b0aea260..48bd955c 100644 --- a/README.md +++ b/README.md @@ -597,11 +597,11 @@ This will be followed by the statistics themselves at regular intervals, for exa 1.99, -22.7, 22.7, 54162, 0, 0, 0, 0, 8567, 216, 225, 44100.00, 44103.12, 44099.87, 24.57, 92, -27.45 ``` -"Sync error in milliseconds" is the average deviation from exact synchronisation. The first line of the example above indicates that the output is on average 0.7 milliseconds behind exact synchronisation. Sync is allowed to drift by the `general` `drift_tolerance_in_seconds` setting — (± 0.002 seconds) by default — before a correction will be made. +"Sync error in milliseconds" is the average deviation from exact synchronisation. The first line of the example above indicates that the output is on average 0.15 milliseconds behind exact synchronisation. Sync is allowed to drift by the `general` `drift_tolerance_in_seconds` setting — (± 0.002 seconds) by default — before a correction will be made. "Net correction in ppm" is actually the net sum of corrections — the number of frame insertions less the number of frame deletions — given as a moving average in parts per million. After an initial settling period, it represents the drift — the divergence between the rate at which frames are generated at the source and the rate at which the output device consumes them. In this case, the drift is negligible, but it can routinely be up to 150 ppm, especially with older machines. -"Corrections in ppm" is the number of frame insertions plus the number of frame deletions (i.e. the total number of corrections), given as a moving average in parts per million. The closer this is to the net corrections, the fewer "unnecessary" corrections that are being made. Third party programs tend to have much larger levels of corrections. +"Corrections in ppm" is the number of frame insertions plus the number of frame deletions (i.e. the total number of corrections), given as a moving average in parts per million. The closer this is to the net corrections, the fewer "unnecessary" corrections are being made. Third party programs tend to have much larger levels of corrections. "Min DAC queue size" is the minimum size the queue of samples in the output device's hardware buffer was measured at. It is meant to stand at 0.2 seconds = 8,820 frames at 44,100 frames per second, and will go low if the processor is very busy. If it goes below about 2,000 then it's an indication that the processor can't really keep up. From fe8198a23b1083272dbf3dbf3e06b41e7b3ed123 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 20 Dec 2019 17:17:21 +0000 Subject: [PATCH 004/186] Add SetAirplayVolume to the D-Bus RemoteControl interface. Add SetVolume to the MPRIS interface. Hook up the Volume property in the MPRIS interface. Modify RemoteCommand in the D-Bus interface to return the HTTP status and response. Change the type of airplay_volume from int to double in the metadata hub. Add a few sample commands in the D-Bus document. --- dacp.c | 2 ++ dacp.h | 2 ++ dbus-service.c | 40 +++++++++++++++++++++++++++++++--- documents/sample dbus commands | 15 +++++++++++++ metadata_hub.h | 2 +- mpris-service.c | 36 ++++++++++++++++++++++++++++-- org.gnome.ShairportSync.xml | 5 +++++ org.mpris.MediaPlayer2.xml | 3 +++ 8 files changed, 99 insertions(+), 6 deletions(-) diff --git a/dacp.c b/dacp.c index 79447afb..7797f794 100644 --- a/dacp.c +++ b/dacp.c @@ -592,6 +592,8 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { // debug(1,"dacp_monitor_thread_code: command: \"%s\"",command); result = dacp_send_command(command, &response, &le); // debug(1,"Response to \"%s\" is %d.",command,result); + // remember: unless the revision_number you pass in is 1, + // response will be 200 only if there's something new to report. if (result == 200) { // if (0) { char *sp = response; diff --git a/dacp.h b/dacp.h index ce1e1278..ea44aacb 100644 --- a/dacp.h +++ b/dacp.h @@ -32,6 +32,8 @@ void relinquish_dacp_server_information(rtsp_conn_info *conn); // tell the DACP // longer associated with it. void dacp_monitor_port_update_callback( char *dacp_id, uint16_t port); // a callback to say the port is no longer in use + +int dacp_send_command(const char *command, char **body, ssize_t *bodysize); int send_simple_dacp_command(const char *command); int dacp_set_include_speaker_volume(int64_t machine_number, int32_t vo); diff --git a/dbus-service.c b/dbus-service.c index 0c85a981..3ebd88f7 100644 --- a/dbus-service.c +++ b/dbus-service.c @@ -337,6 +337,19 @@ static gboolean on_handle_volume_down(ShairportSyncRemoteControl *skeleton, return TRUE; } +static gboolean on_handle_set_airplay_volume(ShairportSyncRemoteControl *skeleton, + GDBusMethodInvocation *invocation, const gdouble volume, + __attribute__((unused)) gpointer user_data) { + debug(2, "Set airplay volume to %.6f.", volume); + char command[256] = ""; + snprintf(command, sizeof(command), "setproperty?dmcp.device-volume=%.6f", volume); + send_simple_dacp_command(command); + shairport_sync_remote_control_complete_set_airplay_volume(skeleton, invocation); + return TRUE; +} + + + gboolean notify_elapsed_time_callback(ShairportSyncDiagnostics *skeleton, __attribute__((unused)) gpointer user_data) { // debug(1, "\"notify_elapsed_time_callback\" called."); @@ -714,12 +727,30 @@ static gboolean on_handle_quit(ShairportSync *skeleton, GDBusMethodInvocation *i static gboolean on_handle_remote_command(ShairportSync *skeleton, GDBusMethodInvocation *invocation, const gchar *command, __attribute__((unused)) gpointer user_data) { - debug(1, "RemoteCommand with command \"%s\".", command); - send_simple_dacp_command((const char *)command); - shairport_sync_complete_remote_command(skeleton, invocation); + debug(1, "RemoteCommand with command \"%s\".", command); + int reply = 0; + char *server_reply = NULL; + ssize_t reply_size = 0; + reply = dacp_send_command((const char *)command, &server_reply, &reply_size); + char *server_reply_hex = alloca(reply_size * 2 + 1); + if (server_reply_hex) { + char *p = server_reply_hex; + if (server_reply) { + char *q = server_reply; + int i; + for (i = 0; i < reply_size; i++) { + snprintf(p, 3, "%02X", *q); + p += 2; + q++; + } + } + *p = '\0'; + } + shairport_sync_complete_remote_command(skeleton, invocation, reply, server_reply_hex); return TRUE; } + static void on_dbus_name_acquired(GDBusConnection *connection, const gchar *name, __attribute__((unused)) gpointer user_data) { @@ -814,6 +845,9 @@ static void on_dbus_name_acquired(GDBusConnection *connection, const gchar *name G_CALLBACK(on_handle_volume_up), NULL); g_signal_connect(shairportSyncRemoteControlSkeleton, "handle-volume-down", G_CALLBACK(on_handle_volume_down), NULL); + g_signal_connect(shairportSyncRemoteControlSkeleton, "handle-set-airplay-volume", + G_CALLBACK(on_handle_set_airplay_volume), NULL); + g_signal_connect(shairportSyncAdvancedRemoteControlSkeleton, "handle-set-volume", G_CALLBACK(on_handle_set_volume), NULL); diff --git a/documents/sample dbus commands b/documents/sample dbus commands index 5b173d4a..96219c19 100644 --- a/documents/sample dbus commands +++ b/documents/sample dbus commands @@ -1,3 +1,6 @@ +# For the "Native" Shairport Sync D-Bus Interface support, Shairport Sync must be built with the D-Bus interface support. Add the '--with-dbus-interface' flag at the ./configure stage. +# There is a simple test client that you can have built -- add the '--with-dbus-test-client' flag at the ./configure stage. You'll get an executable called shairport-sync-dbus-test-client + # Get Log Verbosity dbus-send --print-reply --system --dest=org.gnome.ShairportSync /org/gnome/ShairportSync org.freedesktop.DBus.Properties.Get string:org.gnome.ShairportSync.Diagnostics string:Verbosity # Return Log Verbosity @@ -59,6 +62,11 @@ dbus-send --print-reply --system --dest=org.gnome.ShairportSync /org/gnome/Shair # Set Convolution Impulse Response File: dbus-send --print-reply --system --dest=org.gnome.ShairportSync /org/gnome/ShairportSync org.freedesktop.DBus.Properties.Set string:org.gnome.ShairportSync string:ConvolutionImpulseResponseFile variant:string:"/etc/shairport-sync/boom.wav" +# Set Airplay Volume using Remote Control. Airplay Volume is between -30.0 and 0.0 and maps linearly onto the slider, with -30.0 being lowest and 0.0 being highest. +dbus-send --system --print-reply --type=method_call --dest=org.gnome.ShairportSync '/org/gnome/ShairportSync' org.gnome.ShairportSync.RemoteControl.SetAirplayVolume double:-10.0 + + +# AdvancedRemoteControl interface. # Some commands and properties are accessible only through the AdvancedRemoteControl interface. # However, only iTunes or the macOS Music app from 10.15.2 onwards # provide the functionality needed for the AdvancedRemoteControl interface to work. @@ -70,3 +78,10 @@ dbus-send --print-reply --system --dest=org.gnome.ShairportSync /org/gnome/Shair # Set Volume using Advanced Remote Control -- only works if the org.gnome.ShairportSync.AdvancedRemoteControl is available. dbus-send --system --print-reply --type=method_call --dest=org.gnome.ShairportSync '/org/gnome/ShairportSync' org.gnome.ShairportSync.AdvancedRemoteControl.SetVolume int32:50 + +# MPRIS interface commands. +# For MPRIS support, Shairport Sync must be built with the MPRIS interface support. Add the '--with-mpris-interface' flag at the ./configure stage. +# There is a simple test client that you can have built -- add the '--with-mpris-test-client' flag at the ./configure stage. You'll get an executable called shairport-sync-mpris-test-client. +# This is mostly compatible with the MPRIS standard, except that Volume is read-only, with a separate SetVolume method. +# Set Volume, which must be between 0.0 and 1.0 and maps linearly onto the slider. +dbus-send --system --print-reply --type=method_call --dest=org.mpris.MediaPlayer2.ShairportSync '/org/mpris/MediaPlayer2' org.mpris.MediaPlayer2.Player.SetVolume double:0.3 diff --git a/metadata_hub.h b/metadata_hub.h index 5b565bba..ed7d947d 100644 --- a/metadata_hub.h +++ b/metadata_hub.h @@ -129,7 +129,7 @@ typedef struct metadata_bundle { int speaker_volume; // this is the actual speaker volume, allowing for the main volume and the // speaker volume control - int airplay_volume; + double airplay_volume; metadata_watcher watchers[number_of_watchers]; // functions to call if the metadata is changed. void *watchers_data[number_of_watchers]; // their individual data diff --git a/mpris-service.c b/mpris-service.c index 37e7bf8a..707c063c 100644 --- a/mpris-service.c +++ b/mpris-service.c @@ -14,10 +14,28 @@ #include "metadata_hub.h" #include "mpris-service.h" +double airplay_volume_to_mpris_volume(double sp) { + if (sp < -30.0) + sp = -30.0; + if (sp > 0.0) + sp = 0.0; + sp = (sp/30.0)+1; + return sp; +} + +double mpris_volume_to_airplay_volume(double sp) { + sp = (sp-1.0)*30.0; + if (sp < -30.0) + sp = -30.0; + if (sp > 0.0) + sp = 0.0; + return sp; +} + void mpris_metadata_watcher(struct metadata_bundle *argc, __attribute__((unused)) void *userdata) { // debug(1, "MPRIS metadata watcher called"); char response[100]; - + media_player2_player_set_volume(mprisPlayerPlayerSkeleton, airplay_volume_to_mpris_volume(argc->airplay_volume)); switch (argc->repeat_status) { case RS_NOT_AVAILABLE: strcpy(response, "Not Available"); @@ -229,6 +247,18 @@ static gboolean on_handle_play(MediaPlayer2Player *skeleton, GDBusMethodInvocati return TRUE; } +static gboolean on_handle_set_volume(MediaPlayer2Player *skeleton, + GDBusMethodInvocation *invocation, const gdouble volume, + __attribute__((unused)) gpointer user_data) { + double ap_volume = mpris_volume_to_airplay_volume(volume); + debug(2, "Set mpris volume to %.6f, i.e. airplay volume to %.6f.", volume, ap_volume); + char command[256] = ""; + snprintf(command, sizeof(command), "setproperty?dmcp.device-volume=%.6f", ap_volume); + send_simple_dacp_command(command); + media_player2_player_complete_play(skeleton, invocation); + return TRUE; +} + static void on_mpris_name_acquired(GDBusConnection *connection, const gchar *name, __attribute__((unused)) gpointer user_data) { @@ -254,7 +284,6 @@ static void on_mpris_name_acquired(GDBusConnection *connection, const gchar *nam media_player2_player_set_playback_status(mprisPlayerPlayerSkeleton, "Stopped"); media_player2_player_set_loop_status(mprisPlayerPlayerSkeleton, "None"); - media_player2_player_set_volume(mprisPlayerPlayerSkeleton, 0.5); media_player2_player_set_minimum_rate(mprisPlayerPlayerSkeleton, 1.0); media_player2_player_set_maximum_rate(mprisPlayerPlayerSkeleton, 1.0); media_player2_player_set_can_go_next(mprisPlayerPlayerSkeleton, TRUE); @@ -274,6 +303,9 @@ static void on_mpris_name_acquired(GDBusConnection *connection, const gchar *nam g_signal_connect(mprisPlayerPlayerSkeleton, "handle-next", G_CALLBACK(on_handle_next), NULL); g_signal_connect(mprisPlayerPlayerSkeleton, "handle-previous", G_CALLBACK(on_handle_previous), NULL); + g_signal_connect(mprisPlayerPlayerSkeleton, "handle-set-volume", G_CALLBACK(on_handle_set_volume), + NULL); + add_metadata_watcher(mpris_metadata_watcher, NULL); diff --git a/org.gnome.ShairportSync.xml b/org.gnome.ShairportSync.xml index 15317faa..91c31671 100644 --- a/org.gnome.ShairportSync.xml +++ b/org.gnome.ShairportSync.xml @@ -13,6 +13,8 @@ + + @@ -45,6 +47,9 @@ + + + diff --git a/org.mpris.MediaPlayer2.xml b/org.mpris.MediaPlayer2.xml index ae34e1e5..64ee2cca 100755 --- a/org.mpris.MediaPlayer2.xml +++ b/org.mpris.MediaPlayer2.xml @@ -41,6 +41,9 @@ + + + From 32646f22e2b40ae1b212fc1fe00486d0dc90d982 Mon Sep 17 00:00:00 2001 From: Pieter De Gendt Date: Mon, 23 Dec 2019 09:10:08 +0100 Subject: [PATCH 005/186] jack: Rename soxr "recipe" to "quality" and configurable buffer size --- audio_jack.c | 39 +++++++++++++++++++++---------------- common.h | 2 +- scripts/shairport-sync.conf | 2 ++ 3 files changed, 25 insertions(+), 18 deletions(-) diff --git a/audio_jack.c b/audio_jack.c index 499687a3..db6e1899 100644 --- a/audio_jack.c +++ b/audio_jack.c @@ -41,8 +41,6 @@ typedef jack_default_audio_sample_t sample_t; // Two-channel, 32bit audio: static const int bytes_per_frame = NPORTS * jack_sample_size; -// Four seconds buffer -- should be plenty -#define buffer_size (48000u * 4u * bytes_per_frame) static pthread_mutex_t buffer_mutex = PTHREAD_MUTEX_INITIALIZER; static pthread_mutex_t client_mutex = PTHREAD_MUTEX_INITIALIZER; @@ -86,12 +84,12 @@ static jack_latency_range_t latest_latency_range[NPORTS]; static int64_t time_of_latest_transfer; #ifdef CONFIG_SOXR -typedef struct soxr_recipe { - int recipe; +typedef struct soxr_quality { + int quality; const char *name; -} soxr_recipe_t; +} soxr_quality_t; -static soxr_recipe_t soxr_quality_table[] = { +static soxr_quality_t soxr_quality_table[] = { { SOXR_VHQ, "very high" }, { SOXR_HQ, "high" }, { SOXR_MQ, "medium" }, @@ -100,10 +98,10 @@ static soxr_recipe_t soxr_quality_table[] = { { -1, NULL } }; -static int parse_soxr_recipe_name(const char *name) { - for (soxr_recipe_t *s = soxr_quality_table; s->name != NULL; ++s) { +static int parse_soxr_quality_name(const char *name) { + for (soxr_quality_t *s = soxr_quality_table; s->name != NULL; ++s) { if (!strcmp(s->name, name)) { - return s->recipe; + return s->quality; } } return -1; @@ -213,6 +211,7 @@ static void info(const char *desc) { inform("JACK information: \"%s\"", desc); } int jack_init(__attribute__((unused)) int argc, __attribute__((unused)) char **argv) { int i; + int bufsz = -1; config.audio_backend_latency_offset = 0; config.audio_backend_buffer_desired_length = 0.500; // Below this, soxr interpolation will not occur -- it'll be basic interpolation @@ -222,7 +221,7 @@ int jack_init(__attribute__((unused)) int argc, __attribute__((unused)) char **a // Do the "general" audio options. Note, these options are in the "general" stanza! parse_general_audio_options(); #ifdef CONFIG_SOXR - config.jack_soxr_resample_recipe = -1; // don't resample by default + config.jack_soxr_resample_quality = -1; // don't resample by default #endif // Now the options specific to the backend, from the "jack" stanza: @@ -235,18 +234,24 @@ int jack_init(__attribute__((unused)) int argc, __attribute__((unused)) char **a config.jack_autoconnect_pattern = (char *)str; } #ifdef CONFIG_SOXR - if (config_lookup_string(config.cfg, "jack.soxr_resample_recipe", &str)) { + if (config_lookup_string(config.cfg, "jack.soxr_resample_quality", &str)) { debug(1, "SOXR quality %s", str); - config.jack_soxr_resample_recipe = parse_soxr_recipe_name(str); + config.jack_soxr_resample_quality = parse_soxr_quality_name(str); } #endif + if (config_lookup_int(config.cfg, "jack.bufsz", &bufsz) && bufsz <= 0) + die("jack: bufsz must be > 0"); } if (config.jack_client_name == NULL) config.jack_client_name = strdup("shairport-sync"); - jackbuf = jack_ringbuffer_create(buffer_size); + // by default a buffer that can hold up to 4 seconds of 48kHz samples + if (bufsz <= 0) + bufsz = 48000 * 4 * bytes_per_frame; + + jackbuf = jack_ringbuffer_create((size_t)bufsz); if (jackbuf == NULL) - die("Can't allocate %d bytes for the JACK ringbuffer.", buffer_size); + die("Can't allocate %d bytes for the JACK ringbuffer.", bufsz); // Lock the ringbuffer into memory so that it never gets paged out, which would // break realtime constraints. jack_ringbuffer_mlock(jackbuf); @@ -261,8 +266,8 @@ int jack_init(__attribute__((unused)) int argc, __attribute__((unused)) char **a } sample_rate = jack_get_sample_rate(client); #ifdef CONFIG_SOXR - if (config.jack_soxr_resample_recipe >= SOXR_QQ) { - quality_spec = soxr_quality_spec(config.jack_soxr_resample_recipe, 0); + if (config.jack_soxr_resample_quality >= SOXR_QQ) { + quality_spec = soxr_quality_spec(config.jack_soxr_resample_quality, 0); io_spec = soxr_io_spec(SOXR_INT16_I, SOXR_FLOAT32_I); } else #endif @@ -354,7 +359,7 @@ void jack_start(int i_sample_rate, // Also, we have no say over the sample rate or sample format of JACK, // We convert the 16bit samples to float, and die if the sample rate is != 44k1 without soxr. #ifdef CONFIG_SOXR - if (config.jack_soxr_resample_recipe >= SOXR_QQ) { + if (config.jack_soxr_resample_quality >= SOXR_QQ) { // we might improve a bit with soxr_clear if the sample_rate doesn't change if (soxr) { soxr_delete(soxr); diff --git a/common.h b/common.h index 45fc9683..37217ab8 100644 --- a/common.h +++ b/common.h @@ -271,7 +271,7 @@ typedef struct { char *jack_client_name; char *jack_autoconnect_pattern; #ifdef CONFIG_SOXR - int jack_soxr_resample_recipe; + int jack_soxr_resample_quality; #endif #endif diff --git a/scripts/shairport-sync.conf b/scripts/shairport-sync.conf index bec0d4fc..9c875156 100644 --- a/scripts/shairport-sync.conf +++ b/scripts/shairport-sync.conf @@ -134,6 +134,8 @@ jack = // "jack_mixer:in_2[78]" // Beware: if you make a syntax error, libjack might crash. In that case, fix it and start over. // For a good overview, look here: https://www.ibm.com/support/knowledgecenter/SS8NLW_11.0.1/com.ibm.swg.im.infosphere.dataexpl.engine.doc/c_posix-regex-examples.html +// soxr_resample_quality = "none"; // Enable resampling by setting this to "very high", "high", "medium", "low" or "quick" +// bufsz = ; // advanced optional setting to set the buffer size to this value }; // Parameters for the "pipe" audio back end, a back end that directs raw CD-style audio output to a pipe. No interpolation is done. From 362ae3f7d8660a6858ceda4e02543276f8f86d67 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 23 Dec 2019 12:03:21 +0000 Subject: [PATCH 006/186] Update configure.ac --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index eca3d41b..cc855318 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.6d1], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.6d2], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 26f08b4188ba5490554f92e4efc2927ebed6fa27 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 23 Dec 2019 12:09:05 +0000 Subject: [PATCH 007/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 775d9c69..ebdba4fe 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,12 @@ +Version 3.3.6d2 +==== +**New Features** +* Resampling has been added for the [Jack Audio](http://jackaudio.org) backend. This can be used to up-sample to 192kHz. Thanks to [Pieter De Gendt](https://github.com/pdgendt) for this addition. + +Version 3.3.6d1 +==== +This is release version 3.3.5. + Version 3.3.5d1 ==== **Pesky Changes You May Not Be Able To Ignore** From 272bd2db487065575c1684abfbb707e3913ebc66 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 23 Dec 2019 12:12:43 +0000 Subject: [PATCH 008/186] Add the ability to set the volume directly to the D-Bus RemoteControl interface and to the MPRIS interface. Both use the recently-discovered ability to set the device_volume. --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index cc855318..7b5d83da 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.6d2], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.6d3], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 772ba1178c15ca4b4ff41ebcf637441af9f4a0aa Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 23 Dec 2019 12:18:48 +0000 Subject: [PATCH 009/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index ebdba4fe..4dfe20cf 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,16 @@ +Versioon 3.3.6d3 +==== +**New Features** +* Add `SetAirplayVolume` to the native D-Bus `RemoteControl` interface. +* Add `SetVolume` to the `MPRIS` interface. + +**Bug Fixes** +* Hook up the `Volume` property in the `MPRIS` interface. + +**Enhancements** +* Modify `RemoteCommand` in the D-Bus interface to return the HTTP status and response in hex. +* Add a few sample commands, including `MPRIS` commands, to the [Sample D-Bus Commands](https://github.com/mikebrady/shairport-sync/blob/development/documents/sample%20dbus%20commands) document. + Version 3.3.6d2 ==== **New Features** From 7cd075dd613558bcdaef1d27a3a8cf8c2cc3662d Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 23 Dec 2019 12:19:01 +0000 Subject: [PATCH 010/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 4dfe20cf..bf94f4fd 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,4 +1,4 @@ -Versioon 3.3.6d3 +Version 3.3.6d3 ==== **New Features** * Add `SetAirplayVolume` to the native D-Bus `RemoteControl` interface. From 1f75fe4e7aa0b618f98ad22548217c36b65c7b28 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 27 Dec 2019 14:37:01 +0000 Subject: [PATCH 011/186] Add or update some of the copyright notices --- alac.h | 30 ++++++++++++++++++++++++++++++ apple_alac.cpp | 26 ++++++++++++++++++++++++++ apple_alac.h | 25 +++++++++++++++++++++++++ audio.c | 2 +- dbus-service.c | 26 ++++++++++++++++++++++++++ metadata_hub.c | 2 +- mpris-service.c | 25 +++++++++++++++++++++++++ shairport-sync-dbus-test-client.c | 25 +++++++++++++++++++++++++ shairport-sync-mpris-test-client.c | 26 ++++++++++++++++++++++++++ 9 files changed, 185 insertions(+), 2 deletions(-) diff --git a/alac.h b/alac.h index 33a35928..7cdd507d 100644 --- a/alac.h +++ b/alac.h @@ -1,3 +1,33 @@ +/* + * ALAC (Apple Lossless Audio Codec) decoder + * Copyright (c) 2005 David Hammerton + * All rights reserved. + * + * http://crazney.net/programs/itunes/alac.html + * + * Permission is hereby granted, free of charge, to any person + * obtaining a copy of this software and associated documentation + * files (the "Software"), to deal in the Software without + * restriction, including without limitation the rights to use, + * copy, modify, merge, publish, distribute, sublicense, and/or + * sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES + * OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND + * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + * + */ + + #ifndef __ALAC__DECOMP_H #define __ALAC__DECOMP_H diff --git a/apple_alac.cpp b/apple_alac.cpp index 6d543b2e..485c5bfe 100644 --- a/apple_alac.cpp +++ b/apple_alac.cpp @@ -1,3 +1,29 @@ +/* + * This file is part of Shairport Sync. + * Copyright (c) Mike Brady 2019 + * All rights reserved. + * + * Permission is hereby granted, free of charge, to any person + * obtaining a copy of this software and associated documentation + * files (the "Software"), to deal in the Software without + * restriction, including without limitation the rights to use, + * copy, modify, merge, publish, distribute, sublicense, and/or + * sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES + * OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND + * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + */ + #include // these are headers for the ALAC decoder, utilities and endian utilities diff --git a/apple_alac.h b/apple_alac.h index eb8bcf6c..be11e86e 100644 --- a/apple_alac.h +++ b/apple_alac.h @@ -1,3 +1,28 @@ +/* + * This file is part of Shairport Sync. + * Copyright (c) Mike Brady 2019 + * All rights reserved. + * + * Permission is hereby granted, free of charge, to any person + * obtaining a copy of this software and associated documentation + * files (the "Software"), to deal in the Software without + * restriction, including without limitation the rights to use, + * copy, modify, merge, publish, distribute, sublicense, and/or + * sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES + * OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND + * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + */ #ifndef __APPLE_ALAC_H #define __APPLE_ALAC_H diff --git a/audio.c b/audio.c index c71ac939..8b2762f4 100644 --- a/audio.c +++ b/audio.c @@ -1,7 +1,7 @@ /* * Audio driver handler. This file is part of Shairport. * Copyright (c) James Laird 2013 - * Modifications (c) Mike Brady 2014 -- 2018 + * Modifications (c) Mike Brady 2014 -- 2019 * All rights reserved. * * Permission is hereby granted, free of charge, to any person diff --git a/dbus-service.c b/dbus-service.c index 3ebd88f7..d9d6e95d 100644 --- a/dbus-service.c +++ b/dbus-service.c @@ -1,3 +1,29 @@ +/* + * This file is part of Shairport Sync. + * Copyright (c) Mike Brady 2018 -- 2019 + * All rights reserved. + * + * Permission is hereby granted, free of charge, to any person + * obtaining a copy of this software and associated documentation + * files (the "Software"), to deal in the Software without + * restriction, including without limitation the rights to use, + * copy, modify, merge, publish, distribute, sublicense, and/or + * sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES + * OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND + * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + */ + #include #include #include diff --git a/metadata_hub.c b/metadata_hub.c index 0b5aa412..fb459b6b 100644 --- a/metadata_hub.c +++ b/metadata_hub.c @@ -5,7 +5,7 @@ * then you need a metadata hub, * where everything is stored * This file is part of Shairport Sync. - * Copyright (c) Mike Brady 2017--2018 + * Copyright (c) Mike Brady 2017--2019 * All rights reserved. * * Permission is hereby granted, free of charge, to any person diff --git a/mpris-service.c b/mpris-service.c index 707c063c..387b3a23 100644 --- a/mpris-service.c +++ b/mpris-service.c @@ -1,3 +1,28 @@ +/* + * This file is part of Shairport Sync. + * Copyright (c) Mike Brady 2018 -- 2019 + * All rights reserved. + * + * Permission is hereby granted, free of charge, to any person + * obtaining a copy of this software and associated documentation + * files (the "Software"), to deal in the Software without + * restriction, including without limitation the rights to use, + * copy, modify, merge, publish, distribute, sublicense, and/or + * sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES + * OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND + * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + */ #include #include diff --git a/shairport-sync-dbus-test-client.c b/shairport-sync-dbus-test-client.c index bf968c45..6b8f122c 100644 --- a/shairport-sync-dbus-test-client.c +++ b/shairport-sync-dbus-test-client.c @@ -1,3 +1,28 @@ +/* + * This file is part of Shairport Sync. + * Copyright (c) Mike Brady 2019 + * All rights reserved. + * + * Permission is hereby granted, free of charge, to any person + * obtaining a copy of this software and associated documentation + * files (the "Software"), to deal in the Software without + * restriction, including without limitation the rights to use, + * copy, modify, merge, publish, distribute, sublicense, and/or + * sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES + * OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND + * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + */ #include "dbus-interface.h" #include #include diff --git a/shairport-sync-mpris-test-client.c b/shairport-sync-mpris-test-client.c index c15c71bc..7af29aea 100644 --- a/shairport-sync-mpris-test-client.c +++ b/shairport-sync-mpris-test-client.c @@ -1,3 +1,29 @@ +/* + * This file is part of Shairport Sync. + * Copyright (c) Mike Brady 2019 + * All rights reserved. + * + * Permission is hereby granted, free of charge, to any person + * obtaining a copy of this software and associated documentation + * files (the "Software"), to deal in the Software without + * restriction, including without limitation the rights to use, + * copy, modify, merge, publish, distribute, sublicense, and/or + * sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES + * OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND + * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + */ + #include "mpris-interface.h" #include #include From 85b17cdaef3f5878c7afd793aa8be9001d6b1730 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 10 Jan 2020 08:31:13 +0000 Subject: [PATCH 012/186] Remove timeout on reception of DACP responses to allow for long poll behaviour of some players, e.g. Forked Daapd. Add some examples to the dbus client. --- dacp.c | 608 +++++++++++++++--------------- shairport-sync-dbus-test-client.c | 39 +- 2 files changed, 341 insertions(+), 306 deletions(-) diff --git a/dacp.c b/dacp.c index 7797f794..a47d1608 100644 --- a/dacp.c +++ b/dacp.c @@ -150,12 +150,12 @@ void http_cleanup(void *arg) { int dacp_send_command(const char *command, char **body, ssize_t *bodysize) { int result; - // debug(1,"dacp_send_command: command is: \"%s\".",command); - - if (dacp_server.port == 0) { - debug(1,"No DACP port specified yet"); - result = 490; // no port specified - } else { + // debug(1,"dacp_send_command: command is: \"%s\".",command); + + if (dacp_server.port == 0) { + debug(1, "No DACP port specified yet"); + result = 490; // no port specified + } else { // will malloc space for the body or set it to NULL -- the caller should free it. @@ -213,20 +213,27 @@ int dacp_send_command(const char *command, char **body, ssize_t *bodysize) { sockfd = socket(res->ai_family, res->ai_socktype, res->ai_protocol); if (sockfd == -1) { - // debug(1, "DACP socket could not be created -- error %d: \"%s\".",errno,strerror(errno)); + // debug(1, "DACP socket could not be created -- error %d: + // \"%s\".",errno,strerror(errno)); response.code = 497; // Can't establish a socket to the DACP server } else { pthread_cleanup_push(connect_cleanup, (void *)&sockfd); // debug(2, "dacp_send_command: open socket %d.",sockfd); + + // This is for limiting the time to be spent waiting for a response. + struct timeval tv; tv.tv_sec = 0; tv.tv_usec = 500000; - if (setsockopt(sockfd, SOL_SOCKET, SO_RCVTIMEO, (const char *)&tv, sizeof tv) == -1) - debug(1, "dacp_send_command: error %d setting receive timeout.", errno); + // Don't place a timeout on the receive side of these calls, as some clients may provide + // a "long poll" mechanism whereby a reply will not be sent until a change has occurred + // if (setsockopt(sockfd, SOL_SOCKET, SO_RCVTIMEO, (const char *)&tv, sizeof tv) == -1) + // debug(1, "dacp_send_command: error %d setting receive timeout.", errno); if (setsockopt(sockfd, SOL_SOCKET, SO_SNDTIMEO, (const char *)&tv, sizeof tv) == -1) debug(1, "dacp_send_command: error %d setting send timeout.", errno); + // connect! // debug(1, "DACP socket created."); if (connect(sockfd, res->ai_addr, res->ai_addrlen) < 0) { @@ -240,7 +247,8 @@ int dacp_send_command(const char *command, char **body, ssize_t *bodysize) { snprintf(message, sizeof(message), "GET /ctrl-int/1/%s HTTP/1.1\r\nHost: %s:%u\r\nActive-Remote: %u\r\n\r\n", - command, dacp_server.ip_string, dacp_server.port, dacp_server.active_remote_id); + command, dacp_server.ip_string, dacp_server.port, + dacp_server.active_remote_id); // Send command debug(3, "dacp_send_command: \"%s\".", command); @@ -276,8 +284,12 @@ int dacp_send_command(const char *command, char **body, ssize_t *bodysize) { memset(buffer, 0, sizeof(buffer)); while (needmore && !looperror) { const char *data = buffer; - if (setsockopt(sockfd, SOL_SOCKET, SO_RCVTIMEO, (const char *)&tv, sizeof tv) == -1) - debug(1, "dacp_send_command: error %d setting receive timeout.", errno); + + // Don't place a timeout on the receive, as some clients may provide + // a "long poll" mechanism whereby a reply will not be sent until a change has occurred + + // if (setsockopt(sockfd, SOL_SOCKET, SO_RCVTIMEO, (const char *)&tv, sizeof tv) == -1) + // debug(1, "dacp_send_command: error %d setting receive timeout.", errno); ssize_t ndata = recv(sockfd, buffer, sizeof(buffer), 0); // debug(3, "Received %d bytes: \"%s\".", ndata, buffer); if (ndata <= 0) { @@ -289,10 +301,12 @@ int dacp_send_command(const char *command, char **body, ssize_t *bodysize) { struct linger so_linger; so_linger.l_onoff = 1; // "true" so_linger.l_linger = 0; - int err = setsockopt(sockfd, SOL_SOCKET, SO_LINGER, &so_linger, sizeof so_linger); + int err = + setsockopt(sockfd, SOL_SOCKET, SO_LINGER, &so_linger, sizeof so_linger); if (err) - debug(1, - "Could not set the dacp socket to abort due to a read error on closing."); + debug( + 1, + "Could not set the dacp socket to abort due to a read error on closing."); } free(response.body); @@ -487,7 +501,8 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { metadata_store.dacp_server_active = 0; metadata_store.advanced_dacp_server_active = 0; debug(2, - "setting metadata_store.dacp_server_active and metadata_store.advanced_dacp_server_active to 0 with an update " + "setting metadata_store.dacp_server_active and " + "metadata_store.advanced_dacp_server_active to 0 with an update " "flag value of %d", ch); metadata_hub_modify_epilog(ch); @@ -499,9 +514,8 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { // so dacp_server.scan_enable will be true at this point bad_result_count = 0; idle_scan_count = 0; - } - + result = dacp_get_volume(&the_volume); // just want the http code pthread_cleanup_pop(1); @@ -514,25 +528,27 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { if (bad_result_count < config.scan_max_bad_response_count) // limit to some reasonable value bad_result_count++; } - + // here, do the debouncing calculations to see // if the dacp server and the - // advanced dacp server are available - + // advanced dacp server are available + // -1 means we don't know because some bad statuses have been reported // 0 means definitely no // +1 means definitely yes - + int dacp_server_status_now = -1; int advanced_dacp_server_status_now = -1; - - if (bad_result_count == 0) { + + if (bad_result_count == 0) { dacp_server_status_now = 1; if (result == 200) advanced_dacp_server_status_now = 1; else if (result == 400) advanced_dacp_server_status_now = 0; - } else if (bad_result_count == config.scan_max_bad_response_count) { // if a sequence of bad return codes occurs, then it's gone + } else if (bad_result_count == + config.scan_max_bad_response_count) { // if a sequence of bad return codes occurs, + // then it's gone dacp_server_status_now = 0; advanced_dacp_server_status_now = 0; } @@ -545,14 +561,14 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { debug(3, "Scan Result: %d, Bad Scan Count: %d, Idle Scan Count: %d.", result, bad_result_count, idle_scan_count); -/* not used -// decide if idle for too long - if (idle_scan_count == config.scan_max_inactive_count) { - debug(2, "DACP server status scanning stopped."); - dacp_server.scan_enable = 0; - } -*/ - + /* not used + // decide if idle for too long + if (idle_scan_count == config.scan_max_inactive_count) { + debug(2, "DACP server status scanning stopped."); + dacp_server.scan_enable = 0; + } + */ + int update_needed = 0; metadata_hub_modify_prolog(); if (dacp_server_status_now != -1) { // if dacp_server_status_now is actually known... @@ -562,7 +578,8 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { update_needed = 1; } } - if (advanced_dacp_server_status_now != -1) { // if advanced_dacp_server_status_now is actually known... + if (advanced_dacp_server_status_now != + -1) { // if advanced_dacp_server_status_now is actually known... if (metadata_store.advanced_dacp_server_active != advanced_dacp_server_status_now) { debug(2, "metadata_store.advanced_dacp_server_active set to %d.", dacp_server_status_now); metadata_store.advanced_dacp_server_active = advanced_dacp_server_status_now; @@ -571,286 +588,289 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { } metadata_hub_modify_epilog(update_needed); - + // pthread_mutex_unlock(&dacp_server_information_lock); // debug(1, "DACP Server ID \"%u\" at \"%s:%u\", scan %d.", dacp_server.active_remote_id, // dacp_server.ip_string, dacp_server.port, scan_index); + if (result == 200) { + metadata_hub_modify_prolog(); + int diff = metadata_store.speaker_volume != the_volume; + if (diff) + metadata_store.speaker_volume = the_volume; + metadata_hub_modify_epilog(diff); + + ssize_t le; + char *response = NULL; + int32_t item_size; + char command[1024] = ""; + snprintf(command, sizeof(command) - 1, "playstatusupdate?revision-number=%d", + revision_number); + // debug(1,"dacp_monitor_thread_code: command: \"%s\"",command); + result = dacp_send_command(command, &response, &le); + // debug(1,"Response to \"%s\" is %d.",command,result); + // remember: unless the revision_number you pass in is 1, + // response will be 200 only if there's something new to report. if (result == 200) { - metadata_hub_modify_prolog(); - int diff = metadata_store.speaker_volume != the_volume; - if (diff) - metadata_store.speaker_volume = the_volume; - metadata_hub_modify_epilog(diff); + // if (0) { + char *sp = response; + if (le >= 8) { + // here start looking for the contents of the status update + if (dacp_tlv_crawl(&sp, &item_size) == 'cmst') { // status + // here, we know that we are receiving playerstatusupdates, so set a flag + metadata_hub_modify_prolog(); + // debug(1, "playstatusupdate release track metadata"); + // metadata_hub_reset_track_metadata(); + // metadata_store.playerstatusupdates_are_received = 1; + sp -= item_size; // drop down into the array -- don't skip over it + le -= 8; + // char typestring[5]; + // we need to acquire the metadata data structure and possibly update it + while (le >= 8) { + uint32_t type = dacp_tlv_crawl(&sp, &item_size); + le -= item_size + 8; + char *t; + // char u; + // char *st; + int32_t r; + uint32_t ui; + // uint64_t v; + // int i; - ssize_t le; - char *response = NULL; - int32_t item_size; - char command[1024] = ""; - snprintf(command, sizeof(command) - 1, "playstatusupdate?revision-number=%d", - revision_number); - // debug(1,"dacp_monitor_thread_code: command: \"%s\"",command); - result = dacp_send_command(command, &response, &le); - // debug(1,"Response to \"%s\" is %d.",command,result); - // remember: unless the revision_number you pass in is 1, - // response will be 200 only if there's something new to report. - if (result == 200) { - // if (0) { - char *sp = response; - if (le >= 8) { - // here start looking for the contents of the status update - if (dacp_tlv_crawl(&sp, &item_size) == 'cmst') { // status - // here, we know that we are receiving playerstatusupdates, so set a flag - metadata_hub_modify_prolog(); - // debug(1, "playstatusupdate release track metadata"); - // metadata_hub_reset_track_metadata(); - // metadata_store.playerstatusupdates_are_received = 1; - sp -= item_size; // drop down into the array -- don't skip over it - le -= 8; - // char typestring[5]; - // we need to acquire the metadata data structure and possibly update it - while (le >= 8) { - uint32_t type = dacp_tlv_crawl(&sp, &item_size); - le -= item_size + 8; - char *t; - // char u; - // char *st; - int32_t r; - uint32_t ui; - // uint64_t v; - // int i; - - switch (type) { - case 'cmsr': // revision number - t = sp - item_size; - revision_number = ntohl(*(uint32_t *)(t)); - // debug(1," Serial Number: %d", revision_number); - break; - case 'caps': // play status - t = sp - item_size; - r = *(unsigned char *)(t); - switch (r) { - case 2: - if (metadata_store.play_status != PS_STOPPED) { - metadata_store.play_status = PS_STOPPED; - debug(2, "Play status is \"stopped\"."); - } - break; - case 3: - if (metadata_store.play_status != PS_PAUSED) { - metadata_store.play_status = PS_PAUSED; - debug(2, "Play status is \"paused\"."); - } - break; - case 4: - if (metadata_store.play_status != PS_PLAYING) { - metadata_store.play_status = PS_PLAYING; - debug(2, "Play status changed to \"playing\"."); - } - break; - default: - debug(1, "Unrecognised play status %d received.", r); - break; + switch (type) { + case 'cmsr': // revision number + t = sp - item_size; + revision_number = ntohl(*(uint32_t *)(t)); + // debug(1," Serial Number: %d", revision_number); + break; + case 'caps': // play status + t = sp - item_size; + r = *(unsigned char *)(t); + switch (r) { + case 2: + if (metadata_store.play_status != PS_STOPPED) { + metadata_store.play_status = PS_STOPPED; + debug(2, "Play status is \"stopped\"."); } break; - case 'cash': // shuffle status - t = sp - item_size; - r = *(unsigned char *)(t); - switch (r) { - case 0: - if (metadata_store.shuffle_status != SS_OFF) { - metadata_store.shuffle_status = SS_OFF; - debug(2, "Shuffle status is \"off\"."); - } - break; - case 1: - if (metadata_store.shuffle_status != SS_ON) { - metadata_store.shuffle_status = SS_ON; - debug(2, "Shuffle status is \"on\"."); - } - break; - default: - debug(1, "Unrecognised shuffle status %d received.", r); - break; + case 3: + if (metadata_store.play_status != PS_PAUSED) { + metadata_store.play_status = PS_PAUSED; + debug(2, "Play status is \"paused\"."); } break; - case 'carp': // repeat status - t = sp - item_size; - r = *(unsigned char *)(t); - switch (r) { - case 0: - if (metadata_store.repeat_status != RS_OFF) { - metadata_store.repeat_status = RS_OFF; - debug(2, "Repeat status is \"none\"."); - } - break; - case 1: - if (metadata_store.repeat_status != RS_ONE) { - metadata_store.repeat_status = RS_ONE; - debug(2, "Repeat status is \"one\"."); - } - break; - case 2: - if (metadata_store.repeat_status != RS_ALL) { - metadata_store.repeat_status = RS_ALL; - debug(2, "Repeat status is \"all\"."); - } - break; - default: - debug(1, "Unrecognised repeat status %d received.", r); - break; + case 4: + if (metadata_store.play_status != PS_PLAYING) { + metadata_store.play_status = PS_PLAYING; + debug(2, "Play status changed to \"playing\"."); } break; - case 'cann': // track name - debug(2, "DACP Track Name seen"); - if (string_update_with_size(&metadata_store.track_name, &metadata_store.track_name_changed, - sp - item_size, item_size)) { - debug(2, "DACP Track Name set to: \"%s\"", metadata_store.track_name); - } - break; - case 'cana': // artist name - debug(2, "DACP Artist Name seen"); - if (string_update_with_size(&metadata_store.artist_name, - &metadata_store.artist_name_changed, sp - item_size, - item_size)) { - debug(2, "DACP Artist Name set to: \"%s\"", metadata_store.artist_name); - } - break; - case 'canl': // album name - debug(2, "DACP Album Name seen"); - if (string_update_with_size(&metadata_store.album_name, &metadata_store.album_name_changed, - sp - item_size, item_size)) { - debug(2, "DACP Album Name set to: \"%s\"", metadata_store.album_name); - } - break; - case 'cang': // genre - debug(2, "DACP Genre seen"); - if (string_update_with_size(&metadata_store.genre, &metadata_store.genre_changed, - sp - item_size, item_size)) { - debug(2, "DACP Genre set to: \"%s\"", metadata_store.genre); - } - break; - case 'canp': // nowplaying 4 ids: dbid, plid, playlistItem, itemid (from mellowware - // see reference above) - debug(2, "DACP Composite ID seen"); - if (memcmp(metadata_store.item_composite_id, sp - item_size, - sizeof(metadata_store.item_composite_id)) != 0) { - memcpy(metadata_store.item_composite_id, sp - item_size, - sizeof(metadata_store.item_composite_id)); - char st[33]; - char *pt = st; - int it; - for (it = 0; it < 16; it++) { - snprintf(pt, 3, "%02X", metadata_store.item_composite_id[it]); - pt += 2; - } - *pt = 0; - debug(2, "Item composite ID changed to 0x%s.", st); - metadata_store.item_composite_id_changed = 1; - } - break; - case 'astm': - t = sp - item_size; - ui = ntohl(*(uint32_t *)(t)); - debug(2, "DACP Song Time seen: \"%u\" of length %u.", ui, item_size); - if (ui != metadata_store.songtime_in_milliseconds) { - metadata_store.songtime_in_milliseconds = ui; - metadata_store.songtime_in_milliseconds_changed = 1; - debug(2, "DACP Song Time set to: \"%u\"", - metadata_store.songtime_in_milliseconds); - } - break; - - /* - case 'mstt': - case 'cant': - case 'cast': - case 'cmmk': - case 'caas': - case 'caar': - t = sp - item_size; - r = ntohl(*(uint32_t *)(t)); - printf(" %d", r); - printf(" (0x"); - t = sp - item_size; - for (i = 0; i < item_size; i++) { - printf("%02x", *t & 0xff); - t++; - } - printf(")"); - break; - case 'asai': - t = sp - item_size; - s = ntohl(*(uint32_t *)(t)); - s = s << 32; - t += 4; - v = (ntohl(*(uint32_t *)(t))) & 0xffffffff; - s += v; - printf(" %lu", s); - printf(" (0x"); - t = sp - item_size; - for (i = 0; i < item_size; i++) { - printf("%02x", *t & 0xff); - t++; - } - printf(")"); - break; - */ default: - /* - printf(" 0x"); - t = sp - item_size; - for (i = 0; i < item_size; i++) { - printf("%02x", *t & 0xff); - t++; - } - */ + debug(1, "Unrecognised play status %d received.", r); break; } - // printf("\n"); - } + break; + case 'cash': // shuffle status + t = sp - item_size; + r = *(unsigned char *)(t); + switch (r) { + case 0: + if (metadata_store.shuffle_status != SS_OFF) { + metadata_store.shuffle_status = SS_OFF; + debug(2, "Shuffle status is \"off\"."); + } + break; + case 1: + if (metadata_store.shuffle_status != SS_ON) { + metadata_store.shuffle_status = SS_ON; + debug(2, "Shuffle status is \"on\"."); + } + break; + default: + debug(1, "Unrecognised shuffle status %d received.", r); + break; + } + break; + case 'carp': // repeat status + t = sp - item_size; + r = *(unsigned char *)(t); + switch (r) { + case 0: + if (metadata_store.repeat_status != RS_OFF) { + metadata_store.repeat_status = RS_OFF; + debug(2, "Repeat status is \"none\"."); + } + break; + case 1: + if (metadata_store.repeat_status != RS_ONE) { + metadata_store.repeat_status = RS_ONE; + debug(2, "Repeat status is \"one\"."); + } + break; + case 2: + if (metadata_store.repeat_status != RS_ALL) { + metadata_store.repeat_status = RS_ALL; + debug(2, "Repeat status is \"all\"."); + } + break; + default: + debug(1, "Unrecognised repeat status %d received.", r); + break; + } + break; + case 'cann': // track name + debug(2, "DACP Track Name seen"); + if (string_update_with_size(&metadata_store.track_name, + &metadata_store.track_name_changed, sp - item_size, + item_size)) { + debug(2, "DACP Track Name set to: \"%s\"", metadata_store.track_name); + } + break; + case 'cana': // artist name + debug(2, "DACP Artist Name seen"); + if (string_update_with_size(&metadata_store.artist_name, + &metadata_store.artist_name_changed, sp - item_size, + item_size)) { + debug(2, "DACP Artist Name set to: \"%s\"", metadata_store.artist_name); + } + break; + case 'canl': // album name + debug(2, "DACP Album Name seen"); + if (string_update_with_size(&metadata_store.album_name, + &metadata_store.album_name_changed, sp - item_size, + item_size)) { + debug(2, "DACP Album Name set to: \"%s\"", metadata_store.album_name); + } + break; + case 'cang': // genre + debug(2, "DACP Genre seen"); + if (string_update_with_size(&metadata_store.genre, &metadata_store.genre_changed, + sp - item_size, item_size)) { + debug(2, "DACP Genre set to: \"%s\"", metadata_store.genre); + } + break; + case 'canp': // nowplaying 4 ids: dbid, plid, playlistItem, itemid (from mellowware + // see reference above) + debug(2, "DACP Composite ID seen"); + if (memcmp(metadata_store.item_composite_id, sp - item_size, + sizeof(metadata_store.item_composite_id)) != 0) { + memcpy(metadata_store.item_composite_id, sp - item_size, + sizeof(metadata_store.item_composite_id)); + char st[33]; + char *pt = st; + int it; + for (it = 0; it < 16; it++) { + snprintf(pt, 3, "%02X", metadata_store.item_composite_id[it]); + pt += 2; + } + *pt = 0; + debug(2, "Item composite ID changed to 0x%s.", st); + metadata_store.item_composite_id_changed = 1; + } + break; + case 'astm': + t = sp - item_size; + ui = ntohl(*(uint32_t *)(t)); + debug(2, "DACP Song Time seen: \"%u\" of length %u.", ui, item_size); + if (ui != metadata_store.songtime_in_milliseconds) { + metadata_store.songtime_in_milliseconds = ui; + metadata_store.songtime_in_milliseconds_changed = 1; + debug(2, "DACP Song Time set to: \"%u\"", + metadata_store.songtime_in_milliseconds); + } + break; - // finished possibly writing to the metadata hub - metadata_hub_modify_epilog(1); // should really see if this can be made responsive to changes - } else { - debug(1, "Status Update not found.\n"); + /* + case 'mstt': + case 'cant': + case 'cast': + case 'cmmk': + case 'caas': + case 'caar': + t = sp - item_size; + r = ntohl(*(uint32_t *)(t)); + printf(" %d", r); + printf(" (0x"); + t = sp - item_size; + for (i = 0; i < item_size; i++) { + printf("%02x", *t & 0xff); + t++; + } + printf(")"); + break; + case 'asai': + t = sp - item_size; + s = ntohl(*(uint32_t *)(t)); + s = s << 32; + t += 4; + v = (ntohl(*(uint32_t *)(t))) & 0xffffffff; + s += v; + printf(" %lu", s); + printf(" (0x"); + t = sp - item_size; + for (i = 0; i < item_size; i++) { + printf("%02x", *t & 0xff); + t++; + } + printf(")"); + break; + */ + default: + /* + printf(" 0x"); + t = sp - item_size; + for (i = 0; i < item_size; i++) { + printf("%02x", *t & 0xff); + t++; + } + */ + break; + } + // printf("\n"); } + + // finished possibly writing to the metadata hub + metadata_hub_modify_epilog( + 1); // should really see if this can be made responsive to changes } else { - debug(1, "Can't find any content in playerstatusupdate request"); + debug(1, "Status Update not found.\n"); } - } /* else { - if (result != 403) - debug(1, "Unexpected response %d to playerstatusupdate request", result); - } */ - if (response) { - free(response); - response = NULL; - }; + } else { + debug(1, "Can't find any content in playerstatusupdate request"); + } + } /* else { + if (result != 403) + debug(1, "Unexpected response %d to playerstatusupdate request", result); + } */ + if (response) { + free(response); + response = NULL; }; - /* - strcpy(command,"nowplayingartwork?mw=320&mh=320"); - debug(1,"Command: \"%s\", result is %d",command, dacp_send_command(command, &response, &le)); - if (response) { - free(response); - response = NULL; - } - strcpy(command,"getproperty?properties=dmcp.volume"); - debug(1,"Command: \"%s\", result is %d",command, dacp_send_command(command, &response, &le)); - if (response) { - free(response); - response = NULL; - } - strcpy(command,"setproperty?dmcp.volume=100.000000"); - debug(1,"Command: \"%s\", result is %d",command, dacp_send_command(command, &response, &le)); - if (response) { - free(response); - response = NULL; - } - */ - if (metadata_store.player_thread_active) - sleep(config.scan_interval_when_active); - else - sleep(config.scan_interval_when_inactive); + }; + /* + strcpy(command,"nowplayingartwork?mw=320&mh=320"); + debug(1,"Command: \"%s\", result is %d",command, dacp_send_command(command, &response, &le)); + if (response) { + free(response); + response = NULL; + } + strcpy(command,"getproperty?properties=dmcp.volume"); + debug(1,"Command: \"%s\", result is %d",command, dacp_send_command(command, &response, &le)); + if (response) { + free(response); + response = NULL; + } + strcpy(command,"setproperty?dmcp.volume=100.000000"); + debug(1,"Command: \"%s\", result is %d",command, dacp_send_command(command, &response, &le)); + if (response) { + free(response); + response = NULL; + } + */ + if (metadata_store.player_thread_active) + sleep(config.scan_interval_when_active); + else + sleep(config.scan_interval_when_inactive); } debug(1, "DACP monitor thread exiting -- should never happen."); pthread_exit(NULL); diff --git a/shairport-sync-dbus-test-client.c b/shairport-sync-dbus-test-client.c index 6b8f122c..11100234 100644 --- a/shairport-sync-dbus-test-client.c +++ b/shairport-sync-dbus-test-client.c @@ -65,8 +65,7 @@ void on_properties_changed(__attribute__((unused)) GDBusProxy *proxy, GVariant * } } -void notify_loudness_callback(ShairportSync *proxy, - __attribute__((unused)) gpointer user_data) { +void notify_loudness_callback(ShairportSync *proxy, __attribute__((unused)) gpointer user_data) { // printf("\"notify_loudness_callback\" called with a gpointer of // %lx.\n",(int64_t)user_data); gboolean ebl = shairport_sync_get_loudness(proxy); @@ -187,28 +186,44 @@ int main(int argc, char *argv[]) { g_signal_connect(proxy4, "notify::volume", G_CALLBACK(notify_volume_callback), "ShairportSync.AdvancedRemoteControl"); + + g_print("Starting test...\n"); + g_print("Using the RemoteControl interface, set airplay volume to -30, -20, -10, 0 and -15 for two seconds each...\n"); + shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), -30, NULL, NULL, 0); + sleep(2); + shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), -20, NULL, NULL, 0); + sleep(2); + shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), -10, NULL, NULL, 0); + sleep(2); + shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), 0, NULL, NULL, 0); + sleep(2); + shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), -15, NULL, NULL, 0); + sleep(2); + + g_print("Using the AdvancedRemoteControl interface, set airplay volume to 20%%, 100%%, 40%% and 60%% for two seconds each...\n"); shairport_sync_advanced_remote_control_call_set_volume( SHAIRPORT_SYNC_ADVANCED_REMOTE_CONTROL(proxy4), 20, NULL, NULL, 0); - sleep(5); + sleep(2); shairport_sync_advanced_remote_control_call_set_volume( SHAIRPORT_SYNC_ADVANCED_REMOTE_CONTROL(proxy4), 100, NULL, NULL, 0); - sleep(5); + sleep(2); shairport_sync_advanced_remote_control_call_set_volume( SHAIRPORT_SYNC_ADVANCED_REMOTE_CONTROL(proxy4), 40, NULL, NULL, 0); - sleep(5); + sleep(2); shairport_sync_advanced_remote_control_call_set_volume( SHAIRPORT_SYNC_ADVANCED_REMOTE_CONTROL(proxy4), 60, NULL, NULL, 0); - - sleep(5); - g_print("Volume up for five seconds...\n"); - shairport_sync_remote_control_call_volume_up(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), NULL, NULL, NULL); + sleep(2); + g_print("Using the RemoteControl interface, increase volume for five seconds...\n"); + shairport_sync_remote_control_call_volume_up(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), NULL, NULL, + NULL); sleep(5); - g_print("Volume down\n"); - shairport_sync_remote_control_call_volume_down(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), NULL, NULL, NULL); - + g_print("Using the RemoteControl interface, decrease volume...\n"); + shairport_sync_remote_control_call_volume_down(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), NULL, NULL, + NULL); + /* // sleep(1); shairport_sync_set_loudness_filter_active(SHAIRPORT_SYNC(proxy), TRUE); From 99d99b417af39127911b3314bd48d59cea58f430 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 10 Jan 2020 08:31:40 +0000 Subject: [PATCH 013/186] Remove timeout on reception of DACP responses to allow for long poll behaviour of some players, e.g. Forked Daapd. Add some examples to the dbus client. --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 7b5d83da..15556ea7 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.6d3], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.6d4], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 71e8da22a25cceb42e248da38289a349ff82e9e2 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 10 Jan 2020 08:47:54 +0000 Subject: [PATCH 014/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index bf94f4fd..48b226f8 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,11 @@ +Version 3.3.6d4 +==== +**Bug Fix** +* Fix an incompatibilty with Forked Daapd that was causing Forked Daapd to lock up. Thanks to [@tomgadow](https://github.com/tomgadow) and [@ejurgensen](https://github.com/ejurgensen) for their help in finding and (hopefully) fixing this issue. + +**Enhancements** +* Add examples of using the RemoteControl interface's SetAirplayVolume feature to the shairport-sync-dbus-test-client source. + Version 3.3.6d3 ==== **New Features** From 98a091420cdd163802375b59c0ab73212607522b Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 10 Jan 2020 08:51:42 +0000 Subject: [PATCH 015/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 48b226f8..b56e0b17 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,7 +1,7 @@ Version 3.3.6d4 ==== **Bug Fix** -* Fix an incompatibilty with Forked Daapd that was causing Forked Daapd to lock up. Thanks to [@tomgadow](https://github.com/tomgadow) and [@ejurgensen](https://github.com/ejurgensen) for their help in finding and (hopefully) fixing this issue. +* Fix an incompatibilty with Forked Daapd that was causing Forked Daapd to lock up. Thanks to [@tomgadow](https://github.com/tomgadow) and [@ejurgensen](https://github.com/ejurgensen) for their help in finding and (hopefully) fixing this issue. Addresses Issue [#953](https://github.com/mikebrady/shairport-sync/issues/953) and [Forked Daapd Issue #870](https://github.com/ejurgensen/forked-daapd/issues/870). **Enhancements** * Add examples of using the RemoteControl interface's SetAirplayVolume feature to the shairport-sync-dbus-test-client source. From 27e42a666aa4461a5f9620b65809b189ab95d981 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 10 Jan 2020 17:20:25 +0000 Subject: [PATCH 016/186] Store the UserAgent so as to recognise forked-daapd so as to always send a revision_number of 1 when asking for playerstatus of forked-daapd. --- dacp.c | 40 ++++++++++++++++++++++++++-------------- player.h | 1 + rtsp.c | 14 ++++++++++---- 3 files changed, 37 insertions(+), 18 deletions(-) diff --git a/dacp.c b/dacp.c index a47d1608..f3111a83 100644 --- a/dacp.c +++ b/dacp.c @@ -53,6 +53,7 @@ typedef struct { char dacp_id[256]; // the DACP ID string uint16_t port; // zero if no port discovered short connection_family; // AF_INET6 or AF_INET + int always_use_revision_number_1; // for dealing with forked-daapd; uint32_t scope_id; // if it's an ipv6 connection, this will be its scope id char ip_string[INET6_ADDRSTRLEN]; // the ip string pointing to the client uint32_t active_remote_id; // send this when you want to send remote control commands @@ -226,10 +227,8 @@ int dacp_send_command(const char *command, char **body, ssize_t *bodysize) { struct timeval tv; tv.tv_sec = 0; tv.tv_usec = 500000; - // Don't place a timeout on the receive side of these calls, as some clients may provide - // a "long poll" mechanism whereby a reply will not be sent until a change has occurred - // if (setsockopt(sockfd, SOL_SOCKET, SO_RCVTIMEO, (const char *)&tv, sizeof tv) == -1) - // debug(1, "dacp_send_command: error %d setting receive timeout.", errno); + if (setsockopt(sockfd, SOL_SOCKET, SO_RCVTIMEO, (const char *)&tv, sizeof tv) == -1) + debug(1, "dacp_send_command: error %d setting receive timeout.", errno); if (setsockopt(sockfd, SOL_SOCKET, SO_SNDTIMEO, (const char *)&tv, sizeof tv) == -1) debug(1, "dacp_send_command: error %d setting send timeout.", errno); @@ -284,12 +283,8 @@ int dacp_send_command(const char *command, char **body, ssize_t *bodysize) { memset(buffer, 0, sizeof(buffer)); while (needmore && !looperror) { const char *data = buffer; - - // Don't place a timeout on the receive, as some clients may provide - // a "long poll" mechanism whereby a reply will not be sent until a change has occurred - - // if (setsockopt(sockfd, SOL_SOCKET, SO_RCVTIMEO, (const char *)&tv, sizeof tv) == -1) - // debug(1, "dacp_send_command: error %d setting receive timeout.", errno); + if (setsockopt(sockfd, SOL_SOCKET, SO_RCVTIMEO, (const char *)&tv, sizeof tv) == -1) + debug(1, "dacp_send_command: error %d setting receive timeout.", errno); ssize_t ndata = recv(sockfd, buffer, sizeof(buffer), 0); // debug(3, "Received %d bytes: \"%s\".", ndata, buffer); if (ndata <= 0) { @@ -413,9 +408,21 @@ void set_dacp_server_information(rtsp_conn_info *conn) { dacp_server.connection_family = conn->connection_ip_family; dacp_server.scope_id = conn->self_scope_id; strncpy(dacp_server.ip_string, conn->client_ip_string, INET6_ADDRSTRLEN); - debug(3, "set_dacp_server_information set IP to \"%s\" and DACP id to \"%s\".", + debug(2, "set_dacp_server_information set IP to \"%s\" and DACP id to \"%s\".", dacp_server.ip_string, dacp_server.dacp_id); + // If the client is forked-daapd, then we always use revision number 1 + // because otherwise the return read will hang in a "long poll" if there + // are no changes. + // This is different to other AirPlay clients + // which return immediately with a 403 code if there are no changes. + dacp_server.always_use_revision_number_1 = 0; + char *p = strstr(conn->UserAgent, "forked-daapd"); + if ((p != 0) && (p == conn->UserAgent)) {// must exist and be at the start of the UserAgent string + dacp_server.always_use_revision_number_1 = 1; + } + + mdns_dacp_monitor_set_id(dacp_server.dacp_id); metadata_hub_modify_prolog(); @@ -480,6 +487,7 @@ void dacp_monitor_thread_code_cleanup(__attribute__((unused)) void *arg) { void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { int scan_index = 0; + int always_use_revision_number_1 = 0; // char server_reply[10000]; // debug(1, "DACP monitor thread started."); // wait until we get a valid port number to begin monitoring it @@ -516,6 +524,8 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { idle_scan_count = 0; } + always_use_revision_number_1 = dacp_server.always_use_revision_number_1; // set this while access is locked + result = dacp_get_volume(&the_volume); // just want the http code pthread_cleanup_pop(1); @@ -604,6 +614,8 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { char *response = NULL; int32_t item_size; char command[1024] = ""; + if (always_use_revision_number_1 != 0) // for forked-daapd + revision_number = 1; snprintf(command, sizeof(command) - 1, "playstatusupdate?revision-number=%d", revision_number); // debug(1,"dacp_monitor_thread_code: command: \"%s\"",command); @@ -641,7 +653,7 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { case 'cmsr': // revision number t = sp - item_size; revision_number = ntohl(*(uint32_t *)(t)); - // debug(1," Serial Number: %d", revision_number); + // debug(1,"New revision number received: %d", revision_number); break; case 'caps': // play status t = sp - item_size; @@ -1003,7 +1015,7 @@ int dacp_set_include_speaker_volume(int64_t machine_number, int32_t vo) { snprintf(message, sizeof(message), "setproperty?include-speaker-id=%" PRId64 "&dmcp.volume=%" PRId32 "", machine_number, vo); - debug(1, "sending \"%s\"", message); + debug(2, "sending \"%s\"", message); return send_simple_dacp_command(message); // should return 204 } @@ -1013,7 +1025,7 @@ int dacp_set_speaker_volume(int64_t machine_number, int32_t vo) { memset(message, 0, sizeof(message)); snprintf(message, sizeof(message), "setproperty?speaker-id=%" PRId64 "&dmcp.volume=%" PRId32 "", machine_number, vo); - debug(1, "sending \"%s\"", message); + debug(2, "sending \"%s\"", message); return send_simple_dacp_command(message); // should return 204 } diff --git a/player.h b/player.h index 9246f448..3be249dc 100644 --- a/player.h +++ b/player.h @@ -79,6 +79,7 @@ typedef struct { typedef struct { int connection_number; // for debug ID purposes, nothing else... int resend_interval; // this is really just for debugging + char *UserAgent; // free this on teardown int AirPlayVersion; // zero if not an AirPlay session. Used to help calculate latency uint32_t latency; // the actual latency used for this play session uint32_t minimum_latency; // set if an a=min-latency: line appears in the ANNOUNCE message; zero diff --git a/rtsp.c b/rtsp.c index 757bcd36..0bf68457 100644 --- a/rtsp.c +++ b/rtsp.c @@ -626,16 +626,16 @@ enum rtsp_read_request_response rtsp_read_request(rtsp_conn_info *conn, rtsp_mes goto shutdown; } -/* // this outputs the message received +/* // this outputs the message received { void *pt = malloc(nread+1); memset(pt, 0, nread+1); memcpy(pt, buf + inbuf, nread); debug(1, "Incoming string on port: \"%s\"",pt); - free(pt); + free(pt); } */ - + inbuf += nread; char *next; @@ -1889,7 +1889,7 @@ static void handle_announce(rtsp_conn_info *conn, rtsp_message *req, rtsp_messag unsigned int max_param = sizeof(conn->stream.fmtp) / sizeof(conn->stream.fmtp[0]); char* found; while ((found = strsep(&pfmtp, " \t")) != NULL && i < max_param) { - conn->stream.fmtp[i++] = atoi(found); + conn->stream.fmtp[i++] = atoi(found); } // here we should check the sanity of the fmtp values // for (i = 0; i < sizeof(conn->stream.fmtp) / sizeof(conn->stream.fmtp[0]); i++) @@ -1930,6 +1930,7 @@ static void handle_announce(rtsp_conn_info *conn, rtsp_message *req, rtsp_messag } hdr = msg_get_header(req, "User-Agent"); if (hdr) { + conn->UserAgent = strdup(hdr); debug(2, "Play connection from user agent \"%s\" on RTSP conversation thread %d.", hdr, conn->connection_number); // if the user agent is AirPlay and has a version number of 353 or less (from iOS 11.1,2) @@ -2267,6 +2268,11 @@ void rtsp_conversation_thread_cleanup_function(void *arg) { conn->dacp_id = NULL; } + if (conn->UserAgent) { + free(conn->UserAgent); + conn->UserAgent = NULL; + } + // remove flow control and mutexes int rc = pthread_mutex_destroy(&conn->volume_control_mutex); if (rc) From 249a6deff9d401778100a1576d1eddd3c1ac3551 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 10 Jan 2020 17:20:53 +0000 Subject: [PATCH 017/186] Bump copyright date. --- mpris-service.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/mpris-service.c b/mpris-service.c index 387b3a23..168ce396 100644 --- a/mpris-service.c +++ b/mpris-service.c @@ -1,6 +1,6 @@ /* * This file is part of Shairport Sync. - * Copyright (c) Mike Brady 2018 -- 2019 + * Copyright (c) Mike Brady 2018 -- 2020 * All rights reserved. * * Permission is hereby granted, free of charge, to any person @@ -330,7 +330,7 @@ static void on_mpris_name_acquired(GDBusConnection *connection, const gchar *nam NULL); g_signal_connect(mprisPlayerPlayerSkeleton, "handle-set-volume", G_CALLBACK(on_handle_set_volume), NULL); - + add_metadata_watcher(mpris_metadata_watcher, NULL); From ec665d9be45f8d655da1e444d8514a2da4e27106 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 10 Jan 2020 17:21:55 +0000 Subject: [PATCH 018/186] Change from talking about a server to talking about a client. Technically, e.g. iTunes is a client of Shairport Sync. --- dbus-service.c | 34 ++++++++++++++++++---------------- org.gnome.ShairportSync.xml | 4 ++-- 2 files changed, 20 insertions(+), 18 deletions(-) diff --git a/dbus-service.c b/dbus-service.c index d9d6e95d..3efde8f1 100644 --- a/dbus-service.c +++ b/dbus-service.c @@ -56,7 +56,7 @@ guint ownerID = 0; void dbus_metadata_watcher(struct metadata_bundle *argc, __attribute__((unused)) void *userdata) { char response[100]; gboolean current_status, new_status; - + const char *th; shairport_sync_advanced_remote_control_set_volume(shairportSyncAdvancedRemoteControlSkeleton, argc->speaker_volume); @@ -64,8 +64,10 @@ void dbus_metadata_watcher(struct metadata_bundle *argc, __attribute__((unused)) shairport_sync_remote_control_set_airplay_volume(shairportSyncRemoteControlSkeleton, argc->airplay_volume); - shairport_sync_remote_control_set_server(shairportSyncRemoteControlSkeleton, argc->client_ip); + shairport_sync_remote_control_set_client(shairportSyncRemoteControlSkeleton, argc->client_ip); + + // although it's a DACP server, the server is in fact, part of the the AirPlay "client" (their term). if (argc->dacp_server_active) { shairport_sync_remote_control_set_available(shairportSyncRemoteControlSkeleton, TRUE); } else { @@ -161,7 +163,7 @@ void dbus_metadata_watcher(struct metadata_bundle *argc, __attribute__((unused)) shairportSyncAdvancedRemoteControlSkeleton, response); } - + switch (argc->shuffle_status) { case SS_NOT_AVAILABLE: new_status = FALSE; @@ -176,10 +178,10 @@ void dbus_metadata_watcher(struct metadata_bundle *argc, __attribute__((unused)) new_status = FALSE; debug(1, "Unknown shuffle status -- this should never happen."); } - + current_status = shairport_sync_advanced_remote_control_get_shuffle( shairportSyncAdvancedRemoteControlSkeleton); - + // only set this if it's different if (current_status != new_status) { debug(3, "Shuffle State should be changed"); @@ -243,7 +245,7 @@ void dbus_metadata_watcher(struct metadata_bundle *argc, __attribute__((unused)) // debug(1, "Set tracklength to %lu.", track_length_in_microseconds); GVariant *tracklength = g_variant_new("x", track_length_in_microseconds); g_variant_builder_add(dict_builder, "{sv}", "mpris:length", tracklength); - } + } GVariant *dict = g_variant_builder_end(dict_builder); g_variant_builder_unref(dict_builder); @@ -753,16 +755,16 @@ static gboolean on_handle_quit(ShairportSync *skeleton, GDBusMethodInvocation *i static gboolean on_handle_remote_command(ShairportSync *skeleton, GDBusMethodInvocation *invocation, const gchar *command, __attribute__((unused)) gpointer user_data) { - debug(1, "RemoteCommand with command \"%s\".", command); + debug(1, "RemoteCommand with command \"%s\".", command); int reply = 0; - char *server_reply = NULL; + char *client_reply = NULL; ssize_t reply_size = 0; - reply = dacp_send_command((const char *)command, &server_reply, &reply_size); - char *server_reply_hex = alloca(reply_size * 2 + 1); - if (server_reply_hex) { - char *p = server_reply_hex; - if (server_reply) { - char *q = server_reply; + reply = dacp_send_command((const char *)command, &client_reply, &reply_size); + char *client_reply_hex = alloca(reply_size * 2 + 1); + if (client_reply_hex) { + char *p = client_reply_hex; + if (client_reply) { + char *q = client_reply; int i; for (i = 0; i < reply_size; i++) { snprintf(p, 3, "%02X", *q); @@ -772,7 +774,7 @@ static gboolean on_handle_remote_command(ShairportSync *skeleton, GDBusMethodInv } *p = '\0'; } - shairport_sync_complete_remote_command(skeleton, invocation, reply, server_reply_hex); + shairport_sync_complete_remote_command(skeleton, invocation, reply, client_reply_hex); return TRUE; } @@ -974,7 +976,7 @@ static void on_dbus_name_acquired(GDBusConnection *connection, const gchar *name // else // shairport_sync_set_convolution_impulse_response_file(SHAIRPORT_SYNC(shairportSyncSkeleton), NULL); #endif - + shairport_sync_set_version(SHAIRPORT_SYNC(shairportSyncSkeleton), PACKAGE_VERSION); char *vs = get_version_string(); shairport_sync_set_version_string(SHAIRPORT_SYNC(shairportSyncSkeleton), vs); diff --git a/org.gnome.ShairportSync.xml b/org.gnome.ShairportSync.xml index 91c31671..10eb5477 100644 --- a/org.gnome.ShairportSync.xml +++ b/org.gnome.ShairportSync.xml @@ -13,7 +13,7 @@ - + @@ -45,7 +45,7 @@ - + From c75311c934d888c4272a1ab7881d953558eed8a8 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 10 Jan 2020 17:22:28 +0000 Subject: [PATCH 019/186] Add a few more commands and clarify some text. --- shairport-sync-dbus-test-client.c | 44 ++++++++++++++++++++++--------- 1 file changed, 32 insertions(+), 12 deletions(-) diff --git a/shairport-sync-dbus-test-client.c b/shairport-sync-dbus-test-client.c index 11100234..d0b15bb2 100644 --- a/shairport-sync-dbus-test-client.c +++ b/shairport-sync-dbus-test-client.c @@ -190,32 +190,51 @@ int main(int argc, char *argv[]) { g_print("Starting test...\n"); - g_print("Using the RemoteControl interface, set airplay volume to -30, -20, -10, 0 and -15 for two seconds each...\n"); + g_print("Using the RemoteControl interface, play for five seconds, pause for five seconds and then resume play...\n"); + g_print("Play...\n"); + shairport_sync_remote_control_call_play(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), NULL, NULL, 0); + sleep(5); + g_print("Pause...\n"); + shairport_sync_remote_control_call_pause(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), NULL, NULL, 0); + sleep(5); + g_print("Play...\n"); + shairport_sync_remote_control_call_play(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), NULL, NULL, 0); + sleep(5); + g_print("Using the RemoteControl interface, set AirPlay Volume (range -30 to 0) to -30, -20, -10, 0 and -15 for five seconds each...\n"); + g_print("Set AirPlay Volume (range -30 to 0) to -30\n"); shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), -30, NULL, NULL, 0); - sleep(2); + sleep(5); + g_print("Set AirPlay Volume (range -30 to 0) to -20\n"); shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), -20, NULL, NULL, 0); - sleep(2); + sleep(5); + g_print("Set AirPlay Volume (range -30 to 0) to -10\n"); shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), -10, NULL, NULL, 0); - sleep(2); + sleep(5); + g_print("Set AirPlay Volume (range -30 to 0) to -0\n"); shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), 0, NULL, NULL, 0); - sleep(2); + sleep(5); + g_print("Set AirPlay Volume (range -30 to 0) to -15\n"); shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), -15, NULL, NULL, 0); - sleep(2); + sleep(5); - g_print("Using the AdvancedRemoteControl interface, set airplay volume to 20%%, 100%%, 40%% and 60%% for two seconds each...\n"); + g_print("Using the AdvancedRemoteControl interface, set Volume to 20%%, 100%%, 40%% and 60%% for five seconds each...\n"); + g_print("Set Volume to 20%%\n"); shairport_sync_advanced_remote_control_call_set_volume( SHAIRPORT_SYNC_ADVANCED_REMOTE_CONTROL(proxy4), 20, NULL, NULL, 0); - sleep(2); + sleep(5); + g_print("Set Volume to 100%%\n"); shairport_sync_advanced_remote_control_call_set_volume( SHAIRPORT_SYNC_ADVANCED_REMOTE_CONTROL(proxy4), 100, NULL, NULL, 0); - sleep(2); + sleep(5); + g_print("Set Volume to 40%%\n"); shairport_sync_advanced_remote_control_call_set_volume( SHAIRPORT_SYNC_ADVANCED_REMOTE_CONTROL(proxy4), 40, NULL, NULL, 0); - sleep(2); + sleep(5); + g_print("Set Volume to 50%%\n"); shairport_sync_advanced_remote_control_call_set_volume( - SHAIRPORT_SYNC_ADVANCED_REMOTE_CONTROL(proxy4), 60, NULL, NULL, 0); + SHAIRPORT_SYNC_ADVANCED_REMOTE_CONTROL(proxy4), 50, NULL, NULL, 0); - sleep(2); + sleep(5); g_print("Using the RemoteControl interface, increase volume for five seconds...\n"); shairport_sync_remote_control_call_volume_up(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), NULL, NULL, NULL); @@ -241,6 +260,7 @@ int main(int argc, char *argv[]) { shairport_sync_call_remote_command(SHAIRPORT_SYNC(proxy), "string",NULL,NULL,NULL); */ + sleep(1); g_print("Finished test. Listening for property changes...\n"); // g_main_loop_quit(loop); pthread_join(dbus_thread, NULL); From a7018e4cb8ad6ea4d780cb14cb92e2307e8a305f Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 10 Jan 2020 17:23:31 +0000 Subject: [PATCH 020/186] Use a different workaround for forked-daapd remote access and clean up a few small issues. --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 15556ea7..03177b7f 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.6d4], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.6d5], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From adbba7de161c50416fa66808404b559589c729a7 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 10 Jan 2020 17:44:33 +0000 Subject: [PATCH 021/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index b56e0b17..abcaf344 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,10 +1,17 @@ -Version 3.3.6d4 +Version 3.3.6d5 ==== **Bug Fix** * Fix an incompatibilty with Forked Daapd that was causing Forked Daapd to lock up. Thanks to [@tomgadow](https://github.com/tomgadow) and [@ejurgensen](https://github.com/ejurgensen) for their help in finding and (hopefully) fixing this issue. Addresses Issue [#953](https://github.com/mikebrady/shairport-sync/issues/953) and [Forked Daapd Issue #870](https://github.com/ejurgensen/forked-daapd/issues/870). **Enhancements** -* Add examples of using the RemoteControl interface's SetAirplayVolume feature to the shairport-sync-dbus-test-client source. +* Add some extra examples of using the RemoteControl interface's SetAirplayVolume feature to the shairport-sync-dbus-test-client source. + +**Pesky Changes You May Not Be Able To Ignore** +* Renamed the `Server` property in the D-Bus interface to `Client`. The rather strange language in use has it that a player like iTunes is a "client" of the AirPlay device like Shairport Sync, which is therefore considered the "server". The newly-renamed "Client" property is the IP number of the player. + +Version 3.3.6d4 +==== +A fix for issue #953 that wouldn't work. Version 3.3.6d3 ==== From 400bd752272fd3d53cee59599c2b1bd962c57f6f Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 26 Jan 2020 12:39:36 +0000 Subject: [PATCH 022/186] Update and rename CONTRIBUTING.md to REPORTING ISSUES.md --- CONTRIBUTING.md => REPORTING ISSUES.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) rename CONTRIBUTING.md => REPORTING ISSUES.md (97%) diff --git a/CONTRIBUTING.md b/REPORTING ISSUES.md similarity index 97% rename from CONTRIBUTING.md rename to REPORTING ISSUES.md index 2b879f79..5c502091 100644 --- a/CONTRIBUTING.md +++ b/REPORTING ISSUES.md @@ -1,5 +1,5 @@ -Contributing to Shairport Sync +Reporting Issues and Making Pull Requests ==== Pull Requests From 537158571f97eefaf399331d55daef45b8226cfe Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 11 Feb 2020 22:44:07 +0000 Subject: [PATCH 023/186] Add a SIGCHLD handler to reap zombie processes generated after script invocations where wait_for_completion is set to "no". --- shairport.c | 46 ++++++++++++++++++++++++++++++++++------------ 1 file changed, 34 insertions(+), 12 deletions(-) diff --git a/shairport.c b/shairport.c index d2a1a494..5e375916 100644 --- a/shairport.c +++ b/shairport.c @@ -170,7 +170,7 @@ void *soxr_time_check(__attribute__((unused)) void *arg) { outbuffer, buffer_length + 1, &odone, // Output. &io_spec, // Input, output and transfer spec. NULL, NULL); // Default configuration. - + io_spec.itype = SOXR_INT32_I; io_spec.otype = SOXR_INT32_I; io_spec.scale = 1.0; // this seems to crash if not = 1.0 @@ -766,8 +766,8 @@ int parse_options(int argc, char **argv) { } else die("Invalid alac_decoder option choice \"%s\". It should be \"hammerton\" or \"apple\""); } - - + + /* Get the resend control settings. */ if (config_lookup_float(config.cfg, "general.resend_control_first_check_time", &dvalue)) { @@ -1197,14 +1197,14 @@ int parse_options(int argc, char **argv) { char hostname[100]; gethostname(hostname, 100); - - + + char *i0; if (raw_service_name == NULL) i0 = strdup("%H"); // this is the default it the Service Name wasn't specified else i0 = strdup(raw_service_name); - + // here, do the substitutions for %h, %H, %v and %V char *i1 = str_replace(i0, "%h", hostname); if ((hostname[0] >= 'a') && (hostname[0] <= 'z')) @@ -1284,7 +1284,7 @@ void exit_function() { Actually, there is no terminate_mqtt() function. #ifdef CONFIG_MQTT if (config.mqtt_enabled) { - terminate_mqtt(); + terminate_mqtt(); } #endif */ @@ -1333,14 +1333,14 @@ Actually, there is no stop_mpris_service() function. pthread_join(soxr_time_check_thread, NULL); #endif - + if (conns) free(conns); // make sure the connections have been deleted first if (config.service_name) free(config.service_name); - -#ifdef CONFIG_CONVOLUTION + +#ifdef CONFIG_CONVOLUTION if (config.convolution_ir_file) free(config.convolution_ir_file); #endif @@ -1364,6 +1364,16 @@ Actually, there is no stop_mpris_service() function. // probably should be freeing malloc'ed memory here, including strdup-created strings... } +// for removing zombie script processes +// see: http://www.microhowto.info/howto/reap_zombie_processes_using_a_sigchld_handler.html +// used with thanks. + +void handle_sigchld(__attribute__((unused)) int sig) { + int saved_errno = errno; + while (waitpid((pid_t)(-1), 0, WNOHANG) > 0) {} + errno = saved_errno; +} + int main(int argc, char **argv) { /* Check if we are called with -V or --version parameter */ if (argc >= 2 && ((strcmp(argv[1], "-V") == 0) || (strcmp(argv[1], "--version") == 0))) { @@ -1581,8 +1591,8 @@ int main(int argc, char **argv) { } return ret; } else { /* pid == 0 means we are the daemon */ - - this_is_the_daemon_process = 1; // + + this_is_the_daemon_process = 1; // /* Close FDs */ if (daemon_close_all(-1) < 0) { @@ -1624,6 +1634,18 @@ int main(int argc, char **argv) { #endif debug(1, "Started!"); + + // install a zombie process reaper + // see: http://www.microhowto.info/howto/reap_zombie_processes_using_a_sigchld_handler.html + struct sigaction sa; + sa.sa_handler = &handle_sigchld; + sigemptyset(&sa.sa_mask); + sa.sa_flags = SA_RESTART | SA_NOCLDSTOP; + if (sigaction(SIGCHLD, &sa, 0) == -1) { + perror(0); + exit(1); + } + main_thread_id = pthread_self(); if (!main_thread_id) debug(1, "Main thread is set up to be NULL!"); From 17759c97edd35f5464763ca0b89a0a29a6a265e1 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 11 Feb 2020 22:44:50 +0000 Subject: [PATCH 024/186] Add a SIGCHLD handler to reap zombie processes generated after script invocations where wait_for_completion is set to "no". --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 03177b7f..a47a248d 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.6d5], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.6d6], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 89af8e88c1d8b1e109b7b8a1051496000d47fa7e Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 11 Feb 2020 22:49:56 +0000 Subject: [PATCH 025/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index abcaf344..c452c22a 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,8 @@ +Version 3.3.6d6 +==== +**Bug Fix** +* Add a SIGCHLD handler to remove completed processes that were used to perform and program "hooks" without waiting for completion. Thanks to [patrickjane](https://github.com/patrickjane) for reporting the bug and for suggesting a solution. Addresses Issue [#968](https://github.com/mikebrady/shairport-sync/issues/968). + Version 3.3.6d5 ==== **Bug Fix** From f7717745198fc4748d9d646cfbdfc90adb668801 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 13 Feb 2020 17:37:41 +0000 Subject: [PATCH 026/186] Compiles under gcc-10 with -fno-common --- audio_alsa.c | 20 ++++++++++---------- common.c | 8 ++++++-- common.h | 52 ++++++++++++++++++++++++++-------------------------- mdns.h | 2 +- player.c | 6 +++--- player.h | 6 +++--- rtsp.c | 3 ++- rtsp.h | 4 ++-- 8 files changed, 53 insertions(+), 48 deletions(-) diff --git a/audio_alsa.c b/audio_alsa.c index 27e35ed8..39e4d051 100644 --- a/audio_alsa.c +++ b/audio_alsa.c @@ -123,7 +123,7 @@ int frame_size; // in bytes for interleaved stereo int alsa_device_initialised; // boolean to ensure the initialisation is only // done once -enum yndk_type precision_delay_available_status = +yndk_type precision_delay_available_status = YNDK_DONT_KNOW; // initially, we don't know if the device can do precision delay snd_pcm_t *alsa_handle = NULL; @@ -159,14 +159,14 @@ int volume_based_mute_is_active = snd_pcm_sframes_t (*alsa_pcm_write)(snd_pcm_t *, const void *, snd_pcm_uframes_t) = snd_pcm_writei; int precision_delay_and_status(snd_pcm_state_t *state, snd_pcm_sframes_t *delay, - enum yndk_type *using_update_timestamps); + yndk_type *using_update_timestamps); int standard_delay_and_status(snd_pcm_state_t *state, snd_pcm_sframes_t *delay, - enum yndk_type *using_update_timestamps); + yndk_type *using_update_timestamps); // use this to allow the use of standard or precision delay calculations, with standard the, uh, // standard. int (*delay_and_status)(snd_pcm_state_t *state, snd_pcm_sframes_t *delay, - enum yndk_type *using_update_timestamps) = standard_delay_and_status; + yndk_type *using_update_timestamps) = standard_delay_and_status; // this will return true if the DAC can return precision delay information and false if not // if it is not yet known, it will test the output device to find out @@ -216,7 +216,7 @@ int precision_delay_available() { do_play(silence, frames_of_silence); pthread_cleanup_pop(1); // now we can get the delay, and we'll note if it uses update timestamps - enum yndk_type uses_update_timestamps; + yndk_type uses_update_timestamps; snd_pcm_state_t state; snd_pcm_sframes_t delay; int ret = precision_delay_and_status(&state, &delay, &uses_update_timestamps); @@ -392,7 +392,7 @@ format_record fr[] = { // be added at the lowest possible level. // Hence, selecting the greatest bit depth is always either beneficial or neutral. -enum sps_format_t auto_format_check_sequence[] = { +sps_format_t auto_format_check_sequence[] = { SPS_FORMAT_S32, SPS_FORMAT_S32_LE, SPS_FORMAT_S32_BE, SPS_FORMAT_S24, SPS_FORMAT_S24_LE, SPS_FORMAT_S24_BE, SPS_FORMAT_S24_3LE, SPS_FORMAT_S24_3BE, SPS_FORMAT_S16, SPS_FORMAT_S16_LE, SPS_FORMAT_S16_BE, SPS_FORMAT_S8, SPS_FORMAT_U8, @@ -508,12 +508,12 @@ int actual_open_alsa_device(int do_auto_setup) { } } else { // auto format int number_of_formats_to_try; - enum sps_format_t *formats; + sps_format_t *formats; formats = auto_format_check_sequence; number_of_formats_to_try = sizeof(auto_format_check_sequence) / sizeof(sps_format_t); int i = 0; int format_found = 0; - enum sps_format_t trial_format = SPS_FORMAT_UNKNOWN; + sps_format_t trial_format = SPS_FORMAT_UNKNOWN; while ((i < number_of_formats_to_try) && (format_found == 0)) { trial_format = formats[i]; sf = fr[trial_format].alsa_code; @@ -1422,7 +1422,7 @@ static void start(__attribute__((unused)) int i_sample_rate, } int standard_delay_and_status(snd_pcm_state_t *state, snd_pcm_sframes_t *delay, - enum yndk_type *using_update_timestamps) { + yndk_type *using_update_timestamps) { int ret = 0; if (using_update_timestamps) *using_update_timestamps = YNDK_NO; @@ -1444,7 +1444,7 @@ int standard_delay_and_status(snd_pcm_state_t *state, snd_pcm_sframes_t *delay, } int precision_delay_and_status(snd_pcm_state_t *state, snd_pcm_sframes_t *delay, - enum yndk_type *using_update_timestamps) { + yndk_type *using_update_timestamps) { snd_pcm_status_t *alsa_snd_pcm_status; snd_pcm_status_alloca(&alsa_snd_pcm_status); diff --git a/common.c b/common.c index ff643ef3..3d41cb6c 100644 --- a/common.c +++ b/common.c @@ -87,6 +87,10 @@ void set_alsa_out_dev(char *); #endif +config_t config_file_stuff; +pthread_t main_thread_id; +uint64_t fp_time_at_startup, fp_time_at_last_debug_message; + // always lock use this when accessing the fp_time_at_last_debug_message static pthread_mutex_t debug_timing_lock = PTHREAD_MUTEX_INITIALIZER; @@ -96,7 +100,7 @@ const char *sps_format_description_string_array[] = { "unknown", "S8", "U8", "S16", "S16_LE", "S16_BE", "S24", "S24_LE", "S24_BE", "S24_3LE", "S24_3BE", "S32", "S32_LE", "S32_BE", "auto", "invalid"}; -const char *sps_format_description_string(enum sps_format_t format) { +const char *sps_format_description_string(sps_format_t format) { if ((format >= SPS_FORMAT_UNKNOWN) && (format <= SPS_FORMAT_AUTO)) return sps_format_description_string_array[format]; else @@ -1381,7 +1385,7 @@ char *get_version_string() { return version_string; } -int64_t generate_zero_frames(char *outp, size_t number_of_frames, enum sps_format_t format, +int64_t generate_zero_frames(char *outp, size_t number_of_frames, sps_format_t format, int with_dither, int64_t random_number_in) { // return the last random number used // assuming the buffer has been assigned diff --git a/common.h b/common.h index 37217ab8..35b21bd9 100644 --- a/common.h +++ b/common.h @@ -34,24 +34,24 @@ enum dbus_session_type { #define sps_extra_code_output_state_cannot_make_ready 32769 // yeah/no/auto -enum yna_type { YNA_AUTO = -1, YNA_NO = 0, YNA_YES = 1 } yna_type; +typedef enum { YNA_AUTO = -1, YNA_NO = 0, YNA_YES = 1 } yna_type; // yeah/no/dont-care -enum yndk_type { YNDK_DONT_KNOW = -1, YNDK_NO = 0, YNDK_YES = 1 } yndk_type; +typedef enum { YNDK_DONT_KNOW = -1, YNDK_NO = 0, YNDK_YES = 1 } yndk_type; -enum endian_type { +typedef enum { SS_LITTLE_ENDIAN = 0, SS_PDP_ENDIAN, SS_BIG_ENDIAN, } endian_type; -enum stuffing_type { +typedef enum { ST_basic = 0, // straight deletion or insertion of a frame in a 352-frame packet ST_soxr, // use libsoxr to make a 352 frame packet one frame longer or shorter ST_auto, // use soxr if compiled for it and if the soxr_index is low enough -} s_type; +} stuffing_type; -enum playback_mode_type { +typedef enum { ST_stereo = 0, ST_mono, ST_reverse_stereo, @@ -59,26 +59,26 @@ enum playback_mode_type { ST_right_only, } playback_mode_type; -enum volume_control_profile_type { +typedef enum { VCP_standard = 0, VCP_flat, } volume_control_profile_type; -enum decoders_supported_type { +typedef enum { decoder_hammerton = 0, decoder_apple_alac, } decoders_supported_type; -enum disable_standby_mode_type { +typedef enum { disable_standby_off = 0, disable_standby_auto, disable_standby_always -}; +} disable_standby_mode_type; // the following enum is for the formats recognised -- currently only S16LE is recognised for input, // so these are output only for the present -enum sps_format_t { +typedef enum { SPS_FORMAT_UNKNOWN = 0, SPS_FORMAT_S8, SPS_FORMAT_U8, @@ -97,7 +97,7 @@ enum sps_format_t { SPS_FORMAT_INVALID, } sps_format_t; -const char *sps_format_description_string(enum sps_format_t format); +const char *sps_format_description_string(sps_format_t format); typedef struct { double resend_control_first_check_time; // wait this long before asking for a missing packet to be resent @@ -182,12 +182,12 @@ typedef struct { int debugger_show_relative_time; // in the debug message, display the time since the last one int debugger_show_file_and_line; // in the debug message, display the filename and line number int statistics_requested, use_negotiated_latencies; - enum playback_mode_type playback_mode; + playback_mode_type playback_mode; char *cmd_start, *cmd_stop, *cmd_set_volume, *cmd_unfixable; char *cmd_active_start, *cmd_active_stop; int cmd_blocking, cmd_start_returns_output; double tolerance; // allow this much drift before attempting to correct it - enum stuffing_type packet_stuffing; + stuffing_type packet_stuffing; int soxr_delay_index; int soxr_delay_threshold; // the soxr delay must be less or equal to this for soxr interpolation // to be enabled under the auto setting @@ -221,10 +221,10 @@ typedef struct { // attenuators, lowering the volume, use all the hw attenuation // before using // sw attenuation - enum volume_control_profile_type volume_control_profile; + volume_control_profile_type volume_control_profile; int output_format_auto_requested; // true if the configuration requests auto configuration - enum sps_format_t output_format; + sps_format_t output_format; int output_rate_auto_requested; // true if the configuration requests auto configuration unsigned int output_rate; @@ -240,9 +240,9 @@ typedef struct { float loudness_reference_volume_db; int alsa_use_hardware_mute; double alsa_maximum_stall_time; - enum disable_standby_mode_type disable_standby_mode; + disable_standby_mode_type disable_standby_mode; volatile int keep_dac_busy; - enum yna_type use_precision_timing; // defaults to no + yna_type use_precision_timing; // defaults to no #if defined(CONFIG_DBUS_INTERFACE) enum dbus_session_type dbus_service_bus_type; @@ -321,7 +321,7 @@ int64_t r64i(); void resetFreeUDPPort(); uint16_t nextFreeUDPPort(); -volatile int debuglev; +extern volatile int debuglev; void _die(const char *filename, const int linenumber, const char *format, ...); void _warn(const char *filename, const int linenumber, const char *format, ...); @@ -352,17 +352,17 @@ double vol2attn(double vol, long max_db, long min_db); uint64_t get_absolute_time_in_fp(void); // time at startup for debugging timing -uint64_t fp_time_at_startup, fp_time_at_last_debug_message; +extern uint64_t fp_time_at_startup, fp_time_at_last_debug_message; // this is for reading an unsigned 32 bit number, such as an RTP timestamp uint32_t uatoi(const char *nptr); // this is for allowing us to cancel the whole program -pthread_t main_thread_id; +extern pthread_t main_thread_id; -shairport_cfg config; -config_t config_file_stuff; +extern shairport_cfg config; +extern config_t config_file_stuff; int config_set_lookup_bool(config_t *cfg, char *where, int *dst); @@ -377,7 +377,7 @@ void shairport_shutdown(); extern sigset_t pselect_sigset; -pthread_mutex_t the_conn_lock; +extern pthread_mutex_t the_conn_lock; #define conn_lock(arg) \ pthread_mutex_lock(&the_conn_lock); \ @@ -412,7 +412,7 @@ void pthread_cleanup_debug_mutex_unlock(void *arg); #define config_unlock pthread_mutex_unlock(&config.lock) -pthread_mutex_t r64_mutex; +extern pthread_mutex_t r64_mutex; #define r64_lock pthread_mutex_lock(&r64_mutex) @@ -423,7 +423,7 @@ char *get_version_string(); // mallocs a string space -- remember to free it aft void sps_nanosleep(const time_t sec, const long nanosec); // waits for this time, even through interruptions -int64_t generate_zero_frames(char *outp, size_t number_of_frames, enum sps_format_t format, +int64_t generate_zero_frames(char *outp, size_t number_of_frames, sps_format_t format, int with_dither, int64_t random_number_in); void malloc_cleanup(void *arg); diff --git a/mdns.h b/mdns.h index 8dc919f2..b69f4c85 100644 --- a/mdns.h +++ b/mdns.h @@ -2,7 +2,7 @@ #define _MDNS_H #include "config.h" -#include +#include "player.h" #include extern int mdns_pid; diff --git a/player.c b/player.c index 21eb432f..5876bd33 100644 --- a/player.c +++ b/player.c @@ -660,7 +660,7 @@ int32_t rand_in_range(int32_t exclusive_range_limit) { return sp >> 32; } -static inline void process_sample(int32_t sample, char **outp, enum sps_format_t format, int volume, +static inline void process_sample(int32_t sample, char **outp, sps_format_t format, int volume, int dither, rtsp_conn_info *conn) { /* { @@ -1392,7 +1392,7 @@ static inline int32_t mean_32(int32_t a, int32_t b) { // formats accepted so far include U8, S8, S16, S24, S24_3LE, S24_3BE and S32 // stuff: 1 means add 1; 0 means do nothing; -1 means remove 1 -static int stuff_buffer_basic_32(int32_t *inptr, int length, enum sps_format_t l_output_format, +static int stuff_buffer_basic_32(int32_t *inptr, int length, sps_format_t l_output_format, char *outptr, int stuff, int dither, rtsp_conn_info *conn) { int tstuff = stuff; char *l_outptr = outptr; @@ -1458,7 +1458,7 @@ double longest_soxr_execution_time_us = 0.0; int64_t packets_processed = 0; int stuff_buffer_soxr_32(int32_t *inptr, int32_t *scratchBuffer, int length, - enum sps_format_t l_output_format, char *outptr, int stuff, int dither, + sps_format_t l_output_format, char *outptr, int stuff, int dither, rtsp_conn_info *conn) { if (scratchBuffer == NULL) { die("soxr scratchBuffer not initialised."); diff --git a/player.h b/player.h index 3be249dc..b6cae8a9 100644 --- a/player.h +++ b/player.h @@ -63,17 +63,17 @@ typedef struct audio_buffer_entry { // decoded audio packets #define BUFFER_FRAMES 1024 -enum audio_stream_type { +typedef enum { ast_unknown, ast_uncompressed, // L16/44100/2 ast_apple_lossless, -} ast_type; +} audio_stream_type; typedef struct { int encrypted; uint8_t aesiv[16], aeskey[16]; int32_t fmtp[12]; - enum audio_stream_type type; + audio_stream_type type; } stream_cfg; typedef struct { diff --git a/rtsp.c b/rtsp.c index 0bf68457..6cf53927 100644 --- a/rtsp.c +++ b/rtsp.c @@ -89,7 +89,8 @@ enum rtsp_read_request_response { rtsp_read_request_response_error }; -// Mike Brady's part... +rtsp_conn_info *playing_conn; +rtsp_conn_info **conns; int metadata_running = 0; diff --git a/rtsp.h b/rtsp.h index 71e5b22c..703c1006 100644 --- a/rtsp.h +++ b/rtsp.h @@ -3,8 +3,8 @@ #include "player.h" -rtsp_conn_info *playing_conn; -rtsp_conn_info **conns; +extern rtsp_conn_info *playing_conn; +extern rtsp_conn_info **conns; void rtsp_listen_loop(void); // void rtsp_shutdown_stream(void); From 85ba78b61989670b00c9fb33501d606aa4e7a25f Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 13 Feb 2020 17:57:30 +0000 Subject: [PATCH 027/186] More fixes to allow gcc-10 to compile without error --- audio_sndio.c | 2 +- common.h | 8 ++++---- dbus-service.c | 2 ++ dbus-service.h | 2 +- metadata_hub.c | 2 ++ metadata_hub.h | 20 ++++++++++---------- mpris-service.c | 3 +++ mpris-service.h | 4 ++-- 8 files changed, 25 insertions(+), 18 deletions(-) diff --git a/audio_sndio.c b/audio_sndio.c index 9fd497ac..ac5cfa38 100644 --- a/audio_sndio.c +++ b/audio_sndio.c @@ -65,7 +65,7 @@ struct sio_par par; struct sndio_formats { const char *name; - enum sps_format_t fmt; + sps_format_t fmt; unsigned int rate; unsigned int bits; unsigned int bps; diff --git a/common.h b/common.h index 35b21bd9..2c9c05bd 100644 --- a/common.h +++ b/common.h @@ -24,10 +24,10 @@ #endif #if defined(CONFIG_DBUS_INTERFACE) || defined(CONFIG_MPRIS_INTERFACE) -enum dbus_session_type { +typedef enum { DBT_system = 0, // use the session bus DBT_session, // use the system bus -} dbt_type; +} dbus_session_type; #endif #define sps_extra_code_output_stalled 32768 @@ -245,10 +245,10 @@ typedef struct { yna_type use_precision_timing; // defaults to no #if defined(CONFIG_DBUS_INTERFACE) - enum dbus_session_type dbus_service_bus_type; + dbus_session_type dbus_service_bus_type; #endif #if defined(CONFIG_MPRIS_INTERFACE) - enum dbus_session_type mpris_service_bus_type; + dbus_session_type mpris_service_bus_type; #endif #ifdef CONFIG_METADATA_HUB diff --git a/dbus-service.c b/dbus-service.c index 3efde8f1..1fe59c35 100644 --- a/dbus-service.c +++ b/dbus-service.c @@ -45,6 +45,8 @@ #include #endif +ShairportSync *shairportSyncSkeleton; + int service_is_running = 0; ShairportSyncDiagnostics *shairportSyncDiagnosticsSkeleton = NULL; diff --git a/dbus-service.h b/dbus-service.h index 5cfc02e2..1373e9cb 100644 --- a/dbus-service.h +++ b/dbus-service.h @@ -3,7 +3,7 @@ #define DBUS_SERVICE_H #include "dbus-interface.h" -ShairportSync *shairportSyncSkeleton; +extern ShairportSync *shairportSyncSkeleton; int start_dbus_service(); void stop_dbus_service(); diff --git a/metadata_hub.c b/metadata_hub.c index fb459b6b..c09759f2 100644 --- a/metadata_hub.c +++ b/metadata_hub.c @@ -58,6 +58,8 @@ #include #endif +struct metadata_bundle metadata_store; + int metadata_hub_initialised = 0; pthread_rwlock_t metadata_hub_re_lock = PTHREAD_RWLOCK_INITIALIZER; diff --git a/metadata_hub.h b/metadata_hub.h index ed7d947d..e7e6988a 100644 --- a/metadata_hub.h +++ b/metadata_hub.h @@ -5,25 +5,25 @@ #define number_of_watchers 2 -enum play_status_type { +typedef enum { PS_NOT_AVAILABLE = 0, PS_STOPPED, PS_PAUSED, PS_PLAYING, } play_status_type; -enum active_state_type { +typedef enum { AM_INACTIVE = 0, AM_ACTIVE, } active_state_type; -enum shuffle_status_type { +typedef enum { SS_NOT_AVAILABLE = 0, SS_OFF, SS_ON, } shuffle_status_type; -enum repeat_status_type { +typedef enum { RS_NOT_AVAILABLE = 0, RS_OFF, RS_ONE, @@ -59,9 +59,9 @@ typedef struct metadata_bundle { // used detect transitions between server activity being on or off // e.g. to reease metadata when a server goes inactive, but not if it's permanently // inactive. - enum play_status_type play_status; - enum shuffle_status_type shuffle_status; - enum repeat_status_type repeat_status; + play_status_type play_status; + shuffle_status_type shuffle_status; + repeat_status_type repeat_status; // the following pertain to the track playing @@ -123,9 +123,9 @@ typedef struct metadata_bundle { // end - enum play_status_type + play_status_type player_state; // this is the state of the actual player itself, which can be a bit noisy. - enum active_state_type active_state; + active_state_type active_state; int speaker_volume; // this is the actual speaker volume, allowing for the main volume and the // speaker volume control @@ -136,7 +136,7 @@ typedef struct metadata_bundle { } metadata_bundle; -struct metadata_bundle metadata_store; +extern struct metadata_bundle metadata_store; void add_metadata_watcher(metadata_watcher fn, void *userdata); diff --git a/mpris-service.c b/mpris-service.c index 168ce396..76a04c76 100644 --- a/mpris-service.c +++ b/mpris-service.c @@ -39,6 +39,9 @@ #include "metadata_hub.h" #include "mpris-service.h" +MediaPlayer2 *mprisPlayerSkeleton; +MediaPlayer2Player *mprisPlayerPlayerSkeleton; + double airplay_volume_to_mpris_volume(double sp) { if (sp < -30.0) sp = -30.0; diff --git a/mpris-service.h b/mpris-service.h index f290d79c..160e55d4 100644 --- a/mpris-service.h +++ b/mpris-service.h @@ -4,8 +4,8 @@ #include "mpris-interface.h" -MediaPlayer2 *mprisPlayerSkeleton; -MediaPlayer2Player *mprisPlayerPlayerSkeleton; +extern MediaPlayer2 *mprisPlayerSkeleton; +extern MediaPlayer2Player *mprisPlayerPlayerSkeleton; int start_mpris_service(); From f0c0a2310e4d8c3d1a6fbe2ebefcd5f835c34fab Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 14 Feb 2020 16:15:28 +0000 Subject: [PATCH 028/186] Make some changes to remove new warnings from tinysvcmdns.c in gcc-10 --- tinysvcmdns.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/tinysvcmdns.c b/tinysvcmdns.c index 4cb6fb4c..069a6436 100644 --- a/tinysvcmdns.c +++ b/tinysvcmdns.c @@ -114,8 +114,8 @@ uint8_t *join_nlabel(const uint8_t *n1, const uint8_t *n2) { s = malloc(len1 + len2 + 1); if (s) { - strncpy((char *)s, (char *)n1, len1); - strncpy((char *)s + len1, (char *)n2, len2); + memcpy((char *)s, (char *)n1, len1); + memcpy((char *)s + len1, (char *)n2, len2); s[len1 + len2] = '\0'; } else { die("can not allocate memory for \"s\" in tinysvcmdns"); @@ -200,7 +200,7 @@ uint8_t *create_label(const char *txt) { s = malloc(len + 2); if (s) { s[0] = len; - strncpy((char *)s + 1, txt, len); + memcpy((char *)s + 1, txt, len); s[len + 1] = '\0'; } else { die("can not allocate memory for \"s\" 2 in tinysvcmdns."); @@ -222,7 +222,7 @@ uint8_t *create_nlabel(const char *name) { if (label == NULL) return NULL; - strncpy((char *)label + 1, name, len); + memcpy((char *)label + 1, name, len); label[len + 1] = '\0'; p = label; @@ -1466,7 +1466,7 @@ int close_pipe(int s) { // main loop to receive, process and send out MDNS replies // also handles MDNS service announces -static void main_loop(struct mdnsd *svr) { +void* main_loop(struct mdnsd *svr) { fd_set sockfd_set; int max_fd = svr->sockfd; char notify_buf[2]; // buffer for reading of notify_pipe @@ -1568,6 +1568,7 @@ static void main_loop(struct mdnsd *svr) { close_pipe(svr->sockfd); svr->stop_flag = 2; + return NULL; } ///////////////////////////////////////////////////// @@ -1723,7 +1724,7 @@ struct mdnsd *mdnsd_start() { pthread_attr_init(&attr); pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED); - if (pthread_create(&tid, &attr, (void *(*)(void *))main_loop, (void *)server) != 0) { + if (pthread_create(&tid, &attr, (void * (*)(void *))&main_loop, (void *)server) != 0) { pthread_mutex_destroy(&server->data_lock); free(server); return NULL; From 0902cdfe1a7b5e8c8520c1154b1ad13f8256ce22 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 14 Feb 2020 16:28:45 +0000 Subject: [PATCH 029/186] Make some changes to remove new warnings from tinysvcmdns.c in gcc-10 --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index a47a248d..7e0d7407 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.6d6], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.6d7], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 9dee8756601b80b00c045317a3fefd47e075b8ce Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 15 Feb 2020 12:35:02 +0000 Subject: [PATCH 030/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index c452c22a..3ecb0cbd 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,12 +1,19 @@ +Version 3.3.6d7 +==== +**Bug Fix** +* Make Shairport Sync compile in the forthcoming Fedora 32. Fedora 32 uses GCC-10 which [defaults to `-fno-common`](https://gcc.gnu.org/gcc-10/porting_to.html) exposing a number of issues with Shairport Sync – multiple definitions of some `enum`s and failure to define certain variables as `extern`. + + Version 3.3.6d7 is an attempt to clear up all these errors. Many thanks to [Bill Peck](https://github.com/p3ck) for his ongoing support, for bringing this issue to notice and for developing a workaround. Address issue [#973](https://github.com/mikebrady/shairport-sync/issues/973). + Version 3.3.6d6 ==== **Bug Fix** -* Add a SIGCHLD handler to remove completed processes that were used to perform and program "hooks" without waiting for completion. Thanks to [patrickjane](https://github.com/patrickjane) for reporting the bug and for suggesting a solution. Addresses Issue [#968](https://github.com/mikebrady/shairport-sync/issues/968). +* Add a `SIGCHLD` handler to remove completed processes that were used to perform and program "hooks" without waiting for completion and thus prevent them from becoming zombie processes. Thanks to [patrickjane](https://github.com/patrickjane) for reporting the bug and for suggesting a solution. Addresses issue [#968](https://github.com/mikebrady/shairport-sync/issues/968). Version 3.3.6d5 ==== **Bug Fix** -* Fix an incompatibilty with Forked Daapd that was causing Forked Daapd to lock up. Thanks to [@tomgadow](https://github.com/tomgadow) and [@ejurgensen](https://github.com/ejurgensen) for their help in finding and (hopefully) fixing this issue. Addresses Issue [#953](https://github.com/mikebrady/shairport-sync/issues/953) and [Forked Daapd Issue #870](https://github.com/ejurgensen/forked-daapd/issues/870). +* Fix an incompatibilty with Forked Daapd that was causing Forked Daapd to lock up. Thanks to [@tomgadow](https://github.com/tomgadow) and [@ejurgensen](https://github.com/ejurgensen) for their help in finding and (hopefully) fixing this issue. Addresses issue [#953](https://github.com/mikebrady/shairport-sync/issues/953) and [Forked Daapd Issue #870](https://github.com/ejurgensen/forked-daapd/issues/870). **Enhancements** * Add some extra examples of using the RemoteControl interface's SetAirplayVolume feature to the shairport-sync-dbus-test-client source. From 0c3ec38c97768a8770cb01c9a044c90a4c1876b4 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 18 Feb 2020 14:07:59 +0000 Subject: [PATCH 031/186] Add the -fno-common flag to the compiler --- Makefile.am | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Makefile.am b/Makefile.am index 2bbc387c..c9ccb75e 100644 --- a/Makefile.am +++ b/Makefile.am @@ -18,8 +18,8 @@ if BUILD_FOR_OPENBSD AM_CXXFLAGS = -I/usr/local/include -Wno-multichar -Wall -Wextra -Wno-clobbered -Wno-psabi -pthread -DSYSCONFDIR=\"$(sysconfdir)\" AM_CFLAGS = -Wno-multichar -Wall -Wextra -pthread -DSYSCONFDIR=\"$(sysconfdir)\" else - AM_CXXFLAGS = -Wno-multichar -Wall -Wextra -Wno-clobbered -Wno-psabi -pthread -DSYSCONFDIR=\"$(sysconfdir)\" - AM_CFLAGS = -Wno-multichar -Wall -Wextra -Wno-clobbered -Wno-psabi -pthread -DSYSCONFDIR=\"$(sysconfdir)\" + AM_CXXFLAGS = -fno-common -Wno-multichar -Wall -Wextra -Wno-clobbered -Wno-psabi -pthread -DSYSCONFDIR=\"$(sysconfdir)\" + AM_CFLAGS = -fno-common -Wno-multichar -Wall -Wextra -Wno-clobbered -Wno-psabi -pthread -DSYSCONFDIR=\"$(sysconfdir)\" endif endif From 789ba1ee5e8ab17b70bd2419a08c27ce6fa604fb Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 18 Feb 2020 14:09:43 +0000 Subject: [PATCH 032/186] Treat the "mper" attribute as the 64-bit item that it is rather than a 32-bit item as hithereto. Output it as a hexadecimal number to correspond with the format of the track id obtained from AppleScript --- dbus-service.c | 5 +++-- metadata_hub.c | 42 ++++++++++++++++++++++++------------------ metadata_hub.h | 2 +- mpris-service.c | 4 ++-- 4 files changed, 30 insertions(+), 23 deletions(-) diff --git a/dbus-service.c b/dbus-service.c index 1fe59c35..457aefed 100644 --- a/dbus-service.c +++ b/dbus-service.c @@ -27,6 +27,7 @@ #include #include #include +#include #include "config.h" @@ -204,7 +205,7 @@ void dbus_metadata_watcher(struct metadata_bundle *argc, __attribute__((unused)) // Add in the Track ID based on the 'mper' metadata if it is non-zero if (argc->item_id != 0) { char trackidstring[128]; - snprintf(trackidstring, sizeof(trackidstring), "/org/gnome/ShairportSync/mper_%u", + snprintf(trackidstring, sizeof(trackidstring), "/org/gnome/ShairportSync/%" PRIX64 "", argc->item_id); GVariant *trackid = g_variant_new("o", trackidstring); g_variant_builder_add(dict_builder, "{sv}", "mpris:trackid", trackid); @@ -517,7 +518,7 @@ gboolean notify_convolution_impulse_response_file_callback(ShairportSync *skelet #else gboolean notify_convolution_impulse_response_file_callback(__attribute__((unused)) ShairportSync *skeleton, __attribute__((unused)) gpointer user_data) { - char *th = (char *)shairport_sync_get_convolution_impulse_response_file(skeleton); + __attribute__((unused)) char *th = (char *)shairport_sync_get_convolution_impulse_response_file(skeleton); return TRUE; } #endif diff --git a/metadata_hub.c b/metadata_hub.c index c09759f2..d52241a0 100644 --- a/metadata_hub.c +++ b/metadata_hub.c @@ -5,7 +5,7 @@ * then you need a metadata hub, * where everything is stored * This file is part of Shairport Sync. - * Copyright (c) Mike Brady 2017--2019 + * Copyright (c) Mike Brady 2017--2020 * All rights reserved. * * Permission is hereby granted, free of charge, to any person @@ -38,6 +38,7 @@ #include #include #include +#include #include "config.h" @@ -303,28 +304,33 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin // all the following items of metadata are contained in one metadata packet // they are preceded by an 'ssnc' 'mdst' item and followed by an 'ssnc 'mden' item. - uint32_t ui; char *cs; int changed = 0; if (type == 'core') { switch (code) { - case 'mper': - ui = ntohl(*(uint32_t *)data); - debug(2, "MH Item ID seen: \"%u\" of length %u.", ui, length); - if (ui != metadata_store.item_id) { - metadata_store.item_id = ui; - metadata_store.item_id_changed = 1; - metadata_store.item_id_received = 1; - debug(2, "MH Item ID set to: \"%u\"", metadata_store.item_id); - } + case 'mper': { + // get the 64-bit number as a uint64_t by reading two uint32_t s and combining them + uint64_t vl = ntohl(*(uint32_t*)data); // get the high order 32 bits + vl = vl << 32; // shift them into the correct location + uint64_t ul = ntohl(*(uint32_t*)(data+sizeof(uint32_t))); // and the low order 32 bits + vl = vl + ul; + debug(2, "MH Item ID seen: \"%" PRIx64 "\" of length %u.", vl, length); + if (vl != metadata_store.item_id) { + metadata_store.item_id = vl; + metadata_store.item_id_changed = 1; + metadata_store.item_id_received = 1; + debug(2, "MH Item ID set to: \"%" PRIx64 "\"", metadata_store.item_id); + } + } break; - case 'astm': - ui = ntohl(*(uint32_t *)data); - debug(2, "MH Song Time seen: \"%u\" of length %u.", ui, length); - if (ui != metadata_store.songtime_in_milliseconds) { - metadata_store.songtime_in_milliseconds = ui; - metadata_store.songtime_in_milliseconds_changed = 1; - debug(2, "MH Song Time set to: \"%u\"", metadata_store.songtime_in_milliseconds); + case 'astm': { + uint32_t ui = ntohl(*(uint32_t *)data); + debug(2, "MH Song Time seen: \"%u\" of length %u.", ui, length); + if (ui != metadata_store.songtime_in_milliseconds) { + metadata_store.songtime_in_milliseconds = ui; + metadata_store.songtime_in_milliseconds_changed = 1; + debug(2, "MH Song Time set to: \"%u\"", metadata_store.songtime_in_milliseconds); + } } break; case 'asal': diff --git a/metadata_hub.h b/metadata_hub.h index e7e6988a..75d2f852 100644 --- a/metadata_hub.h +++ b/metadata_hub.h @@ -68,7 +68,7 @@ typedef struct metadata_bundle { char *cover_art_pathname; int cover_art_pathname_changed; - uint32_t item_id; // seems to be a track ID -- see itemid in DACP.c + uint64_t item_id; // seems to be a track ID -- see itemid in DACP.c int item_id_changed; int item_id_received; // important for deciding if the track information should be ignored. diff --git a/mpris-service.c b/mpris-service.c index 76a04c76..125398a1 100644 --- a/mpris-service.c +++ b/mpris-service.c @@ -25,6 +25,7 @@ */ #include #include +#include #include "config.h" @@ -174,8 +175,7 @@ void mpris_metadata_watcher(struct metadata_bundle *argc, __attribute__((unused) // Add in the Track ID based on the 'mper' metadata if it is non-zero if (argc->item_id != 0) { char trackidstring[128]; - snprintf(trackidstring, sizeof(trackidstring), "/org/gnome/ShairportSync/mper_%u", - argc->item_id); + snprintf(trackidstring, sizeof(trackidstring), "/org/gnome/ShairportSync/%" PRIX64 "", argc->item_id); GVariant *trackid = g_variant_new("o", trackidstring); g_variant_builder_add(dict_builder, "{sv}", "mpris:trackid", trackid); } From a3cf72c8ced25801028fdc5ccf6e2dc948e566db Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 18 Feb 2020 14:11:32 +0000 Subject: [PATCH 033/186] Add the -fno-common flag to compilation. Treat the "mper" attribute correctly as a 64 bit item and output it in hex to correspond with AppleScript. --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 7e0d7407..d6fa5ef0 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.6d7], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.6d8], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From a1cf11317144bc56170a3ba9999441aef92be62d Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 18 Feb 2020 15:03:14 +0000 Subject: [PATCH 034/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 3ecb0cbd..1796eefb 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,7 +1,24 @@ Version 3.3.6d7 ==== **Bug Fix** -* Make Shairport Sync compile in the forthcoming Fedora 32. Fedora 32 uses GCC-10 which [defaults to `-fno-common`](https://gcc.gnu.org/gcc-10/porting_to.html) exposing a number of issues with Shairport Sync – multiple definitions of some `enum`s and failure to define certain variables as `extern`. +* Treat the `mper` (Persistent ID of a track) metadata attribute as the 64-bit item that it really is rather than a 32-bit item as hithereto. Output it as a hexadecimal number on the MPRIS and D-Bus interfaces to correspond with the format of the persistent id obtained from AppleScript. E.g: + +``` +tell application "Music" + get the {persistent ID} of the current track +end tell +``` +resulting in: +``` +{"FD84B4B40FA33A85"} +``` +**Other Changes** +* Include the `-fno-common` flag in the compilation options to detect incorrect definitions of global variables and multiple definitions of `enum` data types. + +Version 3.3.6d7 +==== +**Bug Fix** +* Make Shairport Sync compile in the forthcoming Fedora 33. Fedora 33 uses GCC-10 which [defaults to `-fno-common`](https://gcc.gnu.org/gcc-10/porting_to.html) exposing a number of issues with Shairport Sync – multiple definitions of some `enum`s and failure to define certain variables as `extern`. Version 3.3.6d7 is an attempt to clear up all these errors. Many thanks to [Bill Peck](https://github.com/p3ck) for his ongoing support, for bringing this issue to notice and for developing a workaround. Address issue [#973](https://github.com/mikebrady/shairport-sync/issues/973). From 28c3c4252204c14c229276722f813ee762f22d76 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 18 Feb 2020 15:27:29 +0000 Subject: [PATCH 035/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 1796eefb..608f1432 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -12,6 +12,10 @@ resulting in: ``` {"FD84B4B40FA33A85"} ``` +The MPRIS `trackid` corresponding to this would be `/org/gnome/ShairportSync/FD84B4B40FA33A85`. + +Thanks to to Scott Simon (https://github.com/zutroy97) for finding a related bug in the [shairport-sync-metadata-reader](https://github.com/mikebrady/shairport-sync-metadata-reader). + **Other Changes** * Include the `-fno-common` flag in the compilation options to detect incorrect definitions of global variables and multiple definitions of `enum` data types. From 4c6b0e83c5399014066923aa182e5d4b91a64d5c Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 18 Feb 2020 16:25:28 +0000 Subject: [PATCH 036/186] Remove some unwanted and unused variable declarations from audio_dummy.c and audio_soundio.c --- audio_dummy.c | 6 ------ audio_soundio.c | 6 ------ 2 files changed, 12 deletions(-) diff --git a/audio_dummy.c b/audio_dummy.c index 026ef78a..5d64ecf2 100644 --- a/audio_dummy.c +++ b/audio_dummy.c @@ -34,17 +34,11 @@ #include #include -int Fs; -long long starttime, samples_played; - static int init(__attribute__((unused)) int argc, __attribute__((unused)) char **argv) { return 0; } static void deinit(void) {} static void start(int sample_rate, __attribute__((unused)) int sample_format) { - Fs = sample_rate; - starttime = 0; - samples_played = 0; debug(1, "dummy audio output started at Fs=%d Hz\n", sample_rate); } diff --git a/audio_soundio.c b/audio_soundio.c index 53de2acb..119f7670 100644 --- a/audio_soundio.c +++ b/audio_soundio.c @@ -7,9 +7,6 @@ #include -int Fs; -long long starttime, samples_played; - struct SoundIoOutStream *outstream; struct SoundIo *soundio; struct SoundIoDevice *device; @@ -129,9 +126,6 @@ static void deinit(void) { } static void start(int sample_rate, int sample_format) { - Fs = sample_rate; - starttime = 0; - samples_played = 0; int err; debug(1, "soundion rate: %d, format: %d", sample_rate, sample_format); From 1d0464adbc2b966752ad70c7a9ce2f1b3094bfbb Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 18 Feb 2020 16:45:37 +0000 Subject: [PATCH 037/186] removbe an extra declaration of a variable called "changed" which might allow it to be used uninitalised. --- metadata_hub.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/metadata_hub.c b/metadata_hub.c index d52241a0..f079ea2a 100644 --- a/metadata_hub.c +++ b/metadata_hub.c @@ -547,7 +547,7 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin case 'pffr': // this is sent when the first frame has been received case 'prsm': metadata_hub_modify_prolog(); - int changed = (metadata_store.player_state != PS_PLAYING); + changed = (metadata_store.player_state != PS_PLAYING); metadata_store.player_state = PS_PLAYING; metadata_hub_modify_epilog(changed); break; From f6ce0262a231ce8398716a00b86e98d75388ebab Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 19 Feb 2020 09:34:17 +0000 Subject: [PATCH 038/186] fix compaison between old and new metadata string -- check for lenghts, huh. --- common.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/common.c b/common.c index 3d41cb6c..1f56430e 100644 --- a/common.c +++ b/common.c @@ -1554,7 +1554,7 @@ int64_t generate_zero_frames(char *outp, size_t number_of_frames, sps_format_t f int string_update_with_size(char **str, int *flag, char *s, size_t len) { if (*str) { if ((s) && (len)) { - if (strncmp(*str, s, len) != 0) { + if ((len == strlen(*str)) && (strncmp(*str, s, len) != 0)) { free(*str); *str = strndup(s, len); *flag = 1; From d237d72c9746e76351819e19aca57463dc9aef97 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 19 Feb 2020 09:36:59 +0000 Subject: [PATCH 039/186] fix comparison between old and new metadata strings by checking they have equal lengths as well. Remove a redundant definition to prevent use on an uninitialised copy, clean up some of the less-used backends --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index d6fa5ef0..0d617d46 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.6d8], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.6d9], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 52da14a295ad7aceababe46a35d5dddbd51db617 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 19 Feb 2020 10:28:19 +0000 Subject: [PATCH 040/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 608f1432..aa0b57d8 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,14 @@ +Version 3.3.6d9 +==== +**Bug Fixes** +* Fix a bug in the provision of metadata which would on occasion cause metadata from the previous track to be provided. Thanks to [HiFiBerry](https://github.com/hifiberry) for reporting. This bugfix addresses issue [#972](https://github.com/mikebrady/shairport-sync/issues/972). + +Version 3.3.6d8 +==== +**Bug Fixes** +* Tidy up compilation of the `audio-dummy` and `audio-soundio` back ends when the `fno-common` flag is used. +* Remove a superflous extra definition of a variable which would casue it to be used uninitialised. + Version 3.3.6d7 ==== **Bug Fix** From 1e5adac012524a148fd5216f4a331cb63de7fa89 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 19 Feb 2020 10:28:34 +0000 Subject: [PATCH 041/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index aa0b57d8..556301f6 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,6 +1,6 @@ Version 3.3.6d9 ==== -**Bug Fixes** +**Bug Fix** * Fix a bug in the provision of metadata which would on occasion cause metadata from the previous track to be provided. Thanks to [HiFiBerry](https://github.com/hifiberry) for reporting. This bugfix addresses issue [#972](https://github.com/mikebrady/shairport-sync/issues/972). Version 3.3.6d8 From 98e444857bca8020231eb0c4862cbfc44064fd20 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 19 Feb 2020 10:28:54 +0000 Subject: [PATCH 042/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 556301f6..3577c2fe 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -7,7 +7,7 @@ Version 3.3.6d8 ==== **Bug Fixes** * Tidy up compilation of the `audio-dummy` and `audio-soundio` back ends when the `fno-common` flag is used. -* Remove a superflous extra definition of a variable which would casue it to be used uninitialised. +* Remove a superflous extra definition of a variable which would cause it to be used uninitialised. Version 3.3.6d7 ==== From 2cd77c690a55cfa6a65c4eedb607a97345b82e47 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 19 Feb 2020 10:29:46 +0000 Subject: [PATCH 043/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 3577c2fe..e26385e2 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -6,7 +6,7 @@ Version 3.3.6d9 Version 3.3.6d8 ==== **Bug Fixes** -* Tidy up compilation of the `audio-dummy` and `audio-soundio` back ends when the `fno-common` flag is used. +* Tidy up compilation of the `audio-dummy` and `audio-soundio` back ends when the `-fno-common` flag is used. * Remove a superflous extra definition of a variable which would cause it to be used uninitialised. Version 3.3.6d7 From d64a8d1e7576127df5999cd63e8f27c93598952d Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 19 Feb 2020 10:43:01 +0000 Subject: [PATCH 044/186] Update FREEBSD.md --- FREEBSD.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/FREEBSD.md b/FREEBSD.md index 63e57685..057a96f2 100644 --- a/FREEBSD.md +++ b/FREEBSD.md @@ -60,7 +60,7 @@ Next, configure the build and compile it: ``` $ autoreconf -i -f -./configure --with-avahi --with-ssl=openssl --with-sndio --with-libdaemon --with-os=freebsd --with-freebsd-service +./configure --with-libdaemon --with-avahi --with-ssl=openssl --with-sndio --with-libdaemon --with-os=freebsd --with-freebsd-service $ make ``` Add `--with-alsa` if you wish to include the ALSA back end. Omit the `--with-sndio` if you don't want the `sndio` back end. Omit the `--with-freebsd-service` if you don't want to install a FreeBSD startup script, runtime folder and user and group -- see below for more details. From 6c72e52a07b2f63579e72a59826072d82a212b0e Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 19 Feb 2020 14:34:29 +0000 Subject: [PATCH 045/186] Brown bag -- this is what 3.3.6d9 was supposed to do! --- common.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/common.c b/common.c index 1f56430e..84b31793 100644 --- a/common.c +++ b/common.c @@ -1554,7 +1554,7 @@ int64_t generate_zero_frames(char *outp, size_t number_of_frames, sps_format_t f int string_update_with_size(char **str, int *flag, char *s, size_t len) { if (*str) { if ((s) && (len)) { - if ((len == strlen(*str)) && (strncmp(*str, s, len) != 0)) { + if ((len != strlen(*str)) || (strncmp(*str, s, len) != 0)) { free(*str); *str = strndup(s, len); *flag = 1; From d017ac3948b45a2bb8a21ba3d446e622f6c62dbe Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 19 Feb 2020 14:34:51 +0000 Subject: [PATCH 046/186] Brown bag -- this is what 3.3.6d9 was supposed to do! --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 0d617d46..2d9a0496 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.6d9], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.6d10], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 6c9177a77866fc583ea4e74b2f19873228f45492 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 19 Feb 2020 14:35:50 +0000 Subject: [PATCH 047/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index e26385e2..51ee6516 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,8 @@ +Version 3.3.6d10 +==== +**Bug Fix** +* Brown-bag fix -- do what 3.3.6d9 was supposed to do! + Version 3.3.6d9 ==== **Bug Fix** From 4fc161cf0c4d0bbf500b5887da5704a600d7ab5e Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 19 Feb 2020 20:18:09 +0000 Subject: [PATCH 048/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 51ee6516..04d192c4 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -6,7 +6,7 @@ Version 3.3.6d10 Version 3.3.6d9 ==== **Bug Fix** -* Fix a bug in the provision of metadata which would on occasion cause metadata from the previous track to be provided. Thanks to [HiFiBerry](https://github.com/hifiberry) for reporting. This bugfix addresses issue [#972](https://github.com/mikebrady/shairport-sync/issues/972). +* Fix a bug in the provision of metadata which would on occasion cause metadata from the previous track to be provided. Thanks to [Tuomas Hämäläinen](https://github.com/tuomashamalainen) and [HiFiBerry](https://github.com/hifiberry) for reporting. This bugfix addresses issue [#972](https://github.com/mikebrady/shairport-sync/issues/972). Version 3.3.6d8 ==== @@ -38,7 +38,7 @@ Thanks to to Scott Simon (https://github.com/zutroy97) for finding a related bug Version 3.3.6d7 ==== **Bug Fix** -* Make Shairport Sync compile in the forthcoming Fedora 33. Fedora 33 uses GCC-10 which [defaults to `-fno-common`](https://gcc.gnu.org/gcc-10/porting_to.html) exposing a number of issues with Shairport Sync – multiple definitions of some `enum`s and failure to define certain variables as `extern`. +* Make Shairport Sync compile in the forthcoming Fedora 32. Fedora 32 uses GCC-10 which [defaults to `-fno-common`](https://gcc.gnu.org/gcc-10/porting_to.html) exposing a number of issues with Shairport Sync – multiple definitions of some `enum`s and failure to define certain variables as `extern`. Version 3.3.6d7 is an attempt to clear up all these errors. Many thanks to [Bill Peck](https://github.com/p3ck) for his ongoing support, for bringing this issue to notice and for developing a workaround. Address issue [#973](https://github.com/mikebrady/shairport-sync/issues/973). From d673f274cbb0b9aa06132715cd587899948956be Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 20 Feb 2020 18:24:04 +0000 Subject: [PATCH 049/186] Don't use strndup as old versions of OpenWrt don't seem to have it --- common.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/common.c b/common.c index 84b31793..228dfec9 100644 --- a/common.c +++ b/common.c @@ -1556,7 +1556,11 @@ int string_update_with_size(char **str, int *flag, char *s, size_t len) { if ((s) && (len)) { if ((len != strlen(*str)) || (strncmp(*str, s, len) != 0)) { free(*str); - *str = strndup(s, len); + //*str = strndup(s, len); // it seems that OpenWrt 12 doesn't have this + char *p = malloc(len + 1); + memcpy(p,s,len); + p[len] = '\0'; + *str = p; *flag = 1; } else { *flag = 0; @@ -1569,7 +1573,11 @@ int string_update_with_size(char **str, int *flag, char *s, size_t len) { } } else { // old string is NULL if ((s) && (len)) { - *str = strndup(s, len); + //*str = strndup(s, len); // it seems that OpenWrt 12 doesn't have this + char *p = malloc(len + 1); + memcpy(p,s,len); + p[len] = '\0'; + *str = p; *flag = 1; } else { // old string is NULL and new string is NULL or length 0 From d6c6e8c1b7f68ada55e7b3098e3b8e67fb5fe90a Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 20 Feb 2020 18:24:32 +0000 Subject: [PATCH 050/186] Don't use strndup as old versions of OpenWrt don't seem to have it --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 2d9a0496..19f60933 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.6d10], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.6d11], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From db12e0a8e3608e3e1dd4cc25d5719c418307a5e1 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 20 Feb 2020 21:25:16 +0000 Subject: [PATCH 051/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 04d192c4..2c275823 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,8 @@ +Version 3.3.6d11 +==== +**Bug Fix** +* Avoid using `strndup` as it seems older versions of OpenWrt do not recognise it. + Version 3.3.6d10 ==== **Bug Fix** From 20ca91d016f51789651e05f55e9f8762375e3f83 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 20 Feb 2020 22:26:04 +0000 Subject: [PATCH 052/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 2c275823..d0146e3d 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,4 +1,4 @@ -Version 3.3.6d11 +Version 3.3.6d11 = Version 3.3.6 ==== **Bug Fix** * Avoid using `strndup` as it seems older versions of OpenWrt do not recognise it. From 528b5141ac40b59bc47b1a00474966547e2e4b3c Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 21 Feb 2020 13:36:07 +0000 Subject: [PATCH 053/186] Update FREEBSD.md --- FREEBSD.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/FREEBSD.md b/FREEBSD.md index 057a96f2..fe3a2a5e 100644 --- a/FREEBSD.md +++ b/FREEBSD.md @@ -60,7 +60,7 @@ Next, configure the build and compile it: ``` $ autoreconf -i -f -./configure --with-libdaemon --with-avahi --with-ssl=openssl --with-sndio --with-libdaemon --with-os=freebsd --with-freebsd-service +$ ./configure --with-libdaemon --with-avahi --with-ssl=openssl --with-sndio --with-libdaemon --with-os=freebsd --with-freebsd-service $ make ``` Add `--with-alsa` if you wish to include the ALSA back end. Omit the `--with-sndio` if you don't want the `sndio` back end. Omit the `--with-freebsd-service` if you don't want to install a FreeBSD startup script, runtime folder and user and group -- see below for more details. From ebb1d69ce0a1d42794752a2c4aeffe239847ccca Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 22 Feb 2020 18:24:57 +0000 Subject: [PATCH 054/186] Remove a redundant comparison --- common.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/common.c b/common.c index 228dfec9..78f89f9c 100644 --- a/common.c +++ b/common.c @@ -101,7 +101,7 @@ const char *sps_format_description_string_array[] = { "S24_BE", "S24_3LE", "S24_3BE", "S32", "S32_LE", "S32_BE", "auto", "invalid"}; const char *sps_format_description_string(sps_format_t format) { - if ((format >= SPS_FORMAT_UNKNOWN) && (format <= SPS_FORMAT_AUTO)) + if (format <= SPS_FORMAT_AUTO) return sps_format_description_string_array[format]; else return sps_format_description_string_array[SPS_FORMAT_INVALID]; From 1f526938841e979526c8e7e2148c83d31aeb7afe Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 22 Feb 2020 18:25:14 +0000 Subject: [PATCH 055/186] This is 3.3.6 with some documentation updates and a tiny edit --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 19f60933..c1e393f7 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.6d11], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.7d1], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 820ee1c9f2099ebb9aa8978859d221c56999b9e3 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 23 Feb 2020 11:46:07 +0000 Subject: [PATCH 056/186] Add a call to get the time in nanoseconds. --- common.c | 40 ++++++++++++++++++++++++++++++++++++++++ common.h | 1 + 2 files changed, 41 insertions(+) diff --git a/common.c b/common.c index 78f89f9c..3bdf17de 100644 --- a/common.c +++ b/common.c @@ -1013,6 +1013,46 @@ uint64_t get_absolute_time_in_fp() { return time_now_fp; } +uint64_t get_absolute_time_in_ns() { + uint64_t time_now_ns; + +#ifdef COMPILE_FOR_LINUX_AND_FREEBSD_AND_CYGWIN_AND_OPENBSD + struct timespec tn; + // can't use CLOCK_MONOTONIC_RAW as it's not implemented in OpenWrt + clock_gettime(CLOCK_MONOTONIC, &tn); + uint64_t tnnsec = tn.tv_sec; + tnnsec = tnnsec * 1000000000; + uint64_t tnjnsec = tn.tv_nsec; + time_now_ns = tnnsec + tnjnsec; +#endif + +#ifdef COMPILE_FOR_OSX + uint64_t time_now_mach; + uint64_t elapsedNano; + static mach_timebase_info_data_t sTimebaseInfo = {0, 0}; + + time_now_mach = mach_absolute_time(); + + // If this is the first time we've run, get the timebase. + // We can use denom == 0 to indicate that sTimebaseInfo is + // uninitialised because it makes no sense to have a zero + // denominator in a fraction. + + if (sTimebaseInfo.denom == 0) { + debug(1, "Mac initialise timebase info."); + (void)mach_timebase_info(&sTimebaseInfo); + } + + // Do the maths. We hope that the multiplication doesn't + // overflow; the price you pay for working in fixed point. + + // this gives us nanoseconds + time_now_ns = time_now_mach * sTimebaseInfo.numer / sTimebaseInfo.denom; +#endif + + return time_now_ns; +} + ssize_t non_blocking_write_with_timeout(int fd, const void *buf, size_t count, int timeout) { // timeout is in milliseconds void *ibuf = (void *)buf; diff --git a/common.h b/common.h index 2c9c05bd..96d59b7d 100644 --- a/common.h +++ b/common.h @@ -350,6 +350,7 @@ double vol2attn(double vol, long max_db, long min_db); // return a monolithic (always increasing) time in nanoseconds uint64_t get_absolute_time_in_fp(void); +uint64_t get_absolute_time_in_ns(void); // time at startup for debugging timing extern uint64_t fp_time_at_startup, fp_time_at_last_debug_message; From 67e9b1b6cd2efd23532a618ee7e9a0651f55ce4a Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 24 Feb 2020 19:40:44 +0000 Subject: [PATCH 057/186] partial move to using a nanosecond-based system timer. More to come. --- activity_monitor.c | 16 +++---- audio_alsa.c | 110 +++++++++++++++++++++------------------------ audio_jack.c | 8 ++-- common.c | 75 +++++++++++++++---------------- common.h | 2 +- dacp.c | 7 ++- player.c | 70 ++++++++++++++--------------- rtsp.c | 8 ++-- shairport.c | 16 +++---- 9 files changed, 145 insertions(+), 167 deletions(-) diff --git a/activity_monitor.c b/activity_monitor.c index bcc25d24..ce7edc5f 100644 --- a/activity_monitor.c +++ b/activity_monitor.c @@ -196,21 +196,19 @@ void *activity_monitor_thread_code(void *arg) { } else { state = am_timing_out; - uint64_t time_to_wait_for_wakeup_fp = - (uint64_t)(config.active_state_timeout * 1000000); // resolution of microseconds - time_to_wait_for_wakeup_fp = time_to_wait_for_wakeup_fp << 32; - time_to_wait_for_wakeup_fp = time_to_wait_for_wakeup_fp / 1000000; + uint64_t time_to_wait_for_wakeup_ns = (uint64_t)(config.active_state_timeout * 1000000000); #ifdef COMPILE_FOR_LINUX_AND_FREEBSD_AND_CYGWIN_AND_OPENBSD - uint64_t time_of_wakeup_fp = get_absolute_time_in_fp() + time_to_wait_for_wakeup_fp; - sec = time_of_wakeup_fp >> 32; - nsec = ((time_of_wakeup_fp & 0xffffffff) * 1000000000) >> 32; + uint64_t time_of_wakeup_ns = get_absolute_time_in_ns() + time_to_wait_for_wakeup_ns; + sec = time_of_wakeup_ns / 1000000000; + nsec = time_of_wakeup_ns % 1000000000; time_for_wait.tv_sec = sec; time_for_wait.tv_nsec = nsec; #endif + #ifdef COMPILE_FOR_OSX - sec = time_to_wait_for_wakeup_fp >> 32; - nsec = ((time_to_wait_for_wakeup_fp & 0xffffffff) * 1000000000) >> 32; + sec = time_to_wait_for_wakeup_ns / 1000000000; + nsec = time_to_wait_for_wakeup_ns % 1000000000; time_for_wait.tv_sec = sec; time_for_wait.tv_nsec = nsec; #endif diff --git a/audio_alsa.c b/audio_alsa.c index 39e4d051..40287909 100644 --- a/audio_alsa.c +++ b/audio_alsa.c @@ -210,9 +210,6 @@ int precision_delay_available() { generate_zero_frames(silence, frames_of_silence, config.output_format, use_dither, // i.e. with dither dither_random_number_store); - // debug(1,"Play %d frames of silence with most_recent_write_time of - // %" PRIx64 ".", - // frames_of_silence,most_recent_write_time); do_play(silence, frames_of_silence); pthread_cleanup_pop(1); // now we can get the delay, and we'll note if it uses update timestamps @@ -255,8 +252,6 @@ static uint64_t frames_played_at_measurement_start_time; static uint64_t measurement_time; static uint64_t frames_played_at_measurement_time; -volatile uint64_t most_recent_write_time; - static uint64_t frames_sent_for_playing; static uint64_t frame_index; static int measurement_data_is_valid; @@ -1350,7 +1345,6 @@ static int init(int argc, char **argv) { // length of the queue // if the queue gets too short, stuff it with silence - most_recent_write_time = 0; // could be used by the alsa_buffer_monitor_thread_code pthread_create(&alsa_buffer_monitor_thread, NULL, &alsa_buffer_monitor_thread_code, NULL); return response; @@ -1605,6 +1599,7 @@ int delay(long *the_delay) { } int get_rate_information(uint64_t *elapsed_time, uint64_t *frames_played) { + // elapsed_time is in nanoseconds int response = 0; // zero means okay if (measurement_data_is_valid) { *elapsed_time = measurement_time - measurement_start_time; @@ -1657,7 +1652,7 @@ int do_play(void *buf, int samples) { if ((frame_index == start_measurement_from_this_frame) || ((frame_index > start_measurement_from_this_frame) && (frame_index % 32 == 0))) { - measurement_time = get_absolute_time_in_fp(); + measurement_time = get_absolute_time_in_ns(); frames_played_at_measurement_time = frames_sent_for_playing - my_delay - samples; if (frame_index == start_measurement_from_this_frame) { @@ -1744,6 +1739,7 @@ int do_close() { // debug(1,"alsa: do_close() -- closing the output device"); if ((derr = snd_pcm_drop(alsa_handle))) debug(1, "Error %d (\"%s\") dropping output device.", derr, snd_strerror(derr)); + usleep(5000); if ((derr = snd_pcm_hw_free(alsa_handle))) debug(1, "Error %d (\"%s\") freeing the output device hardware.", derr, snd_strerror(derr)); @@ -1946,7 +1942,7 @@ void *alsa_buffer_monitor_thread_code(__attribute__((unused)) void *arg) { alsa_device_init(); alsa_device_initialised = 1; } - int sleep_time_ms = (int)(config.disable_standby_mode_silence_scan_interval * 1000); + int sleep_time_us = (int)(config.disable_standby_mode_silence_scan_interval * 1000000); pthread_cleanup_debug_mutex_lock(&alsa_mutex, 200000, 0); // check possible state transitions here if ((alsa_backend_state == abm_disconnected) && (config.keep_dac_busy != 0)) { @@ -1974,57 +1970,51 @@ void *alsa_buffer_monitor_thread_code(__attribute__((unused)) void *arg) { int reply; long buffer_size = 0; snd_pcm_state_t state; - uint64_t present_time = get_absolute_time_in_fp(); - if ((most_recent_write_time == 0) || (present_time > most_recent_write_time)) { - reply = delay_and_status(&state, &buffer_size, NULL); - if (reply != 0) { - buffer_size = 0; - char errorstring[1024]; - strerror_r(-reply, (char *)errorstring, sizeof(errorstring)); - debug(1, "alsa: alsa_buffer_monitor_thread_code delay error %d: \"%s\".", reply, - (char *)errorstring); - } - long buffer_size_threshold = - (long)(config.disable_standby_mode_silence_threshold * config.output_rate); - size_t size_of_silence_buffer; - if (buffer_size < buffer_size_threshold) { - uint64_t sleep_time_in_fp = sleep_time_ms; - sleep_time_in_fp = sleep_time_in_fp << 32; - sleep_time_in_fp = sleep_time_in_fp / 1000; - int frames_of_silence = 1024; - size_of_silence_buffer = frames_of_silence * frame_size; - void *silence = malloc(size_of_silence_buffer); - if (silence == NULL) { - warn("disable_standby_mode has been turned off because a memory allocation error " - "occurred."); - error_detected = 1; - } else { - int ret; - pthread_cleanup_push(malloc_cleanup, silence); - int use_dither = 0; - if ((alsa_mix_ctrl == NULL) && (config.ignore_volume_control == 0) && - (config.airplay_volume != 0.0)) - use_dither = 1; - dither_random_number_store = - generate_zero_frames(silence, frames_of_silence, config.output_format, - use_dither, // i.e. with dither - dither_random_number_store); - ret = do_play(silence, frames_of_silence); - frame_count++; - pthread_cleanup_pop(1); // free malloced buffer - if (ret < 0) { - error_count++; - char errorstring[1024]; - strerror_r(-ret, (char *)errorstring, sizeof(errorstring)); - debug(2, "alsa: alsa_buffer_monitor_thread_code error %d (\"%s\") writing %d samples " - "to alsa device -- %d errors in %d trials.", - ret, (char *)errorstring, frames_of_silence, error_count, frame_count); - if ((error_count > 40) && (frame_count < 100)) { - warn("disable_standby_mode has been turned off because too many underruns " - "occurred. Is Shairport Sync outputting to a virtual device or running in a " - "virtual machine?"); - error_detected = 1; - } + reply = delay_and_status(&state, &buffer_size, NULL); + if (reply != 0) { + buffer_size = 0; + char errorstring[1024]; + strerror_r(-reply, (char *)errorstring, sizeof(errorstring)); + debug(1, "alsa: alsa_buffer_monitor_thread_code delay error %d: \"%s\".", reply, + (char *)errorstring); + } + long buffer_size_threshold = + (long)(config.disable_standby_mode_silence_threshold * config.output_rate); + size_t size_of_silence_buffer; + if (buffer_size < buffer_size_threshold) { + int frames_of_silence = 1024; + size_of_silence_buffer = frames_of_silence * frame_size; + void *silence = malloc(size_of_silence_buffer); + if (silence == NULL) { + warn("disable_standby_mode has been turned off because a memory allocation error " + "occurred."); + error_detected = 1; + } else { + int ret; + pthread_cleanup_push(malloc_cleanup, silence); + int use_dither = 0; + if ((alsa_mix_ctrl == NULL) && (config.ignore_volume_control == 0) && + (config.airplay_volume != 0.0)) + use_dither = 1; + dither_random_number_store = + generate_zero_frames(silence, frames_of_silence, config.output_format, + use_dither, // i.e. with dither + dither_random_number_store); + ret = do_play(silence, frames_of_silence); + frame_count++; + pthread_cleanup_pop(1); // free malloced buffer + if (ret < 0) { + error_count++; + char errorstring[1024]; + strerror_r(-ret, (char *)errorstring, sizeof(errorstring)); + debug(2, "alsa: alsa_buffer_monitor_thread_code error %d (\"%s\") writing %d samples " + "to alsa device -- %d errors in %d trials.", + ret, (char *)errorstring, frames_of_silence, error_count, frame_count); + if ((error_count > 40) && (frame_count < 100)) { + warn("disable_standby_mode has been turned off because too many underruns " + "occurred. Is Shairport Sync outputting to a virtual device or running in a " + "virtual machine?"); + error_detected = 1; } } } @@ -2032,7 +2022,7 @@ void *alsa_buffer_monitor_thread_code(__attribute__((unused)) void *arg) { } debug_mutex_unlock(&alsa_mutex, 0); pthread_cleanup_pop(0); // release the mutex - usleep(sleep_time_ms * 1000); // has a cancellation point in it + usleep(sleep_time_us); // has a cancellation point in it } pthread_exit(NULL); } diff --git a/audio_jack.c b/audio_jack.c index db6e1899..5caa6c68 100644 --- a/audio_jack.c +++ b/audio_jack.c @@ -395,13 +395,13 @@ int jack_delay(long *the_delay) { // occupancy after another transfer had occurred, so we could "lose" a full transfer // (e.g. 1024 frames @ 44,100 fps ~ 23.2 milliseconds) pthread_mutex_lock(&buffer_mutex); - int64_t time_now = get_absolute_time_in_fp(); - int64_t delta = time_now - time_of_latest_transfer; + int64_t time_now = get_absolute_time_in_ns(); + int64_t delta = time_now - time_of_latest_transfer; // nanoseconds size_t audio_occupancy_now = jack_ringbuffer_read_space(jackbuf) / bytes_per_frame; debug(2, "audio_occupancy_now is %d.", audio_occupancy_now); pthread_mutex_unlock(&buffer_mutex); - int64_t frames_processed_since_latest_latency_check = (delta * sample_rate) >> 32; + int64_t frames_processed_since_latest_latency_check = (delta * sample_rate) / 1000000000; // debug(1,"delta: %" PRId64 " frames.",frames_processed_since_latest_latency_check); // jack_latency is set by the graph() callback, it's the average of the maximum // latencies of all our output ports. Adjust this constant baseline delay according @@ -456,7 +456,7 @@ int play(void *buf, int samples) { } #endif } - time_of_latest_transfer = get_absolute_time_in_fp(); + time_of_latest_transfer = get_absolute_time_in_ns(); pthread_mutex_unlock(&buffer_mutex); if (samples) { warn("JACK ringbuffer overrun. Dropped %d samples.", samples); diff --git a/common.c b/common.c index 3bdf17de..b3942f78 100644 --- a/common.c +++ b/common.c @@ -89,9 +89,9 @@ void set_alsa_out_dev(char *); config_t config_file_stuff; pthread_t main_thread_id; -uint64_t fp_time_at_startup, fp_time_at_last_debug_message; +uint64_t ns_time_at_startup, ns_time_at_last_debug_message; -// always lock use this when accessing the fp_time_at_last_debug_message +// always lock use this when accessing the ns_time_at_last_debug_message static pthread_mutex_t debug_timing_lock = PTHREAD_MUTEX_INITIALIZER; pthread_mutex_t the_conn_lock = PTHREAD_MUTEX_INITIALIZER; @@ -218,14 +218,13 @@ void _die(const char *filename, const int linenumber, const char *format, ...) { char *s; if (debuglev) { pthread_mutex_lock(&debug_timing_lock); - uint64_t time_now = get_absolute_time_in_fp(); - uint64_t time_since_start = time_now - fp_time_at_startup; - uint64_t time_since_last_debug_message = time_now - fp_time_at_last_debug_message; - fp_time_at_last_debug_message = time_now; + uint64_t time_now = get_absolute_time_in_ns(); + uint64_t time_since_start = time_now - ns_time_at_startup; + uint64_t time_since_last_debug_message = time_now - ns_time_at_last_debug_message; + ns_time_at_last_debug_message = time_now; pthread_mutex_unlock(&debug_timing_lock); - uint64_t divisor = (uint64_t)1 << 32; - s = generate_preliminary_string(b, sizeof(b), 1.0 * time_since_start / divisor, - 1.0 * time_since_last_debug_message / divisor, filename, + s = generate_preliminary_string(b, sizeof(b), 1.0 * time_since_start / 1000000000, + 1.0 * time_since_last_debug_message / 1000000000, filename, linenumber, " *fatal error: "); } else { s = b; @@ -247,14 +246,13 @@ void _warn(const char *filename, const int linenumber, const char *format, ...) char *s; if (debuglev) { pthread_mutex_lock(&debug_timing_lock); - uint64_t time_now = get_absolute_time_in_fp(); - uint64_t time_since_start = time_now - fp_time_at_startup; - uint64_t time_since_last_debug_message = time_now - fp_time_at_last_debug_message; - fp_time_at_last_debug_message = time_now; + uint64_t time_now = get_absolute_time_in_ns(); + uint64_t time_since_start = time_now - ns_time_at_startup; + uint64_t time_since_last_debug_message = time_now - ns_time_at_last_debug_message; + ns_time_at_last_debug_message = time_now; pthread_mutex_unlock(&debug_timing_lock); - uint64_t divisor = (uint64_t)1 << 32; - s = generate_preliminary_string(b, sizeof(b), 1.0 * time_since_start / divisor, - 1.0 * time_since_last_debug_message / divisor, filename, + s = generate_preliminary_string(b, sizeof(b), 1.0 * time_since_start / 1000000000, + 1.0 * time_since_last_debug_message / 1000000000, filename, linenumber, " *warning: "); } else { s = b; @@ -275,14 +273,13 @@ void _debug(const char *filename, const int linenumber, int level, const char *f char b[1024]; b[0] = 0; pthread_mutex_lock(&debug_timing_lock); - uint64_t time_now = get_absolute_time_in_fp(); - uint64_t time_since_start = time_now - fp_time_at_startup; - uint64_t time_since_last_debug_message = time_now - fp_time_at_last_debug_message; - fp_time_at_last_debug_message = time_now; + uint64_t time_now = get_absolute_time_in_ns(); + uint64_t time_since_start = time_now - ns_time_at_startup; + uint64_t time_since_last_debug_message = time_now - ns_time_at_last_debug_message; + ns_time_at_last_debug_message = time_now; pthread_mutex_unlock(&debug_timing_lock); - uint64_t divisor = (uint64_t)1 << 32; - char *s = generate_preliminary_string(b, sizeof(b), 1.0 * time_since_start / divisor, - 1.0 * time_since_last_debug_message / divisor, filename, + char *s = generate_preliminary_string(b, sizeof(b), 1.0 * time_since_start / 1000000000, + 1.0 * time_since_last_debug_message / 1000000000, filename, linenumber, " "); va_list args; va_start(args, format); @@ -300,14 +297,13 @@ void _inform(const char *filename, const int linenumber, const char *format, ... char *s; if (debuglev) { pthread_mutex_lock(&debug_timing_lock); - uint64_t time_now = get_absolute_time_in_fp(); - uint64_t time_since_start = time_now - fp_time_at_startup; - uint64_t time_since_last_debug_message = time_now - fp_time_at_last_debug_message; - fp_time_at_last_debug_message = time_now; + uint64_t time_now = get_absolute_time_in_ns(); + uint64_t time_since_start = time_now - ns_time_at_startup; + uint64_t time_since_last_debug_message = time_now - ns_time_at_last_debug_message; + ns_time_at_last_debug_message = time_now; pthread_mutex_unlock(&debug_timing_lock); - uint64_t divisor = (uint64_t)1 << 32; - s = generate_preliminary_string(b, sizeof(b), 1.0 * time_since_start / divisor, - 1.0 * time_since_last_debug_message / divisor, filename, + s = generate_preliminary_string(b, sizeof(b), 1.0 * time_since_start / 1000000000, + 1.0 * time_since_last_debug_message / 1000000000, filename, linenumber, " "); } else { s = b; @@ -1133,7 +1129,9 @@ char *str_replace(const char *string, const char *substr, const char *replacemen /* from http://burtleburtle.net/bob/rand/smallprng.html */ -// this is not thread-safe, so we need a mutex on it to use it properly// always lock use this when accessing the fp_time_at_last_debug_message +// this is not thread-safe, so we need a mutex on it to use it properly. +// always lock use this when accessing the fp_time_at_last_debug_message + pthread_mutex_t r64_mutex = PTHREAD_MUTEX_INITIALIZER; // typedef uint64_t u8; @@ -1241,18 +1239,17 @@ int sps_pthread_mutex_timedlock(pthread_mutex_t *mutex, useconds_t dally_time, timeoutTime.tv_sec = time_then; timeoutTime.tv_nsec = time_then_nsec; - int64_t start_time = get_absolute_time_in_fp(); + int64_t start_time = get_absolute_time_in_ns(); int r = pthread_mutex_timedlock(mutex, &timeoutTime); - int64_t et = get_absolute_time_in_fp() - start_time; + int64_t et = get_absolute_time_in_ns() - start_time; if ((debuglevel != 0) && (r != 0) && (debugmessage != NULL)) { - et = (et * 1000000) >> 32; // microseconds char errstr[1000]; if (r == ETIMEDOUT) debug(debuglevel, "timed out waiting for a mutex, having waiting %f seconds, with a maximum " "waiting time of %d microseconds. \"%s\".", - (1.0 * et) / 1000000, dally_time, debugmessage); + (1.0 * et) / 1000000000, dally_time, debugmessage); else debug(debuglevel, "error %d: \"%s\" waiting for a mutex: \"%s\".", r, strerror_r(r, errstr, sizeof(errstr)), debugmessage); @@ -1301,7 +1298,7 @@ int _debug_mutex_lock(pthread_mutex_t *mutex, useconds_t dally_time, const char return pthread_mutex_lock(mutex); int oldState; pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); - uint64_t time_at_start = get_absolute_time_in_fp(); + uint64_t time_at_start = get_absolute_time_in_ns(); char dstring[1000]; memset(dstring, 0, sizeof(dstring)); snprintf(dstring, sizeof(dstring), "%s:%d", filename, line); @@ -1310,12 +1307,10 @@ int _debug_mutex_lock(pthread_mutex_t *mutex, useconds_t dally_time, const char int result = sps_pthread_mutex_timedlock(mutex, dally_time, dstring, debuglevel); if (result == ETIMEDOUT) { result = pthread_mutex_lock(mutex); - uint64_t time_delay = get_absolute_time_in_fp() - time_at_start; - uint64_t divisor = (uint64_t)1 << 32; - double delay = 1.0 * time_delay / divisor; + uint64_t time_delay = get_absolute_time_in_ns() - time_at_start; debug(debuglevel, "mutex_lock \"%s\" at \"%s\" expected max wait: %0.9f, actual wait: %0.9f sec.", - mutexname, dstring, (1.0 * dally_time) / 1000000, delay); + mutexname, dstring, (1.0 * dally_time) / 1000000, 0.000000001 * time_delay); } pthread_setcancelstate(oldState, NULL); return result; diff --git a/common.h b/common.h index 96d59b7d..92095387 100644 --- a/common.h +++ b/common.h @@ -353,7 +353,7 @@ uint64_t get_absolute_time_in_fp(void); uint64_t get_absolute_time_in_ns(void); // time at startup for debugging timing -extern uint64_t fp_time_at_startup, fp_time_at_last_debug_message; +extern uint64_t ns_time_at_startup, ns_time_at_last_debug_message; // this is for reading an unsigned 32 bit number, such as an RTP timestamp diff --git a/dacp.c b/dacp.c index f3111a83..4c36de34 100644 --- a/dacp.c +++ b/dacp.c @@ -201,7 +201,7 @@ int dacp_send_command(const char *command, char **body, ssize_t *bodysize) { // debug(1,"Error %d \"%s\" at getaddrinfo.",ires,gai_strerror(ires)); response.code = 498; // Bad Address information for the DACP server } else { - uint64_t start_time = get_absolute_time_in_fp(); + uint64_t start_time = get_absolute_time_in_ns(); pthread_cleanup_push(addrinfo_cleanup, (void *)&res); // only do this one at a time -- not sure it is necessary, but better safe than sorry @@ -352,10 +352,9 @@ int dacp_send_command(const char *command, char **body, ssize_t *bodysize) { } pthread_cleanup_pop(1); // this should free the addrinfo // freeaddrinfo(res); - uint64_t et = get_absolute_time_in_fp() - start_time; - et = (et * 1000000) >> 32; // microseconds + uint64_t et = get_absolute_time_in_ns() - start_time; // this will be in nanoseconds debug(3, "dacp_send_command: %f seconds, response code %d, command \"%s\".", - (1.0 * et) / 1000000, response.code, command); + (1.0 * et) / 1000000000, response.code, command); } *body = response.body; *bodysize = response.size; diff --git a/player.c b/player.c index 5876bd33..a034346c 100644 --- a/player.c +++ b/player.c @@ -124,7 +124,7 @@ static void ab_resync(rtsp_conn_info *conn) { conn->audio_buffer[i].ready = 0; conn->audio_buffer[i].resend_request_number = 0; conn->audio_buffer[i].resend_time = 0; // this is either zero or the time the last resend was requested. - conn->audio_buffer[i].initialisation_time = 0; // this is either the time the packet was received or the time it was noticed the packet was missing. + conn->audio_buffer[i].initialisation_time = 0; // this is either the time the packet was received or the time it was noticed the packet was missing. conn->audio_buffer[i].sequence_number = 0; } conn->ab_synced = 0; @@ -564,7 +564,7 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in int rc = pthread_cond_signal(&conn->flowcontrol); if (rc) debug(1, "Error signalling flowcontrol."); - + // resend checks { uint64_t minimum_wait_time = (uint64_t)(config.resend_control_first_check_time * (uint64_t)0x100000000); @@ -572,7 +572,7 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in uint64_t minimum_remaining_time = (uint64_t)((config.resend_control_last_check_time + config.audio_backend_buffer_desired_length)* (uint64_t)0x100000000); uint64_t latency_time = (uint64_t)(conn->latency * (uint64_t)0x100000000); latency_time = latency_time / (uint64_t)conn->input_rate; - + int x; // this is the first frame to be checked // if we detected a first empty frame before and if it's still in the buffer! if ((first_possibly_missing_frame >= 0) && (position_in_modulo_uint16_t_buffer(first_possibly_missing_frame, conn->ab_read, conn->ab_write, NULL))) { @@ -580,9 +580,9 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in } else { x = conn->ab_read; } - + first_possibly_missing_frame = -1; // has not been set - + int missing_frame_run_count = 0; int start_of_missing_frame_run = -1; int number_of_missing_frames = 0; @@ -604,12 +604,12 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in if (too_early) check_buf->status |= 1<<3; // too early else - check_buf->status &= 0xFF-(1<<3); // not too early + check_buf->status &= 0xFF-(1<<3); // not too early if (too_soon_after_last_request) check_buf->status |= 1<<4; // too soon after last request else check_buf->status &= 0xFF-(1<<4); // not too soon after last request - + if ((!too_soon_after_last_request) && (!too_late) && (!too_early)){ if (start_of_missing_frame_run == -1) { start_of_missing_frame_run = x; @@ -619,7 +619,7 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in } check_buf->resend_time = time_now; // setting the time to now because we are definitely going to take action check_buf->resend_request_number++; - debug(3,"Frame %d is missing with ab_read of %u and ab_write of %u.", x, conn->ab_read, conn->ab_write); + debug(3,"Frame %d is missing with ab_read of %u and ab_write of %u.", x, conn->ab_read, conn->ab_write); } // if (too_late) { // debug(1,"too late to get missing frame %u.", x); @@ -631,13 +631,13 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in if (((check_buf->ready) || (x == conn->ab_write)) && (missing_frame_run_count > 0)) { // send a resend request if (missing_frame_run_count > 1) - debug(2,"request resend of %d packets starting at seqno %u.", missing_frame_run_count, start_of_missing_frame_run); + debug(2,"request resend of %d packets starting at seqno %u.", missing_frame_run_count, start_of_missing_frame_run); if (config.disable_resend_requests == 0) { debug_mutex_unlock(&conn->ab_mutex, 3); rtp_request_resend(start_of_missing_frame_run, missing_frame_run_count, conn); debug_mutex_lock(&conn->ab_mutex, 20000, 1); conn->resend_requests++; - } + } start_of_missing_frame_run = -1; missing_frame_run_count = 0; } @@ -1454,7 +1454,7 @@ static int stuff_buffer_basic_32(int32_t *inptr, int length, sps_format_t l_outp int32_t stat_n = 0; double stat_mean = 0.0; double stat_M2 = 0.0; -double longest_soxr_execution_time_us = 0.0; +double longest_soxr_execution_time = 0.0; int64_t packets_processed = 0; int stuff_buffer_soxr_32(int32_t *inptr, int32_t *scratchBuffer, int length, @@ -1482,7 +1482,7 @@ int stuff_buffer_soxr_32(int32_t *inptr, int32_t *scratchBuffer, int length, size_t odone; - uint64_t soxr_start_time = get_absolute_time_in_fp(); + uint64_t soxr_start_time = get_absolute_time_in_ns(); soxr_error_t error = soxr_oneshot(length, length + tstuff, 2, // Rates and # of chans. inptr, length, NULL, // Input. @@ -1499,15 +1499,15 @@ int stuff_buffer_soxr_32(int32_t *inptr, int32_t *scratchBuffer, int length, // mean and variance calculations from "online_variance" algorithm at // https://en.wikipedia.org/wiki/Algorithms_for_calculating_variance#Online_algorithm - double soxr_execution_time_us = - (((get_absolute_time_in_fp() - soxr_start_time) * 1000000) >> 32) * 1.0; + double soxr_execution_time = + (get_absolute_time_in_ns() - soxr_start_time) * 0.000000001; // debug(1,"soxr_execution_time_us: %10.1f",soxr_execution_time_us); - if (soxr_execution_time_us > longest_soxr_execution_time_us) - longest_soxr_execution_time_us = soxr_execution_time_us; + if (soxr_execution_time > longest_soxr_execution_time) + longest_soxr_execution_time = soxr_execution_time; stat_n += 1; - double stat_delta = soxr_execution_time_us - stat_mean; + double stat_delta = soxr_execution_time - stat_mean; stat_mean += stat_delta / stat_n; - stat_M2 += stat_delta * (soxr_execution_time_us - stat_mean); + stat_M2 += stat_delta * (soxr_execution_time - stat_mean); int i; int32_t *ip, *op; @@ -1553,15 +1553,15 @@ int stuff_buffer_soxr_32(int32_t *inptr, int32_t *scratchBuffer, int length, } if (packets_processed % 1250 == 0) { - debug(3, "soxr_oneshot execution time in microseconds: mean, standard deviation and max " + debug(3, "soxr_oneshot execution time in nanoseconds: mean, standard deviation and max " "for %" PRId32 " interpolations in the last " - "1250 packets. %10.1f, %10.1f, %10.1f.", + "1250 packets. %10.6f, %10.6f, %10.6f.", stat_n, stat_mean, stat_n <= 1 ? 0.0 : sqrtf(stat_M2 / (stat_n - 1)), - longest_soxr_execution_time_us); + longest_soxr_execution_time); stat_n = 0; stat_mean = 0.0; stat_M2 = 0.0; - longest_soxr_execution_time_us = 0.0; + longest_soxr_execution_time = 0.0; } conn->amountStuffed = tstuff; @@ -2365,11 +2365,11 @@ void *player_thread_func(void *arg) { amount_to_stuff = 0; // no stuffing if it's been disabled // Apply DSP here - + // check the state of loudness and convolution flags here and don't change them for the frame - + int do_loudness = config.loudness; - + #ifdef CONFIG_CONVOLUTION int do_convolution = 0; @@ -2377,12 +2377,12 @@ void *player_thread_func(void *arg) { do_convolution = 1; // we will apply the convolution gain if convolution is enabled, even if there is no valid convolution happening - + int convolution_is_enabled = 0; if (config.convolution) convolution_is_enabled = 1; #endif - + if (do_loudness #ifdef CONFIG_CONVOLUTION || convolution_is_enabled @@ -2533,7 +2533,7 @@ void *player_thread_func(void *arg) { // update the watchdog if ((config.dont_check_timeout == 0) && (config.timeout != 0)) { - uint64_t time_now = get_absolute_time_in_fp(); + uint64_t time_now = get_absolute_time_in_ns(); debug_mutex_lock(&conn->watchdog_mutex, 1000, 0); conn->watchdog_bark_time = time_now; debug_mutex_unlock(&conn->watchdog_mutex, 0); @@ -2620,12 +2620,8 @@ void *player_thread_func(void *arg) { conn->frame_rate_status = 0; if (conn->frame_rate_status) { conn->frame_rate = - (1.0 * frames_played) / + (1000000000.0 * frames_played) / elapsed_play_time; // an IEEE double calculation with two 64-bit integers - conn->frame_rate = - conn->frame_rate * (uint64_t)0x100000000; // this should just change the [binary] - // exponent in the IEEE FP representation; - // the mantissa should be unaffected. } else { conn->frame_rate = 0.0; } @@ -2758,7 +2754,7 @@ void player_volume_without_notification(double airplay_volume, rtsp_conn_info *c int32_t hw_max_db = 0, hw_min_db = 0; // zeroed to quieten an incorrect uninitialised warning int32_t sw_max_db = 0, sw_min_db = -9630; - + if (config.output->parameters) { volume_mode = vol_hw_only; audio_parameters audio_information; @@ -2928,7 +2924,7 @@ void player_volume_without_notification(double airplay_volume, rtsp_conn_info *c if (config.logOutputLevel) { inform("Output Level set to: %.2f dB.", scaled_attenuation / 100.0); } - + #ifdef CONFIG_METADATA // here, send the 'pvol' metadata message when the airplay volume information // is being used by shairport sync to control the output volume @@ -2938,7 +2934,7 @@ void player_volume_without_notification(double airplay_volume, rtsp_conn_info *c if (volume_mode == vol_both) { // normalise the maximum output to the hardware device's max output snprintf(dv, 127, "%.2f,%.2f,%.2f,%.2f", airplay_volume, (scaled_attenuation - max_db + hw_max_db) / 100.0, - (min_db - max_db + hw_max_db) / 100.0, (max_db - max_db + hw_max_db) / 100.0); + (min_db - max_db + hw_max_db) / 100.0, (max_db - max_db + hw_max_db) / 100.0); } else { snprintf(dv, 127, "%.2f,%.2f,%.2f,%.2f", airplay_volume, scaled_attenuation / 100.0, min_db / 100.0, max_db / 100.0); @@ -2972,7 +2968,7 @@ void player_volume_without_notification(double airplay_volume, rtsp_conn_info *c } #endif - + // here, store the volume for possible use in the future config.airplay_volume = airplay_volume; debug_mutex_unlock(&conn->volume_control_mutex, 3); diff --git a/rtsp.c b/rtsp.c index 6cf53927..dca0ecb5 100644 --- a/rtsp.c +++ b/rtsp.c @@ -286,7 +286,7 @@ void *player_watchdog_thread_code(void *arg) { uint64_t last_watchdog_bark_time = conn->watchdog_bark_time; debug_mutex_unlock(&conn->watchdog_mutex, 0); if (last_watchdog_bark_time != 0) { - uint64_t time_since_last_bark = (get_absolute_time_in_fp() - last_watchdog_bark_time) >> 32; + uint64_t time_since_last_bark = (get_absolute_time_in_ns() - last_watchdog_bark_time) / 1000000000; uint64_t ct = config.timeout; // go from int to 64-bit int if (time_since_last_bark >= ct) { @@ -666,7 +666,7 @@ enum rtsp_read_request_response rtsp_read_request(rtsp_conn_info *conn, rtsp_mes } uint64_t threshold_time = - get_absolute_time_in_fp() + ((uint64_t)15 << 32); // i.e. fifteen seconds from now + get_absolute_time_in_ns() + ((uint64_t)15000000000); // i.e. fifteen seconds from now int warning_message_sent = 0; const size_t max_read_chunk = 1024 * 1024 / 16; @@ -679,7 +679,7 @@ enum rtsp_read_request_response rtsp_read_request(rtsp_conn_info *conn, rtsp_mes // metadata if (warning_message_sent == 0) { - uint64_t time_now = get_absolute_time_in_fp(); + uint64_t time_now = get_absolute_time_in_ns(); if (time_now > threshold_time) { // it's taking too long debug(1, "Error receiving metadata from source -- transmission seems " "to be stalled."); @@ -2319,7 +2319,7 @@ static void *rtsp_conversation_thread_func(void *pconn) { rtsp_conn_info *conn = pconn; // create the watchdog mutex, initialise the watchdog time and start the watchdog thread; - conn->watchdog_bark_time = get_absolute_time_in_fp(); + conn->watchdog_bark_time = get_absolute_time_in_ns(); pthread_mutex_init(&conn->watchdog_mutex, NULL); pthread_create(&conn->player_watchdog_thread, NULL, &player_watchdog_thread_code, (void *)conn); diff --git a/shairport.c b/shairport.c index 5e375916..881ecd97 100644 --- a/shairport.c +++ b/shairport.c @@ -143,10 +143,10 @@ void *soxr_time_check(__attribute__((unused)) void *arg) { int i; int number_of_iterations = 0; - uint64_t soxr_start_time = get_absolute_time_in_fp(); + uint64_t soxr_start_time = get_absolute_time_in_ns(); uint64_t loop_until_time = - (uint64_t)0x180000000 + soxr_start_time; // loop for a second and a half, max -- no need to be able to cancel it, do _don't even try_! - while (get_absolute_time_in_fp() < loop_until_time) { + (uint64_t)1500000000 + soxr_start_time; // loop for a second and a half, max -- no need to be able to cancel it, do _don't even try_! + while (get_absolute_time_in_ns() < loop_until_time) { number_of_iterations++; for (i = 0; i < buffer_length; i++) { @@ -184,11 +184,11 @@ void *soxr_time_check(__attribute__((unused)) void *arg) { NULL, NULL); // Default configuration. } - double soxr_execution_time_us = - (((get_absolute_time_in_fp() - soxr_start_time) * 1000000) >> 32) * 1.0; + double soxr_execution_time_ns = + (get_absolute_time_in_ns() - soxr_start_time) * 1.0; // free(outbuffer); // free(inbuffer); - config.soxr_delay_index = (int)(0.9 + soxr_execution_time_us / (number_of_iterations * 1000)); + config.soxr_delay_index = (int)(0.9 + soxr_execution_time_ns / (number_of_iterations * 1000000)); debug(2, "soxr_delay_index: %d.", config.soxr_delay_index); if ((config.packet_stuffing == ST_soxr) && (config.soxr_delay_index > config.soxr_delay_threshold)) @@ -1393,8 +1393,8 @@ int main(int argc, char **argv) { conns = NULL; // no connections active memset((void *)&main_thread_id, 0, sizeof(main_thread_id)); memset(&config, 0, sizeof(config)); // also clears all strings, BTW - fp_time_at_startup = get_absolute_time_in_fp(); - fp_time_at_last_debug_message = fp_time_at_startup; + ns_time_at_startup = get_absolute_time_in_ns(); + ns_time_at_last_debug_message = ns_time_at_startup; // this is a bit weird, but necessary -- basename() may modify the argument passed in char *basec = strdup(argv[0]); char *bname = basename(basec); From 32427423af836734c8997247e500c7bd942bf741 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 24 Feb 2020 19:58:35 +0000 Subject: [PATCH 058/186] update packet reception statistics to use nanosecond timing --- rtp.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/rtp.c b/rtp.c index 6a90a5cd..6932420a 100644 --- a/rtp.c +++ b/rtp.c @@ -108,7 +108,7 @@ void *rtp_audio_receiver(void *arg) { int32_t last_seqno = -1; uint8_t packet[2048], *pktp; - uint64_t time_of_previous_packet_fp = 0; + uint64_t time_of_previous_packet_ns = 0; float longest_packet_time_interval_us = 0.0; // mean and variance calculations from "online_variance" algorithm at @@ -125,11 +125,11 @@ void *rtp_audio_receiver(void *arg) { frame_count++; - uint64_t local_time_now_fp = get_absolute_time_in_fp(); - if (time_of_previous_packet_fp) { + uint64_t local_time_now_ns = get_absolute_time_in_ns(); + if (time_of_previous_packet_ns) { float time_interval_us = - (((local_time_now_fp - time_of_previous_packet_fp) * 1000000) >> 32) * 1.0; - time_of_previous_packet_fp = local_time_now_fp; + (local_time_now_ns - time_of_previous_packet_ns) * 0.001; + time_of_previous_packet_ns = local_time_now_ns; if (time_interval_us > longest_packet_time_interval_us) longest_packet_time_interval_us = time_interval_us; stat_n += 1; @@ -143,11 +143,11 @@ void *rtp_audio_receiver(void *arg) { stat_n = 0; stat_mean = 0.0; stat_M2 = 0.0; - time_of_previous_packet_fp = 0; + time_of_previous_packet_ns = 0; longest_packet_time_interval_us = 0.0; } } else { - time_of_previous_packet_fp = local_time_now_fp; + time_of_previous_packet_ns = local_time_now_ns; } if (nread >= 0) { @@ -270,8 +270,8 @@ void *rtp_control_receiver(void *arg) { obfp += 2; }; *obfp = 0; - - + + // get raw timestamp information // I think that a good way to understand these timestamps is that // (1) the rtlt below is the timestamp of the frame that should be playing at the @@ -282,19 +282,19 @@ void *rtp_control_receiver(void *arg) { // Thus, (3) the latency can be calculated by subtracting the second from the // first. // There must be more to it -- there something missing. - + // In addition, it seems that if the value of the short represented by the second // pair of bytes in the packet is 7 // then an extra time lag is expected to be added, presumably by // the AirPort Express. - + // Best guess is that this delay is 11,025 frames. - + uint32_t rtlt = nctohl(&packet[4]); // raw timestamp less latency uint32_t rt = nctohl(&packet[16]); // raw timestamp - + uint32_t fl = nctohs(&packet[2]); // - + debug(1,"Sync Packet of %d bytes received: \"%s\", flags: %d, timestamps %u and %u, giving a latency of %d frames.",plen,obf,fl,rt,rtlt,rt-rtlt); //debug(1,"Monotonic timestamps are: %" PRId64 " and %" PRId64 " From 40289ca6d38bc5969a049e307c5ad875349180f7 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 24 Feb 2020 22:23:08 +0000 Subject: [PATCH 059/186] functional with full removal of get_absolute_time_fp() but not quite starting correctly --- player.c | 63 ++++++++------------ player.h | 2 +- rtp.c | 179 ++++++++++++++++++------------------------------------- 3 files changed, 85 insertions(+), 159 deletions(-) diff --git a/player.c b/player.c index a034346c..0e2a9956 100644 --- a/player.c +++ b/player.c @@ -478,7 +478,7 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in } debug_mutex_lock(&conn->ab_mutex, 30000, 0); - uint64_t time_now = get_absolute_time_in_fp(); + uint64_t time_now = get_absolute_time_in_ns(); conn->packet_count++; conn->packet_count_since_flush++; conn->time_of_last_audio_packet = time_now; @@ -567,10 +567,10 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in // resend checks { - uint64_t minimum_wait_time = (uint64_t)(config.resend_control_first_check_time * (uint64_t)0x100000000); - uint64_t resend_repeat_interval = (uint64_t)(config.resend_control_check_interval_time * (uint64_t)0x100000000); - uint64_t minimum_remaining_time = (uint64_t)((config.resend_control_last_check_time + config.audio_backend_buffer_desired_length)* (uint64_t)0x100000000); - uint64_t latency_time = (uint64_t)(conn->latency * (uint64_t)0x100000000); + uint64_t minimum_wait_time = (uint64_t)(config.resend_control_first_check_time * (uint64_t)1000000000); + uint64_t resend_repeat_interval = (uint64_t)(config.resend_control_check_interval_time * (uint64_t)1000000000); + uint64_t minimum_remaining_time = (uint64_t)((config.resend_control_last_check_time + config.audio_backend_buffer_desired_length)* (uint64_t)1000000000); + uint64_t latency_time = (uint64_t)(conn->latency * (uint64_t)1000000000); latency_time = latency_time / (uint64_t)conn->input_rate; int x; // this is the first frame to be checked @@ -902,7 +902,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { (void *)conn); // undo what's been done so far do { // get the time - local_time_now = get_absolute_time_in_fp(); // type okay + local_time_now = get_absolute_time_in_ns(); // type okay // debug(3, "buffer_get_frame is iterating"); int rco = get_requested_connection_state_to_output(); @@ -1057,8 +1057,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { if (local_time_now > conn->first_packet_time_to_play) { uint64_t lateness = local_time_now - conn->first_packet_time_to_play; - lateness = (lateness * 1000000) >> 32; // microseconds - debug(3, "First packet is %" PRIu64 " microseconds late! Flushing 0.5 seconds", + debug(3, "First packet is %" PRIu64 " nanoseconds late! Flushing 0.5 seconds", lateness); do_flush(conn->first_packet_timestamp + 5 * 4410, conn); } @@ -1097,8 +1096,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { if (local_time_now > conn->first_packet_time_to_play) { uint64_t lateness = local_time_now - conn->first_packet_time_to_play; - lateness = (lateness * 1000000) >> 32; // microseconds - debug(3, "Gone past starting time by %" PRIu64 " microseconds.", lateness); + debug(3, "Gone past starting time by %" PRIu64 " nanoseconds.", lateness); have_sent_prefiller_silence = 1; conn->ab_buffering = 0; @@ -1123,7 +1121,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { int64_t lead_in_time = -1; if (config.audio_backend_silent_lead_in_time >= 0) lead_in_time = - (int64_t)(config.audio_backend_silent_lead_in_time * (int64_t)0x100000000); + (int64_t)(config.audio_backend_silent_lead_in_time * (int64_t)1000000000); // debug(1,"Lead time is %llx at fpttp // %llx.",lead_time,conn->first_packet_time_to_play); if ((lead_in_time < 0) || (lead_time <= lead_in_time)) { @@ -1330,15 +1328,15 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { wait = (conn->ab_buffering || (do_wait != 0) || (!conn->ab_synced)); if (wait) { - uint64_t time_to_wait_for_wakeup_fp = - ((uint64_t)1 << 32) / conn->input_rate; // this is time period of one frame - time_to_wait_for_wakeup_fp *= 2 * 352; // two full 352-frame packets - time_to_wait_for_wakeup_fp /= 3; // two thirds of a packet time + uint64_t time_to_wait_for_wakeup_ns = + 1000000000 / conn->input_rate; // this is time period of one frame + time_to_wait_for_wakeup_ns *= 2 * 352; // two full 352-frame packets + time_to_wait_for_wakeup_ns /= 3; // two thirds of a packet time #ifdef COMPILE_FOR_LINUX_AND_FREEBSD_AND_CYGWIN_AND_OPENBSD - uint64_t time_of_wakeup_fp = local_time_now + time_to_wait_for_wakeup_fp; - uint64_t sec = time_of_wakeup_fp >> 32; - uint64_t nsec = ((time_of_wakeup_fp & 0xffffffff) * 1000000000) >> 32; + uint64_t time_of_wakeup_ns = local_time_now + time_to_wait_for_wakeup_ns; + uint64_t sec = time_of_wakeup_ns / 1000000000; + uint64_t nsec = time_of_wakeup_ns % 1000000000; struct timespec time_of_wakeup; time_of_wakeup.tv_sec = sec; @@ -1350,8 +1348,8 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { debug(3, "pthread_cond_timedwait returned error code %d.", rc); #endif #ifdef COMPILE_FOR_OSX - uint64_t sec = time_to_wait_for_wakeup_fp >> 32; - uint64_t nsec = ((time_to_wait_for_wakeup_fp & 0xffffffff) * 1000000000) >> 32; + uint64_t sec = time_to_wait_for_wakeup_ns / 1000000000; + uint64_t nsec = time_to_wait_for_wakeup_ns % 1000000000; struct timespec time_to_wait; time_to_wait.tv_sec = sec; time_to_wait.tv_nsec = nsec; @@ -2102,11 +2100,7 @@ void *player_thread_func(void *arg) { rt = rt * conn->output_sample_ratio; nt = nt * conn->output_sample_ratio; - uint64_t local_time_now = get_absolute_time_in_fp(); // types okay - // struct timespec tn; - // clock_gettime(CLOCK_MONOTONIC,&tn); - // uint64_t - // local_time_now=((uint64_t)tn.tv_sec<<32)+((uint64_t)tn.tv_nsec<<32)/1000000000; + uint64_t local_time_now = get_absolute_time_in_ns(); // types okay int64_t td = 0; // td is the time difference between the reference timestamp time and the // present time. Only used to calculate td_in_frames @@ -2117,14 +2111,13 @@ void *player_thread_func(void *arg) { td = local_time_now - reference_timestamp_time; // this is the positive value. // Conversion is positive uint64_t to // int64_t, thus okay - td_in_frames = (td * config.output_rate) >> 32; + td_in_frames = (td * config.output_rate) / 1000000000; } else { td = reference_timestamp_time - local_time_now; // this is the absolute value, which // should be negated. Conversion is // positive uint64_t to int64_t, thus // okay. - td_in_frames = (td * config.output_rate) >> - 32; // use the absolute td value for the present. Types okay + td_in_frames = (td * config.output_rate) / 1000000000 ; // use the absolute td value for the present. Types okay td_in_frames = -td_in_frames; td = -td; // should be okay, as the range of values should be very small w.r.t 64 bits } @@ -2346,8 +2339,7 @@ void *player_thread_func(void *arg) { if ((local_time_now) && (conn->first_packet_time_to_play) && (local_time_now >= conn->first_packet_time_to_play)) { - int64_t tp = (local_time_now - conn->first_packet_time_to_play) >> - 32; // seconds int64_t from uint64_t which is always positive, so ok + int64_t tp = (local_time_now - conn->first_packet_time_to_play) / 1000000000; // seconds int64_t from uint64_t which is always positive, so ok if (tp < 5) amount_to_stuff = 0; // wait at least five seconds @@ -2601,14 +2593,9 @@ void *player_thread_func(void *arg) { frames_received = conn->frames_inward_frames_received_at_measurement_time - conn->frames_inward_frames_received_at_measurement_start_time; conn->input_frame_rate = - (1.0 * frames_received) / + (1.0E9 * frames_received) / elapsed_reception_time; // an IEEE double calculation with two 64-bit integers - conn->input_frame_rate = - conn->input_frame_rate * (uint64_t)0x100000000; // this should just change the - // [binary] exponent in the IEEE FP - // representation; the mantissa - // should be unaffected. - } else { + } else { conn->input_frame_rate = 0.0; } @@ -2620,7 +2607,7 @@ void *player_thread_func(void *arg) { conn->frame_rate_status = 0; if (conn->frame_rate_status) { conn->frame_rate = - (1000000000.0 * frames_played) / + (1.0E9 * frames_played) / elapsed_play_time; // an IEEE double calculation with two 64-bit integers } else { conn->frame_rate = 0.0; diff --git a/player.h b/player.h index b6cae8a9..3845ca9f 100644 --- a/player.h +++ b/player.h @@ -177,7 +177,7 @@ typedef struct { // RTP stuff // only one RTP session can be active at a time. int rtp_running; - uint64_t rtp_time_of_last_resend_request_error_fp; + uint64_t rtp_time_of_last_resend_request_error_ns; char client_ip_string[INET6_ADDRSTRLEN]; // the ip string pointing to the client char self_ip_string[INET6_ADDRSTRLEN]; // the ip string being used by this program -- it diff --git a/rtp.c b/rtp.c index 6932420a..70a54a36 100644 --- a/rtp.c +++ b/rtp.c @@ -49,7 +49,7 @@ uint64_t local_to_remote_time_jitter; uint64_t local_to_remote_time_jitter_count; void rtp_initialise(rtsp_conn_info *conn) { - conn->rtp_time_of_last_resend_request_error_fp = 0; + conn->rtp_time_of_last_resend_request_error_ns = 0; conn->rtp_running = 0; // initialise the timer mutex int rc = pthread_mutex_init(&conn->reference_time_mutex, NULL); @@ -69,32 +69,16 @@ uint64_t local_to_remote_time_difference_now(rtsp_conn_info *conn) { // this is an attempt to compensate for clock drift since the last time ping that was used // so, if we have a non-zero clock drift, we will calculate the drift there would // be from the time of the last time ping - uint64_t local_time_now_fp = get_absolute_time_in_fp(); uint64_t time_since_last_local_to_remote_time_difference_measurement = - local_time_now_fp - conn->local_to_remote_time_difference_measurement_time; + get_absolute_time_in_ns() - conn->local_to_remote_time_difference_measurement_time; - uint64_t remote_time_since_last_local_to_remote_time_difference_measurement = - (uint64_t)(conn->local_to_remote_time_gradient * - time_since_last_local_to_remote_time_difference_measurement); - - double drift; - if (remote_time_since_last_local_to_remote_time_difference_measurement >= - time_since_last_local_to_remote_time_difference_measurement) - drift = (1.0 * (remote_time_since_last_local_to_remote_time_difference_measurement - - time_since_last_local_to_remote_time_difference_measurement)) / - (uint64_t)0x100000000; - else - drift = -((1.0 * (time_since_last_local_to_remote_time_difference_measurement - - remote_time_since_last_local_to_remote_time_difference_measurement)) / - (uint64_t)0x100000000); - - // double interval_ms = - // 1.0*(((time_since_last_local_to_remote_time_difference_measurement)*1000)>>32); - // debug(1,"Measurement drift is %.2f microseconds (0x%" PRIx64 " in 64-bit fp) over %.2f - // milliseconds with drift of %.2f - // ppm.",drift*1000000,(uint64_t)(drift*(uint64_t)0x100000000),interval_ms,(1.0-conn->local_to_remote_time_gradient)*1000000); - // return conn->local_to_remote_time_difference + (uint64_t)(drift*(uint64_t 0x100000000)); - return conn->local_to_remote_time_difference + (uint64_t)(drift * (uint64_t)0x100000000); + uint64_t result = conn->local_to_remote_time_difference; + if (conn->local_to_remote_time_gradient >= 1.0) { + result = conn->local_to_remote_time_difference + (uint64_t)((conn->local_to_remote_time_gradient - 1.0) * time_since_last_local_to_remote_time_difference_measurement); + } else { + result = conn->local_to_remote_time_difference - (uint64_t)((1.0 - conn->local_to_remote_time_gradient) * time_since_last_local_to_remote_time_difference_measurement); + } + return result; } void rtp_audio_receiver_cleanup_handler(__attribute__((unused)) void *arg) { @@ -248,9 +232,6 @@ void *rtp_control_receiver(void *arg) { ssize_t nread; while (1) { nread = recv(conn->control_socket, packet, sizeof(packet), 0); - // local_time_now = get_absolute_time_in_fp(); - // clock_gettime(CLOCK_MONOTONIC,&tn); - // local_time_now=((uint64_t)tn.tv_sec<<32)+((uint64_t)tn.tv_nsec<<32)/1000000000; if (nread >= 0) { @@ -303,11 +284,16 @@ void *rtp_control_receiver(void *arg) { */ if (conn->local_to_remote_time_difference) { // need a time packet to be interchanged // first... + uint64_t ps, pn; - remote_time_of_sync = (uint64_t)nctohl(&packet[8]) << 32; - remote_time_of_sync += nctohl(&packet[12]); + ps = nctohl(&packet[8]); + ps = ps * 1000000000; // this many nanoseconds from the whole seconds + pn = nctohl(&packet[12]); + pn = pn * 1000000000; + pn = pn >> 32; // this many nanoseconds from the fractional part + remote_time_of_sync = ps + pn; - // debug(1,"Remote Sync Time: %0llx.",remote_time_of_sync); + // debug(1,"Remote Sync Time: " PRIu64 "",remote_time_of_sync); sync_rtp_timestamp = nctohl(&packet[16]); uint32_t rtp_timestamp_less_latency = nctohl(&packet[4]); @@ -390,15 +376,8 @@ void *rtp_control_receiver(void *arg) { conn->initial_reference_time; // here, this should never be zero if (remote_frame_time_interval) { conn->remote_frame_rate = - (1.0 * (conn->reference_timestamp - conn->initial_reference_timestamp)) / - remote_frame_time_interval; // an IEEE double calculation with a 32-bit - // numerator and 64-bit denominator - // integers - conn->remote_frame_rate = - conn->remote_frame_rate * (uint64_t)0x100000000; // this should just change the - // [binary] exponent in the IEEE - // FP representation; the - // mantissa should be unaffected. + (1.0E9 * (conn->reference_timestamp - conn->initial_reference_timestamp)) / + remote_frame_time_interval; } else { conn->remote_frame_rate = 0.0; // use as a flag. } @@ -423,28 +402,6 @@ void *rtp_control_receiver(void *arg) { else conn->reference_to_previous_frame_difference = sync_rtp_timestamp - old_reference_timestamp; - - // int64_t delayed_frame_difference = rtp_timestamp_less_latency - - // old_latency_delayed_timestamp; - - /* - if (old_remote_reference_time) - debug(1,"Time difference: %" PRIu64 " reference and delayed frame differences: %" - PRId64 " and %" PRId64 ", giving rates _at source!!_ of %f and %f respectively.", - (conn->reference_to_previous_time_difference*1000000)>>32,conn->reference_to_previous_frame_difference,delayed_frame_difference, - (1.0*(conn->reference_to_previous_frame_difference*10000000))/((conn->reference_to_previous_time_difference*10000000)>>32),(1.0*(delayed_frame_difference*10000000))/((conn->reference_to_previous_time_difference*10000000)>>32)); - else - debug(1,"First sync received"); - */ - - // debug(1,"New Reference timestamp and timestamp time..."); - // get estimated remote time now - // remote_time_now = local_time_now + local_to_remote_time_difference; - - // debug(1,"Sync Time is %lld us late (remote - // times).",((remote_time_now-remote_time_of_sync)*1000000)>>32); - // debug(1,"Sync Time is %lld us late (local - // times).",((local_time_now-reference_timestamp_time)*1000000)>>32); } else { debug(2, "Sync packet received before we got a timing packet back."); } @@ -520,8 +477,7 @@ void *rtp_timing_sender(void *arg) { req.filler = 0; req.origin = req.receive = req.transmit = 0; - // clock_gettime(CLOCK_MONOTONIC,&dtt); - conn->departure_time = get_absolute_time_in_fp(); + conn->departure_time = get_absolute_time_in_ns(); socklen_t msgsize = sizeof(struct sockaddr_in); #ifdef AF_INET6 if (conn->rtp_client_timing_socket.SAFAMILY == AF_INET6) { @@ -596,45 +552,39 @@ void *rtp_timing_receiver(void *arg) { if ((config.diagnostic_drop_packet_fraction == 0.0) || (drand48() > config.diagnostic_drop_packet_fraction)) { - arrival_time = get_absolute_time_in_fp(); + arrival_time = get_absolute_time_in_ns(); // ssize_t plen = nread; // debug(1,"Packet Received on Timing Port."); if (packet[1] == 0xd3) { // timing reply - /* - char obf[4096]; - char *obfp = obf; - int obfc; - for (obfc=0;obfcdeparture_time; - // uint64_t rtus = (return_time * 1000000) >> 32; + if (return_time < 300000000) { // must be less than 0.3 seconds - if (((return_time * 1000000) >> 32) < 300000) { - - // debug(2,"Synchronisation ping return time is %f milliseconds.",(rtus*1.0)/1000); + // debug(1,"Synchronisation ping return time is %" PRIu64 " nanoseconds.",return_time); // distant_receive_time = // ((uint64_t)ntohl(*((uint32_t*)&packet[16])))<<32+ntohl(*((uint32_t*)&packet[20])); - distant_receive_time = (uint64_t)nctohl(&packet[16]) << 32; - distant_receive_time += nctohl(&packet[20]); + uint64_t ps, pn; + + ps = nctohl(&packet[16]); + ps = ps * 1000000000; // this many nanoseconds from the whole seconds + pn = nctohl(&packet[20]); + pn = pn * 1000000000; + pn = pn >> 32; // this many nanoseconds from the fractional part + distant_receive_time = ps + pn; // distant_transmit_time = // ((uint64_t)ntohl(*((uint32_t*)&packet[24])))<<32+ntohl(*((uint32_t*)&packet[28])); - distant_transmit_time = (uint64_t)nctohl(&packet[24]) << 32; - distant_transmit_time += nctohl(&packet[28]); + ps = nctohl(&packet[24]); + ps = ps * 1000000000; // this many nanoseconds from the whole seconds + pn = nctohl(&packet[28]); + pn = pn * 1000000000; + pn = pn >> 32; // this many nanoseconds from the fractional part + distant_transmit_time = ps + pn; uint64_t remote_processing_time = 0; @@ -644,8 +594,8 @@ void *rtp_timing_receiver(void *arg) { debug(1, "Yikes: distant_transmit_time is before distant_receive_time; remote " "processing time set to zero."); } - // debug(1,"Return trip time: %" PRIu64 " uS, remote processing time: %" PRIu64 " - // uS.",(return_time*1000000)>>32,(remote_processing_time*1000000)>>32); + // debug(1,"Return trip time: %" PRIu64 " nS, remote processing time: %" PRIu64 " + // nS.",return_time, remote_processing_time); uint64_t local_time_by_remote_clock = distant_transmit_time + return_time / 2; @@ -685,14 +635,13 @@ void *rtp_timing_receiver(void *arg) { // mean and variance calculations from "online_variance" algorithm at // https://en.wikipedia.org/wiki/Algorithms_for_calculating_variance#Online_algorithm - double rtfus = 1.0 * ((return_time * 1000000) >> 32); stat_n += 1; - double stat_delta = rtfus - stat_mean; + double stat_delta = return_time - stat_mean; stat_mean += stat_delta / stat_n; - stat_M2 += stat_delta * (rtfus - stat_mean); + stat_M2 += stat_delta * (return_time - stat_mean); // debug(1, "Timing packet return time stats: current, mean and standard deviation over - // %d packets: %.1f, %.1f, %.1f (microseconds).", - // stat_n,rtfus,stat_mean, sqrtf(stat_M2 / (stat_n - 1))); + // %d packets: %.1f, %.1f, %.1f (nanoseconds).", + // stat_n,return_time,stat_mean, sqrtf(stat_M2 / (stat_n - 1))); // here, pick the record with the least dispersion, and record that it's been chosen @@ -742,16 +691,14 @@ void *rtp_timing_receiver(void *arg) { ltd = -(conn->local_to_remote_time_difference_measurement_time-lt); if (ltd) { - debug(1,"Jitter: %" PRId64 " microseconds in %" PRId64 " microseconds.", (ji * - (int64_t)1000000)>>32, (ltd * (int64_t)1000000)>>32); + debug(1,"Jitter: %" PRId64 " nanoseconds in %" PRId64 " nanoseconds.", ji, ltd); debug(1,"Source clock to local clock drift: %.2f ppm.",((1.0*ji)/ltd)*1000000.0); } // uncomment below to print jitter between client's clock and our clock if (ji) { - int64_t rtus = (tld*1000000)>>32; - debug(1,"Choosing time difference[%d] with dispersion of %" PRId64 " us with an - adjustment of %" PRId64 " us",chosen, rtus, (ji*1000000)>>32); + debug(1,"Choosing time difference[%d] with dispersion of %" PRId64 " ns with an + adjustment of %" PRId64 " ns",chosen, tld, ji); } */ conn->local_to_remote_time_difference = @@ -830,8 +777,8 @@ void *rtp_timing_receiver(void *arg) { // debug(1,"local to remote time gradient is %12.2f ppm, based on %d // samples.",conn->local_to_remote_time_gradient*1000000,sample_count); } else { - debug(2, "Time ping turnaround time: %lld us -- it looks like a timing ping was lost.", - (return_time * 1000000) >> 32); + debug(2, "Time ping turnaround time: " PRIu64 " ns -- it looks like a timing ping was lost.", + return_time); } } else { debug(1, "Timing port -- Unknown RTP packet of type 0x%02X length %d.", packet[1], nread); @@ -1039,9 +986,6 @@ void rtp_setup(SOCKADDR *local, SOCKADDR *remote, uint16_t cport, uint16_t tport conn->local_control_port, conn->local_timing_port); conn->reference_timestamp = 0; - // pthread_create(&rtp_audio_thread, NULL, &rtp_audio_receiver, NULL); - // pthread_create(&rtp_control_thread, NULL, &rtp_control_receiver, NULL); - // pthread_create(&rtp_timing_thread, NULL, &rtp_timing_receiver, NULL); conn->request_sent = 0; conn->rtp_running = 1; @@ -1061,9 +1005,6 @@ void get_reference_timestamp_stuff(uint32_t *timestamp, uint64_t *timestamp_time *remote_timestamp_time = conn->remote_reference_timestamp_time; *timestamp_time = conn->remote_reference_timestamp_time - local_to_remote_time_difference_now(conn); - // if ((*timestamp == 0) && (*timestamp_time == 0)) { - // debug(1,"Reference timestamp is invalid."); - //} debug_mutex_unlock(&conn->reference_time_mutex, 0); } @@ -1088,9 +1029,8 @@ const int use_nominal_rate = 0; // specify whether to use the nominal input rate int sanitised_source_rate_information(uint32_t *frames, uint64_t *time, rtsp_conn_info *conn) { int result = 1; uint32_t fs = conn->input_rate; - *frames = fs; - uint64_t one_fp = (uint64_t)(0x100000000); // one second in fp form - *time = one_fp; + *frames = fs; // default value to return + *time = 1000000000; // default value to return if ((conn->initial_reference_time) && (conn->initial_reference_timestamp)) { // uint32_t local_frames = conn->reference_timestamp - conn->initial_reference_timestamp; uint32_t local_frames = @@ -1101,7 +1041,7 @@ int sanitised_source_rate_information(uint32_t *frames, uint64_t *time, rtsp_con } else { double calculated_frame_rate = conn->input_rate; if (local_time) - calculated_frame_rate = ((1.0 * local_frames) / local_time) * one_fp; + calculated_frame_rate = (1.0E9 * local_frames) / local_time; else debug(1, "sanitised_source_rate_information: local_time is zero"); if ((local_time == 0) || ((calculated_frame_rate / conn->input_rate) > 1.002) || @@ -1173,7 +1113,7 @@ int local_time_to_frame(uint64_t time, uint32_t *frame, rtsp_conn_info *conn) { // here, we calculate the time interval, in terms of remote time uint64_t offset = modulo_64_offset(conn->remote_reference_timestamp_time, remote_time); - int reference_time_was_earlier = (offset <= (uint64_t)0x100000000 * 3600); + int reference_time_was_earlier = (offset <= (uint64_t)3600000000000); if (reference_time_was_earlier) // if we haven't had a reference within the last hour, it'll be // taken as afterwards time_interval = remote_time - conn->remote_reference_timestamp_time; @@ -1217,11 +1157,10 @@ void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn) { msgsize = sizeof(struct sockaddr_in6); } #endif - uint64_t time_of_sending_fp = get_absolute_time_in_fp(); - uint64_t resend_error_backoff_time = (uint64_t)1000000 * 0.3; // 0.3 seconds - resend_error_backoff_time = (resend_error_backoff_time << 32) / 1000000; - if ((conn->rtp_time_of_last_resend_request_error_fp == 0) || - ((time_of_sending_fp - conn->rtp_time_of_last_resend_request_error_fp) > + uint64_t time_of_sending_ns = get_absolute_time_in_ns(); + uint64_t resend_error_backoff_time = 300000000; // 0.3 seconds + if ((conn->rtp_time_of_last_resend_request_error_ns == 0) || + ((time_of_sending_ns - conn->rtp_time_of_last_resend_request_error_ns) > resend_error_backoff_time)) { if ((config.diagnostic_drop_packet_fraction == 0.0) || (drand48() > config.diagnostic_drop_packet_fraction)) { @@ -1241,9 +1180,9 @@ void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn) { strerror_r(errno, em, sizeof(em)); debug(2, "Error %d using sendto to request a resend: \"%s\".", errno, em); - conn->rtp_time_of_last_resend_request_error_fp = time_of_sending_fp; + conn->rtp_time_of_last_resend_request_error_ns = time_of_sending_ns; } else { - conn->rtp_time_of_last_resend_request_error_fp = 0; + conn->rtp_time_of_last_resend_request_error_ns = 0; } } else { @@ -1251,7 +1190,7 @@ void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn) { 3, "Dropping resend request packet to simulate a bad network. Backing off for 0.3 " "second."); - conn->rtp_time_of_last_resend_request_error_fp = time_of_sending_fp; + conn->rtp_time_of_last_resend_request_error_ns = time_of_sending_ns; } } else { debug(1, "Suppressing a resend request due to a resend sendto error in the last 0.3 seconds."); From 8bd6d8d45f58b12818f1ba048898c840f40322ce Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 25 Feb 2020 14:44:13 +0000 Subject: [PATCH 060/186] Fix calculation of initial delay --- player.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/player.c b/player.c index 0e2a9956..483fbee8 100644 --- a/player.c +++ b/player.c @@ -1137,8 +1137,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { resp = config.output->delay(&dac_delay); if (resp == 0) { int64_t gross_frame_gap = - ((conn->first_packet_time_to_play - local_time_now) * config.output_rate) >> - 32; + ((conn->first_packet_time_to_play - local_time_now) * config.output_rate) / 1000000000; int64_t exact_frame_gap = gross_frame_gap - dac_delay; // debug(1,"Exact and gross frame gaps are %" PRId64 " and %" PRId64 " frames, // and the dac delay is %ld.", exact_frame_gap, gross_frame_gap, dac_delay); From c2655b0388b455ddce1f21b1c72f4ce3c17161d8 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 25 Feb 2020 15:53:56 +0000 Subject: [PATCH 061/186] Use nanosecond time in audio_sndio --- audio_sndio.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/audio_sndio.c b/audio_sndio.c index ac5cfa38..e7aaaf42 100644 --- a/audio_sndio.c +++ b/audio_sndio.c @@ -247,7 +247,7 @@ static void stop() { } static void onmove_cb(__attribute__((unused)) void *arg, int delta) { - time_of_last_onmove_cb = get_absolute_time_in_fp(); + time_of_last_onmove_cb = get_absolute_time_in_ns(); at_least_one_onmove_cb_seen = 1; played += delta; } @@ -258,10 +258,14 @@ static int delay(long *_delay) { if (at_least_one_onmove_cb_seen) { // when output starts, the onmove_cb callback will be made // calculate the difference in time between now and when the last callback occurred, // and use it to estimate the frames that would have been output - uint64_t time_difference = get_absolute_time_in_fp() - time_of_last_onmove_cb; - uint64_t frame_difference = time_difference * par.rate; - uint64_t frame_difference_big_integer = frame_difference >> 32; - estimated_extra_frames_output = frame_difference_big_integer; + uint64_t time_difference = get_absolute_time_in_ns() - time_of_last_onmove_cb; + uint64_t frame_difference = (time_difference * par.rate) / 1000000000; + estimated_extra_frames_output = frame_difference; + // sanity check -- total estimate can not exceed frames written. + if ((estimated_extra_frames_output + played) > written/framesize) { + // debug(1,"play estimate fails sanity check, possibly due to running on a VM"); + estimated_extra_frames_output = 0; // can't make any sensible guess + } // debug(1,"Frames played to last cb: %d, estimated to current time: // %d.",played,estimated_extra_frames_output); } From e1028b2c896f72a6d91f612187317dffb454c6e0 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 25 Feb 2020 15:56:53 +0000 Subject: [PATCH 062/186] Hide get_absolute_time_in_fp as it is now obselete --- common.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/common.h b/common.h index 92095387..b9e9cb52 100644 --- a/common.h +++ b/common.h @@ -349,7 +349,7 @@ double flat_vol2attn(double vol, long max_db, long min_db); double vol2attn(double vol, long max_db, long min_db); // return a monolithic (always increasing) time in nanoseconds -uint64_t get_absolute_time_in_fp(void); +// uint64_t get_absolute_time_in_fp(void); // obselete uint64_t get_absolute_time_in_ns(void); // time at startup for debugging timing From 514a21e911c812cf110abcc5bd6eff4e9fcd2cb2 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 25 Feb 2020 16:04:55 +0000 Subject: [PATCH 063/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index d0146e3d..e8c62920 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,15 @@ +Version 3.3.7d2 +==== +**Pesky Changes** +* The underlying timing system has been moved from 64-bit fixed-point time representation (a là NTP) to a 64-bit unsigned nanoseconds representation. This should make precisely no difference to the functionality of Shairport Sync but the transition might behabve inadeventely introduced bugs. Problem reports gratefully received. + +**Enhancement** +* The timing software in the `sndio` backend does some extra sanity checking on certain time estimates, it may help a little whne running on virtual machines. + +Version 3.3.7d1 +==== +This is 3.3.6 with some documentation updates. + Version 3.3.6d11 = Version 3.3.6 ==== **Bug Fix** From db34ec8d3cbc6ac39ec3d0d525d7fbe75ae6910e Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 25 Feb 2020 16:13:39 +0000 Subject: [PATCH 064/186] Move from fixed-point 64-bit timing to unsigned nanosecond timing --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index c1e393f7..1abafedc 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.7d1], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.7d2], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 218a4385f49b3a4cc0856bfccd28a050e86d7767 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 24 Mar 2020 15:57:16 +0000 Subject: [PATCH 065/186] Tidy up timed mutex messages. --- common.c | 8 ++++---- common.h | 3 --- 2 files changed, 4 insertions(+), 7 deletions(-) diff --git a/common.c b/common.c index b3942f78..2958cfe6 100644 --- a/common.c +++ b/common.c @@ -1247,9 +1247,9 @@ int sps_pthread_mutex_timedlock(pthread_mutex_t *mutex, useconds_t dally_time, char errstr[1000]; if (r == ETIMEDOUT) debug(debuglevel, - "timed out waiting for a mutex, having waiting %f seconds, with a maximum " + "timed out waiting for a mutex, having waited %f microseconds, with a maximum " "waiting time of %d microseconds. \"%s\".", - (1.0 * et) / 1000000000, dally_time, debugmessage); + (1.0E6 * et) / 1000000000, dally_time, debugmessage); else debug(debuglevel, "error %d: \"%s\" waiting for a mutex: \"%s\".", r, strerror_r(r, errstr, sizeof(errstr)), debugmessage); @@ -1309,8 +1309,8 @@ int _debug_mutex_lock(pthread_mutex_t *mutex, useconds_t dally_time, const char result = pthread_mutex_lock(mutex); uint64_t time_delay = get_absolute_time_in_ns() - time_at_start; debug(debuglevel, - "mutex_lock \"%s\" at \"%s\" expected max wait: %0.9f, actual wait: %0.9f sec.", - mutexname, dstring, (1.0 * dally_time) / 1000000, 0.000000001 * time_delay); + "mutex_lock \"%s\" at \"%s\" expected max wait: %0.9f, actual wait: %0.9f microseconds.", + mutexname, dstring, (1.0 * dally_time), 0.001 * time_delay); } pthread_setcancelstate(oldState, NULL); return result; diff --git a/common.h b/common.h index b9e9cb52..ba20078a 100644 --- a/common.h +++ b/common.h @@ -421,9 +421,6 @@ extern pthread_mutex_t r64_mutex; char *get_version_string(); // mallocs a string space -- remember to free it afterwards -void sps_nanosleep(const time_t sec, - const long nanosec); // waits for this time, even through interruptions - int64_t generate_zero_frames(char *outp, size_t number_of_frames, sps_format_t format, int with_dither, int64_t random_number_in); From a6acb6062afa51a6550d7dde872269000f80aaf7 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 25 Mar 2020 10:16:21 +0000 Subject: [PATCH 066/186] If the DACP port is zero, do a zeroconf re-broswe every two seconds in case it has been missed. --- common.h | 1 + dacp.c | 753 ++++++++++++++++++++++++++++------------------------ rtsp.c | 1 + shairport.c | 1 + 4 files changed, 404 insertions(+), 352 deletions(-) diff --git a/common.h b/common.h index ba20078a..95f4decc 100644 --- a/common.h +++ b/common.h @@ -100,6 +100,7 @@ typedef enum { const char *sps_format_description_string(sps_format_t format); typedef struct { + double missing_port_dacp_scan_interval_seconds; // if no DACP port number can be found, check at these intervals double resend_control_first_check_time; // wait this long before asking for a missing packet to be resent double resend_control_check_interval_time; // wait this long between making requests double resend_control_last_check_time; // if the packet is missing this close to the time of use, give up diff --git a/dacp.c b/dacp.c index 4c36de34..dc794d4d 100644 --- a/dacp.c +++ b/dacp.c @@ -1,6 +1,6 @@ /* * DACP protocol handler. This file is part of Shairport Sync. - * Copyright (c) Mike Brady 2017 -- 2019 + * Copyright (c) Mike Brady 2017 -- 2020 * All rights reserved. * * Permission is hereby granted, free of charge, to any person @@ -49,7 +49,7 @@ typedef struct { int players_connection_thread_index; // the connection thread index when a player thread is // associated with this, zero otherwise - int scan_enable; // set to 1 if if sacanning should be considered + int scan_enable; // set to 1 if scanning should be considered char dacp_id[256]; // the DACP ID string uint16_t port; // zero if no port discovered short connection_family; // AF_INET6 or AF_INET @@ -123,7 +123,7 @@ static const struct http_funcs responseFuncs = { // static pthread_mutex_t dacp_server_information_lock = PTHREAD_MUTEX_INITIALIZER; static pthread_mutex_t dacp_conversation_lock; static pthread_mutex_t dacp_server_information_lock; -static pthread_cond_t dacp_server_information_cv = PTHREAD_COND_INITIALIZER; +static pthread_cond_t dacp_server_information_cv; void addrinfo_cleanup(void *arg) { // debug(1, "addrinfo cleanup called."); @@ -154,7 +154,7 @@ int dacp_send_command(const char *command, char **body, ssize_t *bodysize) { // debug(1,"dacp_send_command: command is: \"%s\".",command); if (dacp_server.port == 0) { - debug(1, "No DACP port specified yet"); + debug(2, "No DACP port specified yet"); result = 490; // no port specified } else { @@ -169,6 +169,7 @@ int dacp_send_command(const char *command, char **body, ssize_t *bodysize) { // 493 Client failed to send a message // 492 Argument out of range // 491 Client refused connection + // 490 No port specified struct addrinfo hints, *res; int sockfd; @@ -421,9 +422,6 @@ void set_dacp_server_information(rtsp_conn_info *conn) { dacp_server.always_use_revision_number_1 = 1; } - - mdns_dacp_monitor_set_id(dacp_server.dacp_id); - metadata_hub_modify_prolog(); int ch = metadata_store.dacp_server_active != dacp_server.scan_enable; metadata_store.dacp_server_active = dacp_server.scan_enable; @@ -513,14 +511,40 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { "flag value of %d", ch); metadata_hub_modify_epilog(ch); - while (dacp_server.scan_enable == 0) { - // debug(1, "dacp_monitor_thread_code wait for an event to possible enable scan"); - pthread_cond_wait(&dacp_server_information_cv, &dacp_server_information_lock); - // debug(1,"dacp_monitor_thread_code wake up."); + + uint64_t time_to_wait_for_wakeup_ns = (uint64_t)(1000000000 * config.missing_port_dacp_scan_interval_seconds); + +#ifdef COMPILE_FOR_LINUX_AND_FREEBSD_AND_CYGWIN_AND_OPENBSD + uint64_t time_of_wakeup_ns = get_absolute_time_in_ns() + time_to_wait_for_wakeup_ns; + uint64_t sec = time_of_wakeup_ns / 1000000000; + uint64_t nsec = time_of_wakeup_ns % 1000000000; +#endif +#ifdef COMPILE_FOR_OSX + uint64_t sec = time_to_wait_for_wakeup_ns / 1000000000; + uint64_t nsec = time_to_wait_for_wakeup_ns % 1000000000; +#endif + + struct timespec time_to_wait; + time_to_wait.tv_sec = sec; + time_to_wait.tv_nsec = nsec; + + while ((dacp_server.scan_enable == 0) && (result != ETIMEDOUT)) { + // debug(1, "dacp_monitor_thread_code wait for an event to possibly enable scan"); + +#ifdef COMPILE_FOR_LINUX_AND_FREEBSD_AND_CYGWIN_AND_OPENBSD + result = pthread_cond_timedwait(&dacp_server_information_cv, &dacp_server_information_lock, + &time_to_wait); // this is a pthread cancellation point + // debug(1, "result is %d, dacp_server.scan_enable is %d, time_to_wait_for_wakeup_ns is %" PRId64 ".", result, dacp_server.scan_enable, time_to_wait_for_wakeup_ns); + +#endif +#ifdef COMPILE_FOR_OSX + result = pthread_cond_timedwait_relative_np(&dacp_server_information_cv, &dacp_server_information_lock, &time_to_wait); +#endif + } + if (dacp_server.scan_enable == 1) { + bad_result_count = 0; + idle_scan_count = 0; } - // so dacp_server.scan_enable will be true at this point - bad_result_count = 0; - idle_scan_count = 0; } always_use_revision_number_1 = dacp_server.always_use_revision_number_1; // set this while access is locked @@ -528,360 +552,367 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { result = dacp_get_volume(&the_volume); // just want the http code pthread_cleanup_pop(1); - scan_index++; - // debug(1,"DACP Scan Result: %d.", result); - - if ((result == 200) || (result == 400)) { - bad_result_count = 0; + if (result == 490) { // 490 means no port was specified + if ((dacp_server.dacp_id != NULL) && (strlen(dacp_server.dacp_id) != 0)) { + // debug(1,"mdns_dacp_monitor_set_id"); + mdns_dacp_monitor_set_id(dacp_server.dacp_id); + } } else { - if (bad_result_count < config.scan_max_bad_response_count) // limit to some reasonable value - bad_result_count++; - } + scan_index++; + // debug(1,"DACP Scan Result: %d.", result); - // here, do the debouncing calculations to see - // if the dacp server and the - // advanced dacp server are available + if ((result == 200) || (result == 400)) { + bad_result_count = 0; + } else { + if (bad_result_count < config.scan_max_bad_response_count) // limit to some reasonable value + bad_result_count++; + } - // -1 means we don't know because some bad statuses have been reported - // 0 means definitely no - // +1 means definitely yes + // here, do the debouncing calculations to see + // if the dacp server and the + // advanced dacp server are available - int dacp_server_status_now = -1; - int advanced_dacp_server_status_now = -1; + // -1 means we don't know because some bad statuses have been reported + // 0 means definitely no + // +1 means definitely yes - if (bad_result_count == 0) { - dacp_server_status_now = 1; - if (result == 200) - advanced_dacp_server_status_now = 1; - else if (result == 400) + int dacp_server_status_now = -1; + int advanced_dacp_server_status_now = -1; + + if (bad_result_count == 0) { + dacp_server_status_now = 1; + if (result == 200) + advanced_dacp_server_status_now = 1; + else if (result == 400) + advanced_dacp_server_status_now = 0; + } else if (bad_result_count == + config.scan_max_bad_response_count) { // if a sequence of bad return codes occurs, + // then it's gone + dacp_server_status_now = 0; advanced_dacp_server_status_now = 0; - } else if (bad_result_count == - config.scan_max_bad_response_count) { // if a sequence of bad return codes occurs, - // then it's gone - dacp_server_status_now = 0; - advanced_dacp_server_status_now = 0; - } - - if (metadata_store.player_thread_active == 0) - idle_scan_count++; - else - idle_scan_count = 0; - - debug(3, "Scan Result: %d, Bad Scan Count: %d, Idle Scan Count: %d.", result, bad_result_count, - idle_scan_count); - - /* not used - // decide if idle for too long - if (idle_scan_count == config.scan_max_inactive_count) { - debug(2, "DACP server status scanning stopped."); - dacp_server.scan_enable = 0; - } - */ - - int update_needed = 0; - metadata_hub_modify_prolog(); - if (dacp_server_status_now != -1) { // if dacp_server_status_now is actually known... - if (metadata_store.dacp_server_active != dacp_server_status_now) { - debug(2, "metadata_store.dacp_server_active set to %d.", dacp_server_status_now); - metadata_store.dacp_server_active = dacp_server_status_now; - update_needed = 1; } - } - if (advanced_dacp_server_status_now != - -1) { // if advanced_dacp_server_status_now is actually known... - if (metadata_store.advanced_dacp_server_active != advanced_dacp_server_status_now) { - debug(2, "metadata_store.advanced_dacp_server_active set to %d.", dacp_server_status_now); - metadata_store.advanced_dacp_server_active = advanced_dacp_server_status_now; - update_needed = 1; - } - } - metadata_hub_modify_epilog(update_needed); + if (metadata_store.player_thread_active == 0) + idle_scan_count++; + else + idle_scan_count = 0; - // pthread_mutex_unlock(&dacp_server_information_lock); - // debug(1, "DACP Server ID \"%u\" at \"%s:%u\", scan %d.", dacp_server.active_remote_id, - // dacp_server.ip_string, dacp_server.port, scan_index); + debug(3, "Scan Result: %d, Bad Scan Count: %d, Idle Scan Count: %d.", result, bad_result_count, + idle_scan_count); - if (result == 200) { - metadata_hub_modify_prolog(); - int diff = metadata_store.speaker_volume != the_volume; - if (diff) - metadata_store.speaker_volume = the_volume; - metadata_hub_modify_epilog(diff); - - ssize_t le; - char *response = NULL; - int32_t item_size; - char command[1024] = ""; - if (always_use_revision_number_1 != 0) // for forked-daapd - revision_number = 1; - snprintf(command, sizeof(command) - 1, "playstatusupdate?revision-number=%d", - revision_number); - // debug(1,"dacp_monitor_thread_code: command: \"%s\"",command); - result = dacp_send_command(command, &response, &le); - // debug(1,"Response to \"%s\" is %d.",command,result); - // remember: unless the revision_number you pass in is 1, - // response will be 200 only if there's something new to report. - if (result == 200) { - // if (0) { - char *sp = response; - if (le >= 8) { - // here start looking for the contents of the status update - if (dacp_tlv_crawl(&sp, &item_size) == 'cmst') { // status - // here, we know that we are receiving playerstatusupdates, so set a flag - metadata_hub_modify_prolog(); - // debug(1, "playstatusupdate release track metadata"); - // metadata_hub_reset_track_metadata(); - // metadata_store.playerstatusupdates_are_received = 1; - sp -= item_size; // drop down into the array -- don't skip over it - le -= 8; - // char typestring[5]; - // we need to acquire the metadata data structure and possibly update it - while (le >= 8) { - uint32_t type = dacp_tlv_crawl(&sp, &item_size); - le -= item_size + 8; - char *t; - // char u; - // char *st; - int32_t r; - uint32_t ui; - // uint64_t v; - // int i; - - switch (type) { - case 'cmsr': // revision number - t = sp - item_size; - revision_number = ntohl(*(uint32_t *)(t)); - // debug(1,"New revision number received: %d", revision_number); - break; - case 'caps': // play status - t = sp - item_size; - r = *(unsigned char *)(t); - switch (r) { - case 2: - if (metadata_store.play_status != PS_STOPPED) { - metadata_store.play_status = PS_STOPPED; - debug(2, "Play status is \"stopped\"."); - } - break; - case 3: - if (metadata_store.play_status != PS_PAUSED) { - metadata_store.play_status = PS_PAUSED; - debug(2, "Play status is \"paused\"."); - } - break; - case 4: - if (metadata_store.play_status != PS_PLAYING) { - metadata_store.play_status = PS_PLAYING; - debug(2, "Play status changed to \"playing\"."); - } - break; - default: - debug(1, "Unrecognised play status %d received.", r); - break; - } - break; - case 'cash': // shuffle status - t = sp - item_size; - r = *(unsigned char *)(t); - switch (r) { - case 0: - if (metadata_store.shuffle_status != SS_OFF) { - metadata_store.shuffle_status = SS_OFF; - debug(2, "Shuffle status is \"off\"."); - } - break; - case 1: - if (metadata_store.shuffle_status != SS_ON) { - metadata_store.shuffle_status = SS_ON; - debug(2, "Shuffle status is \"on\"."); - } - break; - default: - debug(1, "Unrecognised shuffle status %d received.", r); - break; - } - break; - case 'carp': // repeat status - t = sp - item_size; - r = *(unsigned char *)(t); - switch (r) { - case 0: - if (metadata_store.repeat_status != RS_OFF) { - metadata_store.repeat_status = RS_OFF; - debug(2, "Repeat status is \"none\"."); - } - break; - case 1: - if (metadata_store.repeat_status != RS_ONE) { - metadata_store.repeat_status = RS_ONE; - debug(2, "Repeat status is \"one\"."); - } - break; - case 2: - if (metadata_store.repeat_status != RS_ALL) { - metadata_store.repeat_status = RS_ALL; - debug(2, "Repeat status is \"all\"."); - } - break; - default: - debug(1, "Unrecognised repeat status %d received.", r); - break; - } - break; - case 'cann': // track name - debug(2, "DACP Track Name seen"); - if (string_update_with_size(&metadata_store.track_name, - &metadata_store.track_name_changed, sp - item_size, - item_size)) { - debug(2, "DACP Track Name set to: \"%s\"", metadata_store.track_name); - } - break; - case 'cana': // artist name - debug(2, "DACP Artist Name seen"); - if (string_update_with_size(&metadata_store.artist_name, - &metadata_store.artist_name_changed, sp - item_size, - item_size)) { - debug(2, "DACP Artist Name set to: \"%s\"", metadata_store.artist_name); - } - break; - case 'canl': // album name - debug(2, "DACP Album Name seen"); - if (string_update_with_size(&metadata_store.album_name, - &metadata_store.album_name_changed, sp - item_size, - item_size)) { - debug(2, "DACP Album Name set to: \"%s\"", metadata_store.album_name); - } - break; - case 'cang': // genre - debug(2, "DACP Genre seen"); - if (string_update_with_size(&metadata_store.genre, &metadata_store.genre_changed, - sp - item_size, item_size)) { - debug(2, "DACP Genre set to: \"%s\"", metadata_store.genre); - } - break; - case 'canp': // nowplaying 4 ids: dbid, plid, playlistItem, itemid (from mellowware - // see reference above) - debug(2, "DACP Composite ID seen"); - if (memcmp(metadata_store.item_composite_id, sp - item_size, - sizeof(metadata_store.item_composite_id)) != 0) { - memcpy(metadata_store.item_composite_id, sp - item_size, - sizeof(metadata_store.item_composite_id)); - char st[33]; - char *pt = st; - int it; - for (it = 0; it < 16; it++) { - snprintf(pt, 3, "%02X", metadata_store.item_composite_id[it]); - pt += 2; - } - *pt = 0; - debug(2, "Item composite ID changed to 0x%s.", st); - metadata_store.item_composite_id_changed = 1; - } - break; - case 'astm': - t = sp - item_size; - ui = ntohl(*(uint32_t *)(t)); - debug(2, "DACP Song Time seen: \"%u\" of length %u.", ui, item_size); - if (ui != metadata_store.songtime_in_milliseconds) { - metadata_store.songtime_in_milliseconds = ui; - metadata_store.songtime_in_milliseconds_changed = 1; - debug(2, "DACP Song Time set to: \"%u\"", - metadata_store.songtime_in_milliseconds); - } - break; - - /* - case 'mstt': - case 'cant': - case 'cast': - case 'cmmk': - case 'caas': - case 'caar': - t = sp - item_size; - r = ntohl(*(uint32_t *)(t)); - printf(" %d", r); - printf(" (0x"); - t = sp - item_size; - for (i = 0; i < item_size; i++) { - printf("%02x", *t & 0xff); - t++; - } - printf(")"); - break; - case 'asai': - t = sp - item_size; - s = ntohl(*(uint32_t *)(t)); - s = s << 32; - t += 4; - v = (ntohl(*(uint32_t *)(t))) & 0xffffffff; - s += v; - printf(" %lu", s); - printf(" (0x"); - t = sp - item_size; - for (i = 0; i < item_size; i++) { - printf("%02x", *t & 0xff); - t++; - } - printf(")"); - break; - */ - default: - /* - printf(" 0x"); - t = sp - item_size; - for (i = 0; i < item_size; i++) { - printf("%02x", *t & 0xff); - t++; - } - */ - break; - } - // printf("\n"); - } - - // finished possibly writing to the metadata hub - metadata_hub_modify_epilog( - 1); // should really see if this can be made responsive to changes - } else { - debug(1, "Status Update not found.\n"); + /* not used + // decide if idle for too long + if (idle_scan_count == config.scan_max_inactive_count) { + debug(2, "DACP server status scanning stopped."); + dacp_server.scan_enable = 0; } - } else { - debug(1, "Can't find any content in playerstatusupdate request"); + */ + + int update_needed = 0; + metadata_hub_modify_prolog(); + if (dacp_server_status_now != -1) { // if dacp_server_status_now is actually known... + if (metadata_store.dacp_server_active != dacp_server_status_now) { + debug(2, "metadata_store.dacp_server_active set to %d.", dacp_server_status_now); + metadata_store.dacp_server_active = dacp_server_status_now; + update_needed = 1; } - } /* else { - if (result != 403) - debug(1, "Unexpected response %d to playerstatusupdate request", result); - } */ + } + if (advanced_dacp_server_status_now != + -1) { // if advanced_dacp_server_status_now is actually known... + if (metadata_store.advanced_dacp_server_active != advanced_dacp_server_status_now) { + debug(2, "metadata_store.advanced_dacp_server_active set to %d.", dacp_server_status_now); + metadata_store.advanced_dacp_server_active = advanced_dacp_server_status_now; + update_needed = 1; + } + } + + metadata_hub_modify_epilog(update_needed); + + // pthread_mutex_unlock(&dacp_server_information_lock); + // debug(1, "DACP Server ID \"%u\" at \"%s:%u\", scan %d.", dacp_server.active_remote_id, + // dacp_server.ip_string, dacp_server.port, scan_index); + + if (result == 200) { + metadata_hub_modify_prolog(); + int diff = metadata_store.speaker_volume != the_volume; + if (diff) + metadata_store.speaker_volume = the_volume; + metadata_hub_modify_epilog(diff); + + ssize_t le; + char *response = NULL; + int32_t item_size; + char command[1024] = ""; + if (always_use_revision_number_1 != 0) // for forked-daapd + revision_number = 1; + snprintf(command, sizeof(command) - 1, "playstatusupdate?revision-number=%d", + revision_number); + // debug(1,"dacp_monitor_thread_code: command: \"%s\"",command); + result = dacp_send_command(command, &response, &le); + // debug(1,"Response to \"%s\" is %d.",command,result); + // remember: unless the revision_number you pass in is 1, + // response will be 200 only if there's something new to report. + if (result == 200) { + // if (0) { + char *sp = response; + if (le >= 8) { + // here start looking for the contents of the status update + if (dacp_tlv_crawl(&sp, &item_size) == 'cmst') { // status + // here, we know that we are receiving playerstatusupdates, so set a flag + metadata_hub_modify_prolog(); + // debug(1, "playstatusupdate release track metadata"); + // metadata_hub_reset_track_metadata(); + // metadata_store.playerstatusupdates_are_received = 1; + sp -= item_size; // drop down into the array -- don't skip over it + le -= 8; + // char typestring[5]; + // we need to acquire the metadata data structure and possibly update it + while (le >= 8) { + uint32_t type = dacp_tlv_crawl(&sp, &item_size); + le -= item_size + 8; + char *t; + // char u; + // char *st; + int32_t r; + uint32_t ui; + // uint64_t v; + // int i; + + switch (type) { + case 'cmsr': // revision number + t = sp - item_size; + revision_number = ntohl(*(uint32_t *)(t)); + // debug(1,"New revision number received: %d", revision_number); + break; + case 'caps': // play status + t = sp - item_size; + r = *(unsigned char *)(t); + switch (r) { + case 2: + if (metadata_store.play_status != PS_STOPPED) { + metadata_store.play_status = PS_STOPPED; + debug(2, "Play status is \"stopped\"."); + } + break; + case 3: + if (metadata_store.play_status != PS_PAUSED) { + metadata_store.play_status = PS_PAUSED; + debug(2, "Play status is \"paused\"."); + } + break; + case 4: + if (metadata_store.play_status != PS_PLAYING) { + metadata_store.play_status = PS_PLAYING; + debug(2, "Play status changed to \"playing\"."); + } + break; + default: + debug(1, "Unrecognised play status %d received.", r); + break; + } + break; + case 'cash': // shuffle status + t = sp - item_size; + r = *(unsigned char *)(t); + switch (r) { + case 0: + if (metadata_store.shuffle_status != SS_OFF) { + metadata_store.shuffle_status = SS_OFF; + debug(2, "Shuffle status is \"off\"."); + } + break; + case 1: + if (metadata_store.shuffle_status != SS_ON) { + metadata_store.shuffle_status = SS_ON; + debug(2, "Shuffle status is \"on\"."); + } + break; + default: + debug(1, "Unrecognised shuffle status %d received.", r); + break; + } + break; + case 'carp': // repeat status + t = sp - item_size; + r = *(unsigned char *)(t); + switch (r) { + case 0: + if (metadata_store.repeat_status != RS_OFF) { + metadata_store.repeat_status = RS_OFF; + debug(2, "Repeat status is \"none\"."); + } + break; + case 1: + if (metadata_store.repeat_status != RS_ONE) { + metadata_store.repeat_status = RS_ONE; + debug(2, "Repeat status is \"one\"."); + } + break; + case 2: + if (metadata_store.repeat_status != RS_ALL) { + metadata_store.repeat_status = RS_ALL; + debug(2, "Repeat status is \"all\"."); + } + break; + default: + debug(1, "Unrecognised repeat status %d received.", r); + break; + } + break; + case 'cann': // track name + debug(2, "DACP Track Name seen"); + if (string_update_with_size(&metadata_store.track_name, + &metadata_store.track_name_changed, sp - item_size, + item_size)) { + debug(2, "DACP Track Name set to: \"%s\"", metadata_store.track_name); + } + break; + case 'cana': // artist name + debug(2, "DACP Artist Name seen"); + if (string_update_with_size(&metadata_store.artist_name, + &metadata_store.artist_name_changed, sp - item_size, + item_size)) { + debug(2, "DACP Artist Name set to: \"%s\"", metadata_store.artist_name); + } + break; + case 'canl': // album name + debug(2, "DACP Album Name seen"); + if (string_update_with_size(&metadata_store.album_name, + &metadata_store.album_name_changed, sp - item_size, + item_size)) { + debug(2, "DACP Album Name set to: \"%s\"", metadata_store.album_name); + } + break; + case 'cang': // genre + debug(2, "DACP Genre seen"); + if (string_update_with_size(&metadata_store.genre, &metadata_store.genre_changed, + sp - item_size, item_size)) { + debug(2, "DACP Genre set to: \"%s\"", metadata_store.genre); + } + break; + case 'canp': // nowplaying 4 ids: dbid, plid, playlistItem, itemid (from mellowware + // see reference above) + debug(2, "DACP Composite ID seen"); + if (memcmp(metadata_store.item_composite_id, sp - item_size, + sizeof(metadata_store.item_composite_id)) != 0) { + memcpy(metadata_store.item_composite_id, sp - item_size, + sizeof(metadata_store.item_composite_id)); + char st[33]; + char *pt = st; + int it; + for (it = 0; it < 16; it++) { + snprintf(pt, 3, "%02X", metadata_store.item_composite_id[it]); + pt += 2; + } + *pt = 0; + debug(2, "Item composite ID changed to 0x%s.", st); + metadata_store.item_composite_id_changed = 1; + } + break; + case 'astm': + t = sp - item_size; + ui = ntohl(*(uint32_t *)(t)); + debug(2, "DACP Song Time seen: \"%u\" of length %u.", ui, item_size); + if (ui != metadata_store.songtime_in_milliseconds) { + metadata_store.songtime_in_milliseconds = ui; + metadata_store.songtime_in_milliseconds_changed = 1; + debug(2, "DACP Song Time set to: \"%u\"", + metadata_store.songtime_in_milliseconds); + } + break; + + /* + case 'mstt': + case 'cant': + case 'cast': + case 'cmmk': + case 'caas': + case 'caar': + t = sp - item_size; + r = ntohl(*(uint32_t *)(t)); + printf(" %d", r); + printf(" (0x"); + t = sp - item_size; + for (i = 0; i < item_size; i++) { + printf("%02x", *t & 0xff); + t++; + } + printf(")"); + break; + case 'asai': + t = sp - item_size; + s = ntohl(*(uint32_t *)(t)); + s = s << 32; + t += 4; + v = (ntohl(*(uint32_t *)(t))) & 0xffffffff; + s += v; + printf(" %lu", s); + printf(" (0x"); + t = sp - item_size; + for (i = 0; i < item_size; i++) { + printf("%02x", *t & 0xff); + t++; + } + printf(")"); + break; + */ + default: + /* + printf(" 0x"); + t = sp - item_size; + for (i = 0; i < item_size; i++) { + printf("%02x", *t & 0xff); + t++; + } + */ + break; + } + // printf("\n"); + } + + // finished possibly writing to the metadata hub + metadata_hub_modify_epilog( + 1); // should really see if this can be made responsive to changes + } else { + debug(1, "Status Update not found.\n"); + } + } else { + debug(1, "Can't find any content in playerstatusupdate request"); + } + } /* else { + if (result != 403) + debug(1, "Unexpected response %d to playerstatusupdate request", result); + } */ + if (response) { + free(response); + response = NULL; + }; + }; + /* + strcpy(command,"nowplayingartwork?mw=320&mh=320"); + debug(1,"Command: \"%s\", result is %d",command, dacp_send_command(command, &response, &le)); if (response) { free(response); response = NULL; - }; - }; - /* - strcpy(command,"nowplayingartwork?mw=320&mh=320"); - debug(1,"Command: \"%s\", result is %d",command, dacp_send_command(command, &response, &le)); - if (response) { - free(response); - response = NULL; + } + strcpy(command,"getproperty?properties=dmcp.volume"); + debug(1,"Command: \"%s\", result is %d",command, dacp_send_command(command, &response, &le)); + if (response) { + free(response); + response = NULL; + } + strcpy(command,"setproperty?dmcp.volume=100.000000"); + debug(1,"Command: \"%s\", result is %d",command, dacp_send_command(command, &response, &le)); + if (response) { + free(response); + response = NULL; + } + */ + if (metadata_store.player_thread_active) + sleep(config.scan_interval_when_active); + else + sleep(config.scan_interval_when_inactive); } - strcpy(command,"getproperty?properties=dmcp.volume"); - debug(1,"Command: \"%s\", result is %d",command, dacp_send_command(command, &response, &le)); - if (response) { - free(response); - response = NULL; - } - strcpy(command,"setproperty?dmcp.volume=100.000000"); - debug(1,"Command: \"%s\", result is %d",command, dacp_send_command(command, &response, &le)); - if (response) { - free(response); - response = NULL; - } - */ - if (metadata_store.player_thread_active) - sleep(config.scan_interval_when_active); - else - sleep(config.scan_interval_when_inactive); } debug(1, "DACP monitor thread exiting -- should never happen."); pthread_exit(NULL); @@ -889,6 +920,22 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { void dacp_monitor_start() { int rc; + +#ifdef COMPILE_FOR_LINUX_AND_FREEBSD_AND_CYGWIN_AND_OPENBSD + pthread_condattr_t attr; + pthread_condattr_init(&attr); + pthread_condattr_setclock(&attr, CLOCK_MONOTONIC); // can't do this in OS X, and don't need it. + rc = pthread_cond_init(&dacp_server_information_cv, &attr); + if (rc) + debug(1, "Error initialising the DACP Server Information Condition Variable"); + pthread_condattr_destroy(&attr); +#endif +#ifdef COMPILE_FOR_OSX + rc = pthread_cond_init(&dacp_server_information_cv, NULL); +#endif + + + pthread_mutexattr_t mta; rc = pthread_mutexattr_init(&mta); @@ -947,6 +994,8 @@ void dacp_monitor_stop() { pthread_mutex_destroy(&dacp_server_information_lock); debug(3, "DACP Conversation Lock Mutex Destroyed"); pthread_mutex_destroy(&dacp_conversation_lock); + pthread_cond_destroy(&dacp_server_information_cv); + debug(1, "DACP Server Information Condition Variable destroyed."); } } diff --git a/rtsp.c b/rtsp.c index dca0ecb5..4013acb9 100644 --- a/rtsp.c +++ b/rtsp.c @@ -2335,6 +2335,7 @@ static void *rtsp_conversation_thread_func(void *pconn) { pthread_condattr_init(&attr); pthread_condattr_setclock(&attr, CLOCK_MONOTONIC); // can't do this in OS X, and don't need it. rc = pthread_cond_init(&conn->flowcontrol, &attr); + pthread_condattr_destroy(&attr); #endif #ifdef COMPILE_FOR_OSX rc = pthread_cond_init(&conn->flowcontrol, NULL); diff --git a/shairport.c b/shairport.c index 881ecd97..8f80ebdb 100644 --- a/shairport.c +++ b/shairport.c @@ -405,6 +405,7 @@ int parse_options(int argc, char **argv) { config.resend_control_first_check_time = 0.10; // wait this many seconds before requesting the resending of a missing packet config.resend_control_check_interval_time = 0.25; // wait this many seconds before again requesting the resending of a missing packet config.resend_control_last_check_time = 0.10; // give up if the packet is still missing this close to when it's needed + config.missing_port_dacp_scan_interval_seconds = 2.0; // check at this interval if no DACP port number is known #ifdef CONFIG_METADATA_HUB config.cover_art_cache_dir = "/tmp/shairport-sync/.cache/coverart"; From ff3cef44714b1b298bc28849d3ed8e1cf79a5443 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 25 Mar 2020 10:17:39 +0000 Subject: [PATCH 067/186] If the DACP port is zero, do a zeroconf re-broswe every two seconds in case it has been missed. --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 1abafedc..3e041734 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.7d2], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.7d3], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 31f5e8b9ca9173244b1e64d13945acebcd0a5d1e Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 25 Mar 2020 10:31:19 +0000 Subject: [PATCH 068/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index e8c62920..9aff68db 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,9 @@ +Version 3.3.7d3 +==== + +**Enhancement** +* While a player is active, the DACP port number to which to send remote commands should be broadcast over ZEROCONF/Bonjour. However, if that information is not available, Shairport Sync will now check for it every two seconds. + Version 3.3.7d2 ==== **Pesky Changes** From 1b7e23c093b6159b1f05a91e80a7c9ae4a216012 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 25 Mar 2020 10:40:05 +0000 Subject: [PATCH 069/186] Add ability to set the missing DACP port scan interval. --- scripts/shairport-sync.conf | 2 +- shairport.c | 12 ++++++++++++ 2 files changed, 13 insertions(+), 1 deletion(-) diff --git a/scripts/shairport-sync.conf b/scripts/shairport-sync.conf index 9c875156..170fafc2 100644 --- a/scripts/shairport-sync.conf +++ b/scripts/shairport-sync.conf @@ -54,7 +54,7 @@ general = // resend_control_first_check_time = 0.10; // Use this optional advanced setting to set the wait time in seconds before deciding a packet is missing. // resend_control_check_interval_time = 0.25; // Use this optional advanced setting to set the time in seconds between requests for a missing packet. // resend_control_last_check_time = 0.10; // Use this optional advanced setting to set the latest time, in seconds, by which the last check should be done before the estimated time of a missing packet's transfer to the output buffer. -// +// missing_port_dacp_scan_interval_seconds = 2.0; // Use this optional advanced setting to set the time interval between scans for a DACP port number if no port number has been provided by the player for remote control commands }; // Advanced parameters for controlling how Shairport Sync stays active and how it runs a session diff --git a/shairport.c b/shairport.c index 8f80ebdb..ec5a259a 100644 --- a/shairport.c +++ b/shairport.c @@ -806,6 +806,18 @@ int parse_options(int argc, char **argv) { dvalue, config.resend_control_last_check_time); } + if (config_lookup_float(config.cfg, "general.missing_port_dacp_scan_interval_seconds", + &dvalue)) { + if ((dvalue >= 0.0) && (dvalue <= 300.0)) + config.missing_port_dacp_scan_interval_seconds = dvalue; + else + warn("Invalid general missing_port_dacp_scan_interval_seconds setting \"%d\". It should " + "be " + "between 0.0 and 300.0, " + "inclusive. The setting remains at %f seconds.", + dvalue, config.missing_port_dacp_scan_interval_seconds); + } + /* Get the default latency. Deprecated! */ if (config_lookup_int(config.cfg, "latencies.default", &value)) config.userSuppliedLatency = value; From eef6bbf9839ff5c0434ff23fdcf8587bf1d17411 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 26 Mar 2020 17:03:53 +0000 Subject: [PATCH 070/186] Remove leading zeroes from the DACP ID provided by mDNS --- mdns_avahi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/mdns_avahi.c b/mdns_avahi.c index caec909d..575eba08 100644 --- a/mdns_avahi.c +++ b/mdns_avahi.c @@ -96,6 +96,8 @@ static void resolve_callback(AvahiServiceResolver *r, AVAHI_GCC_UNUSED AvahiIfIn char *dacpid = strstr(name, "iTunes_Ctrl_"); if (dacpid) { dacpid += strlen("iTunes_Ctrl_"); + while (*dacpid == '0') + dacpid++; // remove any leading zeroes if (strcmp(dacpid, dbs->dacp_id) == 0) { debug(3, "resolve_callback: client dacp_id \"%s\" dacp port: %u.", dbs->dacp_id, port); #ifdef CONFIG_DACP_CLIENT From 1f9be7d6aa0484726d68f806b05293018aa6addf Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 26 Mar 2020 17:04:25 +0000 Subject: [PATCH 071/186] Remove leading zeroes from the DACP ID provided by mDNS --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 3e041734..ebc1122f 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.7d3], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.7d4], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 4eaf23aa22277e603f3e55487c9fa3c0130d672e Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 26 Mar 2020 17:11:31 +0000 Subject: [PATCH 072/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 9aff68db..0d769e66 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,6 +1,10 @@ +Version 3.3.7d4 +==== +**Bug Fix** +* The DACP ID looks like a 64-bit number expressed in hexadecimal. It is normally quoted with leading zeroes removed, but in the `_dacp._tcp` service string, leading zeros are not removed from the DACP ID. The bug fix removes those leading zeroes. + Version 3.3.7d3 ==== - **Enhancement** * While a player is active, the DACP port number to which to send remote commands should be broadcast over ZEROCONF/Bonjour. However, if that information is not available, Shairport Sync will now check for it every two seconds. @@ -10,7 +14,7 @@ Version 3.3.7d2 * The underlying timing system has been moved from 64-bit fixed-point time representation (a là NTP) to a 64-bit unsigned nanoseconds representation. This should make precisely no difference to the functionality of Shairport Sync but the transition might behabve inadeventely introduced bugs. Problem reports gratefully received. **Enhancement** -* The timing software in the `sndio` backend does some extra sanity checking on certain time estimates, it may help a little whne running on virtual machines. +* The timing software in the `sndio` backend does some extra sanity checking on certain time estimates, it may help a little when running on virtual machines. Version 3.3.7d1 ==== From aacc4e802bdaffbfbe4aaffa5dc5a3f9f43cab7c Mon Sep 17 00:00:00 2001 From: Guillaume Revaillot Date: Thu, 26 Mar 2020 16:13:36 +0100 Subject: [PATCH 073/186] Allow to specify pulseaudio server instead of using default one. Add "server" entry to pa config section, allowing to connect to a remote or a local system pulseaudio instance instead of letting pulseaudio choose. --- audio_pa.c | 8 +++++++- common.h | 1 + scripts/shairport-sync.conf | 1 + 3 files changed, 9 insertions(+), 1 deletion(-) diff --git a/audio_pa.c b/audio_pa.c index e6a5b576..951860be 100644 --- a/audio_pa.c +++ b/audio_pa.c @@ -85,6 +85,11 @@ static int init(__attribute__((unused)) int argc, __attribute__((unused)) char * if (config.cfg != NULL) { const char *str; + /* Get the PulseAudio server name. */ + if (config_lookup_string(config.cfg, "pa.server", &str)) { + config.pa_server = (char *)str; + } + /* Get the Application Name. */ if (config_lookup_string(config.cfg, "pa.application_name", &str)) { config.pa_application_name = (char *)str; @@ -126,7 +131,8 @@ static int init(__attribute__((unused)) int argc, __attribute__((unused)) char * // Start the mainloop if (pa_threaded_mainloop_start(mainloop) != 0) die("could not start the pulseaudio threaded mainloop"); - if (pa_context_connect(context, NULL, 0, NULL) != 0) + + if (pa_context_connect(context, config.pa_server, 0, NULL) != 0) die("failed to connect to the pulseaudio context -- the error message is \"%s\".", pa_strerror(pa_context_errno(context))); diff --git a/common.h b/common.h index 95f4decc..d5aad230 100644 --- a/common.h +++ b/common.h @@ -114,6 +114,7 @@ typedef struct { // on host %h" #ifdef CONFIG_PA + char *pa_server; // the pulseaudio server address that Shairport Sync will play on. char *pa_application_name; // the name under which Shairport Sync shows up as an "Application" in // the Sound Preferences in most desktop Linuxes. // Defaults to "Shairport Sync". Shairport Sync must be playing to see it. diff --git a/scripts/shairport-sync.conf b/scripts/shairport-sync.conf index 170fafc2..992c4b95 100644 --- a/scripts/shairport-sync.conf +++ b/scripts/shairport-sync.conf @@ -118,6 +118,7 @@ sndio = // --with-pa pa = { +// server = "host"; // Set this to override the default pulseaudio server that should be used. // application_name = "Shairport Sync"; //Set this to the name that should appear in the Sounds "Applications" tab when Shairport Sync is active. }; From 13794352b272f2c09b12ade4247bd4dd027bc3d0 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 31 Mar 2020 14:40:18 +0100 Subject: [PATCH 074/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 3 +++ 1 file changed, 3 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 0d769e66..962a33f2 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,6 @@ +**New Feature** +* For the PulseAudio backend `pa`, add a new `server` entry to `pa` section of the configuration file, allowing you to specify a connection to a remote or a local system PulseAudio instance instead of letting PulseAudio choose. Thanks to [Guillaume Revaillot](https://github.com/grevaillot) for this new feature. + Version 3.3.7d4 ==== **Bug Fix** From ca72b570df0aac111ef36488650f9ec586db8e28 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 15 Apr 2020 14:57:39 +0100 Subject: [PATCH 075/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 962a33f2..a1d93710 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -14,7 +14,7 @@ Version 3.3.7d3 Version 3.3.7d2 ==== **Pesky Changes** -* The underlying timing system has been moved from 64-bit fixed-point time representation (a là NTP) to a 64-bit unsigned nanoseconds representation. This should make precisely no difference to the functionality of Shairport Sync but the transition might behabve inadeventely introduced bugs. Problem reports gratefully received. +* The underlying timing system has been moved from 64-bit fixed-point time representation (a là NTP) to a 64-bit unsigned nanoseconds representation. This should make precisely no difference to the functionality of Shairport Sync but the transition might inadvertently have introduced bugs. Problem reports gratefully received. **Enhancement** * The timing software in the `sndio` backend does some extra sanity checking on certain time estimates, it may help a little when running on virtual machines. From 0ad9d3d85eef97b81c51593d994d593fc2f71d2c Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 15 Apr 2020 14:59:16 +0100 Subject: [PATCH 076/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index a1d93710..f72f53f6 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -4,7 +4,7 @@ Version 3.3.7d4 ==== **Bug Fix** -* The DACP ID looks like a 64-bit number expressed in hexadecimal. It is normally quoted with leading zeroes removed, but in the `_dacp._tcp` service string, leading zeros are not removed from the DACP ID. The bug fix removes those leading zeroes. +* The DACP ID looks like a 64-bit number expressed in hexadecimal. It is normally quoted with leading zeroes removed, but in the `_dacp._tcp` service string, leading zeros are not removed from the DACP ID. The bug fix removes those leading zeroes. Thanks to [julianc1969](https://github.com/juliandc1969) for tenaciously tracking down this bug! Version 3.3.7d3 ==== From f0f6276d64b3728309bb440f97fc0704c5948fe9 Mon Sep 17 00:00:00 2001 From: Tucker Kern Date: Mon, 27 Apr 2020 15:22:06 -0600 Subject: [PATCH 077/186] Use signed integers when calculating backend latency adjustment --- player.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/player.c b/player.c index 483fbee8..ee1d1a4f 100644 --- a/player.c +++ b/player.c @@ -1048,7 +1048,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { uint64_t should_be_time; frame_to_local_time(conn->first_packet_timestamp + conn->latency + - (uint32_t)(config.audio_backend_latency_offset * + (int32_t)(config.audio_backend_latency_offset * conn->input_rate), // this will go modulo 2^32 &should_be_time, conn); @@ -1069,7 +1069,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { uint64_t should_be_time; frame_to_local_time(conn->first_packet_timestamp + conn->latency + - (uint32_t)(config.audio_backend_latency_offset * + (int32_t)(config.audio_backend_latency_offset * conn->input_rate), // this should go modulo 2^32 &should_be_time, conn); @@ -1304,7 +1304,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { uint64_t time_to_play; frame_to_local_time(curframe->given_timestamp + conn->latency + - (uint32_t)(config.audio_backend_latency_offset * conn->input_rate) - + (int32_t)(config.audio_backend_latency_offset * conn->input_rate) - (uint32_t)(config.audio_backend_buffer_desired_length * conn->input_rate), // this will go modulo 2^32 &time_to_play, From 72d715352951efe3b4cecd412ec7528659ad18f2 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 14 May 2020 19:51:32 +0100 Subject: [PATCH 078/186] Fix calculation of latency -- honour the signs of the offset and account for the DAC buffer size. --- audio.c | 12 +++--------- common.h | 1 + dacp.c | 2 +- player.h | 1 + shairport.c | 4 ++++ 5 files changed, 10 insertions(+), 10 deletions(-) diff --git a/audio.c b/audio.c index 8b2762f4..f2195279 100644 --- a/audio.c +++ b/audio.c @@ -147,10 +147,10 @@ void parse_general_audio_options(void) { /* Get the desired buffer size setting in seconds. */ if (config_lookup_float(config.cfg, "general.audio_backend_buffer_desired_length_in_seconds", &dvalue)) { - if ((dvalue < 0) || (dvalue > 1.5)) { + if (dvalue < 0) { die("Invalid audio_backend_buffer_desired_length_in_seconds value: \"%f\". It " - "should be between 0 and " - "1.5, default is %.3f seconds", + "should be 0.0 or greater." + " The default is %.3f seconds", dvalue, config.audio_backend_buffer_desired_length); } else { config.audio_backend_buffer_desired_length = dvalue; @@ -190,13 +190,7 @@ void parse_general_audio_options(void) { /* Get the latency offset in seconds. */ if (config_lookup_float(config.cfg, "general.audio_backend_latency_offset_in_seconds", &dvalue)) { - if ((dvalue < -1.75) || (dvalue > 1.75)) { - die("Invalid audio_backend_latency_offset_in_seconds \"%f\". It " - "should be between -1.75 and +1.75, default is 0 seconds", - dvalue); - } else { config.audio_backend_latency_offset = dvalue; - } } /* Get the desired length of the silent lead-in. */ diff --git a/common.h b/common.h index d5aad230..dd267550 100644 --- a/common.h +++ b/common.h @@ -215,6 +215,7 @@ typedef struct { double audio_backend_latency_offset; // this will be the offset in seconds to compensate for any // fixed latency there might be in the audio path double audio_backend_silent_lead_in_time; // the length of the silence that should precede a play. + uint32_t minimum_free_buffer_headroom; // when effective latency is calculated, ensure this number of buffers are unallocated double active_state_timeout; // the amount of time from when play ends to when the system leaves // into the "active" mode. uint32_t volume_range_db; // the range, in dB, from max dB to min dB. Zero means use the mixer's diff --git a/dacp.c b/dacp.c index dc794d4d..b8e41d67 100644 --- a/dacp.c +++ b/dacp.c @@ -1248,7 +1248,7 @@ int dacp_get_volume(int32_t *the_actual_volume) { } else { debug(1, "Unexpected return code %d from dacp_get_speaker_list.", http_response); } - } else { + } else if (http_response != 400){ debug(3, "Unexpected return code %d from dacp_get_client_volume.", http_response); } if (the_actual_volume) { diff --git a/player.h b/player.h index 3845ca9f..483d4fee 100644 --- a/player.h +++ b/player.h @@ -87,6 +87,7 @@ typedef struct { uint32_t maximum_latency; // set if an a=max-latency: line appears in the ANNOUNCE message; zero // otherwise int software_mute_enabled; // if we don't have a real mute that we can use + int unachievable_audio_backend_latency_offset_notified; // set when a latency warning is issued int fd; int authorized; // set if a password is required and has been supplied diff --git a/shairport.c b/shairport.c index ec5a259a..867938ed 100644 --- a/shairport.c +++ b/shairport.c @@ -407,6 +407,10 @@ int parse_options(int argc, char **argv) { config.resend_control_last_check_time = 0.10; // give up if the packet is still missing this close to when it's needed config.missing_port_dacp_scan_interval_seconds = 2.0; // check at this interval if no DACP port number is known + config.minimum_free_buffer_headroom = 125; // leave approximately one second's worth of buffers free after calculating the effective latency. + // e.g. if we have 1024 buffers or 352 frames = 8.17 seconds and we have a nominal latency of 2.0 seconds + // then we can add an offset of 5.17 seconds and still leave a second's worth of buffers for unexpected circumstances + #ifdef CONFIG_METADATA_HUB config.cover_art_cache_dir = "/tmp/shairport-sync/.cache/coverart"; config.scan_interval_when_active = From 14d75679bedab7fe4bd9ae63fb037d6cc7e8d407 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 14 May 2020 20:45:52 +0100 Subject: [PATCH 079/186] Fix calculation of latency -- honour the signs of the offset and account for the DAC buffer size. --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index ebc1122f..7f148707 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.7d4], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.7d5], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From a4edc649f8f4255a7d84495f2a72771424813a9d Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 16 May 2020 12:23:08 +0100 Subject: [PATCH 080/186] Add code to do calculations involving the audio_backend_latency_offset_in_seconds and audio_backend_silent_lead_in_time settings correctly. Clean up the audio_backend_silent_lead_in_time setting by adding an "auto" setting. --- audio.c | 28 ++++++- common.h | 1 + player.c | 158 +++++++++++++++++++++++++++--------- scripts/shairport-sync.conf | 6 +- shairport.c | 7 +- 5 files changed, 155 insertions(+), 45 deletions(-) diff --git a/audio.c b/audio.c index f2195279..4d5df37b 100644 --- a/audio.c +++ b/audio.c @@ -126,6 +126,7 @@ void parse_general_audio_options(void) { * are set before any options are chosen */ int value; double dvalue; + const char *str = 0; if (config.cfg != NULL) { /* Get the desired buffer size setting (deprecated). */ @@ -193,14 +194,37 @@ void parse_general_audio_options(void) { config.audio_backend_latency_offset = dvalue; } + + /* Check if the length of the silent lead-in ia set to \"auto\". */ + if (config_lookup_string(config.cfg, "general.audio_backend_silent_lead_in_time", &str)) { + if (strcasecmp(str, "auto") == 0) { + config.audio_backend_silent_lead_in_time_auto = 1; + } else { + if (config.audio_backend_silent_lead_in_time_auto == 1) + warn("Invalid audio_backend_silent_lead_in_time \"%s\". It should be \"auto\" or the lead-in time in seconds. " + "It remains set to \"auto\". Note: numbers should not be placed in quotes.", + str); + else + warn("Invalid output rate \"%s\". It should be \"auto\" or the lead-in time in seconds. " + "It remains set to %f. Note: numbers should not be placed in quotes.", + str, config.audio_backend_silent_lead_in_time); + } + } + /* Get the desired length of the silent lead-in. */ if (config_lookup_float(config.cfg, "general.audio_backend_silent_lead_in_time", &dvalue)) { if ((dvalue < 0.0) || (dvalue > 4)) { - die("Invalid audio_backend_silent_lead_in_time \"%f\". It " - "must be between 0.0 and 4.0 seconds. Omit setting to use the default value", + if (config.audio_backend_silent_lead_in_time_auto == 1) + warn("Invalid audio_backend_silent_lead_in_time \"%f\". It " + "must be between 0.0 and 4.0 seconds. Omit the setting to use the automatic value. The setting remains at \"auto\".", dvalue); + else + warn("Invalid audio_backend_silent_lead_in_time \"%f\". It " + "must be between 0.0 and 4.0 seconds. Omit the setting to use the automatic value. It remains set to %f.", + dvalue, config.audio_backend_silent_lead_in_time); } else { config.audio_backend_silent_lead_in_time = dvalue; + config.audio_backend_silent_lead_in_time_auto = 0; } } } diff --git a/common.h b/common.h index dd267550..d2d90421 100644 --- a/common.h +++ b/common.h @@ -214,6 +214,7 @@ typedef struct { double audio_backend_latency_offset; // this will be the offset in seconds to compensate for any // fixed latency there might be in the audio path + int audio_backend_silent_lead_in_time_auto; // true if the lead-in time should be from as soon as packets are received double audio_backend_silent_lead_in_time; // the length of the silence that should precede a play. uint32_t minimum_free_buffer_headroom; // when effective latency is calculated, ensure this number of buffers are unallocated double active_state_timeout; // the amount of time from when play ends to when the system leaves diff --git a/player.c b/player.c index ee1d1a4f..fb16a3e8 100644 --- a/player.c +++ b/player.c @@ -4,7 +4,7 @@ * All rights reserved. * * Modifications for audio synchronisation - * and related work, copyright (c) Mike Brady 2014 -- 2019 + * and related work, copyright (c) Mike Brady 2014 -- 2020 * All rights reserved. * * Permission is hereby granted, free of charge, to any person @@ -660,6 +660,28 @@ int32_t rand_in_range(int32_t exclusive_range_limit) { return sp >> 32; } +int get_and_check_effective_latency(rtsp_conn_info *conn, uint32_t *effective_latency, double offset_time) { +// check that the overall effective latency remains positive and is not greater than the capacity of the buffers +// return 0 if okay, -1 if the latency would be negative, +1 if it would be too large + + int result = 0; + if (offset_time >= 0.0) { + uint32_t latency_addition = (uint32_t)(offset_time * conn->input_rate); + // keep about one second of buffers back + if ((*effective_latency + latency_addition) <= (conn->max_frames_per_packet * (BUFFER_FRAMES - config.minimum_free_buffer_headroom))) + *effective_latency += latency_addition; + else + result = 1; + } else { + uint32_t latency_reduction = (uint32_t)((-offset_time) * conn->input_rate); + if (latency_reduction <= *effective_latency) + *effective_latency -= latency_reduction; + else + result = -1; + } + return result; +} + static inline void process_sample(int32_t sample, char **outp, sps_format_t format, int volume, int dither, rtsp_conn_info *conn) { /* @@ -898,6 +920,8 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { int wait; long dac_delay = 0; // long because alsa returns a long + int have_sent_prefiller_silence = 0; // set to true when we have sent at least one silent frame to the DAC + pthread_cleanup_push(buffer_get_frame_cleanup_handler, (void *)conn); // undo what's been done so far do { @@ -989,8 +1013,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { notified_buffer_empty = 0; // at least one buffer now -- diagnostic only. if (conn->ab_buffering) { // if we are getting packets but not yet forwarding them to the // player - int have_sent_prefiller_silence = 1; // set true when we have sent some silent frames to - // the DAC +// int have_sent_prefiller_silence = 1; // set to true when we have sent at least one silent frame to the DAC /* int64_t reference_timestamp; uint64_t reference_timestamp_time, remote_reference_timestamp_time; @@ -1007,7 +1030,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { conn->first_packet_timestamp = curframe->given_timestamp; // we will keep buffering until we are // supposed to start playing this - have_sent_prefiller_silence = 0; + // have_sent_prefiller_silence = 0; // debug(1, "First packet timestamp is %" PRId64 ".", conn->first_packet_timestamp); @@ -1047,9 +1070,12 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { // frame should be played" uint64_t should_be_time; - frame_to_local_time(conn->first_packet_timestamp + conn->latency + - (int32_t)(config.audio_backend_latency_offset * - conn->input_rate), // this will go modulo 2^32 + uint32_t effective_latency = conn->latency; + + get_and_check_effective_latency(conn, &effective_latency,config.audio_backend_latency_offset); + // we are ignoring the returned status because it will be captured on subsequent frames, below. + + frame_to_local_time(conn->first_packet_timestamp + effective_latency, // this will go modulo 2^32 &should_be_time, conn); @@ -1067,14 +1093,34 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { if (conn->first_packet_time_to_play != 0) { // recalculate conn->first_packet_time_to_play -- the latency might change - uint64_t should_be_time; - frame_to_local_time(conn->first_packet_timestamp + conn->latency + - (int32_t)(config.audio_backend_latency_offset * - conn->input_rate), // this should go modulo 2^32 - &should_be_time, - conn); + uint64_t should_be_time; + uint32_t effective_latency = conn->latency; - conn->first_packet_time_to_play = should_be_time; + switch (get_and_check_effective_latency(conn, &effective_latency,config.audio_backend_latency_offset)) { + case -1: + if (conn->unachievable_audio_backend_latency_offset_notified == 0) { + warn("Negative latency! A latency of %d frames requested by the player, when combined with an audio_backend_latency_offset of %f seconds, would make the overall latency negative. The audio_backend_latency_offset setting is ignored.", conn->latency, config.audio_backend_latency_offset); + config.audio_backend_latency_offset = 0; //set it to zero + conn->unachievable_audio_backend_latency_offset_notified = 1; + }; +// effective_latency = 0 ; + break; + case 1: + if (conn->unachievable_audio_backend_latency_offset_notified == 0) { + warn("An audio_backend_latency_offset of %f seconds may exceed the frame buffering capacity -- the setting is ignored.", config.audio_backend_latency_offset); + config.audio_backend_latency_offset = 0; // set it to zero; + conn->unachievable_audio_backend_latency_offset_notified = 1; + }; + break; + default: + break; + } + + frame_to_local_time(conn->first_packet_timestamp + effective_latency, // this will go modulo 2^32 + &should_be_time, + conn); + + conn->first_packet_time_to_play = should_be_time; // we want the frames of silence sent at the start to be fairly large in case the output // device's minimum buffer size is large. But they can't be greater than the silent @@ -1083,21 +1129,35 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { // setting // In fact, if should be some fraction of them, to allow for adjustment. - int64_t max_dac_delay = conn->latency; - if (config.audio_backend_silent_lead_in_time >= 0) + double adjustment_interval_seconds = 0.025; + int64_t minimum_adjustments_to_make = 3; + double initial_silence_seconds = 0.2; + + int64_t adjustment_interval_frames = (int64_t)(adjustment_interval_seconds * conn->input_rate); + int64_t initial_silence_frames = (int64_t)(initial_silence_seconds * conn->input_rate); + + int64_t max_dac_delay = effective_latency; + if (config.audio_backend_silent_lead_in_time_auto == 0) max_dac_delay = (int64_t)(config.audio_backend_silent_lead_in_time * conn->input_rate); - max_dac_delay = max_dac_delay / 4; + // should check that the audio_backend_silent_lead_in_time is greater than the desired_buffer_size - // debug(1,"max_dac_delay is %" PRIu64 " frames.", max_dac_delay); + int64_t filler_size = adjustment_interval_frames; - int64_t filler_size = max_dac_delay; + if (have_sent_prefiller_silence == 0) { + int64_t delay_before_adjustments = max_dac_delay - minimum_adjustments_to_make * adjustment_interval_frames; + if (delay_before_adjustments > initial_silence_frames) + filler_size = initial_silence_frames; + else + filler_size = max_dac_delay / (minimum_adjustments_to_make + 1); + debug(1,"Initial filler size: %" PRId64 " frames.", filler_size); + } if (local_time_now > conn->first_packet_time_to_play) { uint64_t lateness = local_time_now - conn->first_packet_time_to_play; debug(3, "Gone past starting time by %" PRIu64 " nanoseconds.", lateness); - have_sent_prefiller_silence = 1; + // have_sent_prefiller_silence = 1; conn->ab_buffering = 0; // we've gone past the time... @@ -1117,16 +1177,8 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { } else { // do some calculations int64_t lead_time = conn->first_packet_time_to_play - local_time_now; - // an audio_backend_silent_lead_in_time of less than zero means start filling ASAP - int64_t lead_in_time = -1; - if (config.audio_backend_silent_lead_in_time >= 0) - lead_in_time = - (int64_t)(config.audio_backend_silent_lead_in_time * (int64_t)1000000000); - // debug(1,"Lead time is %llx at fpttp - // %llx.",lead_time,conn->first_packet_time_to_play); - if ((lead_in_time < 0) || (lead_time <= lead_in_time)) { - // debug(1,"Lead time is %" PRIx64 ", lead-in time is %" PRIx64 " at fpttp - // %llx.",lead_time,conn->first_packet_time_to_play); + if ((config.audio_backend_silent_lead_in_time_auto == 1) || (lead_time <= (int64_t)(config.audio_backend_silent_lead_in_time * (int64_t)1000000000))) { + debug(1,"Lead time: %" PRId64 ".", lead_time); // debug(1,"Checking"); if (config.output->delay) { @@ -1235,7 +1287,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { void *silence; if (lead_time != 0) { - int64_t frame_gap = (lead_time * config.output_rate) >> 32; + int64_t frame_gap = (lead_time * config.output_rate) / 1000000000; // debug(1,"%d frames needed.",frame_gap); while (frame_gap > 0) { ssize_t fs = config.output_rate / 10; @@ -1303,12 +1355,40 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { if (have_timestamp_timing_information(conn)) { // if we have a reference time uint64_t time_to_play; - frame_to_local_time(curframe->given_timestamp + conn->latency + - (int32_t)(config.audio_backend_latency_offset * conn->input_rate) - - (uint32_t)(config.audio_backend_buffer_desired_length * - conn->input_rate), // this will go modulo 2^32 - &time_to_play, - conn); + uint32_t effective_latency = conn->latency; + + switch (get_and_check_effective_latency(conn, &effective_latency,config.audio_backend_latency_offset - config.audio_backend_buffer_desired_length)) { + case -1: + // this means that the latency is negative, i.e. the packet must be played before its time. This is + // clearly a mistake, and can arise if the combination of a large negative latency offset and the desired buffer length are greater than the + // latency chosen by the player. For example if the latency is 88200 frames (2 seconds), the audio_backend_latency_offset is -1.9 and the audio_backend_buffer_desired_length is 0.2 + // the effective latency is -0.1, so the frame must be played 0.1 seconds before it is time-tagged for, and in reality before it arrives at the player. + + // to deal with the problem, rather than block the packets, we'll just let 'em go... + + // + if (conn->unachievable_audio_backend_latency_offset_notified == 0) { + warn("Negative latency! A latency of %d frames requested by the player, when combined with an audio_backend_latency_offset of %f seconds an audio_backend_buffer_desired_length of %f seconds, would make the overall latency negative. No latency is used. Synchronisation may fail.", conn->latency, config.audio_backend_latency_offset, config.audio_backend_buffer_desired_length); + conn->unachievable_audio_backend_latency_offset_notified = 1; + }; + time_to_play = local_time_now; // pretend the frame should be played now... + break; + case 1: + if (conn->unachievable_audio_backend_latency_offset_notified == 0) { + warn("Latency too long! An audio_backend_latency_offset of %f seconds combined with an audio_backend_buffer_desired_length of %f seconds may exceed the frame buffering capacity.", config.audio_backend_latency_offset, config.audio_backend_buffer_desired_length); + conn->unachievable_audio_backend_latency_offset_notified = 1; + }; + time_to_play = local_time_now; // pretend the frame should be played now... + break; + default: + frame_to_local_time(curframe->given_timestamp + effective_latency, // this will go modulo 2^32 + &time_to_play, + conn); + break; + } + + + if (local_time_now >= time_to_play) { do_wait = 0; @@ -1320,7 +1400,9 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { if (notified_buffer_empty == 0) { debug(3, "Buffers exhausted."); notified_buffer_empty = 1; - reset_input_flow_metrics(conn); + // reset_input_flow_metrics(conn); // don't do a full flush parameters reset + conn->initial_reference_time = 0; + conn->initial_reference_timestamp = 0; } do_wait = 1; } diff --git a/scripts/shairport-sync.conf b/scripts/shairport-sync.conf index 992c4b95..fd20b33f 100644 --- a/scripts/shairport-sync.conf +++ b/scripts/shairport-sync.conf @@ -43,10 +43,10 @@ general = // If you build Shairport Sync with the flag --with-apple-alac, the Apple ALAC decoder will be chosen by default. // interface = "name"; // Use this advanced setting to specify the interface on which Shairport Sync should provide its service. Leave it commented out to get the default, which is to select the interface(s) automatically. -// audio_backend_latency_offset_in_seconds = 0.0; // Set this offset to compensate for a fixed delay in the audio back end. E.g. if the output device delays by 100 ms, set this to -0.1. -// audio_backend_buffer_desired_length_in_seconds = 0.2; // If set too small, buffer underflow occurs on low-powered machines. Too long and the response time to volume changes becomes annoying. Default is 0.15 seconds in the alsa backend, 0.35 seconds in the pa backend and 1.0 seconds otherwise. +// audio_backend_latency_offset_in_seconds = 0.0; // This is added to the latency requested by the player to delay or advance the output by a fixed amount. Use it, for example, to compensate for a fixed delay in the audio back end. E.g. if the output device, e.g. a soundbar, takes 100 ms to process audio, set this to -0.1 to deliver the audio to the output device 100 ms early, allowing it time to process the audio and output it perfectly in sync. +// audio_backend_buffer_desired_length_in_seconds = 0.2; // If set too small, buffer underflow occurs on low-powered machines. Too long and the response time to volume changes becomes annoying. Default is 0.2 seconds in the alsa backend, 0.35 seconds in the pa backend and 1.0 seconds otherwise. // audio_backend_buffer_interpolation_threshold_in_seconds = 0.075; // Advanced feature. If the buffer size drops below this, stop using time-consuming interpolation like soxr to avoid dropouts due to underrun. -// audio_backend_silent_lead_in_time = 2.0; // This optional advanced setting, from 0.0 and 4.0 seconds, sets the length of the period of silence that precedes the start of the audio. The default is the latency, usually 2.0 seconds. Values greater than the latency are ignored. Values that are too low will affect initial synchronisation. +// audio_backend_silent_lead_in_time = "auto"; // This optional advanced setting, either "auto" or a positive number, sets the length of the period of silence that precedes the start of the audio. The default is "auto" -- the silent lead-in starts as soon as the player starts sending packets. Values greater than the latency are ignored. Values that are too low will affect initial synchronisation. // dbus_service_bus = "system"; // The Shairport Sync dbus interface, if selected at compilation, will appear // as "org.gnome.ShairportSync" on the whichever bus you specify here: "system" (default) or "session". // mpris_service_bus = "system"; // The Shairport Sync mpris interface, if selected at compilation, will appear diff --git a/shairport.c b/shairport.c index 867938ed..ead93b2d 100644 --- a/shairport.c +++ b/shairport.c @@ -388,7 +388,7 @@ int parse_options(int argc, char **argv) { config.resyncthreshold = 1.0 * fResyncthreshold / 44100; config.tolerance = 1.0 * fTolerance / 44100; - config.audio_backend_silent_lead_in_time = -1.0; // flag to indicate it has not been set + config.audio_backend_silent_lead_in_time_auto = 1; // start outputting silence as soon as packets start arriving config.airplay_volume = -18.0; // if no volume is ever set, default to initial default value if // nothing else comes in first. config.fixedLatencyOffset = 11025; // this sounds like it works properly. @@ -1788,7 +1788,10 @@ int main(int argc, char **argv) { debug(1, "audio_backend_buffer_interpolation_threshold_in_seconds is %f seconds.", config.audio_backend_buffer_interpolation_threshold_in_seconds); debug(1, "audio backend latency offset is %f seconds.", config.audio_backend_latency_offset); - debug(1, "audio backend silence lead-in time is %f seconds. A value -1.0 means use the default.", + if (config.audio_backend_silent_lead_in_time_auto == 1) + debug(1, "audio backend silence lead-in time is \"auto\"."); + else + debug(1, "audio backend silence lead-in time is %f seconds.", config.audio_backend_silent_lead_in_time); debug(1, "zeroconf regtype is \"%s\".", config.regtype); debug(1, "decoders_supported field is %d.", config.decoders_supported); From 07d956c014217ea7391854ab25ad88a28b9d0e00 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 16 May 2020 12:25:27 +0100 Subject: [PATCH 081/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index f72f53f6..a72ed3c0 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -4,7 +4,7 @@ Version 3.3.7d4 ==== **Bug Fix** -* The DACP ID looks like a 64-bit number expressed in hexadecimal. It is normally quoted with leading zeroes removed, but in the `_dacp._tcp` service string, leading zeros are not removed from the DACP ID. The bug fix removes those leading zeroes. Thanks to [julianc1969](https://github.com/juliandc1969) for tenaciously tracking down this bug! +* The DACP ID looks like a 64-bit number expressed in hexadecimal. It is normally quoted with leading zeroes removed, but in the `_dacp._tcp` service string, leading zeros are not removed from the DACP ID. The bug fix removes those leading zeroes. Thanks to [julianc1969](https://github.com/juliandc1969) for tracking down this bug so tenaciously! Version 3.3.7d3 ==== From e8831f6850378676f9e65e9876f6526309256b4b Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 16 May 2020 12:35:31 +0100 Subject: [PATCH 082/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index a72ed3c0..a1a91eb7 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,10 @@ +Version 3.3.7d5 +==== +**Bug Fixes and Enhancements** +* Added code to do calculations involving the `audio_backend_latency_offset_in_seconds` and `audio_backend_silent_lead_in_time` settings correctly. Many thanks to [Tucker Kern](https://github.com/mill1000) for discovering a number of bugs associated with this and for [proposing a solution](https://github.com/mikebrady/shairport-sync/pull/1001). This prompted a closer investigation and a number of further improvements were made, and a few "hostages to fortune" removed. +* Cleaned up the `audio_backend_silent_lead_in_time` setting by adding an `"auto"` setting. +* Improved synchronisation accuracy with short silence lead-ins. + **New Feature** * For the PulseAudio backend `pa`, add a new `server` entry to `pa` section of the configuration file, allowing you to specify a connection to a remote or a local system PulseAudio instance instead of letting PulseAudio choose. Thanks to [Guillaume Revaillot](https://github.com/grevaillot) for this new feature. From 18749bc3159c8f67c32c90563333e11c2024952e Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 16 May 2020 12:49:00 +0100 Subject: [PATCH 083/186] Add brief message about finding ALSA devices. --- scripts/shairport-sync.conf | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/scripts/shairport-sync.conf b/scripts/shairport-sync.conf index fd20b33f..e4137638 100644 --- a/scripts/shairport-sync.conf +++ b/scripts/shairport-sync.conf @@ -84,7 +84,7 @@ sessioncontrol = // --with-alsa alsa = { -// output_device = "default"; // the name of the alsa output device. Use "alsamixer" or "aplay" to find out the names of devices, mixers, etc. +// output_device = "default"; // the name of the alsa output device. Use "shairport-sync -h" to discover the names of ALSA hardware devices. Use "alsamixer" or "aplay" to find out the names of devices, mixers, etc. // mixer_control_name = "PCM"; // the name of the mixer to use to adjust output volume. If not specified, volume in adjusted in software. // mixer_device = "default"; // the mixer_device default is whatever the output_device is. Normally you wouldn't have to use this. // output_rate = "auto"; // can be "auto", 44100, 88200, 176400 or 352800, but the device must have the capability. From bca704d5b71997a740e15f7a30602068d0217647 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 16 May 2020 12:54:18 +0100 Subject: [PATCH 084/186] Quieten a few debug messages. --- player.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/player.c b/player.c index fb16a3e8..f36417c5 100644 --- a/player.c +++ b/player.c @@ -1151,7 +1151,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { filler_size = initial_silence_frames; else filler_size = max_dac_delay / (minimum_adjustments_to_make + 1); - debug(1,"Initial filler size: %" PRId64 " frames.", filler_size); + // debug(1,"Initial filler size: %" PRId64 " frames.", filler_size); } if (local_time_now > conn->first_packet_time_to_play) { @@ -1178,7 +1178,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { // do some calculations int64_t lead_time = conn->first_packet_time_to_play - local_time_now; if ((config.audio_backend_silent_lead_in_time_auto == 1) || (lead_time <= (int64_t)(config.audio_backend_silent_lead_in_time * (int64_t)1000000000))) { - debug(1,"Lead time: %" PRId64 ".", lead_time); + // debug(1,"Lead time: %" PRId64 ".", lead_time); // debug(1,"Checking"); if (config.output->delay) { From 012dd26c77c6abf89a73076bc085ad97cf3639de Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 17 May 2020 12:48:25 +0100 Subject: [PATCH 085/186] Add the ability to set the silent lead-in parameters initial_period, minimum number of adjustments and adjustment interval, and give them sensible defaults. --- common.h | 4 +++ player.c | 12 +++----- scripts/shairport-sync.conf | 4 +++ shairport.c | 58 ++++++++++++++++++++++++++++++++----- 4 files changed, 63 insertions(+), 15 deletions(-) diff --git a/common.h b/common.h index d2d90421..ec9f35a1 100644 --- a/common.h +++ b/common.h @@ -212,6 +212,10 @@ typedef struct { // buffer double disable_standby_mode_silence_scan_interval; // check the threshold this often + double lead_in_silence_initial_period; // the size of the first block of silence to send to the DAC + int lead_in_silence_minimum_adjustments_to_make; // make at least this number of checks and timing adjustments + double lead_in_silence_adjustment_interval; // make checks and adjustments at this interval + double audio_backend_latency_offset; // this will be the offset in seconds to compensate for any // fixed latency there might be in the audio path int audio_backend_silent_lead_in_time_auto; // true if the lead-in time should be from as soon as packets are received diff --git a/player.c b/player.c index f36417c5..2d8f635e 100644 --- a/player.c +++ b/player.c @@ -1129,12 +1129,8 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { // setting // In fact, if should be some fraction of them, to allow for adjustment. - double adjustment_interval_seconds = 0.025; - int64_t minimum_adjustments_to_make = 3; - double initial_silence_seconds = 0.2; - - int64_t adjustment_interval_frames = (int64_t)(adjustment_interval_seconds * conn->input_rate); - int64_t initial_silence_frames = (int64_t)(initial_silence_seconds * conn->input_rate); + int64_t adjustment_interval_frames = (int64_t)(config.lead_in_silence_adjustment_interval * conn->input_rate); + int64_t initial_silence_frames = (int64_t)(config.lead_in_silence_initial_period * conn->input_rate); int64_t max_dac_delay = effective_latency; if (config.audio_backend_silent_lead_in_time_auto == 0) @@ -1146,11 +1142,11 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { int64_t filler_size = adjustment_interval_frames; if (have_sent_prefiller_silence == 0) { - int64_t delay_before_adjustments = max_dac_delay - minimum_adjustments_to_make * adjustment_interval_frames; + int64_t delay_before_adjustments = max_dac_delay - config.lead_in_silence_minimum_adjustments_to_make * adjustment_interval_frames; if (delay_before_adjustments > initial_silence_frames) filler_size = initial_silence_frames; else - filler_size = max_dac_delay / (minimum_adjustments_to_make + 1); + filler_size = max_dac_delay / (config.lead_in_silence_minimum_adjustments_to_make + 1); // debug(1,"Initial filler size: %" PRId64 " frames.", filler_size); } diff --git a/scripts/shairport-sync.conf b/scripts/shairport-sync.conf index e4137638..33db5d91 100644 --- a/scripts/shairport-sync.conf +++ b/scripts/shairport-sync.conf @@ -47,6 +47,10 @@ general = // audio_backend_buffer_desired_length_in_seconds = 0.2; // If set too small, buffer underflow occurs on low-powered machines. Too long and the response time to volume changes becomes annoying. Default is 0.2 seconds in the alsa backend, 0.35 seconds in the pa backend and 1.0 seconds otherwise. // audio_backend_buffer_interpolation_threshold_in_seconds = 0.075; // Advanced feature. If the buffer size drops below this, stop using time-consuming interpolation like soxr to avoid dropouts due to underrun. // audio_backend_silent_lead_in_time = "auto"; // This optional advanced setting, either "auto" or a positive number, sets the length of the period of silence that precedes the start of the audio. The default is "auto" -- the silent lead-in starts as soon as the player starts sending packets. Values greater than the latency are ignored. Values that are too low will affect initial synchronisation. +// audio_backend_silent_lead_in_initial_period = 0.5; // This optional advanced setting sets the length, in seconds, of the first "block" of silence sent to the DAC -- some DACs need a fairly substantial block to prevent lots of initial underrun. +// audio_backend_silent_lead_in_minimum_adjustments_to_make = 3; // This optional advanced setting sets the minimum number of times Shairport Sync will readjust the length of the silent lead in to ensure that the audio starts at precisely the correct time. +// audio_backend_silent_lead_in_adjustment_interval = 0.1; // This optional advanced setting sets the interval, in seconds, between adjustments of the silent lead in to ensure that the audio starts at precisely the correct time. + // dbus_service_bus = "system"; // The Shairport Sync dbus interface, if selected at compilation, will appear // as "org.gnome.ShairportSync" on the whichever bus you specify here: "system" (default) or "session". // mpris_service_bus = "system"; // The Shairport Sync mpris interface, if selected at compilation, will appear diff --git a/shairport.c b/shairport.c index ead93b2d..60bfb4a0 100644 --- a/shairport.c +++ b/shairport.c @@ -751,9 +751,9 @@ int parse_options(int argc, char **argv) { * range set by the mixer. */ if (config_lookup_int(config.cfg, "general.volume_range_db", &value)) { if ((value < 30) || (value > 150)) - die("Invalid volume range \"%sd\". It should be between 30 and 150 dB. Zero means use " - "the mixer's native range", - value); + die("Invalid volume range %d dB. It should be between 30 and 150 dB. Zero means use " + "the mixer's native range. The setting reamins at %d.", + value, config.volume_range_db); else config.volume_range_db = value; } @@ -779,7 +779,7 @@ int parse_options(int argc, char **argv) { if ((dvalue >= 0.0) && (dvalue <= 3.0)) config.resend_control_first_check_time = dvalue; else - warn("Invalid general resend_control_first_check_time setting \"%d\". It should " + warn("Invalid general resend_control_first_check_time setting \"%f\". It should " "be " "between 0.0 and 3.0, " "inclusive. The setting remains at %f seconds.", @@ -791,7 +791,7 @@ int parse_options(int argc, char **argv) { if ((dvalue >= 0.0) && (dvalue <= 3.0)) config.resend_control_check_interval_time = dvalue; else - warn("Invalid general resend_control_check_interval_time setting \"%d\". It should " + warn("Invalid general resend_control_check_interval_time setting \"%f\". It should " "be " "between 0.0 and 3.0, " "inclusive. The setting remains at %f seconds.", @@ -803,7 +803,7 @@ int parse_options(int argc, char **argv) { if ((dvalue >= 0.0) && (dvalue <= 3.0)) config.resend_control_last_check_time = dvalue; else - warn("Invalid general resend_control_last_check_time setting \"%d\". It should " + warn("Invalid general resend_control_last_check_time setting \"%f\". It should " "be " "between 0.0 and 3.0, " "inclusive. The setting remains at %f seconds.", @@ -815,7 +815,7 @@ int parse_options(int argc, char **argv) { if ((dvalue >= 0.0) && (dvalue <= 300.0)) config.missing_port_dacp_scan_interval_seconds = dvalue; else - warn("Invalid general missing_port_dacp_scan_interval_seconds setting \"%d\". It should " + warn("Invalid general missing_port_dacp_scan_interval_seconds setting \"%f\". It should " "be " "between 0.0 and 300.0, " "inclusive. The setting remains at %f seconds.", @@ -826,6 +826,40 @@ int parse_options(int argc, char **argv) { if (config_lookup_int(config.cfg, "latencies.default", &value)) config.userSuppliedLatency = value; + /* Get the lead-in silence settings. */ + if (config_lookup_float(config.cfg, "general.audio_backend_silent_lead_in_initial_period", + &dvalue)) { + if ((dvalue >= 0.0) && (dvalue <= 3.0)) + config.lead_in_silence_initial_period = dvalue; + else + warn("Invalid general audio_backend_silent_lead_in_initial_period setting \"%f\". It should " + "be " + "between 0.0 and 3.0, " + "inclusive. The setting remains at %f seconds.", + dvalue, config.lead_in_silence_initial_period); + } + + if (config_lookup_int(config.cfg, "general.audio_backend_silent_lead_in_minimum_adjustments_to_make", &value)) { + if (value < 0) + die("Invalid audio_backend_silent_lead_in_minimum_adjustments_to_make range \"%d\". It must be positive. The setting remains at %d.", + value, config.lead_in_silence_minimum_adjustments_to_make); + else + config.lead_in_silence_minimum_adjustments_to_make = value; + } + + if (config_lookup_float(config.cfg, "general.audio_backend_silent_lead_in_adjustment_interval", + &dvalue)) { + if ((dvalue >= 0.0) && (dvalue <= 3.0)) + config.lead_in_silence_adjustment_interval = dvalue; + else + warn("Invalid general audio_backend_silent_lead_in_adjustment_interval setting \"%f\". It should " + "be " + "between 0.0 and 3.0, " + "inclusive. The setting remains at %f seconds.", + dvalue, config.lead_in_silence_adjustment_interval); + } + + #ifdef CONFIG_METADATA /* Get the metadata setting. */ config.metadata_enabled = 1; // if metadata support is included, then enable it by default @@ -1498,6 +1532,9 @@ int main(int argc, char **argv) { config.output_rate_auto_requested = 1; // default auto select format config.decoders_supported = 1 << decoder_hammerton; // David Hammerton's decoder supported by default + config.lead_in_silence_initial_period = 0.5; // the size of the first block of silence to send to the DAC + config.lead_in_silence_minimum_adjustments_to_make = 3; // make at least this number of checks and timing adjustments + config.lead_in_silence_adjustment_interval = 0.025; // make checks and adjustments at this interval #ifdef CONFIG_APPLE_ALAC config.decoders_supported += 1 << decoder_apple_alac; config.use_apple_decoder = 1; // use the ALAC decoder by default if support has been included @@ -1793,6 +1830,13 @@ int main(int argc, char **argv) { else debug(1, "audio backend silence lead-in time is %f seconds.", config.audio_backend_silent_lead_in_time); + + debug(1, "audio backend silent lead-in initial period is %f seconds.", + config.lead_in_silence_initial_period); + debug(1, "audio backend silent lead-in minimum adjustments to make is %d.", config.lead_in_silence_minimum_adjustments_to_make); + debug(1, "audio backend silent lead-in adjustment interval is %f seconds.", + config.lead_in_silence_adjustment_interval); + debug(1, "zeroconf regtype is \"%s\".", config.regtype); debug(1, "decoders_supported field is %d.", config.decoders_supported); debug(1, "use_apple_decoder is %d.", config.use_apple_decoder); From 54d761ff7d0f08b04bb8a8b45588c08a8daf8ca0 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 18 May 2020 16:55:09 +0100 Subject: [PATCH 086/186] Clean up some of the wording and layout of the configuration file Clean up and improve the way lead-in sillence is used to give a more accurate sync time to the first packet of audio. Allow very good sync to occur with 0.3 seconds of lead-in silence. Improve the use of the buffer offset. Format some other files. --- alac.h | 1 - audio.c | 22 +- audio_alsa.c | 260 +++++----- audio_jack.c | 53 +-- audio_sndio.c | 6 +- audio_soundio.c | 5 +- common.c | 10 +- common.h | 21 +- dacp.c | 33 +- dbus-service.c | 65 +-- mdns_avahi.c | 2 +- mdns_dns_sd.c | 2 +- mdns_external.c | 2 +- metadata_hub.c | 89 ++-- mpris-service.c | 17 +- player.c | 766 +++++++++++++++--------------- player.h | 9 +- rtp.c | 47 +- rtsp.c | 75 +-- scripts/shairport-sync.conf | 45 +- shairport-sync-dbus-test-client.c | 26 +- shairport.c | 247 ++++------ tinysvcmdns.c | 7 +- 23 files changed, 907 insertions(+), 903 deletions(-) diff --git a/alac.h b/alac.h index 7cdd507d..51db2c60 100644 --- a/alac.h +++ b/alac.h @@ -27,7 +27,6 @@ * */ - #ifndef __ALAC__DECOMP_H #define __ALAC__DECOMP_H diff --git a/audio.c b/audio.c index 4d5df37b..23b1760d 100644 --- a/audio.c +++ b/audio.c @@ -191,17 +191,17 @@ void parse_general_audio_options(void) { /* Get the latency offset in seconds. */ if (config_lookup_float(config.cfg, "general.audio_backend_latency_offset_in_seconds", &dvalue)) { - config.audio_backend_latency_offset = dvalue; + config.audio_backend_latency_offset = dvalue; } - /* Check if the length of the silent lead-in ia set to \"auto\". */ if (config_lookup_string(config.cfg, "general.audio_backend_silent_lead_in_time", &str)) { if (strcasecmp(str, "auto") == 0) { config.audio_backend_silent_lead_in_time_auto = 1; } else { if (config.audio_backend_silent_lead_in_time_auto == 1) - warn("Invalid audio_backend_silent_lead_in_time \"%s\". It should be \"auto\" or the lead-in time in seconds. " + warn("Invalid audio_backend_silent_lead_in_time \"%s\". It should be \"auto\" or the " + "lead-in time in seconds. " "It remains set to \"auto\". Note: numbers should not be placed in quotes.", str); else @@ -214,14 +214,16 @@ void parse_general_audio_options(void) { /* Get the desired length of the silent lead-in. */ if (config_lookup_float(config.cfg, "general.audio_backend_silent_lead_in_time", &dvalue)) { if ((dvalue < 0.0) || (dvalue > 4)) { - if (config.audio_backend_silent_lead_in_time_auto == 1) - warn("Invalid audio_backend_silent_lead_in_time \"%f\". It " - "must be between 0.0 and 4.0 seconds. Omit the setting to use the automatic value. The setting remains at \"auto\".", - dvalue); + if (config.audio_backend_silent_lead_in_time_auto == 1) + warn("Invalid audio_backend_silent_lead_in_time \"%f\". It " + "must be between 0.0 and 4.0 seconds. Omit the setting to use the automatic value. " + "The setting remains at \"auto\".", + dvalue); else - warn("Invalid audio_backend_silent_lead_in_time \"%f\". It " - "must be between 0.0 and 4.0 seconds. Omit the setting to use the automatic value. It remains set to %f.", - dvalue, config.audio_backend_silent_lead_in_time); + warn("Invalid audio_backend_silent_lead_in_time \"%f\". It " + "must be between 0.0 and 4.0 seconds. Omit the setting to use the automatic value. " + "It remains set to %f.", + dvalue, config.audio_backend_silent_lead_in_time); } else { config.audio_backend_silent_lead_in_time = dvalue; config.audio_backend_silent_lead_in_time_auto = 0; diff --git a/audio_alsa.c b/audio_alsa.c index 40287909..69722d12 100644 --- a/audio_alsa.c +++ b/audio_alsa.c @@ -337,8 +337,9 @@ void actual_close_alsa_device() { if (alsa_handle) { int derr; if ((derr = snd_pcm_hw_free(alsa_handle))) - debug(1, "Error %d (\"%s\") freeing the output device hardware while " - "closing it.", + debug(1, + "Error %d (\"%s\") freeing the output device hardware while " + "closing it.", derr, snd_strerror(derr)); if ((derr = snd_pcm_close(alsa_handle))) @@ -353,7 +354,10 @@ void actual_close_alsa_device() { // The lowest rate that the DAC is capable of is chosen. unsigned int auto_speed_output_rates[] = { - 44100, 88200, 176400, 352800, + 44100, + 88200, + 176400, + 352800, }; // This array is of all the formats known to Shairport Sync, in order of the SPS_FORMAT definitions, @@ -702,9 +706,10 @@ int actual_open_alsa_device(int do_auto_setup) { buffer_size); } */ - debug(1, "The alsa buffer is smaller (%lu bytes) than the desired backend " - "buffer " - "length (%ld) you have chosen.", + debug(1, + "The alsa buffer is smaller (%lu bytes) than the desired backend " + "buffer " + "length (%ld) you have chosen.", actual_buffer_length, config.audio_backend_buffer_desired_length); } @@ -864,120 +869,117 @@ int prepare_mixer() { // do any alsa device initialisation (general case) // at present, this is only needed if a hardware mixer is being used // if there's a hardware mixer, it needs to be initialised before use - if (alsa_mix_ctrl == NULL) { - audio_alsa.volume = NULL; - audio_alsa.parameters = NULL; - audio_alsa.mute = NULL; - } else { - debug(2, "alsa: hardware mixer prepare"); - int oldState; - pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable + if (alsa_mix_ctrl == NULL) { + audio_alsa.volume = NULL; + audio_alsa.parameters = NULL; + audio_alsa.mute = NULL; + } else { + debug(2, "alsa: hardware mixer prepare"); + int oldState; + pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable - if (alsa_mix_dev == NULL) - alsa_mix_dev = alsa_out_dev; + if (alsa_mix_dev == NULL) + alsa_mix_dev = alsa_out_dev; - // Now, start trying to initialise the alsa device with the settings - // obtained - pthread_cleanup_debug_mutex_lock(&alsa_mixer_mutex, 1000, 1); - if (open_mixer() == 1) { - if (snd_mixer_selem_get_playback_volume_range(alsa_mix_elem, &alsa_mix_minv, - &alsa_mix_maxv) < 0) - debug(1, "Can't read mixer's [linear] min and max volumes."); - else { - if (snd_mixer_selem_get_playback_dB_range(alsa_mix_elem, &alsa_mix_mindb, - &alsa_mix_maxdb) == 0) { + // Now, start trying to initialise the alsa device with the settings + // obtained + pthread_cleanup_debug_mutex_lock(&alsa_mixer_mutex, 1000, 1); + if (open_mixer() == 1) { + if (snd_mixer_selem_get_playback_volume_range(alsa_mix_elem, &alsa_mix_minv, &alsa_mix_maxv) < + 0) + debug(1, "Can't read mixer's [linear] min and max volumes."); + else { + if (snd_mixer_selem_get_playback_dB_range(alsa_mix_elem, &alsa_mix_mindb, + &alsa_mix_maxdb) == 0) { - audio_alsa.volume = &volume; // insert the volume function now we - // know it can do dB stuff - audio_alsa.parameters = ¶meters; // likewise the parameters stuff - if (alsa_mix_mindb == SND_CTL_TLV_DB_GAIN_MUTE) { - // For instance, the Raspberry Pi does this - debug(1, "Lowest dB value is a mute"); - mixer_volume_setting_gives_mute = 1; - alsa_mix_mute = SND_CTL_TLV_DB_GAIN_MUTE; // this may not be - // necessary -- it's - // always - // going to be SND_CTL_TLV_DB_GAIN_MUTE, right? - // debug(1, "Try minimum volume + 1 as lowest true attenuation - // value"); - if (snd_mixer_selem_ask_playback_vol_dB(alsa_mix_elem, alsa_mix_minv + 1, - &alsa_mix_mindb) != 0) - debug(1, "Can't get dB value corresponding to a minimum volume " - "+ 1."); - } - debug(3, "Hardware mixer has dB volume from %f to %f.", (1.0 * alsa_mix_mindb) / 100.0, - (1.0 * alsa_mix_maxdb) / 100.0); - } else { - // use the linear scale and do the db conversion ourselves - warn("The hardware mixer specified -- \"%s\" -- does not have " - "a dB volume scale.", - alsa_mix_ctrl); - - if (snd_ctl_open(&ctl, alsa_mix_dev, 0) < 0) { - warn("Cannot open control \"%s\"", alsa_mix_dev); - response = -1; - } - if (snd_ctl_elem_id_malloc(&elem_id) < 0) { - debug(1, "Cannot allocate memory for control \"%s\"", alsa_mix_dev); - elem_id = NULL; - response = -2; - } else { - snd_ctl_elem_id_set_interface(elem_id, SND_CTL_ELEM_IFACE_MIXER); - snd_ctl_elem_id_set_name(elem_id, alsa_mix_ctrl); - - if (snd_ctl_get_dB_range(ctl, elem_id, &alsa_mix_mindb, &alsa_mix_maxdb) == 0) { - debug(1, "alsa: hardware mixer \"%s\" selected, with dB volume " - "from %f to %f.", - alsa_mix_ctrl, (1.0 * alsa_mix_mindb) / 100.0, - (1.0 * alsa_mix_maxdb) / 100.0); - has_softvol = 1; - audio_alsa.volume = &volume; // insert the volume function now - // we know it can do dB stuff - audio_alsa.parameters = ¶meters; // likewise the parameters stuff - } else { - debug(1, "Cannot get the dB range from the volume control \"%s\"", alsa_mix_ctrl); - } - } - /* - debug(1, "Min and max volumes are %d and - %d.",alsa_mix_minv,alsa_mix_maxv); - alsa_mix_maxdb = 0; - if ((alsa_mix_maxv!=0) && (alsa_mix_minv!=0)) - alsa_mix_mindb = - -20*100*(log10(alsa_mix_maxv*1.0)-log10(alsa_mix_minv*1.0)); - else if (alsa_mix_maxv!=0) - alsa_mix_mindb = -20*100*log10(alsa_mix_maxv*1.0); - audio_alsa.volume = &linear_volume; // insert the linear volume - function - audio_alsa.parameters = ¶meters; // likewise the parameters - stuff - debug(1,"Max and min dB calculated are %d and - %d.",alsa_mix_maxdb,alsa_mix_mindb); - */ + audio_alsa.volume = &volume; // insert the volume function now we + // know it can do dB stuff + audio_alsa.parameters = ¶meters; // likewise the parameters stuff + if (alsa_mix_mindb == SND_CTL_TLV_DB_GAIN_MUTE) { + // For instance, the Raspberry Pi does this + debug(1, "Lowest dB value is a mute"); + mixer_volume_setting_gives_mute = 1; + alsa_mix_mute = SND_CTL_TLV_DB_GAIN_MUTE; // this may not be + // necessary -- it's + // always + // going to be SND_CTL_TLV_DB_GAIN_MUTE, right? + // debug(1, "Try minimum volume + 1 as lowest true attenuation + // value"); + if (snd_mixer_selem_ask_playback_vol_dB(alsa_mix_elem, alsa_mix_minv + 1, + &alsa_mix_mindb) != 0) + debug(1, "Can't get dB value corresponding to a minimum volume " + "+ 1."); } - } - if (((config.alsa_use_hardware_mute == 1) && - (snd_mixer_selem_has_playback_switch(alsa_mix_elem))) || - mixer_volume_setting_gives_mute) { - audio_alsa.mute = &mute; // insert the mute function now we know it - // can do muting stuff - // debug(1, "Has mixer and mute ability we will use."); + debug(3, "Hardware mixer has dB volume from %f to %f.", (1.0 * alsa_mix_mindb) / 100.0, + (1.0 * alsa_mix_maxdb) / 100.0); } else { - // debug(1, "Has mixer but not using hardware mute."); + // use the linear scale and do the db conversion ourselves + warn("The hardware mixer specified -- \"%s\" -- does not have " + "a dB volume scale.", + alsa_mix_ctrl); + + if (snd_ctl_open(&ctl, alsa_mix_dev, 0) < 0) { + warn("Cannot open control \"%s\"", alsa_mix_dev); + response = -1; + } + if (snd_ctl_elem_id_malloc(&elem_id) < 0) { + debug(1, "Cannot allocate memory for control \"%s\"", alsa_mix_dev); + elem_id = NULL; + response = -2; + } else { + snd_ctl_elem_id_set_interface(elem_id, SND_CTL_ELEM_IFACE_MIXER); + snd_ctl_elem_id_set_name(elem_id, alsa_mix_ctrl); + + if (snd_ctl_get_dB_range(ctl, elem_id, &alsa_mix_mindb, &alsa_mix_maxdb) == 0) { + debug(1, + "alsa: hardware mixer \"%s\" selected, with dB volume " + "from %f to %f.", + alsa_mix_ctrl, (1.0 * alsa_mix_mindb) / 100.0, (1.0 * alsa_mix_maxdb) / 100.0); + has_softvol = 1; + audio_alsa.volume = &volume; // insert the volume function now + // we know it can do dB stuff + audio_alsa.parameters = ¶meters; // likewise the parameters stuff + } else { + debug(1, "Cannot get the dB range from the volume control \"%s\"", alsa_mix_ctrl); + } + } + /* + debug(1, "Min and max volumes are %d and + %d.",alsa_mix_minv,alsa_mix_maxv); + alsa_mix_maxdb = 0; + if ((alsa_mix_maxv!=0) && (alsa_mix_minv!=0)) + alsa_mix_mindb = + -20*100*(log10(alsa_mix_maxv*1.0)-log10(alsa_mix_minv*1.0)); + else if (alsa_mix_maxv!=0) + alsa_mix_mindb = -20*100*log10(alsa_mix_maxv*1.0); + audio_alsa.volume = &linear_volume; // insert the linear volume + function + audio_alsa.parameters = ¶meters; // likewise the parameters + stuff + debug(1,"Max and min dB calculated are %d and + %d.",alsa_mix_maxdb,alsa_mix_mindb); + */ } - close_mixer(); } - debug_mutex_unlock(&alsa_mixer_mutex, 3); // release the mutex - pthread_cleanup_pop(0); - pthread_setcancelstate(oldState, NULL); + if (((config.alsa_use_hardware_mute == 1) && + (snd_mixer_selem_has_playback_switch(alsa_mix_elem))) || + mixer_volume_setting_gives_mute) { + audio_alsa.mute = &mute; // insert the mute function now we know it + // can do muting stuff + // debug(1, "Has mixer and mute ability we will use."); + } else { + // debug(1, "Has mixer but not using hardware mute."); + } + close_mixer(); } + debug_mutex_unlock(&alsa_mixer_mutex, 3); // release the mutex + pthread_cleanup_pop(0); + pthread_setcancelstate(oldState, NULL); + } return response; } -int alsa_device_init() { - return prepare_mixer(); -} - +int alsa_device_init() { return prepare_mixer(); } static int init(int argc, char **argv) { // for debugging @@ -1135,8 +1137,10 @@ static int init(int argc, char **argv) { "\"S16\", \"S24\", \"S24_LE\", \"S24_BE\", " "\"S24_3LE\", \"S24_3BE\" or " "\"S32\", \"S32_LE\", \"S32_BE\". It remains set to \"%s\".", - str, config.output_format_auto_requested == 1 ? "auto" : sps_format_description_string( - config.output_format)); + str, + config.output_format_auto_requested == 1 + ? "auto" + : sps_format_description_string(config.output_format)); } } @@ -1286,9 +1290,9 @@ static int init(int argc, char **argv) { warn("Invalid use_precision_timing option choice \"%s\". It should be " "\"yes\", \"auto\" or \"no\". " "It remains set to \"%s\".", - config.use_precision_timing == YNA_NO ? "no" : config.use_precision_timing == YNA_AUTO - ? "auto" - : "yes"); + config.use_precision_timing == YNA_NO + ? "no" + : config.use_precision_timing == YNA_AUTO ? "auto" : "yes"); } } @@ -1519,14 +1523,16 @@ int precision_delay_and_status(snd_pcm_state_t *state, snd_pcm_sframes_t *delay, if (((update_timestamp_ns - stall_monitor_start_time) > stall_monitor_error_threshold) || ((time_now_ns - stall_monitor_start_time) > stall_monitor_error_threshold)) { - debug(2, "DAC seems to have stalled with time_now_ns: %" PRIX64 - ", update_timestamp_ns: %" PRIX64 ", stall_monitor_start_time %" PRIX64 - ", stall_monitor_error_threshold %" PRIX64 ".", + debug(2, + "DAC seems to have stalled with time_now_ns: %" PRIX64 + ", update_timestamp_ns: %" PRIX64 ", stall_monitor_start_time %" PRIX64 + ", stall_monitor_error_threshold %" PRIX64 ".", time_now_ns, update_timestamp_ns, stall_monitor_start_time, stall_monitor_error_threshold); - debug(2, "DAC seems to have stalled with time_now: %lx,%lx" - ", update_timestamp: %lx,%lx, stall_monitor_start_time %" PRIX64 - ", stall_monitor_error_threshold %" PRIX64 ".", + debug(2, + "DAC seems to have stalled with time_now: %lx,%lx" + ", update_timestamp: %lx,%lx, stall_monitor_start_time %" PRIX64 + ", stall_monitor_error_threshold %" PRIX64 ".", tn.tv_sec, tn.tv_nsec, update_timestamp.tv_sec, update_timestamp.tv_nsec, stall_monitor_start_time, stall_monitor_error_threshold); ret = sps_extra_code_output_stalled; @@ -1692,8 +1698,9 @@ int do_play(void *buf, int samples) { } } } else { - debug(1, "alsa: device status returns fault status %d and SND_PCM_STATE_* " - "%d for play.", + debug(1, + "alsa: device status returns fault status %d and SND_PCM_STATE_* " + "%d for play.", ret, state); frame_index = 0; measurement_data_is_valid = 0; @@ -2007,8 +2014,9 @@ void *alsa_buffer_monitor_thread_code(__attribute__((unused)) void *arg) { error_count++; char errorstring[1024]; strerror_r(-ret, (char *)errorstring, sizeof(errorstring)); - debug(2, "alsa: alsa_buffer_monitor_thread_code error %d (\"%s\") writing %d samples " - "to alsa device -- %d errors in %d trials.", + debug(2, + "alsa: alsa_buffer_monitor_thread_code error %d (\"%s\") writing %d samples " + "to alsa device -- %d errors in %d trials.", ret, (char *)errorstring, frames_of_silence, error_count, frame_count); if ((error_count > 40) && (frame_count < 100)) { warn("disable_standby_mode has been turned off because too many underruns " @@ -2021,8 +2029,8 @@ void *alsa_buffer_monitor_thread_code(__attribute__((unused)) void *arg) { } } debug_mutex_unlock(&alsa_mutex, 0); - pthread_cleanup_pop(0); // release the mutex - usleep(sleep_time_us); // has a cancellation point in it + pthread_cleanup_pop(0); // release the mutex + usleep(sleep_time_us); // has a cancellation point in it } pthread_exit(NULL); } diff --git a/audio_jack.c b/audio_jack.c index 5caa6c68..21c3e543 100644 --- a/audio_jack.c +++ b/audio_jack.c @@ -89,14 +89,9 @@ typedef struct soxr_quality { const char *name; } soxr_quality_t; -static soxr_quality_t soxr_quality_table[] = { - { SOXR_VHQ, "very high" }, - { SOXR_HQ, "high" }, - { SOXR_MQ, "medium" }, - { SOXR_LQ, "low" }, - { SOXR_QQ, "quick" }, - { -1, NULL } -}; +static soxr_quality_t soxr_quality_table[] = {{SOXR_VHQ, "very high"}, {SOXR_HQ, "high"}, + {SOXR_MQ, "medium"}, {SOXR_LQ, "low"}, + {SOXR_QQ, "quick"}, {-1, NULL}}; static int parse_soxr_quality_name(const char *name) { for (soxr_quality_t *s = soxr_quality_table; s->name != NULL; ++s) { @@ -107,9 +102,9 @@ static int parse_soxr_quality_name(const char *name) { return -1; } -static soxr_t soxr = NULL; +static soxr_t soxr = NULL; static soxr_quality_spec_t quality_spec; -static soxr_io_spec_t io_spec; +static soxr_io_spec_t io_spec; #endif static inline sample_t sample_conv(short sample) { @@ -122,8 +117,7 @@ static inline sample_t sample_conv(short sample) { return ((sample < 0) ? (-1.0 * sample / SHRT_MIN) : (1.0 * sample / SHRT_MAX)); } -static void deinterleave(const char *interleaved_input_buffer, - sample_t *jack_output_buffer[], +static void deinterleave(const char *interleaved_input_buffer, sample_t *jack_output_buffer[], jack_nframes_t offset, jack_nframes_t nframes) { jack_nframes_t f; // We're dealing with 16bit audio here: @@ -271,7 +265,7 @@ int jack_init(__attribute__((unused)) int argc, __attribute__((unused)) char **a io_spec = soxr_io_spec(SOXR_INT16_I, SOXR_FLOAT32_I); } else #endif - if (sample_rate != 44100) { + if (sample_rate != 44100) { die("The JACK server is running at the wrong sample rate (%d) for Shairport Sync." " Must be 44100 Hz.", sample_rate); @@ -353,8 +347,7 @@ void jack_deinit() { #endif } -void jack_start(int i_sample_rate, - __attribute__((unused)) int i_sample_format) { +void jack_start(int i_sample_rate, __attribute__((unused)) int i_sample_format) { // Nothing to do, JACK client has already been set up at jack_init(). // Also, we have no say over the sample rate or sample format of JACK, // We convert the 16bit samples to float, and die if the sample rate is != 44k1 without soxr. @@ -365,13 +358,7 @@ void jack_start(int i_sample_rate, soxr_delete(soxr); } soxr_error_t e = NULL; - soxr = soxr_create(i_sample_rate, - sample_rate, - NPORTS, - &e, - &io_spec, - &quality_spec, - NULL); + soxr = soxr_create(i_sample_rate, sample_rate, NPORTS, &e, &io_spec, &quality_spec, NULL); if (!soxr) { die("Unable to create soxr resampler for JACK: %s", e); } @@ -428,13 +415,7 @@ int play(void *buf, int samples) { size_t i_done, o_done; soxr_error_t e; while (samples > 0 && thisbuf > 0) { - e = soxr_process(soxr, - (soxr_in_t)in, - samples, - &i_done, - (soxr_out_t)out, - thisbuf, - &o_done); + e = soxr_process(soxr, (soxr_in_t)in, samples, &i_done, (soxr_out_t)out, thisbuf, &o_done); if (e) die("Error during soxr process: %s", e); @@ -445,13 +426,13 @@ int play(void *buf, int samples) { } } else { #endif - j = 0; - for (j = 0; j < thisbuf && samples > 0; ++j) { - for (c = 0; c < NPORTS; ++c) - out[j * NPORTS + c] = sample_conv(*in++); - --samples; - } - jack_ringbuffer_write_advance(jackbuf, j * jack_sample_size * NPORTS); + j = 0; + for (j = 0; j < thisbuf && samples > 0; ++j) { + for (c = 0; c < NPORTS; ++c) + out[j * NPORTS + c] = sample_conv(*in++); + --samples; + } + jack_ringbuffer_write_advance(jackbuf, j * jack_sample_size * NPORTS); #ifdef CONFIG_SOXR } #endif diff --git a/audio_sndio.c b/audio_sndio.c index e7aaaf42..f8582c8a 100644 --- a/audio_sndio.c +++ b/audio_sndio.c @@ -260,12 +260,12 @@ static int delay(long *_delay) { // and use it to estimate the frames that would have been output uint64_t time_difference = get_absolute_time_in_ns() - time_of_last_onmove_cb; uint64_t frame_difference = (time_difference * par.rate) / 1000000000; - estimated_extra_frames_output = frame_difference; + estimated_extra_frames_output = frame_difference; // sanity check -- total estimate can not exceed frames written. - if ((estimated_extra_frames_output + played) > written/framesize) { + if ((estimated_extra_frames_output + played) > written / framesize) { // debug(1,"play estimate fails sanity check, possibly due to running on a VM"); estimated_extra_frames_output = 0; // can't make any sensible guess - } + } // debug(1,"Frames played to last cb: %d, estimated to current time: // %d.",played,estimated_extra_frames_output); } diff --git a/audio_soundio.c b/audio_soundio.c index 119f7670..501c6db2 100644 --- a/audio_soundio.c +++ b/audio_soundio.c @@ -24,8 +24,9 @@ static void write_callback(struct SoundIoOutStream *outstream, int frame_count_m int fill_bytes = soundio_ring_buffer_fill_count(ring_buffer); int fill_count = fill_bytes / outstream->bytes_per_frame; - debug(3, "[--->>] frame_count_min: %d , frame_count_max: %d , fill_bytes: %d , fill_count: %d , " - "outstream->bytes_per_frame: %d", + debug(3, + "[--->>] frame_count_min: %d , frame_count_max: %d , fill_bytes: %d , fill_count: %d , " + "outstream->bytes_per_frame: %d", frame_count_min, frame_count_max, fill_bytes, fill_count, outstream->bytes_per_frame); if (frame_count_min > fill_count) { diff --git a/common.c b/common.c index 2958cfe6..4aba7e1a 100644 --- a/common.c +++ b/common.c @@ -1578,9 +1578,11 @@ int64_t generate_zero_frames(char *outp, size_t number_of_frames, sps_format_t f return previous_random_number; } -// This will check the incoming string "s" of length "len" with the existing NUL-terminated string "str" and update "flag" accordingly. +// This will check the incoming string "s" of length "len" with the existing NUL-terminated string +// "str" and update "flag" accordingly. -// Note: if the incoming string length is zero, then the a NULL is used; i.e. no zero-length strings are stored. +// Note: if the incoming string length is zero, then the a NULL is used; i.e. no zero-length strings +// are stored. // If the strings are different, the str is free'd and replaced by a pointer // to a newly strdup'd string and the flag is set @@ -1593,7 +1595,7 @@ int string_update_with_size(char **str, int *flag, char *s, size_t len) { free(*str); //*str = strndup(s, len); // it seems that OpenWrt 12 doesn't have this char *p = malloc(len + 1); - memcpy(p,s,len); + memcpy(p, s, len); p[len] = '\0'; *str = p; *flag = 1; @@ -1610,7 +1612,7 @@ int string_update_with_size(char **str, int *flag, char *s, size_t len) { if ((s) && (len)) { //*str = strndup(s, len); // it seems that OpenWrt 12 doesn't have this char *p = malloc(len + 1); - memcpy(p,s,len); + memcpy(p, s, len); p[len] = '\0'; *str = p; *flag = 1; diff --git a/common.h b/common.h index ec9f35a1..e7efc72e 100644 --- a/common.h +++ b/common.h @@ -100,10 +100,13 @@ typedef enum { const char *sps_format_description_string(sps_format_t format); typedef struct { - double missing_port_dacp_scan_interval_seconds; // if no DACP port number can be found, check at these intervals - double resend_control_first_check_time; // wait this long before asking for a missing packet to be resent + double missing_port_dacp_scan_interval_seconds; // if no DACP port number can be found, check at + // these intervals + double resend_control_first_check_time; // wait this long before asking for a missing packet to be + // resent double resend_control_check_interval_time; // wait this long between making requests - double resend_control_last_check_time; // if the packet is missing this close to the time of use, give up + double resend_control_last_check_time; // if the packet is missing this close to the time of use, + // give up pthread_mutex_t lock; config_t *cfg; int endianness; @@ -114,7 +117,7 @@ typedef struct { // on host %h" #ifdef CONFIG_PA - char *pa_server; // the pulseaudio server address that Shairport Sync will play on. + char *pa_server; // the pulseaudio server address that Shairport Sync will play on. char *pa_application_name; // the name under which Shairport Sync shows up as an "Application" in // the Sound Preferences in most desktop Linuxes. // Defaults to "Shairport Sync". Shairport Sync must be playing to see it. @@ -212,15 +215,13 @@ typedef struct { // buffer double disable_standby_mode_silence_scan_interval; // check the threshold this often - double lead_in_silence_initial_period; // the size of the first block of silence to send to the DAC - int lead_in_silence_minimum_adjustments_to_make; // make at least this number of checks and timing adjustments - double lead_in_silence_adjustment_interval; // make checks and adjustments at this interval - double audio_backend_latency_offset; // this will be the offset in seconds to compensate for any // fixed latency there might be in the audio path - int audio_backend_silent_lead_in_time_auto; // true if the lead-in time should be from as soon as packets are received + int audio_backend_silent_lead_in_time_auto; // true if the lead-in time should be from as soon as + // packets are received double audio_backend_silent_lead_in_time; // the length of the silence that should precede a play. - uint32_t minimum_free_buffer_headroom; // when effective latency is calculated, ensure this number of buffers are unallocated + uint32_t minimum_free_buffer_headroom; // when effective latency is calculated, ensure this number + // of buffers are unallocated double active_state_timeout; // the amount of time from when play ends to when the system leaves // into the "active" mode. uint32_t volume_range_db; // the range, in dB, from max dB to min dB. Zero means use the mixer's diff --git a/dacp.c b/dacp.c index b8e41d67..82fc0dbb 100644 --- a/dacp.c +++ b/dacp.c @@ -154,7 +154,7 @@ int dacp_send_command(const char *command, char **body, ssize_t *bodysize) { // debug(1,"dacp_send_command: command is: \"%s\".",command); if (dacp_server.port == 0) { - debug(2, "No DACP port specified yet"); + debug(3, "No DACP port specified yet"); result = 490; // no port specified } else { @@ -222,18 +222,16 @@ int dacp_send_command(const char *command, char **body, ssize_t *bodysize) { pthread_cleanup_push(connect_cleanup, (void *)&sockfd); // debug(2, "dacp_send_command: open socket %d.",sockfd); - // This is for limiting the time to be spent waiting for a response. struct timeval tv; tv.tv_sec = 0; tv.tv_usec = 500000; if (setsockopt(sockfd, SOL_SOCKET, SO_RCVTIMEO, (const char *)&tv, sizeof tv) == -1) - debug(1, "dacp_send_command: error %d setting receive timeout.", errno); + debug(1, "dacp_send_command: error %d setting receive timeout.", errno); if (setsockopt(sockfd, SOL_SOCKET, SO_SNDTIMEO, (const char *)&tv, sizeof tv) == -1) debug(1, "dacp_send_command: error %d setting send timeout.", errno); - // connect! // debug(1, "DACP socket created."); if (connect(sockfd, res->ai_addr, res->ai_addrlen) < 0) { @@ -418,7 +416,8 @@ void set_dacp_server_information(rtsp_conn_info *conn) { // which return immediately with a 403 code if there are no changes. dacp_server.always_use_revision_number_1 = 0; char *p = strstr(conn->UserAgent, "forked-daapd"); - if ((p != 0) && (p == conn->UserAgent)) {// must exist and be at the start of the UserAgent string + if ((p != 0) && + (p == conn->UserAgent)) { // must exist and be at the start of the UserAgent string dacp_server.always_use_revision_number_1 = 1; } @@ -505,14 +504,15 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { (metadata_store.advanced_dacp_server_active != 0); metadata_store.dacp_server_active = 0; metadata_store.advanced_dacp_server_active = 0; - debug(2, + debug(3, "setting metadata_store.dacp_server_active and " "metadata_store.advanced_dacp_server_active to 0 with an update " "flag value of %d", ch); metadata_hub_modify_epilog(ch); - uint64_t time_to_wait_for_wakeup_ns = (uint64_t)(1000000000 * config.missing_port_dacp_scan_interval_seconds); + uint64_t time_to_wait_for_wakeup_ns = + (uint64_t)(1000000000 * config.missing_port_dacp_scan_interval_seconds); #ifdef COMPILE_FOR_LINUX_AND_FREEBSD_AND_CYGWIN_AND_OPENBSD uint64_t time_of_wakeup_ns = get_absolute_time_in_ns() + time_to_wait_for_wakeup_ns; @@ -534,11 +534,13 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { #ifdef COMPILE_FOR_LINUX_AND_FREEBSD_AND_CYGWIN_AND_OPENBSD result = pthread_cond_timedwait(&dacp_server_information_cv, &dacp_server_information_lock, &time_to_wait); // this is a pthread cancellation point - // debug(1, "result is %d, dacp_server.scan_enable is %d, time_to_wait_for_wakeup_ns is %" PRId64 ".", result, dacp_server.scan_enable, time_to_wait_for_wakeup_ns); + // debug(1, "result is %d, dacp_server.scan_enable is %d, time_to_wait_for_wakeup_ns is %" + // PRId64 ".", result, dacp_server.scan_enable, time_to_wait_for_wakeup_ns); #endif #ifdef COMPILE_FOR_OSX - result = pthread_cond_timedwait_relative_np(&dacp_server_information_cv, &dacp_server_information_lock, &time_to_wait); + result = pthread_cond_timedwait_relative_np(&dacp_server_information_cv, + &dacp_server_information_lock, &time_to_wait); #endif } if (dacp_server.scan_enable == 1) { @@ -547,12 +549,13 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { } } - always_use_revision_number_1 = dacp_server.always_use_revision_number_1; // set this while access is locked + always_use_revision_number_1 = + dacp_server.always_use_revision_number_1; // set this while access is locked result = dacp_get_volume(&the_volume); // just want the http code pthread_cleanup_pop(1); - if (result == 490) { // 490 means no port was specified + if (result == 490) { // 490 means no port was specified if ((dacp_server.dacp_id != NULL) && (strlen(dacp_server.dacp_id) != 0)) { // debug(1,"mdns_dacp_monitor_set_id"); mdns_dacp_monitor_set_id(dacp_server.dacp_id); @@ -597,8 +600,8 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { else idle_scan_count = 0; - debug(3, "Scan Result: %d, Bad Scan Count: %d, Idle Scan Count: %d.", result, bad_result_count, - idle_scan_count); + debug(3, "Scan Result: %d, Bad Scan Count: %d, Idle Scan Count: %d.", result, + bad_result_count, idle_scan_count); /* not used // decide if idle for too long @@ -934,8 +937,6 @@ void dacp_monitor_start() { rc = pthread_cond_init(&dacp_server_information_cv, NULL); #endif - - pthread_mutexattr_t mta; rc = pthread_mutexattr_init(&mta); @@ -1248,7 +1249,7 @@ int dacp_get_volume(int32_t *the_actual_volume) { } else { debug(1, "Unexpected return code %d from dacp_get_speaker_list.", http_response); } - } else if (http_response != 400){ + } else if (http_response != 400) { debug(3, "Unexpected return code %d from dacp_get_client_volume.", http_response); } if (the_actual_volume) { diff --git a/dbus-service.c b/dbus-service.c index 457aefed..3c4e6daf 100644 --- a/dbus-service.c +++ b/dbus-service.c @@ -24,10 +24,10 @@ * OTHER DEALINGS IN THE SOFTWARE. */ +#include #include #include #include -#include #include "config.h" @@ -69,8 +69,8 @@ void dbus_metadata_watcher(struct metadata_bundle *argc, __attribute__((unused)) shairport_sync_remote_control_set_client(shairportSyncRemoteControlSkeleton, argc->client_ip); - - // although it's a DACP server, the server is in fact, part of the the AirPlay "client" (their term). + // although it's a DACP server, the server is in fact, part of the the AirPlay "client" (their + // term). if (argc->dacp_server_active) { shairport_sync_remote_control_set_available(shairportSyncRemoteControlSkeleton, TRUE); } else { @@ -166,7 +166,6 @@ void dbus_metadata_watcher(struct metadata_bundle *argc, __attribute__((unused)) shairportSyncAdvancedRemoteControlSkeleton, response); } - switch (argc->shuffle_status) { case SS_NOT_AVAILABLE: new_status = FALSE; @@ -188,8 +187,8 @@ void dbus_metadata_watcher(struct metadata_bundle *argc, __attribute__((unused)) // only set this if it's different if (current_status != new_status) { debug(3, "Shuffle State should be changed"); - shairport_sync_advanced_remote_control_set_shuffle( - shairportSyncAdvancedRemoteControlSkeleton, new_status); + shairport_sync_advanced_remote_control_set_shuffle(shairportSyncAdvancedRemoteControlSkeleton, + new_status); } // Build the metadata array @@ -369,8 +368,9 @@ static gboolean on_handle_volume_down(ShairportSyncRemoteControl *skeleton, } static gboolean on_handle_set_airplay_volume(ShairportSyncRemoteControl *skeleton, - GDBusMethodInvocation *invocation, const gdouble volume, - __attribute__((unused)) gpointer user_data) { + GDBusMethodInvocation *invocation, + const gdouble volume, + __attribute__((unused)) gpointer user_data) { debug(2, "Set airplay volume to %.6f.", volume); char command[256] = ""; snprintf(command, sizeof(command), "setproperty?dmcp.device-volume=%.6f", volume); @@ -379,8 +379,6 @@ static gboolean on_handle_set_airplay_volume(ShairportSyncRemoteControl *skeleto return TRUE; } - - gboolean notify_elapsed_time_callback(ShairportSyncDiagnostics *skeleton, __attribute__((unused)) gpointer user_data) { // debug(1, "\"notify_elapsed_time_callback\" called."); @@ -408,7 +406,7 @@ gboolean notify_delta_time_callback(ShairportSyncDiagnostics *skeleton, } gboolean notify_file_and_line_callback(ShairportSyncDiagnostics *skeleton, - __attribute__((unused)) gpointer user_data) { + __attribute__((unused)) gpointer user_data) { // debug(1, "\"notify_file_and_line_callback\" called."); if (shairport_sync_diagnostics_get_file_and_line(skeleton)) { config.debugger_show_file_and_line = 1; @@ -463,12 +461,13 @@ gboolean notify_disable_standby_callback(ShairportSync *skeleton, #ifdef CONFIG_CONVOLUTION gboolean notify_convolution_callback(ShairportSync *skeleton, - __attribute__((unused)) gpointer user_data) { + __attribute__((unused)) gpointer user_data) { // debug(1, "\"notify_convolution_callback\" called."); if (shairport_sync_get_convolution(skeleton)) { debug(1, ">> activating convolution"); config.convolution = 1; - config.convolver_valid = convolver_init(config.convolution_ir_file, config.convolution_max_length); + config.convolver_valid = + convolver_init(config.convolution_ir_file, config.convolution_max_length); } else { debug(1, ">> deactivating convolution"); config.convolution = 0; @@ -477,7 +476,7 @@ gboolean notify_convolution_callback(ShairportSync *skeleton, } #else gboolean notify_convolution_callback(__attribute__((unused)) ShairportSync *skeleton, - __attribute__((unused)) gpointer user_data) { + __attribute__((unused)) gpointer user_data) { warn(">> Convolution support is not built in to this build of Shairport Sync."); return TRUE; } @@ -485,7 +484,7 @@ gboolean notify_convolution_callback(__attribute__((unused)) ShairportSync *skel #ifdef CONFIG_CONVOLUTION gboolean notify_convolution_gain_callback(ShairportSync *skeleton, - __attribute__((unused)) gpointer user_data) { + __attribute__((unused)) gpointer user_data) { gdouble th = shairport_sync_get_convolution_gain(skeleton); if ((th <= 0.0) && (th >= -100.0)) { @@ -499,34 +498,38 @@ gboolean notify_convolution_gain_callback(ShairportSync *skeleton, } #else gboolean notify_convolution_gain_callback(__attribute__((unused)) ShairportSync *skeleton, - __attribute__((unused)) gpointer user_data) { + __attribute__((unused)) gpointer user_data) { warn(">> Convolution support is not built in to this build of Shairport Sync."); return TRUE; } #endif #ifdef CONFIG_CONVOLUTION gboolean notify_convolution_impulse_response_file_callback(ShairportSync *skeleton, - __attribute__((unused)) gpointer user_data) { + __attribute__((unused)) + gpointer user_data) { char *th = (char *)shairport_sync_get_convolution_impulse_response_file(skeleton); if (config.convolution_ir_file) free(config.convolution_ir_file); config.convolution_ir_file = strdup(th); - debug(1, ">> setting configuration impulse response filter file to \"%s\".", config.convolution_ir_file); - config.convolver_valid = convolver_init(config.convolution_ir_file, config.convolution_max_length); + debug(1, ">> setting configuration impulse response filter file to \"%s\".", + config.convolution_ir_file); + config.convolver_valid = + convolver_init(config.convolution_ir_file, config.convolution_max_length); return TRUE; } #else -gboolean notify_convolution_impulse_response_file_callback(__attribute__((unused)) ShairportSync *skeleton, - __attribute__((unused)) gpointer user_data) { - __attribute__((unused)) char *th = (char *)shairport_sync_get_convolution_impulse_response_file(skeleton); +gboolean notify_convolution_impulse_response_file_callback(__attribute__((unused)) + ShairportSync *skeleton, + __attribute__((unused)) + gpointer user_data) { + __attribute__((unused)) char *th = + (char *)shairport_sync_get_convolution_impulse_response_file(skeleton); return TRUE; } #endif - - gboolean notify_loudness_callback(ShairportSync *skeleton, - __attribute__((unused)) gpointer user_data) { + __attribute__((unused)) gpointer user_data) { // debug(1, "\"notify_loudness_callback\" called."); if (shairport_sync_get_loudness(skeleton)) { debug(1, ">> activating loudness"); @@ -781,7 +784,6 @@ static gboolean on_handle_remote_command(ShairportSync *skeleton, GDBusMethodInv return TRUE; } - static void on_dbus_name_acquired(GDBusConnection *connection, const gchar *name, __attribute__((unused)) gpointer user_data) { @@ -823,8 +825,8 @@ static void on_dbus_name_acquired(GDBusConnection *connection, const gchar *name G_CALLBACK(notify_convolution_gain_callback), NULL); g_signal_connect(shairportSyncSkeleton, "notify::convolution-impulse-response-file", G_CALLBACK(notify_convolution_impulse_response_file_callback), NULL); - g_signal_connect(shairportSyncSkeleton, "notify::loudness", - G_CALLBACK(notify_loudness_callback), NULL); + g_signal_connect(shairportSyncSkeleton, "notify::loudness", G_CALLBACK(notify_loudness_callback), + NULL); g_signal_connect(shairportSyncSkeleton, "notify::loudness-threshold", G_CALLBACK(notify_loudness_threshold_callback), NULL); g_signal_connect(shairportSyncSkeleton, "notify::drift-tolerance", @@ -879,7 +881,6 @@ static void on_dbus_name_acquired(GDBusConnection *connection, const gchar *name g_signal_connect(shairportSyncRemoteControlSkeleton, "handle-set-airplay-volume", G_CALLBACK(on_handle_set_airplay_volume), NULL); - g_signal_connect(shairportSyncAdvancedRemoteControlSkeleton, "handle-set-volume", G_CALLBACK(on_handle_set_volume), NULL); @@ -975,9 +976,11 @@ static void on_dbus_name_acquired(GDBusConnection *connection, const gchar *name shairport_sync_set_convolution(SHAIRPORT_SYNC(shairportSyncSkeleton), TRUE); } if (config.convolution_ir_file) - shairport_sync_set_convolution_impulse_response_file(SHAIRPORT_SYNC(shairportSyncSkeleton), config.convolution_ir_file); + shairport_sync_set_convolution_impulse_response_file(SHAIRPORT_SYNC(shairportSyncSkeleton), + config.convolution_ir_file); // else -// shairport_sync_set_convolution_impulse_response_file(SHAIRPORT_SYNC(shairportSyncSkeleton), NULL); +// shairport_sync_set_convolution_impulse_response_file(SHAIRPORT_SYNC(shairportSyncSkeleton), +// NULL); #endif shairport_sync_set_version(SHAIRPORT_SYNC(shairportSyncSkeleton), PACKAGE_VERSION); diff --git a/mdns_avahi.c b/mdns_avahi.c index 575eba08..65bd6477 100644 --- a/mdns_avahi.c +++ b/mdns_avahi.c @@ -97,7 +97,7 @@ static void resolve_callback(AvahiServiceResolver *r, AVAHI_GCC_UNUSED AvahiIfIn if (dacpid) { dacpid += strlen("iTunes_Ctrl_"); while (*dacpid == '0') - dacpid++; // remove any leading zeroes + dacpid++; // remove any leading zeroes if (strcmp(dacpid, dbs->dacp_id) == 0) { debug(3, "resolve_callback: client dacp_id \"%s\" dacp port: %u.", dbs->dacp_id, port); #ifdef CONFIG_DACP_CLIENT diff --git a/mdns_dns_sd.c b/mdns_dns_sd.c index 95bae443..cf178f1d 100644 --- a/mdns_dns_sd.c +++ b/mdns_dns_sd.c @@ -24,8 +24,8 @@ * OTHER DEALINGS IN THE SOFTWARE. */ -#include "mdns.h" #include "common.h" +#include "mdns.h" #include #include #include diff --git a/mdns_external.c b/mdns_external.c index 2ae23c61..974aa85b 100644 --- a/mdns_external.c +++ b/mdns_external.c @@ -25,8 +25,8 @@ * OTHER DEALINGS IN THE SOFTWARE. */ -#include "mdns.h" #include "common.h" +#include "mdns.h" #include #include #include diff --git a/metadata_hub.c b/metadata_hub.c index f079ea2a..b2f370ac 100644 --- a/metadata_hub.c +++ b/metadata_hub.c @@ -35,10 +35,10 @@ #include #include #include +#include #include #include #include -#include #include "config.h" @@ -173,39 +173,39 @@ char *metadata_write_image_file(const char *buf, int len) { // it will return a path to the image file allocated with malloc. // free it if you don't need it. - char *path = NULL; // this will be what is returned - if (strcmp(config.cover_art_cache_dir,"") != 0) { // an empty string means do not write the file + char *path = NULL; // this will be what is returned + if (strcmp(config.cover_art_cache_dir, "") != 0) { // an empty string means do not write the file uint8_t img_md5[16]; // uint8_t ap_md5[16]; - #ifdef CONFIG_OPENSSL +#ifdef CONFIG_OPENSSL MD5_CTX ctx; MD5_Init(&ctx); MD5_Update(&ctx, buf, len); MD5_Final(img_md5, &ctx); - #endif +#endif - #ifdef CONFIG_MBEDTLS - #if MBEDTLS_VERSION_MINOR >= 7 +#ifdef CONFIG_MBEDTLS +#if MBEDTLS_VERSION_MINOR >= 7 mbedtls_md5_context tctx; mbedtls_md5_starts_ret(&tctx); mbedtls_md5_update_ret(&tctx, (const unsigned char *)buf, len); mbedtls_md5_finish_ret(&tctx, img_md5); - #else +#else mbedtls_md5_context tctx; mbedtls_md5_starts(&tctx); mbedtls_md5_update(&tctx, (const unsigned char *)buf, len); mbedtls_md5_finish(&tctx, img_md5); - #endif - #endif +#endif +#endif - #ifdef CONFIG_POLARSSL +#ifdef CONFIG_POLARSSL md5_context tctx; md5_starts(&tctx); md5_update(&tctx, (const unsigned char *)buf, len); md5_finish(&tctx, img_md5); - #endif +#endif char img_md5_str[33]; memset(img_md5_str, 0, sizeof(img_md5_str)); @@ -232,8 +232,8 @@ char *metadata_write_image_file(const char *buf, int len) { // if it exists, we're done char *prefix = "cover-"; - size_t pl = strlen(config.cover_art_cache_dir) + 1 + strlen(prefix) + strlen(img_md5_str) + 1 + - strlen(ext); + size_t pl = strlen(config.cover_art_cache_dir) + 1 + strlen(prefix) + strlen(img_md5_str) + + 1 + strlen(ext); path = malloc(pl + 1); snprintf(path, pl + 1, "%s/%s%s.%s", config.cover_art_cache_dir, prefix, img_md5_str, ext); @@ -309,30 +309,28 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin if (type == 'core') { switch (code) { case 'mper': { - // get the 64-bit number as a uint64_t by reading two uint32_t s and combining them - uint64_t vl = ntohl(*(uint32_t*)data); // get the high order 32 bits - vl = vl << 32; // shift them into the correct location - uint64_t ul = ntohl(*(uint32_t*)(data+sizeof(uint32_t))); // and the low order 32 bits - vl = vl + ul; - debug(2, "MH Item ID seen: \"%" PRIx64 "\" of length %u.", vl, length); - if (vl != metadata_store.item_id) { - metadata_store.item_id = vl; - metadata_store.item_id_changed = 1; - metadata_store.item_id_received = 1; - debug(2, "MH Item ID set to: \"%" PRIx64 "\"", metadata_store.item_id); - } - } - break; - case 'astm': { - uint32_t ui = ntohl(*(uint32_t *)data); - debug(2, "MH Song Time seen: \"%u\" of length %u.", ui, length); - if (ui != metadata_store.songtime_in_milliseconds) { - metadata_store.songtime_in_milliseconds = ui; - metadata_store.songtime_in_milliseconds_changed = 1; - debug(2, "MH Song Time set to: \"%u\"", metadata_store.songtime_in_milliseconds); - } + // get the 64-bit number as a uint64_t by reading two uint32_t s and combining them + uint64_t vl = ntohl(*(uint32_t *)data); // get the high order 32 bits + vl = vl << 32; // shift them into the correct location + uint64_t ul = ntohl(*(uint32_t *)(data + sizeof(uint32_t))); // and the low order 32 bits + vl = vl + ul; + debug(2, "MH Item ID seen: \"%" PRIx64 "\" of length %u.", vl, length); + if (vl != metadata_store.item_id) { + metadata_store.item_id = vl; + metadata_store.item_id_changed = 1; + metadata_store.item_id_received = 1; + debug(2, "MH Item ID set to: \"%" PRIx64 "\"", metadata_store.item_id); } - break; + } break; + case 'astm': { + uint32_t ui = ntohl(*(uint32_t *)data); + debug(2, "MH Song Time seen: \"%u\" of length %u.", ui, length); + if (ui != metadata_store.songtime_in_milliseconds) { + metadata_store.songtime_in_milliseconds = ui; + metadata_store.songtime_in_milliseconds_changed = 1; + debug(2, "MH Song Time set to: \"%u\"", metadata_store.songtime_in_milliseconds); + } + } break; case 'asal': cs = strndup(data, length); if (string_update(&metadata_store.album_name, &metadata_store.album_name_changed, cs)) { @@ -467,7 +465,8 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin metadata_hub_modify_prolog(); debug(2, "MH Picture received, length %u bytes.", length); char uri[2048]; - if ((length > 16) && (strcmp(config.cover_art_cache_dir,"")!=0)) { // if it's okay to write the file + if ((length > 16) && + (strcmp(config.cover_art_cache_dir, "") != 0)) { // if it's okay to write the file char *pathname = metadata_write_image_file(data, length); snprintf(uri, sizeof(uri), "file://%s", pathname); free(pathname); @@ -526,14 +525,16 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin break; case 'pbeg': metadata_hub_modify_prolog(); - changed = ((metadata_store.player_state != PS_PLAYING) || (metadata_store.player_thread_active == 0)); + changed = ((metadata_store.player_state != PS_PLAYING) || + (metadata_store.player_thread_active == 0)); metadata_store.player_state = PS_PLAYING; metadata_store.player_thread_active = 1; metadata_hub_modify_epilog(changed); break; case 'pend': metadata_hub_modify_prolog(); - changed = ((metadata_store.player_state != PS_STOPPED) || (metadata_store.player_thread_active == 1)); + changed = ((metadata_store.player_state != PS_STOPPED) || + (metadata_store.player_thread_active == 1)); metadata_store.player_state = PS_STOPPED; metadata_store.player_thread_active = 0; metadata_hub_modify_epilog(changed); @@ -554,10 +555,10 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin case 'pvol': { metadata_hub_modify_prolog(); // Note: it's assumed that the config.airplay volume has already been correctly set. - //int32_t actual_volume; - //int gv = dacp_get_volume(&actual_volume); - //metadata_hub_modify_prolog(); - //if ((gv == 200) && (metadata_store.speaker_volume != actual_volume)) { + // int32_t actual_volume; + // int gv = dacp_get_volume(&actual_volume); + // metadata_hub_modify_prolog(); + // if ((gv == 200) && (metadata_store.speaker_volume != actual_volume)) { // metadata_store.speaker_volume = actual_volume; // changed = 1; //} diff --git a/mpris-service.c b/mpris-service.c index 125398a1..8258baf6 100644 --- a/mpris-service.c +++ b/mpris-service.c @@ -23,9 +23,9 @@ * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR * OTHER DEALINGS IN THE SOFTWARE. */ +#include #include #include -#include #include "config.h" @@ -48,12 +48,12 @@ double airplay_volume_to_mpris_volume(double sp) { sp = -30.0; if (sp > 0.0) sp = 0.0; - sp = (sp/30.0)+1; + sp = (sp / 30.0) + 1; return sp; } double mpris_volume_to_airplay_volume(double sp) { - sp = (sp-1.0)*30.0; + sp = (sp - 1.0) * 30.0; if (sp < -30.0) sp = -30.0; if (sp > 0.0) @@ -64,7 +64,8 @@ double mpris_volume_to_airplay_volume(double sp) { void mpris_metadata_watcher(struct metadata_bundle *argc, __attribute__((unused)) void *userdata) { // debug(1, "MPRIS metadata watcher called"); char response[100]; - media_player2_player_set_volume(mprisPlayerPlayerSkeleton, airplay_volume_to_mpris_volume(argc->airplay_volume)); + media_player2_player_set_volume(mprisPlayerPlayerSkeleton, + airplay_volume_to_mpris_volume(argc->airplay_volume)); switch (argc->repeat_status) { case RS_NOT_AVAILABLE: strcpy(response, "Not Available"); @@ -175,7 +176,8 @@ void mpris_metadata_watcher(struct metadata_bundle *argc, __attribute__((unused) // Add in the Track ID based on the 'mper' metadata if it is non-zero if (argc->item_id != 0) { char trackidstring[128]; - snprintf(trackidstring, sizeof(trackidstring), "/org/gnome/ShairportSync/%" PRIX64 "", argc->item_id); + snprintf(trackidstring, sizeof(trackidstring), "/org/gnome/ShairportSync/%" PRIX64 "", + argc->item_id); GVariant *trackid = g_variant_new("o", trackidstring); g_variant_builder_add(dict_builder, "{sv}", "mpris:trackid", trackid); } @@ -276,8 +278,8 @@ static gboolean on_handle_play(MediaPlayer2Player *skeleton, GDBusMethodInvocati } static gboolean on_handle_set_volume(MediaPlayer2Player *skeleton, - GDBusMethodInvocation *invocation, const gdouble volume, - __attribute__((unused)) gpointer user_data) { + GDBusMethodInvocation *invocation, const gdouble volume, + __attribute__((unused)) gpointer user_data) { double ap_volume = mpris_volume_to_airplay_volume(volume); debug(2, "Set mpris volume to %.6f, i.e. airplay volume to %.6f.", volume, ap_volume); char command[256] = ""; @@ -334,7 +336,6 @@ static void on_mpris_name_acquired(GDBusConnection *connection, const gchar *nam g_signal_connect(mprisPlayerPlayerSkeleton, "handle-set-volume", G_CALLBACK(on_handle_set_volume), NULL); - add_metadata_watcher(mpris_metadata_watcher, NULL); debug(1, "MPRIS service started at \"%s\" on the %s bus.", name, diff --git a/player.c b/player.c index 2d8f635e..344662c3 100644 --- a/player.c +++ b/player.c @@ -91,6 +91,14 @@ #include "activity_monitor.h" +// make the first audio packet deliberately early to bias the sync error of +// the very first packet, making the error more likely to be too early +// rather than too late. It it's too early, +// a delay exactly compensating for it can be sent just before the +// first packet. This should exactly compensate for the error. + +int64_t first_frame_early_bias = 8; + // default buffer size // needs to be a power of 2 because of the way BUFIDX(seqno) works //#define BUFFER_FRAMES 512 @@ -123,8 +131,11 @@ static void ab_resync(rtsp_conn_info *conn) { for (i = 0; i < BUFFER_FRAMES; i++) { conn->audio_buffer[i].ready = 0; conn->audio_buffer[i].resend_request_number = 0; - conn->audio_buffer[i].resend_time = 0; // this is either zero or the time the last resend was requested. - conn->audio_buffer[i].initialisation_time = 0; // this is either the time the packet was received or the time it was noticed the packet was missing. + conn->audio_buffer[i].resend_time = + 0; // this is either zero or the time the last resend was requested. + conn->audio_buffer[i].initialisation_time = + 0; // this is either the time the packet was received or the time it was noticed the packet + // was missing. conn->audio_buffer[i].sequence_number = 0; } conn->ab_synced = 0; @@ -323,16 +334,18 @@ int audio_packet_decode(short *dest, int *destlen, uint8_t *buf, int len, rtsp_c } if (outsize > maximum_possible_outsize) { - debug(2, "Output from alac_decode larger (%d bytes, not frames) than expected (%d bytes) -- " - "truncated, but buffer overflow possible! Encrypted = %d.", + debug(2, + "Output from alac_decode larger (%d bytes, not frames) than expected (%d bytes) -- " + "truncated, but buffer overflow possible! Encrypted = %d.", outsize, maximum_possible_outsize, conn->stream.encrypted); reply = -1; // output packet is the wrong size } *destlen = outsize / conn->input_bytes_per_frame; if ((outsize % conn->input_bytes_per_frame) != 0) - debug(1, "Number of audio frames (%d) does not correspond exactly to the number of bytes (%d) " - "and the audio frame size (%d).", + debug(1, + "Number of audio frames (%d) does not correspond exactly to the number of bytes (%d) " + "and the audio frame size (%d).", *destlen, outsize, conn->input_bytes_per_frame); return reply; } @@ -486,9 +499,10 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in if ((conn->flush_rtp_timestamp != 0) && (actual_timestamp != conn->flush_rtp_timestamp) && (modulo_32_offset(actual_timestamp, conn->flush_rtp_timestamp) < conn->input_rate * 10)) { // if it's less than 10 seconds - debug(3, "Dropping flushed packet in player_put_packet, seqno %u, timestamp %" PRIu32 - ", flushing to " - "timestamp: %" PRIu32 ".", + debug(3, + "Dropping flushed packet in player_put_packet, seqno %u, timestamp %" PRIu32 + ", flushing to " + "timestamp: %" PRIu32 ".", seqno, actual_timestamp, conn->flush_rtp_timestamp); conn->initial_reference_time = 0; conn->initial_reference_timestamp = 0; @@ -523,7 +537,8 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in abuf = conn->audio_buffer + BUFIDX(seq_sum(conn->ab_write, i)); abuf->ready = 0; // to be sure, to be sure abuf->resend_request_number = 0; - abuf->initialisation_time = time_now; // this represents when the packet was noticed to be missing + abuf->initialisation_time = + time_now; // this represents when the packet was noticed to be missing abuf->status = 1 << 0; // signifying missing abuf->resend_time = 0; abuf->given_timestamp = 0; @@ -534,7 +549,8 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in // rtp_request_resend(ab_write, gap); // resend_requests++; conn->ab_write = SUCCESSOR(seqno); - } else if (seq_order(conn->ab_read, seqno, conn->ab_read)) { // older than expected but not too late + } else if (seq_order(conn->ab_read, seqno, + conn->ab_read)) { // older than expected but not too late conn->late_packets++; abuf = conn->audio_buffer + BUFIDX(seqno); } else { // too late. @@ -567,15 +583,21 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in // resend checks { - uint64_t minimum_wait_time = (uint64_t)(config.resend_control_first_check_time * (uint64_t)1000000000); - uint64_t resend_repeat_interval = (uint64_t)(config.resend_control_check_interval_time * (uint64_t)1000000000); - uint64_t minimum_remaining_time = (uint64_t)((config.resend_control_last_check_time + config.audio_backend_buffer_desired_length)* (uint64_t)1000000000); - uint64_t latency_time = (uint64_t)(conn->latency * (uint64_t)1000000000); - latency_time = latency_time / (uint64_t)conn->input_rate; + uint64_t minimum_wait_time = + (uint64_t)(config.resend_control_first_check_time * (uint64_t)1000000000); + uint64_t resend_repeat_interval = + (uint64_t)(config.resend_control_check_interval_time * (uint64_t)1000000000); + uint64_t minimum_remaining_time = (uint64_t)( + (config.resend_control_last_check_time + config.audio_backend_buffer_desired_length) * + (uint64_t)1000000000); + uint64_t latency_time = (uint64_t)(conn->latency * (uint64_t)1000000000); + latency_time = latency_time / (uint64_t)conn->input_rate; - int x; // this is the first frame to be checked + int x; // this is the first frame to be checked // if we detected a first empty frame before and if it's still in the buffer! - if ((first_possibly_missing_frame >= 0) && (position_in_modulo_uint16_t_buffer(first_possibly_missing_frame, conn->ab_read, conn->ab_write, NULL))) { + if ((first_possibly_missing_frame >= 0) && + (position_in_modulo_uint16_t_buffer(first_possibly_missing_frame, conn->ab_read, + conn->ab_write, NULL))) { x = first_possibly_missing_frame; } else { x = conn->ab_read; @@ -583,8 +605,8 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in first_possibly_missing_frame = -1; // has not been set - int missing_frame_run_count = 0; - int start_of_missing_frame_run = -1; + int missing_frame_run_count = 0; + int start_of_missing_frame_run = -1; int number_of_missing_frames = 0; while (x != conn->ab_write) { abuf_t *check_buf = conn->audio_buffer + BUFIDX(x); @@ -592,46 +614,57 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in if (first_possibly_missing_frame < 0) first_possibly_missing_frame = x; number_of_missing_frames++; - // debug(1, "frame %u's initialisation_time is 0x%" PRIx64 ", latency_time is 0x%" PRIx64 ", time_now is 0x%" PRIx64 ", minimum_remaining_time is 0x%" PRIx64 ".", x, check_buf->initialisation_time, latency_time, time_now, minimum_remaining_time); - int too_late = ((check_buf->initialisation_time < (time_now - latency_time)) || ((check_buf->initialisation_time - (time_now - latency_time)) < minimum_remaining_time)); + // debug(1, "frame %u's initialisation_time is 0x%" PRIx64 ", latency_time is 0x%" + // PRIx64 ", time_now is 0x%" PRIx64 ", minimum_remaining_time is 0x%" PRIx64 ".", x, + // check_buf->initialisation_time, latency_time, time_now, minimum_remaining_time); + int too_late = ((check_buf->initialisation_time < (time_now - latency_time)) || + ((check_buf->initialisation_time - (time_now - latency_time)) < + minimum_remaining_time)); int too_early = ((time_now - check_buf->initialisation_time) < minimum_wait_time); - int too_soon_after_last_request = ((check_buf->resend_time != 0) && ((time_now - check_buf->resend_time) < resend_repeat_interval)); // time_now can never be less than the time_tag + int too_soon_after_last_request = + ((check_buf->resend_time != 0) && + ((time_now - check_buf->resend_time) < + resend_repeat_interval)); // time_now can never be less than the time_tag if (too_late) - check_buf->status |= 1<<2; // too late + check_buf->status |= 1 << 2; // too late else - check_buf->status &= 0xFF-(1<<2); // not too late + check_buf->status &= 0xFF - (1 << 2); // not too late if (too_early) - check_buf->status |= 1<<3; // too early + check_buf->status |= 1 << 3; // too early else - check_buf->status &= 0xFF-(1<<3); // not too early + check_buf->status &= 0xFF - (1 << 3); // not too early if (too_soon_after_last_request) - check_buf->status |= 1<<4; // too soon after last request + check_buf->status |= 1 << 4; // too soon after last request else - check_buf->status &= 0xFF-(1<<4); // not too soon after last request + check_buf->status &= 0xFF - (1 << 4); // not too soon after last request - if ((!too_soon_after_last_request) && (!too_late) && (!too_early)){ + if ((!too_soon_after_last_request) && (!too_late) && (!too_early)) { if (start_of_missing_frame_run == -1) { start_of_missing_frame_run = x; missing_frame_run_count = 1; } else { missing_frame_run_count++; } - check_buf->resend_time = time_now; // setting the time to now because we are definitely going to take action + check_buf->resend_time = time_now; // setting the time to now because we are + // definitely going to take action check_buf->resend_request_number++; - debug(3,"Frame %d is missing with ab_read of %u and ab_write of %u.", x, conn->ab_read, conn->ab_write); + debug(3, "Frame %d is missing with ab_read of %u and ab_write of %u.", x, + conn->ab_read, conn->ab_write); } // if (too_late) { // debug(1,"too late to get missing frame %u.", x); // } } - //if (number_of_missing_frames != 0) - // debug(1,"check with x = %u, ab_read = %u, ab_write = %u, first_possibly_missing_frame = %d.", x, conn->ab_read, conn->ab_write, first_possibly_missing_frame); + // if (number_of_missing_frames != 0) + // debug(1,"check with x = %u, ab_read = %u, ab_write = %u, first_possibly_missing_frame + // = %d.", x, conn->ab_read, conn->ab_write, first_possibly_missing_frame); x = (x + 1) & 0xffff; if (((check_buf->ready) || (x == conn->ab_write)) && (missing_frame_run_count > 0)) { - // send a resend request - if (missing_frame_run_count > 1) - debug(2,"request resend of %d packets starting at seqno %u.", missing_frame_run_count, start_of_missing_frame_run); + // send a resend request + if (missing_frame_run_count > 1) + debug(2, "request resend of %d packets starting at seqno %u.", + missing_frame_run_count, start_of_missing_frame_run); if (config.disable_resend_requests == 0) { debug_mutex_unlock(&conn->ab_mutex, 3); rtp_request_resend(start_of_missing_frame_run, missing_frame_run_count, conn); @@ -660,26 +693,29 @@ int32_t rand_in_range(int32_t exclusive_range_limit) { return sp >> 32; } -int get_and_check_effective_latency(rtsp_conn_info *conn, uint32_t *effective_latency, double offset_time) { -// check that the overall effective latency remains positive and is not greater than the capacity of the buffers -// return 0 if okay, -1 if the latency would be negative, +1 if it would be too large +int get_and_check_effective_latency(rtsp_conn_info *conn, uint32_t *effective_latency, + double offset_time) { + // check that the overall effective latency remains positive and is not greater than the capacity + // of the buffers return 0 if okay, -1 if the latency would be negative, +1 if it would be too + // large - int result = 0; - if (offset_time >= 0.0) { - uint32_t latency_addition = (uint32_t)(offset_time * conn->input_rate); - // keep about one second of buffers back - if ((*effective_latency + latency_addition) <= (conn->max_frames_per_packet * (BUFFER_FRAMES - config.minimum_free_buffer_headroom))) - *effective_latency += latency_addition; - else - result = 1; - } else { - uint32_t latency_reduction = (uint32_t)((-offset_time) * conn->input_rate); - if (latency_reduction <= *effective_latency) - *effective_latency -= latency_reduction; - else - result = -1; - } - return result; + int result = 0; + if (offset_time >= 0.0) { + uint32_t latency_addition = (uint32_t)(offset_time * conn->input_rate); + // keep about one second of buffers back + if ((*effective_latency + latency_addition) <= + (conn->max_frames_per_packet * (BUFFER_FRAMES - config.minimum_free_buffer_headroom))) + *effective_latency += latency_addition; + else + result = 1; + } else { + uint32_t latency_reduction = (uint32_t)((-offset_time) * conn->input_rate); + if (latency_reduction <= *effective_latency) + *effective_latency -= latency_reduction; + else + result = -1; + } + return result; } static inline void process_sample(int32_t sample, char **outp, sps_format_t format, int volume, @@ -920,7 +956,8 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { int wait; long dac_delay = 0; // long because alsa returns a long - int have_sent_prefiller_silence = 0; // set to true when we have sent at least one silent frame to the DAC + int have_sent_prefiller_silence = + 0; // set to true when we have sent at least one silent frame to the DAC pthread_cleanup_push(buffer_get_frame_cleanup_handler, (void *)conn); // undo what's been done so far @@ -958,6 +995,8 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { conn->first_packet_time_to_play = 0; conn->time_since_play_started = 0; conn->flush_requested = 0; + have_sent_prefiller_silence = 0; + dac_delay = 0; } debug_mutex_unlock(&conn->flush_mutex, 0); @@ -990,9 +1029,10 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { (curframe->given_timestamp != conn->flush_rtp_timestamp) && (modulo_32_offset(curframe->given_timestamp, conn->flush_rtp_timestamp) < conn->input_rate * 10)) { // if it's less than ten seconds - debug(3, "Dropping flushed packet in buffer_get_frame, seqno %u, timestamp %" PRIu32 - ", flushing to " - "timestamp: %" PRIu32 ".", + debug(3, + "Dropping flushed packet in buffer_get_frame, seqno %u, timestamp %" PRIu32 + ", flushing to " + "timestamp: %" PRIu32 ".", curframe->sequence_number, curframe->given_timestamp, conn->flush_rtp_timestamp); curframe->ready = 0; curframe->resend_request_number = 0; @@ -1013,29 +1053,14 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { notified_buffer_empty = 0; // at least one buffer now -- diagnostic only. if (conn->ab_buffering) { // if we are getting packets but not yet forwarding them to the // player -// int have_sent_prefiller_silence = 1; // set to true when we have sent at least one silent frame to the DAC - /* - int64_t reference_timestamp; - uint64_t reference_timestamp_time, remote_reference_timestamp_time; - get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time, - &remote_reference_timestamp_time, conn); - reference_timestamp *= conn->output_sample_ratio; - */ - if (conn->first_packet_timestamp == 0) { // if this is the very first packet - // debug(1,"First frame seen, time %u, with %d - // frames...",curframe->timestamp,seq_diff(ab_read, ab_write)); - + if (conn->first_packet_timestamp == 0) { // if this is the very first packet if (have_timestamp_timing_information(conn)) { // if we have a reference time // debug(1,"First frame seen with timestamp..."); conn->first_packet_timestamp = curframe->given_timestamp; // we will keep buffering until we are // supposed to start playing this - // have_sent_prefiller_silence = 0; - -// debug(1, "First packet timestamp is %" PRId64 ".", conn->first_packet_timestamp); - -// say we have started playing here #ifdef CONFIG_METADATA + // say we have started receiving frames here debug(2, "pffr"); send_ssnc_metadata( 'pffr', NULL, 0, @@ -1046,9 +1071,8 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { // We will send packets of silence from now until that time and then we will send the // first packet, which will be followed by the subsequent packets. - // we will get a fix every second or so, which will be stored as a pair consisting of - // the time when the packet with a particular timestamp should be played, neglecting - // latencies, etc. + // every second or so, we get a reference on when a particular packet should be + // played. // It probably won't be the timestamp of our first packet, however, so we might have // to do some calculations. @@ -1061,29 +1085,30 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { // that the audio back end has a latency of 100 ms, we would // ask for the first packet to be emitted 100 ms earlier than it should, i.e. -4410 // frames, so that when it got through the audio back end, - // if would be in sync. To do this, we would give it a latency offset of -100 ms, i.e. - // -4410 frames. - // debug(1, "Output sample ratio is %d", conn->output_sample_ratio); + // if would be in sync. To do this, we would give it a latency offset of -0.1 sec, + // i.e. -4410 frames. + + uint64_t should_be_time; + uint32_t effective_latency = conn->latency; + + get_and_check_effective_latency(conn, &effective_latency, + config.audio_backend_latency_offset); + // we are ignoring the returned status because it will be captured on subsequent + // frames, below. // what we are asking for here is "what is the local time at which time the calculated // frame should be played" - uint64_t should_be_time; - uint32_t effective_latency = conn->latency; - - get_and_check_effective_latency(conn, &effective_latency,config.audio_backend_latency_offset); - // we are ignoring the returned status because it will be captured on subsequent frames, below. - - frame_to_local_time(conn->first_packet_timestamp + effective_latency, // this will go modulo 2^32 - &should_be_time, - conn); + frame_to_local_time(conn->first_packet_timestamp + + effective_latency, // this will go modulo 2^32 + &should_be_time, conn); conn->first_packet_time_to_play = should_be_time; if (local_time_now > conn->first_packet_time_to_play) { uint64_t lateness = local_time_now - conn->first_packet_time_to_play; - debug(3, "First packet is %" PRIu64 " nanoseconds late! Flushing 0.5 seconds", + debug(2, "First packet is %" PRIu64 " nanoseconds late! Flushing 0.5 seconds", lateness); do_flush(conn->first_packet_timestamp + 5 * 4410, conn); } @@ -1093,192 +1118,120 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { if (conn->first_packet_time_to_play != 0) { // recalculate conn->first_packet_time_to_play -- the latency might change - uint64_t should_be_time; - uint32_t effective_latency = conn->latency; + uint64_t should_be_time; + uint32_t effective_latency = conn->latency; - switch (get_and_check_effective_latency(conn, &effective_latency,config.audio_backend_latency_offset)) { - case -1: - if (conn->unachievable_audio_backend_latency_offset_notified == 0) { - warn("Negative latency! A latency of %d frames requested by the player, when combined with an audio_backend_latency_offset of %f seconds, would make the overall latency negative. The audio_backend_latency_offset setting is ignored.", conn->latency, config.audio_backend_latency_offset); - config.audio_backend_latency_offset = 0; //set it to zero - conn->unachievable_audio_backend_latency_offset_notified = 1; - }; -// effective_latency = 0 ; - break; - case 1: - if (conn->unachievable_audio_backend_latency_offset_notified == 0) { - warn("An audio_backend_latency_offset of %f seconds may exceed the frame buffering capacity -- the setting is ignored.", config.audio_backend_latency_offset); - config.audio_backend_latency_offset = 0; // set it to zero; - conn->unachievable_audio_backend_latency_offset_notified = 1; - }; - break; - default: - break; - } - - frame_to_local_time(conn->first_packet_timestamp + effective_latency, // this will go modulo 2^32 - &should_be_time, - conn); - - conn->first_packet_time_to_play = should_be_time; - - // we want the frames of silence sent at the start to be fairly large in case the output - // device's minimum buffer size is large. But they can't be greater than the silent - // lead_in time - // which is either the agreed latency or the silent lead-in time specified by the - // setting - // In fact, if should be some fraction of them, to allow for adjustment. - - int64_t adjustment_interval_frames = (int64_t)(config.lead_in_silence_adjustment_interval * conn->input_rate); - int64_t initial_silence_frames = (int64_t)(config.lead_in_silence_initial_period * conn->input_rate); - - int64_t max_dac_delay = effective_latency; - if (config.audio_backend_silent_lead_in_time_auto == 0) - max_dac_delay = - (int64_t)(config.audio_backend_silent_lead_in_time * conn->input_rate); - - // should check that the audio_backend_silent_lead_in_time is greater than the desired_buffer_size - - int64_t filler_size = adjustment_interval_frames; - - if (have_sent_prefiller_silence == 0) { - int64_t delay_before_adjustments = max_dac_delay - config.lead_in_silence_minimum_adjustments_to_make * adjustment_interval_frames; - if (delay_before_adjustments > initial_silence_frames) - filler_size = initial_silence_frames; - else - filler_size = max_dac_delay / (config.lead_in_silence_minimum_adjustments_to_make + 1); - // debug(1,"Initial filler size: %" PRId64 " frames.", filler_size); + switch (get_and_check_effective_latency(conn, &effective_latency, + config.audio_backend_latency_offset)) { + case -1: + if (conn->unachievable_audio_backend_latency_offset_notified == 0) { + warn("Negative latency! A latency of %d frames requested by the player, when " + "combined with an audio_backend_latency_offset of %f seconds, would make the " + "overall latency negative. The audio_backend_latency_offset setting is " + "ignored.", + conn->latency, config.audio_backend_latency_offset); + config.audio_backend_latency_offset = 0; // set it to zero + conn->unachievable_audio_backend_latency_offset_notified = 1; + }; + break; + case 1: + if (conn->unachievable_audio_backend_latency_offset_notified == 0) { + warn("An audio_backend_latency_offset of %f seconds may exceed the frame buffering " + "capacity -- the setting is ignored.", + config.audio_backend_latency_offset); + config.audio_backend_latency_offset = 0; // set it to zero; + conn->unachievable_audio_backend_latency_offset_notified = 1; + }; + break; + default: + break; } + frame_to_local_time(conn->first_packet_timestamp + + effective_latency, // this will go modulo 2^32 + &should_be_time, conn); + + conn->first_packet_time_to_play = should_be_time; + if (local_time_now > conn->first_packet_time_to_play) { uint64_t lateness = local_time_now - conn->first_packet_time_to_play; - debug(3, "Gone past starting time by %" PRIu64 " nanoseconds.", lateness); + debug(2, "Gone past starting time by %" PRIu64 " nanoseconds.", lateness); // have_sent_prefiller_silence = 1; conn->ab_buffering = 0; - - // we've gone past the time... - // debug(1,"Run past the exact start time by %llu frames, with time now of %llx, fpttp - // of %llx and dac_delay of %d and %d packets; - // flush.",(((tn-conn->first_packet_time_to_play)*config.output_rate)>>32)+dac_delay,tn,conn->first_packet_time_to_play,dac_delay,seq_diff(ab_read, - // ab_write)); - - /* - if (config.output->flush) - config.output->flush(); - ab_resync(conn); - conn->first_packet_timestamp = 0; - conn->first_packet_time_to_play = 0; - conn->time_since_play_started = 0; - */ } else { // do some calculations int64_t lead_time = conn->first_packet_time_to_play - local_time_now; - if ((config.audio_backend_silent_lead_in_time_auto == 1) || (lead_time <= (int64_t)(config.audio_backend_silent_lead_in_time * (int64_t)1000000000))) { - // debug(1,"Lead time: %" PRId64 ".", lead_time); - - // debug(1,"Checking"); - if (config.output->delay) { - // conn->first_packet_time_to_play is definitely later than local_time_now + if ((config.audio_backend_silent_lead_in_time_auto == 1) || + (lead_time <= + (int64_t)(config.audio_backend_silent_lead_in_time * (int64_t)1000000000))) { + debug(3, "Lead time: %" PRId64 ".", lead_time); + if (config.output->delay) { // if the output device has a delay function + debug(3, "Checking"); int resp = 0; dac_delay = 0; if (have_sent_prefiller_silence != 0) resp = config.output->delay(&dac_delay); if (resp == 0) { int64_t gross_frame_gap = - ((conn->first_packet_time_to_play - local_time_now) * config.output_rate) / 1000000000; + ((conn->first_packet_time_to_play - local_time_now) * config.output_rate) / + 1000000000; int64_t exact_frame_gap = gross_frame_gap - dac_delay; - // debug(1,"Exact and gross frame gaps are %" PRId64 " and %" PRId64 " frames, - // and the dac delay is %ld.", exact_frame_gap, gross_frame_gap, dac_delay); - if (exact_frame_gap < 0) { - // we've gone past the time... - // debug(1,"Run past time."); + int64_t frames_needed_to_maintain_desired_buffer = + (int64_t)(config.audio_backend_buffer_desired_length * config.output_rate) - + dac_delay; + // below, remember that exact_frame_gap and + // frames_needed_to_maintain_desired_buffer could both be negative + int64_t fs = frames_needed_to_maintain_desired_buffer; - // this might happen if a big clock adjustment was made at just the wrong - // time. + // if there isn't enough time to have the desired buffer size + if (exact_frame_gap <= frames_needed_to_maintain_desired_buffer) { + fs = conn->max_frames_per_packet * 2; + } - debug(1, "Run a bit past the exact start time by %" PRId64 - " frames with a DAC delay of %ld frames.", - -exact_frame_gap, dac_delay); - if (config.output->flush) - config.output->flush(); - ab_resync(conn); - conn->first_packet_timestamp = 0; - conn->first_packet_time_to_play = 0; - } else { - int64_t fs = filler_size; - if (fs > (max_dac_delay - dac_delay)) - fs = max_dac_delay - dac_delay; - if (fs < 0) { - // this could happen if the dac delay mysteriously grows between samples, - // which could happen in a transition between having no interpolation and - // having interpolated buffer numbers. - - // this will happen benignly if standby is being prevented, because a - // thread in the alsa back end will be stuffing frames of silence in there - // to keep it busy - - debug(3, - "frame size (fs) < 0 with max_dac_delay of %lld and dac_delay of %ld", - max_dac_delay, dac_delay); - fs = 0; + // if we are very close to the end of buffering, i.e. within to frame-lengths, + // add the remaining silence needed and end buffering + if (exact_frame_gap <= conn->max_frames_per_packet * 2) { + fs = exact_frame_gap; + if (fs > first_frame_early_bias) + fs = fs - first_frame_early_bias; // deliberately make the first packet a tiny bit early so that the player may compensate for it at the last minute + conn->ab_buffering = 0; + } + void *silence; + if (fs > 0) { + silence = malloc(conn->output_bytes_per_frame * fs); + if (silence == NULL) + debug(1, "Failed to allocate %d byte silence buffer.", fs); + else { + // generate frames of silence with dither if necessary + conn->previous_random_number = + generate_zero_frames(silence, fs, config.output_format, + conn->enable_dither, conn->previous_random_number); + config.output->play(silence, fs); + debug(3, "Sent %" PRId64 " frames of silence", fs); + free(silence); + have_sent_prefiller_silence = 1; } - if ((exact_frame_gap <= fs) || - (exact_frame_gap <= conn->max_frames_per_packet * 2)) { - fs = exact_frame_gap; - // debug(1,"Exact frame gap is %llu; play %d frames of silence. Dac_delay is - // %d, - // with %d packets, ab_read is %04x, ab_write is - // %04x.",exact_frame_gap,fs,dac_delay,seq_diff(ab_read, - // ab_write),ab_read,ab_write); - conn->ab_buffering = 0; - } - void *silence; - // if (fs==0) - // debug(2,"Zero length silence buffer needed with gross_frame_gap of %lld - // and - // dac_delay of %lld.",gross_frame_gap,dac_delay); - // the fs (number of frames of silence to play) can be zero in the DAC doesn't - // start - // outputting frames for a while -- it could get loaded up but not start - // responding - // for many milliseconds. - if (fs > 0) { - silence = malloc(conn->output_bytes_per_frame * fs); - if (silence == NULL) - debug(1, "Failed to allocate %d byte silence buffer.", fs); - else { - - conn->previous_random_number = generate_zero_frames( - silence, fs, config.output_format, conn->enable_dither, - conn->previous_random_number); - - // debug(1,"Frames to start: %llu, DAC delay %d, buffer: %d - // packets.",exact_frame_gap,dac_delay,seq_diff(conn->ab_read, - // conn->ab_write, conn->ab_read)); - config.output->play(silence, fs); - // debug(1,"Sent %" PRId64 " frames of silence",fs); - free(silence); - } - } - have_sent_prefiller_silence = - 1; // even if we haven't sent silence because it's zero frames long... } } else { - if ((resp == sps_extra_code_output_stalled) && - (conn->unfixable_error_reported == 0)) { - conn->unfixable_error_reported = 1; - if (config.cmd_unfixable) { - command_execute(config.cmd_unfixable, "output_device_stalled", 1); - } else { - warn( - "an unrecoverable error, \"output_device_stalled\", has been detected.", - conn->connection_number); + + if (resp == sps_extra_code_output_stalled) { + if (conn->unfixable_error_reported == 0) { + conn->unfixable_error_reported = 1; + if (config.cmd_unfixable) { + command_execute(config.cmd_unfixable, "output_device_stalled", 1); + } else { + warn("an unrecoverable error, \"output_device_stalled\", has been " + "detected.", + conn->connection_number); + } } + } else { + debug(2, "Unexpected response to getting dac delay: %d.", resp); } } } else { // no delay function on back end -- just send the prefiller silence - // debug(1,"Back end has no delay function."); + // debug(2,"Back end has no delay function."); // send the appropriate prefiller here... void *silence; @@ -1351,40 +1304,50 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { if (have_timestamp_timing_information(conn)) { // if we have a reference time uint64_t time_to_play; - uint32_t effective_latency = conn->latency; - - switch (get_and_check_effective_latency(conn, &effective_latency,config.audio_backend_latency_offset - config.audio_backend_buffer_desired_length)) { - case -1: - // this means that the latency is negative, i.e. the packet must be played before its time. This is - // clearly a mistake, and can arise if the combination of a large negative latency offset and the desired buffer length are greater than the - // latency chosen by the player. For example if the latency is 88200 frames (2 seconds), the audio_backend_latency_offset is -1.9 and the audio_backend_buffer_desired_length is 0.2 - // the effective latency is -0.1, so the frame must be played 0.1 seconds before it is time-tagged for, and in reality before it arrives at the player. - - // to deal with the problem, rather than block the packets, we'll just let 'em go... - - // - if (conn->unachievable_audio_backend_latency_offset_notified == 0) { - warn("Negative latency! A latency of %d frames requested by the player, when combined with an audio_backend_latency_offset of %f seconds an audio_backend_buffer_desired_length of %f seconds, would make the overall latency negative. No latency is used. Synchronisation may fail.", conn->latency, config.audio_backend_latency_offset, config.audio_backend_buffer_desired_length); - conn->unachievable_audio_backend_latency_offset_notified = 1; - }; - time_to_play = local_time_now; // pretend the frame should be played now... - break; - case 1: - if (conn->unachievable_audio_backend_latency_offset_notified == 0) { - warn("Latency too long! An audio_backend_latency_offset of %f seconds combined with an audio_backend_buffer_desired_length of %f seconds may exceed the frame buffering capacity.", config.audio_backend_latency_offset, config.audio_backend_buffer_desired_length); - conn->unachievable_audio_backend_latency_offset_notified = 1; - }; - time_to_play = local_time_now; // pretend the frame should be played now... - break; - default: - frame_to_local_time(curframe->given_timestamp + effective_latency, // this will go modulo 2^32 - &time_to_play, - conn); - break; - } + uint32_t effective_latency = conn->latency; + switch (get_and_check_effective_latency(conn, &effective_latency, + config.audio_backend_latency_offset - + config.audio_backend_buffer_desired_length)) { + case -1: + // this means that the latency is negative, i.e. the packet must be played before its + // time. This is clearly a mistake, and can arise if the combination of a large negative + // latency offset and the desired buffer length are greater than the latency chosen by the + // player. For example if the latency is 88200 frames (2 seconds), the + // audio_backend_latency_offset is -1.9 and the audio_backend_buffer_desired_length is 0.2 + // the effective latency is -0.1, so the frame must be played 0.1 seconds before it is + // time-tagged for, and in reality before it arrives at the player. + // to deal with the problem, rather than block the packets, we'll just let 'em go... + // + if (conn->unachievable_audio_backend_latency_offset_notified == 0) { + warn("Negative latency! A latency of %d frames requested by the player, when combined " + "with an audio_backend_latency_offset of %f seconds an " + "audio_backend_buffer_desired_length of %f seconds, would make the overall " + "latency negative. No latency is used. Synchronisation may fail.", + conn->latency, config.audio_backend_latency_offset, + config.audio_backend_buffer_desired_length); + conn->unachievable_audio_backend_latency_offset_notified = 1; + }; + time_to_play = local_time_now; // pretend the frame should be played now... + break; + case 1: + if (conn->unachievable_audio_backend_latency_offset_notified == 0) { + warn("Latency too long! An audio_backend_latency_offset of %f seconds combined with an " + "audio_backend_buffer_desired_length of %f seconds may exceed the frame " + "buffering capacity.", + config.audio_backend_latency_offset, config.audio_backend_buffer_desired_length); + conn->unachievable_audio_backend_latency_offset_notified = 1; + }; + time_to_play = local_time_now; // pretend the frame should be played now... + break; + default: + frame_to_local_time(curframe->given_timestamp + + effective_latency, // this will go modulo 2^32 + &time_to_play, conn); + break; + } if (local_time_now >= time_to_play) { do_wait = 0; @@ -1398,7 +1361,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { notified_buffer_empty = 1; // reset_input_flow_metrics(conn); // don't do a full flush parameters reset conn->initial_reference_time = 0; - conn->initial_reference_timestamp = 0; + conn->initial_reference_timestamp = 0; } do_wait = 1; } @@ -1406,9 +1369,9 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { if (wait) { uint64_t time_to_wait_for_wakeup_ns = - 1000000000 / conn->input_rate; // this is time period of one frame - time_to_wait_for_wakeup_ns *= 2 * 352; // two full 352-frame packets - time_to_wait_for_wakeup_ns /= 3; // two thirds of a packet time + 1000000000 / conn->input_rate; // this is time period of one frame + time_to_wait_for_wakeup_ns *= 2 * 352; // two full 352-frame packets + time_to_wait_for_wakeup_ns /= 3; // two thirds of a packet time #ifdef COMPILE_FOR_LINUX_AND_FREEBSD_AND_CYGWIN_AND_OPENBSD uint64_t time_of_wakeup_ns = local_time_now + time_to_wait_for_wakeup_ns; @@ -1574,8 +1537,7 @@ int stuff_buffer_soxr_32(int32_t *inptr, int32_t *scratchBuffer, int length, // mean and variance calculations from "online_variance" algorithm at // https://en.wikipedia.org/wiki/Algorithms_for_calculating_variance#Online_algorithm - double soxr_execution_time = - (get_absolute_time_in_ns() - soxr_start_time) * 0.000000001; + double soxr_execution_time = (get_absolute_time_in_ns() - soxr_start_time) * 0.000000001; // debug(1,"soxr_execution_time_us: %10.1f",soxr_execution_time_us); if (soxr_execution_time > longest_soxr_execution_time) longest_soxr_execution_time = soxr_execution_time; @@ -1628,9 +1590,10 @@ int stuff_buffer_soxr_32(int32_t *inptr, int32_t *scratchBuffer, int length, } if (packets_processed % 1250 == 0) { - debug(3, "soxr_oneshot execution time in nanoseconds: mean, standard deviation and max " - "for %" PRId32 " interpolations in the last " - "1250 packets. %10.6f, %10.6f, %10.6f.", + debug(3, + "soxr_oneshot execution time in nanoseconds: mean, standard deviation and max " + "for %" PRId32 " interpolations in the last " + "1250 packets. %10.6f, %10.6f, %10.6f.", stat_n, stat_mean, stat_n <= 1 ? 0.0 : sqrtf(stat_M2 / (stat_n - 1)), longest_soxr_execution_time); stat_n = 0; @@ -1826,6 +1789,7 @@ void *player_thread_func(void *arg) { stats_t statistics[trend_interval]; int number_of_statistics, oldest_statistic, newest_statistic; int at_least_one_frame_seen = 0; + int at_least_one_frame_seen_this_session = 0; int64_t tsum_of_sync_errors, tsum_of_corrections, tsum_of_insertions_and_deletions, tsum_of_drifts; int64_t previous_sync_error = 0, previous_correction = 0; @@ -1920,17 +1884,17 @@ void *player_thread_func(void *arg) { // if ((input_rate!=config.output_rate) || (input_bit_depth!=output_bit_depth)) { // debug(1,"Define tbuf of length // %d.",output_bytes_per_frame*(max_frames_per_packet*output_sample_ratio+max_frame_size_change)); - conn->tbuf = - malloc(sizeof(int32_t) * 2 * (conn->max_frames_per_packet * conn->output_sample_ratio + - conn->max_frame_size_change)); + conn->tbuf = malloc( + sizeof(int32_t) * 2 * + (conn->max_frames_per_packet * conn->output_sample_ratio + conn->max_frame_size_change)); if (conn->tbuf == NULL) die("Failed to allocate memory for the transition buffer."); // initialise this, because soxr stuffing might be chosen later - conn->sbuf = - malloc(sizeof(int32_t) * 2 * (conn->max_frames_per_packet * conn->output_sample_ratio + - conn->max_frame_size_change)); + conn->sbuf = malloc( + sizeof(int32_t) * 2 * + (conn->max_frames_per_packet * conn->output_sample_ratio + conn->max_frame_size_change)); if (conn->sbuf == NULL) die("Failed to allocate memory for the sbuf buffer."); @@ -2040,8 +2004,11 @@ void *player_thread_func(void *arg) { // debug(3, "Play frame %d.", play_number); conn->play_number_after_flush++; if (inframe->given_timestamp == 0) { - debug(1, "Player has supplied a silent frame, (possibly frame %u) for play number %d, status 0x%X after %u resend requests.", - SUCCESSOR(conn->last_seqno_read), play_number, inframe->status, inframe->resend_request_number); + debug(1, + "Player has supplied a silent frame, (possibly frame %u) for play number %d, " + "status 0x%X after %u resend requests.", + SUCCESSOR(conn->last_seqno_read), play_number, inframe->status, + inframe->resend_request_number); conn->last_seqno_read = (SUCCESSOR(conn->last_seqno_read) & 0xffff); // manage the packet out of sequence minder @@ -2162,8 +2129,6 @@ void *player_thread_func(void *arg) { // If it's late, we remove an audio frame from this frame to bring a subsequent frame // forward in time - at_least_one_frame_seen = 1; - // now, go back as far as the total latency less, say, 100 ms, and check the presence of // frames from then onwards @@ -2194,7 +2159,8 @@ void *player_thread_func(void *arg) { // should be negated. Conversion is // positive uint64_t to int64_t, thus // okay. - td_in_frames = (td * config.output_rate) / 1000000000 ; // use the absolute td value for the present. Types okay + td_in_frames = (td * config.output_rate) / + 1000000000; // use the absolute td value for the present. Types okay td_in_frames = -td_in_frames; td = -td; // should be okay, as the range of values should be very small w.r.t 64 bits } @@ -2213,8 +2179,9 @@ void *player_thread_func(void *arg) { SUCCESSOR(conn->last_seqno_read); // int32_t from seq_t, i.e. uint16_t, so okay. if (inframe->sequence_number != conn->last_seqno_read) { // seq_t, ei.e. uint16_t and int32_t, so okay - debug(2, "Player: packets out of sequence: expected: %u, got: %u, with ab_read: %u " - "and ab_write: %u.", + debug(2, + "Player: packets out of sequence: expected: %u, got: %u, with ab_read: %u " + "and ab_write: %u.", conn->last_seqno_read, inframe->sequence_number, conn->ab_read, conn->ab_write); conn->last_seqno_read = inframe->sequence_number; // reset warning... } @@ -2306,7 +2273,48 @@ void *player_thread_func(void *arg) { (int64_t)(config.audio_backend_latency_offset * config.output_rate)); // int64_t from int64_t - int32_t, so okay - // debug(1,"%" PRId64 "",sync_error,inbuflength); + if (at_least_one_frame_seen_this_session == 0) { + at_least_one_frame_seen_this_session = 1; + + // debug(2,"first frame real sync error (positive --> late): %" PRId64 " frames.", sync_error); + + // this is a sneaky attempt to make a final adjustment to the timing of the first packet + + // the very first packet generally has a first_frame_early_bias subtracted from its timing + // to make it more likely that it will be early than late, + // making it possible to compensate for it be adding a few frames of silence. + + // debug(2,"first frame real sync error (positive --> late): %" PRId64 " frames.", sync_error); + + // remove the bias when reporting the error to make it the true error + + debug(2,"first frame sync error (positive --> late): %" PRId64 " frames, %.3f mS at %d frames per second output.", sync_error+first_frame_early_bias, (1000.0*(sync_error+first_frame_early_bias))/config.output_rate, config.output_rate); + + // if the packet is early, add the frames needed to put it in sync. + if (sync_error < 0) { + size_t final_adjustment_length_sized = -sync_error; + char *final_adjustment_silence = malloc(conn->output_bytes_per_frame * final_adjustment_length_sized); + if (final_adjustment_silence) { + + conn->previous_random_number = + generate_zero_frames(final_adjustment_silence, final_adjustment_length_sized, config.output_format, + conn->enable_dither, conn->previous_random_number); + int final_adjustment = -sync_error; + final_adjustment = final_adjustment - first_frame_early_bias; + debug(2, "final sync adjustment: %d frames.", final_adjustment); + config.output->play(final_adjustment_silence, final_adjustment_length_sized); + free(final_adjustment_silence); + } else { + warn("Failed to allocate memory for a final_adjustment_silence buffer of %d frames for a " + "sync error of %d frames.", + final_adjustment_length_sized, sync_error); + } + sync_error = 0; // say the error was fixed! + } + } + + + at_least_one_frame_seen = 1; // not too sure if abs() is implemented for int64_t, so we'll do it manually int64_t abs_sync_error = sync_error; @@ -2416,7 +2424,9 @@ void *player_thread_func(void *arg) { if ((local_time_now) && (conn->first_packet_time_to_play) && (local_time_now >= conn->first_packet_time_to_play)) { - int64_t tp = (local_time_now - conn->first_packet_time_to_play) / 1000000000; // seconds int64_t from uint64_t which is always positive, so ok + int64_t tp = + (local_time_now - conn->first_packet_time_to_play) / + 1000000000; // seconds int64_t from uint64_t which is always positive, so ok if (tp < 5) amount_to_stuff = 0; // wait at least five seconds @@ -2435,17 +2445,18 @@ void *player_thread_func(void *arg) { // Apply DSP here - // check the state of loudness and convolution flags here and don't change them for the frame + // check the state of loudness and convolution flags here and don't change them for + // the frame int do_loudness = config.loudness; - #ifdef CONFIG_CONVOLUTION int do_convolution = 0; if ((config.convolution) && (config.convolver_valid)) do_convolution = 1; - // we will apply the convolution gain if convolution is enabled, even if there is no valid convolution happening + // we will apply the convolution gain if convolution is enabled, even if there is no + // valid convolution happening int convolution_is_enabled = 0; if (config.convolution) @@ -2456,7 +2467,7 @@ void *player_thread_func(void *arg) { #ifdef CONFIG_CONVOLUTION || convolution_is_enabled #endif - ) { + ) { int32_t *tbuf32 = (int32_t *)conn->tbuf; float fbuf_l[inbuflength]; float fbuf_r[inbuflength]; @@ -2511,7 +2522,7 @@ void *player_thread_func(void *arg) { ((config.packet_stuffing == ST_auto) && (config.soxr_delay_index > config.soxr_delay_threshold)) // if the CPU is deemed too slow - ) { + ) { #endif play_samples = stuff_buffer_basic_32((int32_t *)conn->tbuf, inbuflength, config.output_format, @@ -2672,7 +2683,7 @@ void *player_thread_func(void *arg) { conn->input_frame_rate = (1.0E9 * frames_received) / elapsed_reception_time; // an IEEE double calculation with two 64-bit integers - } else { + } else { conn->input_frame_rate = 0.0; } @@ -2704,42 +2715,41 @@ void *player_thread_func(void *arg) { if ((config.output->delay)) { if (config.no_sync == 0) { - inform("%*.2f," /* Sync error in milliseconds */ - "%*.1f," /* net correction in ppm */ - "%*.1f," /* corrections in ppm */ - "%*d," /* total packets */ - "%*" PRIu64 "," /* missing packets */ - "%*" PRIu64 "," /* late packets */ - "%*" PRIu64 "," /* too late packets */ - "%*" PRIu64 "," /* resend requests */ - "%*" PRId64 "," /* min DAC queue size */ - "%*" PRId32 "," /* min buffer occupancy */ - "%*" PRId32 "," /* max buffer occupancy */ - "%*.2f," /* source nominal frame rate */ - "%*.2f," /* source actual (average) frame rate */ - "%*.2f," /* output frame rate */ - "%*.2f," /* source clock drift */ - "%*d," /* source clock drift sample count */ - "%*.2f", /* rough calculated correction in ppm */ - 10, - 1000 * moving_average_sync_error / config.output_rate, 10, - moving_average_correction * 1000000 / (352 * conn->output_sample_ratio), - 10, moving_average_insertions_plus_deletions * 1000000 / - (352 * conn->output_sample_ratio), - 12, play_number, 7, conn->missing_packets, 7, conn->late_packets, 7, - conn->too_late_packets, 7, conn->resend_requests, 7, - minimum_dac_queue_size, 5, minimum_buffer_occupancy, 5, - maximum_buffer_occupancy, 11, conn->remote_frame_rate, 11, - conn->input_frame_rate, 11, conn->frame_rate, 10, - (conn->local_to_remote_time_gradient - 1.0) * 1000000, 6, - conn->local_to_remote_time_gradient_sample_count, 10, - (conn->frame_rate > 0.0) - ? ((conn->frame_rate - - conn->remote_frame_rate * conn->output_sample_ratio * - conn->local_to_remote_time_gradient) * - 1000000) / - conn->frame_rate - : 0.0); + inform( + "%*.2f," /* Sync error in milliseconds */ + "%*.1f," /* net correction in ppm */ + "%*.1f," /* corrections in ppm */ + "%*d," /* total packets */ + "%*" PRIu64 "," /* missing packets */ + "%*" PRIu64 "," /* late packets */ + "%*" PRIu64 "," /* too late packets */ + "%*" PRIu64 "," /* resend requests */ + "%*" PRId64 "," /* min DAC queue size */ + "%*" PRId32 "," /* min buffer occupancy */ + "%*" PRId32 "," /* max buffer occupancy */ + "%*.2f," /* source nominal frame rate */ + "%*.2f," /* source actual (average) frame rate */ + "%*.2f," /* output frame rate */ + "%*.2f," /* source clock drift */ + "%*d," /* source clock drift sample count */ + "%*.2f", /* rough calculated correction in ppm */ + 10, 1000 * moving_average_sync_error / config.output_rate, 10, + moving_average_correction * 1000000 / (352 * conn->output_sample_ratio), 10, + moving_average_insertions_plus_deletions * 1000000 / + (352 * conn->output_sample_ratio), + 12, play_number, 7, conn->missing_packets, 7, conn->late_packets, 7, + conn->too_late_packets, 7, conn->resend_requests, 7, minimum_dac_queue_size, + 5, minimum_buffer_occupancy, 5, maximum_buffer_occupancy, 11, + conn->remote_frame_rate, 11, conn->input_frame_rate, 11, conn->frame_rate, 10, + (conn->local_to_remote_time_gradient - 1.0) * 1000000, 6, + conn->local_to_remote_time_gradient_sample_count, 10, + (conn->frame_rate > 0.0) + ? ((conn->frame_rate - conn->remote_frame_rate * + conn->output_sample_ratio * + conn->local_to_remote_time_gradient) * + 1000000) / + conn->frame_rate + : 0.0); } else { inform("%*.2f," /* Sync error in milliseconds */ "%*d," /* total packets */ @@ -2754,10 +2764,9 @@ void *player_thread_func(void *arg) { "%*.2f," /* source actual (average) frame rate */ "%*.2f," /* source clock drift */ "%*d", /* source clock drift sample count */ - 10, - 1000 * moving_average_sync_error / config.output_rate, 12, play_number, 7, - conn->missing_packets, 7, conn->late_packets, 7, conn->too_late_packets, 7, - conn->resend_requests, 7, minimum_dac_queue_size, 5, + 10, 1000 * moving_average_sync_error / config.output_rate, 12, play_number, + 7, conn->missing_packets, 7, conn->late_packets, 7, conn->too_late_packets, + 7, conn->resend_requests, 7, minimum_dac_queue_size, 5, minimum_buffer_occupancy, 5, maximum_buffer_occupancy, 11, conn->remote_frame_rate, 11, conn->input_frame_rate, 10, (conn->local_to_remote_time_gradient - 1.0) * 1000000, 6, @@ -2776,10 +2785,9 @@ void *player_thread_func(void *arg) { "%*.2f," /* source actual (average) frame rate */ "%*.2f," /* source clock drift */ "%*d", /* source clock drift sample count */ - 10, - 1000 * moving_average_sync_error / config.output_rate, 12, play_number, 7, - conn->missing_packets, 7, conn->late_packets, 7, conn->too_late_packets, 7, - conn->resend_requests, 5, minimum_buffer_occupancy, 5, + 10, 1000 * moving_average_sync_error / config.output_rate, 12, play_number, + 7, conn->missing_packets, 7, conn->late_packets, 7, conn->too_late_packets, + 7, conn->resend_requests, 5, minimum_buffer_occupancy, 5, maximum_buffer_occupancy, 11, conn->remote_frame_rate, 11, conn->input_frame_rate, 10, (conn->local_to_remote_time_gradient - 1.0) * 1000000, 6, @@ -2879,13 +2887,15 @@ void player_volume_without_notification(double airplay_volume, rtsp_conn_info *c if (airplay_volume == -144.0) { if ((config.output->mute) && (config.output->mute(1) == 0)) - debug(2, "player_volume_without_notification: volume mode is %d, airplay_volume is %f, " - "hardware mute is enabled.", + debug(2, + "player_volume_without_notification: volume mode is %d, airplay_volume is %f, " + "hardware mute is enabled.", volume_mode, airplay_volume); else { conn->software_mute_enabled = 1; - debug(2, "player_volume_without_notification: volume mode is %d, airplay_volume is %f, " - "software mute is enabled.", + debug(2, + "player_volume_without_notification: volume mode is %d, airplay_volume is %f, " + "software mute is enabled.", volume_mode, airplay_volume); } @@ -2990,14 +3000,15 @@ void player_volume_without_notification(double airplay_volume, rtsp_conn_info *c } #ifdef CONFIG_METADATA - // here, send the 'pvol' metadata message when the airplay volume information - // is being used by shairport sync to control the output volume + // here, send the 'pvol' metadata message when the airplay volume information + // is being used by shairport sync to control the output volume char *dv = malloc(128); // will be freed in the metadata thread if (dv) { memset(dv, 0, 128); if (volume_mode == vol_both) { // normalise the maximum output to the hardware device's max output - snprintf(dv, 127, "%.2f,%.2f,%.2f,%.2f", airplay_volume, (scaled_attenuation - max_db + hw_max_db) / 100.0, + snprintf(dv, 127, "%.2f,%.2f,%.2f,%.2f", airplay_volume, + (scaled_attenuation - max_db + hw_max_db) / 100.0, (min_db - max_db + hw_max_db) / 100.0, (max_db - max_db + hw_max_db) / 100.0); } else { snprintf(dv, 127, "%.2f,%.2f,%.2f,%.2f", airplay_volume, scaled_attenuation / 100.0, @@ -3007,32 +3018,31 @@ void player_volume_without_notification(double airplay_volume, rtsp_conn_info *c } #endif - if (config.output->mute) config.output->mute(0); conn->software_mute_enabled = 0; - debug(2, "player_volume_without_notification: volume mode is %d, airplay volume is %f, " - "software_attenuation: %f, hardware_attenuation: %f, muting " - "is disabled.", + debug(2, + "player_volume_without_notification: volume mode is %d, airplay volume is %f, " + "software_attenuation: %f, hardware_attenuation: %f, muting " + "is disabled.", volume_mode, airplay_volume, software_attenuation, hardware_attenuation); } } #ifdef CONFIG_METADATA - else { - // here, send the 'pvol' metadata message when the airplay volume information - // is being used by shairport sync to control the output volume - char *dv = malloc(128); // will be freed in the metadata thread - if (dv) { - memset(dv, 0, 128); - snprintf(dv, 127, "%.2f,%.2f,%.2f,%.2f", airplay_volume, 0.0, 0.0, 0.0); - send_ssnc_metadata('pvol', dv, strlen(dv), 1); - } + else { + // here, send the 'pvol' metadata message when the airplay volume information + // is being used by shairport sync to control the output volume + char *dv = malloc(128); // will be freed in the metadata thread + if (dv) { + memset(dv, 0, 128); + snprintf(dv, 127, "%.2f,%.2f,%.2f,%.2f", airplay_volume, 0.0, 0.0, 0.0); + send_ssnc_metadata('pvol', dv, strlen(dv), 1); + } } #endif - // here, store the volume for possible use in the future config.airplay_volume = airplay_volume; debug_mutex_unlock(&conn->volume_control_mutex, 3); diff --git a/player.h b/player.h index 483d4fee..1db4b36f 100644 --- a/player.h +++ b/player.h @@ -43,10 +43,11 @@ typedef struct audio_buffer_entry { // decoded audio packets uint16_t resend_request_number; signed short *data; seq_t sequence_number; - uint64_t initialisation_time; // the time the packet was added or the time it was noticed the packet was missing - uint64_t resend_time; // time of last resend request or zero - uint32_t given_timestamp; // for debugging and checking - int length; // the length of the decoded data + uint64_t initialisation_time; // the time the packet was added or the time it was noticed the + // packet was missing + uint64_t resend_time; // time of last resend request or zero + uint32_t given_timestamp; // for debugging and checking + int length; // the length of the decoded data } abuf_t; // default buffer size diff --git a/rtp.c b/rtp.c index 70a54a36..f7e12c50 100644 --- a/rtp.c +++ b/rtp.c @@ -74,9 +74,13 @@ uint64_t local_to_remote_time_difference_now(rtsp_conn_info *conn) { uint64_t result = conn->local_to_remote_time_difference; if (conn->local_to_remote_time_gradient >= 1.0) { - result = conn->local_to_remote_time_difference + (uint64_t)((conn->local_to_remote_time_gradient - 1.0) * time_since_last_local_to_remote_time_difference_measurement); + result = conn->local_to_remote_time_difference + + (uint64_t)((conn->local_to_remote_time_gradient - 1.0) * + time_since_last_local_to_remote_time_difference_measurement); } else { - result = conn->local_to_remote_time_difference - (uint64_t)((1.0 - conn->local_to_remote_time_gradient) * time_since_last_local_to_remote_time_difference_measurement); + result = conn->local_to_remote_time_difference - + (uint64_t)((1.0 - conn->local_to_remote_time_gradient) * + time_since_last_local_to_remote_time_difference_measurement); } return result; } @@ -111,8 +115,7 @@ void *rtp_audio_receiver(void *arg) { uint64_t local_time_now_ns = get_absolute_time_in_ns(); if (time_of_previous_packet_ns) { - float time_interval_us = - (local_time_now_ns - time_of_previous_packet_ns) * 0.001; + float time_interval_us = (local_time_now_ns - time_of_previous_packet_ns) * 0.001; time_of_previous_packet_ns = local_time_now_ns; if (time_interval_us > longest_packet_time_interval_us) longest_packet_time_interval_us = time_interval_us; @@ -121,8 +124,9 @@ void *rtp_audio_receiver(void *arg) { stat_mean += stat_delta / stat_n; stat_M2 += stat_delta * (time_interval_us - stat_mean); if (stat_n % 2500 == 0) { - debug(2, "Packet reception interval stats: mean, standard deviation and max for the last " - "2,500 packets in microseconds: %10.1f, %10.1f, %10.1f.", + debug(2, + "Packet reception interval stats: mean, standard deviation and max for the last " + "2,500 packets in microseconds: %10.1f, %10.1f, %10.1f.", stat_mean, sqrtf(stat_M2 / (stat_n - 1)), longest_packet_time_interval_us); stat_n = 0; stat_mean = 0.0; @@ -290,7 +294,7 @@ void *rtp_control_receiver(void *arg) { ps = ps * 1000000000; // this many nanoseconds from the whole seconds pn = nctohl(&packet[12]); pn = pn * 1000000000; - pn = pn >> 32; // this many nanoseconds from the fractional part + pn = pn >> 32; // this many nanoseconds from the fractional part remote_time_of_sync = ps + pn; // debug(1,"Remote Sync Time: " PRIu64 "",remote_time_of_sync); @@ -354,9 +358,10 @@ void *rtp_control_receiver(void *arg) { if (la != conn->latency) { conn->latency = la; - debug(3, "New latency detected: %" PRIu32 ", sync latency: %" PRIu32 - ", minimum latency: %" PRIu32 ", maximum " - "latency: %" PRIu32 ", fixed offset: %" PRIu32 ".", + debug(3, + "New latency detected: %" PRIu32 ", sync latency: %" PRIu32 + ", minimum latency: %" PRIu32 ", maximum " + "latency: %" PRIu32 ", fixed offset: %" PRIu32 ".", la, sync_rtp_timestamp - rtp_timestamp_less_latency, conn->minimum_latency, conn->maximum_latency, config.fixedLatencyOffset); } @@ -573,7 +578,7 @@ void *rtp_timing_receiver(void *arg) { ps = ps * 1000000000; // this many nanoseconds from the whole seconds pn = nctohl(&packet[20]); pn = pn * 1000000000; - pn = pn >> 32; // this many nanoseconds from the fractional part + pn = pn >> 32; // this many nanoseconds from the fractional part distant_receive_time = ps + pn; // distant_transmit_time = @@ -583,7 +588,7 @@ void *rtp_timing_receiver(void *arg) { ps = ps * 1000000000; // this many nanoseconds from the whole seconds pn = nctohl(&packet[28]); pn = pn * 1000000000; - pn = pn >> 32; // this many nanoseconds from the fractional part + pn = pn >> 32; // this many nanoseconds from the fractional part distant_transmit_time = ps + pn; uint64_t remote_processing_time = 0; @@ -777,7 +782,9 @@ void *rtp_timing_receiver(void *arg) { // debug(1,"local to remote time gradient is %12.2f ppm, based on %d // samples.",conn->local_to_remote_time_gradient*1000000,sample_count); } else { - debug(2, "Time ping turnaround time: " PRIu64 " ns -- it looks like a timing ping was lost.", + debug(2, + "Time ping turnaround time: %" PRIu64 + " ns -- it looks like a timing ping was lost.", return_time); } } else { @@ -1029,7 +1036,7 @@ const int use_nominal_rate = 0; // specify whether to use the nominal input rate int sanitised_source_rate_information(uint32_t *frames, uint64_t *time, rtsp_conn_info *conn) { int result = 1; uint32_t fs = conn->input_rate; - *frames = fs; // default value to return + *frames = fs; // default value to return *time = 1000000000; // default value to return if ((conn->initial_reference_time) && (conn->initial_reference_timestamp)) { // uint32_t local_frames = conn->reference_timestamp - conn->initial_reference_timestamp; @@ -1178,22 +1185,20 @@ void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn) { (struct sockaddr *)&conn->rtp_client_control_socket, msgsize) == -1) { char em[1024]; strerror_r(errno, em, sizeof(em)); - debug(2, "Error %d using sendto to request a resend: \"%s\".", - errno, em); + debug(2, "Error %d using sendto to request a resend: \"%s\".", errno, em); conn->rtp_time_of_last_resend_request_error_ns = time_of_sending_ns; } else { conn->rtp_time_of_last_resend_request_error_ns = 0; } } else { - debug( - 3, - "Dropping resend request packet to simulate a bad network. Backing off for 0.3 " - "second."); + debug(3, "Dropping resend request packet to simulate a bad network. Backing off for 0.3 " + "second."); conn->rtp_time_of_last_resend_request_error_ns = time_of_sending_ns; } } else { - debug(1, "Suppressing a resend request due to a resend sendto error in the last 0.3 seconds."); + debug(1, + "Suppressing a resend request due to a resend sendto error in the last 0.3 seconds."); } } else { // if (!request_sent) { diff --git a/rtsp.c b/rtsp.c index 4013acb9..a1252133 100644 --- a/rtsp.c +++ b/rtsp.c @@ -286,15 +286,17 @@ void *player_watchdog_thread_code(void *arg) { uint64_t last_watchdog_bark_time = conn->watchdog_bark_time; debug_mutex_unlock(&conn->watchdog_mutex, 0); if (last_watchdog_bark_time != 0) { - uint64_t time_since_last_bark = (get_absolute_time_in_ns() - last_watchdog_bark_time) / 1000000000; + uint64_t time_since_last_bark = + (get_absolute_time_in_ns() - last_watchdog_bark_time) / 1000000000; uint64_t ct = config.timeout; // go from int to 64-bit int if (time_since_last_bark >= ct) { conn->watchdog_barks++; if (conn->watchdog_barks == 1) { // debuglev = 3; // tell us everything. - debug(1, "Connection %d: As Yeats almost said, \"Too long a silence / can make a stone " - "of the heart\".", + debug(1, + "Connection %d: As Yeats almost said, \"Too long a silence / can make a stone " + "of the heart\".", conn->connection_number); conn->stop = 1; pthread_cancel(conn->thread); @@ -571,7 +573,7 @@ int msg_handle_line(rtsp_message **pmsg, char *line) { } fail: - debug(3,"msg_handle_line fail"); + debug(3, "msg_handle_line fail"); msg_free(pmsg); *pmsg = NULL; return 0; @@ -614,7 +616,8 @@ enum rtsp_read_request_response rtsp_read_request(rtsp_conn_info *conn, rtsp_mes if (errno == EINTR) continue; if (errno == EAGAIN) { - debug(1, "Connection %d: getting Error 11 -- EAGAIN from a blocking read!", conn->connection_number); + debug(1, "Connection %d: getting Error 11 -- EAGAIN from a blocking read!", + conn->connection_number); continue; } if (errno != ECONNRESET) { @@ -627,15 +630,15 @@ enum rtsp_read_request_response rtsp_read_request(rtsp_conn_info *conn, rtsp_mes goto shutdown; } -/* // this outputs the message received - { - void *pt = malloc(nread+1); - memset(pt, 0, nread+1); - memcpy(pt, buf + inbuf, nread); - debug(1, "Incoming string on port: \"%s\"",pt); - free(pt); - } -*/ + /* // this outputs the message received + { + void *pt = malloc(nread+1); + memset(pt, 0, nread+1); + memcpy(pt, buf + inbuf, nread); + debug(1, "Incoming string on port: \"%s\"",pt); + free(pt); + } + */ inbuf += nread; @@ -644,7 +647,8 @@ enum rtsp_read_request_response rtsp_read_request(rtsp_conn_info *conn, rtsp_mes msg_size = msg_handle_line(the_packet, buf); if (!(*the_packet)) { - debug(1,"Connection %d: rtsp_read_request can't find an RTSP header.", conn->connection_number); + debug(1, "Connection %d: rtsp_read_request can't find an RTSP header.", + conn->connection_number); reply = rtsp_read_request_response_bad_packet; goto shutdown; } @@ -851,9 +855,10 @@ void handle_options(rtsp_conn_info *conn, __attribute__((unused)) rtsp_message * rtsp_message *resp) { debug(3, "Connection %d: OPTIONS", conn->connection_number); resp->respcode = 200; - msg_add_header(resp, "Public", "ANNOUNCE, SETUP, RECORD, " - "PAUSE, FLUSH, TEARDOWN, " - "OPTIONS, GET_PARAMETER, SET_PARAMETER"); + msg_add_header(resp, "Public", + "ANNOUNCE, SETUP, RECORD, " + "PAUSE, FLUSH, TEARDOWN, " + "OPTIONS, GET_PARAMETER, SET_PARAMETER"); } void handle_teardown(rtsp_conn_info *conn, __attribute__((unused)) rtsp_message *req, @@ -994,8 +999,9 @@ void handle_setup(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp) { msg_add_header(resp, "Session", "1"); resp->respcode = 200; // it all worked out okay - debug(1, "Connection %d: SETUP DACP-ID \"%s\" from %s to %s with UDP ports Control: " - "%d, Timing: %d and Audio: %d.", + debug(1, + "Connection %d: SETUP DACP-ID \"%s\" from %s to %s with UDP ports Control: " + "%d, Timing: %d and Audio: %d.", conn->connection_number, conn->dacp_id, &conn->client_ip_string, &conn->self_ip_string, conn->local_control_port, conn->local_timing_port, conn->local_audio_port); @@ -1590,7 +1596,7 @@ static void handle_set_parameter(rtsp_conn_info *conn, rtsp_message *req, rtsp_m char *ct = msg_get_header(req, "Content-Type"); if (ct) { -// debug(2, "SET_PARAMETER Content-Type:\"%s\".", ct); + // debug(2, "SET_PARAMETER Content-Type:\"%s\".", ct); #ifdef CONFIG_METADATA // It seems that the rtptime of the message is used as a kind of an ID that @@ -1888,7 +1894,7 @@ static void handle_announce(rtsp_conn_info *conn, rtsp_message *req, rtsp_messag unsigned int i = 0; unsigned int max_param = sizeof(conn->stream.fmtp) / sizeof(conn->stream.fmtp[0]); - char* found; + char *found; while ((found = strsep(&pfmtp, " \t")) != NULL && i < max_param) { conn->stream.fmtp[i++] = atoi(found); } @@ -2370,8 +2376,9 @@ static void *rtsp_conversation_thread_func(void *pconn) { if (strcmp(req->method, "OPTIONS") != 0) // the options message is very common, so don't log it until level 3 debug_level = 2; - debug(debug_level, "Connection %d: Received an RTSP Packet of type \"%s\":", - conn->connection_number, req->method), + debug(debug_level, + "Connection %d: Received an RTSP Packet of type \"%s\":", conn->connection_number, + req->method), debug_print_msg_headers(debug_level, req); apple_challenge(conn->fd, req, resp); @@ -2420,8 +2427,9 @@ static void *rtsp_conversation_thread_func(void *pconn) { if (conn->stop == 0) { int err = msg_write_response(conn->fd, resp); if (err) { - debug(1, "Connection %d: Unable to write an RTSP message response. Terminating the " - "connection.", + debug(1, + "Connection %d: Unable to write an RTSP message response. Terminating the " + "connection.", conn->connection_number); struct linger so_linger; so_linger.l_onoff = 1; // "true" @@ -2474,10 +2482,11 @@ static void *rtsp_conversation_thread_func(void *pconn) { if (reply == -1) { char errorstring[1024]; strerror_r(errno, (char *)errorstring, sizeof(errorstring)); - debug(1, "rtsp_read_request_response_bad_packet write response error %d: \"%s\".", errno, (char *)errorstring); + debug(1, "rtsp_read_request_response_bad_packet write response error %d: \"%s\".", errno, + (char *)errorstring); } else if (reply != (ssize_t)strlen(response_text)) { - debug(1, "rtsp_read_request_response_bad_packet write %d bytes requested but %d written.", strlen(response_text), - reply); + debug(1, "rtsp_read_request_response_bad_packet write %d bytes requested but %d written.", + strlen(response_text), reply); } } else { debug(1, "Connection %d: rtsp_read_request error %d, packet ignored.", @@ -2690,7 +2699,7 @@ void rtsp_listen_loop(void) { debug(2, "Connection %d: new connection from %s:%u to self at %s:%u.", conn->connection_number, remote_ip4, rport, ip4, tport); } - #ifdef AF_INET6 +#ifdef AF_INET6 if (local_info->SAFAMILY == AF_INET6) { // IPv6: @@ -2707,7 +2716,7 @@ void rtsp_listen_loop(void) { debug(2, "Connection %d: new connection from [%s]:%u to self at [%s]:%u.", conn->connection_number, remote_ip6, rport, ip6, tport); } - #endif +#endif } else { debug(1, "Error figuring out Shairport Sync's own IP number."); @@ -2735,8 +2744,8 @@ void rtsp_listen_loop(void) { pthread_cleanup_pop(1); // should never happen } else { warn("could not establish a service on port %d -- program terminating. Is another instance of " - "Shairport Sync running on this device?", - config.port); + "Shairport Sync running on this device?", + config.port); } // debug(1, "Oops -- fell out of the RTSP select loop"); } diff --git a/scripts/shairport-sync.conf b/scripts/shairport-sync.conf index 33db5d91..09098fbc 100644 --- a/scripts/shairport-sync.conf +++ b/scripts/shairport-sync.conf @@ -18,11 +18,20 @@ general = // interpolation = "auto"; // aka "stuffing". Default is "auto". Alternatives are "basic" or "soxr". Choose "soxr" only if you have a reasonably fast processor and Shairport Sync has been built with "soxr" support. // output_backend = "alsa"; // Run "shairport-sync -h" to get a list of all output_backends, e.g. "alsa", "pipe", "stdout". The default is the first one. // mdns_backend = "avahi"; // Run "shairport-sync -h" to get a list of all mdns_backends. The default is the first one. +// interface = "name"; // Use this advanced setting to specify the interface on which Shairport Sync should provide its service. Leave it commented out to get the default, which is to select the interface(s) automatically. // port = 5000; // Listen for service requests on this port // udp_port_base = 6001; // start allocating UDP ports from this port number when needed // udp_port_range = 10; // look for free ports in this number of places, starting at the UDP port base. Allow at least 10, though only three are needed in a steady state. +// regtype = "_raop._tcp"; // Use this advanced setting to set the service type and transport to be advertised by Zeroconf/Bonjour. Default is "_raop._tcp". + // drift_tolerance_in_seconds = 0.002; // allow a timing error of this number of seconds of drift away from exact synchronisation before attempting to correct it // resync_threshold_in_seconds = 0.050; // a synchronisation error greater than this number of seconds will cause resynchronisation; 0 disables it + +// playback_mode = "stereo"; // This can be "stereo", "mono", "reverse stereo", "both left" or "both right". Default is "stereo". +// alac_decoder = "hammerton"; // This can be "hammerton" or "apple". This advanced setting allows you to choose +// the original Shairport decoder by David Hammerton or the Apple Lossless Audio Codec (ALAC) decoder written by Apple. +// If you build Shairport Sync with the flag --with-apple-alac, the Apple ALAC decoder will be chosen by default. + // ignore_volume_control = "no"; // set this to "yes" if you want the volume to be at 100% no matter what the source's volume control is set to. // volume_range_db = 60 ; // use this advanced setting to set the range, in dB, you want between the maximum volume and the minimum volume. Range is 30 to 150 dB. Leave it commented out to use mixer's native range. // volume_max_db = 0.0 ; // use this advanced setting, which must have a decimal point in it, to set the maximum volume, in dB, you wish to use. @@ -36,25 +45,23 @@ general = // The desired AirPlay volume is appended to the end of the command line – leave a space if you want it treated as an extra argument. // AirPlay volume goes from 0 to -30 and -144 means "mute". -// regtype = "_raop._tcp"; // Use this advanced setting to set the service type and transport to be advertised by Zeroconf/Bonjour. Default is "_raop._tcp". -// playback_mode = "stereo"; // This can be "stereo", "mono", "reverse stereo", "both left" or "both right". Default is "stereo". -// alac_decoder = "hammerton"; // This can be "hammerton" or "apple". This advanced setting allows you to choose -// the original Shairport decoder by David Hammerton or the Apple Lossless Audio Codec (ALAC) decoder written by Apple. -// If you build Shairport Sync with the flag --with-apple-alac, the Apple ALAC decoder will be chosen by default. -// interface = "name"; // Use this advanced setting to specify the interface on which Shairport Sync should provide its service. Leave it commented out to get the default, which is to select the interface(s) automatically. - -// audio_backend_latency_offset_in_seconds = 0.0; // This is added to the latency requested by the player to delay or advance the output by a fixed amount. Use it, for example, to compensate for a fixed delay in the audio back end. E.g. if the output device, e.g. a soundbar, takes 100 ms to process audio, set this to -0.1 to deliver the audio to the output device 100 ms early, allowing it time to process the audio and output it perfectly in sync. -// audio_backend_buffer_desired_length_in_seconds = 0.2; // If set too small, buffer underflow occurs on low-powered machines. Too long and the response time to volume changes becomes annoying. Default is 0.2 seconds in the alsa backend, 0.35 seconds in the pa backend and 1.0 seconds otherwise. +// audio_backend_latency_offset_in_seconds = 0.0; // This is added to the latency requested by the player to delay or advance the output by a fixed amount. +// Use it, for example, to compensate for a fixed delay in the audio back end. +// E.g. if the output device, e.g. a soundbar, takes 100 ms to process audio, set this to -0.1 to deliver the audio +// to the output device 100 ms early, allowing it time to process the audio and output it perfectly in sync. +// audio_backend_buffer_desired_length_in_seconds = 0.2; // If set too small, buffer underflow occurs on low-powered machines. +// Too long and the response time to volume changes becomes annoying. +// Default is 0.2 seconds in the alsa backend, 0.35 seconds in the pa backend and 1.0 seconds otherwise. // audio_backend_buffer_interpolation_threshold_in_seconds = 0.075; // Advanced feature. If the buffer size drops below this, stop using time-consuming interpolation like soxr to avoid dropouts due to underrun. -// audio_backend_silent_lead_in_time = "auto"; // This optional advanced setting, either "auto" or a positive number, sets the length of the period of silence that precedes the start of the audio. The default is "auto" -- the silent lead-in starts as soon as the player starts sending packets. Values greater than the latency are ignored. Values that are too low will affect initial synchronisation. -// audio_backend_silent_lead_in_initial_period = 0.5; // This optional advanced setting sets the length, in seconds, of the first "block" of silence sent to the DAC -- some DACs need a fairly substantial block to prevent lots of initial underrun. -// audio_backend_silent_lead_in_minimum_adjustments_to_make = 3; // This optional advanced setting sets the minimum number of times Shairport Sync will readjust the length of the silent lead in to ensure that the audio starts at precisely the correct time. -// audio_backend_silent_lead_in_adjustment_interval = 0.1; // This optional advanced setting sets the interval, in seconds, between adjustments of the silent lead in to ensure that the audio starts at precisely the correct time. +// audio_backend_silent_lead_in_time = "auto"; // This optional advanced setting, either "auto" or a positive number, sets the length of the period of silence that precedes the start of the audio. +// The default is "auto" -- the silent lead-in starts as soon as the player starts sending packets. +// Values greater than the latency are ignored. Values that are too low will affect initial synchronisation. // dbus_service_bus = "system"; // The Shairport Sync dbus interface, if selected at compilation, will appear // as "org.gnome.ShairportSync" on the whichever bus you specify here: "system" (default) or "session". // mpris_service_bus = "system"; // The Shairport Sync mpris interface, if selected at compilation, will appear // as "org.gnome.ShairportSync" on the whichever bus you specify here: "system" (default) or "session". + // resend_control_first_check_time = 0.10; // Use this optional advanced setting to set the wait time in seconds before deciding a packet is missing. // resend_control_check_interval_time = 0.25; // Use this optional advanced setting to set the time in seconds between requests for a missing packet. // resend_control_last_check_time = 0.10; // Use this optional advanced setting to set the latest time, in seconds, by which the last check should be done before the estimated time of a missing packet's transfer to the output buffer. @@ -64,14 +71,14 @@ general = // Advanced parameters for controlling how Shairport Sync stays active and how it runs a session sessioncontrol = { -// run_this_before_play_begins = "/full/path/to/application and args"; // make sure the application has executable permission. If it's a script, include the shebang (#!/bin/...) on the first line -// run_this_after_play_ends = "/full/path/to/application and args"; // make sure the application has executable permission. If it's a script, include the shebang (#!/bin/...) on the first line - // "active" state starts when play begins and ends when the active_state_timeout has elapsed after play ends, unless another play session starts before the timeout has fully elapsed. // run_this_before_entering_active_state = "/full/path/to/application and args"; // make sure the application has executable permission. If it's a script, include the shebang (#!/bin/...) on the first line // run_this_after_exiting_active_state = "/full/path/to/application and args"; // make sure the application has executable permission. If it's a script, include the shebang (#!/bin/...) on the first line // active_state_timeout = 10.0; // wait for this number of seconds after play ends before leaving the active state, unless another play session begins. +// run_this_before_play_begins = "/full/path/to/application and args"; // make sure the application has executable permission. If it's a script, include the shebang (#!/bin/...) on the first line +// run_this_after_play_ends = "/full/path/to/application and args"; // make sure the application has executable permission. If it's a script, include the shebang (#!/bin/...) on the first line + // run_this_if_an_unfixable_error_is_detected = "/full/path/to/application and args"; // if a problem occurs that can't be cleared by Shairport Sync itself, hook a program on here to deal with it. An error code-string is passed as the last argument. // Many of these "unfixable" problems are caused by malfunctioning output devices, and sometimes it is necessary to restart the whole device to clear the problem. // You could hook on a program to do this automatically, but beware -- the device may then power off and restart without warning! @@ -91,15 +98,19 @@ alsa = // output_device = "default"; // the name of the alsa output device. Use "shairport-sync -h" to discover the names of ALSA hardware devices. Use "alsamixer" or "aplay" to find out the names of devices, mixers, etc. // mixer_control_name = "PCM"; // the name of the mixer to use to adjust output volume. If not specified, volume in adjusted in software. // mixer_device = "default"; // the mixer_device default is whatever the output_device is. Normally you wouldn't have to use this. + // output_rate = "auto"; // can be "auto", 44100, 88200, 176400 or 352800, but the device must have the capability. // output_format = "auto"; // can be "auto", "U8", "S8", "S16", "S16_LE", "S16_BE", "S24", "S24_LE", "S24_BE", "S24_3LE", "S24_3BE", "S32", "S32_LE" or "S32_BE" but the device must have the capability. Except where stated using (*LE or *BE), endianness matches that of the processor. -// disable_synchronization = "no"; // Set to "yes" to disable synchronization. Default is "no". + +// disable_synchronization = "no"; // Set to "yes" to disable synchronization. Default is "no" This is really meant for troubleshooting. + // period_size = ; // Use this optional advanced setting to set the alsa period size near to this value // buffer_size = ; // Use this optional advanced setting to set the alsa buffer size near to this value // use_mmap_if_available = "yes"; // Use this optional advanced setting to control whether MMAP-based output is used to communicate with the DAC. Default is "yes" // use_hardware_mute_if_available = "no"; // Use this optional advanced setting to control whether the hardware in the DAC is used for muting. Default is "no", for compatibility with other audio players. // maximum_stall_time = 0.200; // Use this optional advanced setting to control how long to wait for data to be consumed by the output device before considering it an error. It should never approach 200 ms. // use_precision_timing = "auto"; // Use this optional advanced setting to control how Shairport Sync gathers timing information. When set to "auto", if the output device is a real hardware device, precision timing will be used. Choose "no" for more compatible standard timing, choose "yes" to force the use of precision timing, which may cause problems. + // disable_standby_mode = "never"; // This setting prevents the DAC from entering the standby mode. Some DACs make small "popping" noises when they go in and out of standby mode. Settings can be: "always", "auto" or "never". Default is "never", but only for backwards compatibility. The "auto" setting prevents entry to standby mode while Shairport Sync is in the "active" mode. You can use "yes" instead of "always" and "no" instead of "never". // disable_standby_mode_silence_threshold = 0.040; // Use this optional advanced setting to control how little audio should remain in the output buffer before the disable_standby code should start sending silence to the output device. // disable_standby_mode_silence_scan_interval = 0.004; // Use this optional advanced setting to control how often the amount of audio remaining in the output buffer should be checked. diff --git a/shairport-sync-dbus-test-client.c b/shairport-sync-dbus-test-client.c index d0b15bb2..2433fd69 100644 --- a/shairport-sync-dbus-test-client.c +++ b/shairport-sync-dbus-test-client.c @@ -186,11 +186,10 @@ int main(int argc, char *argv[]) { g_signal_connect(proxy4, "notify::volume", G_CALLBACK(notify_volume_callback), "ShairportSync.AdvancedRemoteControl"); - - g_print("Starting test...\n"); - g_print("Using the RemoteControl interface, play for five seconds, pause for five seconds and then resume play...\n"); + g_print("Using the RemoteControl interface, play for five seconds, pause for five seconds and " + "then resume play...\n"); g_print("Play...\n"); shairport_sync_remote_control_call_play(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), NULL, NULL, 0); sleep(5); @@ -200,24 +199,31 @@ int main(int argc, char *argv[]) { g_print("Play...\n"); shairport_sync_remote_control_call_play(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), NULL, NULL, 0); sleep(5); - g_print("Using the RemoteControl interface, set AirPlay Volume (range -30 to 0) to -30, -20, -10, 0 and -15 for five seconds each...\n"); + g_print("Using the RemoteControl interface, set AirPlay Volume (range -30 to 0) to -30, -20, " + "-10, 0 and -15 for five seconds each...\n"); g_print("Set AirPlay Volume (range -30 to 0) to -30\n"); - shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), -30, NULL, NULL, 0); + shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), -30, + NULL, NULL, 0); sleep(5); g_print("Set AirPlay Volume (range -30 to 0) to -20\n"); - shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), -20, NULL, NULL, 0); + shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), -20, + NULL, NULL, 0); sleep(5); g_print("Set AirPlay Volume (range -30 to 0) to -10\n"); - shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), -10, NULL, NULL, 0); + shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), -10, + NULL, NULL, 0); sleep(5); g_print("Set AirPlay Volume (range -30 to 0) to -0\n"); - shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), 0, NULL, NULL, 0); + shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), 0, + NULL, NULL, 0); sleep(5); g_print("Set AirPlay Volume (range -30 to 0) to -15\n"); - shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), -15, NULL, NULL, 0); + shairport_sync_remote_control_call_set_airplay_volume(SHAIRPORT_SYNC_REMOTE_CONTROL(proxy3), -15, + NULL, NULL, 0); sleep(5); - g_print("Using the AdvancedRemoteControl interface, set Volume to 20%%, 100%%, 40%% and 60%% for five seconds each...\n"); + g_print("Using the AdvancedRemoteControl interface, set Volume to 20%%, 100%%, 40%% and 60%% for " + "five seconds each...\n"); g_print("Set Volume to 20%%\n"); shairport_sync_advanced_remote_control_call_set_volume( SHAIRPORT_SYNC_ADVANCED_REMOTE_CONTROL(proxy4), 20, NULL, NULL, 0); diff --git a/shairport.c b/shairport.c index 60bfb4a0..e9cd70d3 100644 --- a/shairport.c +++ b/shairport.c @@ -145,7 +145,8 @@ void *soxr_time_check(__attribute__((unused)) void *arg) { int number_of_iterations = 0; uint64_t soxr_start_time = get_absolute_time_in_ns(); uint64_t loop_until_time = - (uint64_t)1500000000 + soxr_start_time; // loop for a second and a half, max -- no need to be able to cancel it, do _don't even try_! + (uint64_t)1500000000 + soxr_start_time; // loop for a second and a half, max -- no need to be + // able to cancel it, do _don't even try_! while (get_absolute_time_in_ns() < loop_until_time) { number_of_iterations++; @@ -184,8 +185,7 @@ void *soxr_time_check(__attribute__((unused)) void *arg) { NULL, NULL); // Default configuration. } - double soxr_execution_time_ns = - (get_absolute_time_in_ns() - soxr_start_time) * 1.0; + double soxr_execution_time_ns = (get_absolute_time_in_ns() - soxr_start_time) * 1.0; // free(outbuffer); // free(inbuffer); config.soxr_delay_index = (int)(0.9 + soxr_execution_time_ns / (number_of_iterations * 1000000)); @@ -388,7 +388,8 @@ int parse_options(int argc, char **argv) { config.resyncthreshold = 1.0 * fResyncthreshold / 44100; config.tolerance = 1.0 * fTolerance / 44100; - config.audio_backend_silent_lead_in_time_auto = 1; // start outputting silence as soon as packets start arriving + config.audio_backend_silent_lead_in_time_auto = + 1; // start outputting silence as soon as packets start arriving config.airplay_volume = -18.0; // if no volume is ever set, default to initial default value if // nothing else comes in first. config.fixedLatencyOffset = 11025; // this sounds like it works properly. @@ -399,17 +400,23 @@ int parse_options(int argc, char **argv) { // automatically. config.volume_range_hw_priority = 0; // if combining software and hardware volume control, give the software priority -// i.e. when reducing volume, reduce the sw first before reducing the software. -// this is because some hw mixers mute at the bottom of their range, and they don't always advertise -// this fact - config.resend_control_first_check_time = 0.10; // wait this many seconds before requesting the resending of a missing packet - config.resend_control_check_interval_time = 0.25; // wait this many seconds before again requesting the resending of a missing packet - config.resend_control_last_check_time = 0.10; // give up if the packet is still missing this close to when it's needed - config.missing_port_dacp_scan_interval_seconds = 2.0; // check at this interval if no DACP port number is known + // i.e. when reducing volume, reduce the sw first before reducing the software. + // this is because some hw mixers mute at the bottom of their range, and they don't always + // advertise this fact + config.resend_control_first_check_time = + 0.10; // wait this many seconds before requesting the resending of a missing packet + config.resend_control_check_interval_time = + 0.25; // wait this many seconds before again requesting the resending of a missing packet + config.resend_control_last_check_time = + 0.10; // give up if the packet is still missing this close to when it's needed + config.missing_port_dacp_scan_interval_seconds = + 2.0; // check at this interval if no DACP port number is known - config.minimum_free_buffer_headroom = 125; // leave approximately one second's worth of buffers free after calculating the effective latency. - // e.g. if we have 1024 buffers or 352 frames = 8.17 seconds and we have a nominal latency of 2.0 seconds - // then we can add an offset of 5.17 seconds and still leave a second's worth of buffers for unexpected circumstances + config.minimum_free_buffer_headroom = 125; // leave approximately one second's worth of buffers + // free after calculating the effective latency. + // e.g. if we have 1024 buffers or 352 frames = 8.17 seconds and we have a nominal latency of 2.0 + // seconds then we can add an offset of 5.17 seconds and still leave a second's worth of buffers + // for unexpected circumstances #ifdef CONFIG_METADATA_HUB config.cover_art_cache_dir = "/tmp/shairport-sync/.cache/coverart"; @@ -419,9 +426,10 @@ int parse_options(int argc, char **argv) { 1; // number of seconds between DACP server scans when playing nothing config.scan_max_bad_response_count = 5; // number of successive bad results to ignore before giving up - //config.scan_max_inactive_count = - // (365 * 24 * 60 * 60) / config.scan_interval_when_inactive; // number of scans to do before stopping if - // not made active again (not used) + // config.scan_max_inactive_count = + // (365 * 24 * 60 * 60) / config.scan_interval_when_inactive; // number of scans to do before + // stopping if + // not made active again (not used) #endif // config_setting_t *setting; @@ -772,42 +780,38 @@ int parse_options(int argc, char **argv) { die("Invalid alac_decoder option choice \"%s\". It should be \"hammerton\" or \"apple\""); } - /* Get the resend control settings. */ - if (config_lookup_float(config.cfg, "general.resend_control_first_check_time", - &dvalue)) { + if (config_lookup_float(config.cfg, "general.resend_control_first_check_time", &dvalue)) { if ((dvalue >= 0.0) && (dvalue <= 3.0)) config.resend_control_first_check_time = dvalue; else warn("Invalid general resend_control_first_check_time setting \"%f\". It should " - "be " - "between 0.0 and 3.0, " - "inclusive. The setting remains at %f seconds.", - dvalue, config.resend_control_first_check_time); + "be " + "between 0.0 and 3.0, " + "inclusive. The setting remains at %f seconds.", + dvalue, config.resend_control_first_check_time); } - if (config_lookup_float(config.cfg, "general.resend_control_check_interval_time", - &dvalue)) { + if (config_lookup_float(config.cfg, "general.resend_control_check_interval_time", &dvalue)) { if ((dvalue >= 0.0) && (dvalue <= 3.0)) config.resend_control_check_interval_time = dvalue; else warn("Invalid general resend_control_check_interval_time setting \"%f\". It should " - "be " - "between 0.0 and 3.0, " - "inclusive. The setting remains at %f seconds.", - dvalue, config.resend_control_check_interval_time); + "be " + "between 0.0 and 3.0, " + "inclusive. The setting remains at %f seconds.", + dvalue, config.resend_control_check_interval_time); } - if (config_lookup_float(config.cfg, "general.resend_control_last_check_time", - &dvalue)) { + if (config_lookup_float(config.cfg, "general.resend_control_last_check_time", &dvalue)) { if ((dvalue >= 0.0) && (dvalue <= 3.0)) config.resend_control_last_check_time = dvalue; else warn("Invalid general resend_control_last_check_time setting \"%f\". It should " - "be " - "between 0.0 and 3.0, " - "inclusive. The setting remains at %f seconds.", - dvalue, config.resend_control_last_check_time); + "be " + "between 0.0 and 3.0, " + "inclusive. The setting remains at %f seconds.", + dvalue, config.resend_control_last_check_time); } if (config_lookup_float(config.cfg, "general.missing_port_dacp_scan_interval_seconds", @@ -816,50 +820,16 @@ int parse_options(int argc, char **argv) { config.missing_port_dacp_scan_interval_seconds = dvalue; else warn("Invalid general missing_port_dacp_scan_interval_seconds setting \"%f\". It should " - "be " - "between 0.0 and 300.0, " - "inclusive. The setting remains at %f seconds.", - dvalue, config.missing_port_dacp_scan_interval_seconds); + "be " + "between 0.0 and 300.0, " + "inclusive. The setting remains at %f seconds.", + dvalue, config.missing_port_dacp_scan_interval_seconds); } /* Get the default latency. Deprecated! */ if (config_lookup_int(config.cfg, "latencies.default", &value)) config.userSuppliedLatency = value; - /* Get the lead-in silence settings. */ - if (config_lookup_float(config.cfg, "general.audio_backend_silent_lead_in_initial_period", - &dvalue)) { - if ((dvalue >= 0.0) && (dvalue <= 3.0)) - config.lead_in_silence_initial_period = dvalue; - else - warn("Invalid general audio_backend_silent_lead_in_initial_period setting \"%f\". It should " - "be " - "between 0.0 and 3.0, " - "inclusive. The setting remains at %f seconds.", - dvalue, config.lead_in_silence_initial_period); - } - - if (config_lookup_int(config.cfg, "general.audio_backend_silent_lead_in_minimum_adjustments_to_make", &value)) { - if (value < 0) - die("Invalid audio_backend_silent_lead_in_minimum_adjustments_to_make range \"%d\". It must be positive. The setting remains at %d.", - value, config.lead_in_silence_minimum_adjustments_to_make); - else - config.lead_in_silence_minimum_adjustments_to_make = value; - } - - if (config_lookup_float(config.cfg, "general.audio_backend_silent_lead_in_adjustment_interval", - &dvalue)) { - if ((dvalue >= 0.0) && (dvalue <= 3.0)) - config.lead_in_silence_adjustment_interval = dvalue; - else - warn("Invalid general audio_backend_silent_lead_in_adjustment_interval setting \"%f\". It should " - "be " - "between 0.0 and 3.0, " - "inclusive. The setting remains at %f seconds.", - dvalue, config.lead_in_silence_adjustment_interval); - } - - #ifdef CONFIG_METADATA /* Get the metadata setting. */ config.metadata_enabled = 1; // if metadata support is included, then enable it by default @@ -1014,7 +984,8 @@ int parse_options(int argc, char **argv) { if (config_lookup_string(config.cfg, "dsp.convolution_ir_file", &str)) { config.convolution_ir_file = strdup(str); - config.convolver_valid = convolver_init(config.convolution_ir_file, config.convolution_max_length); + config.convolver_valid = + convolver_init(config.convolution_ir_file, config.convolution_max_length); } if (config.convolution && config.convolution_ir_file == NULL) { @@ -1210,7 +1181,7 @@ int parse_options(int argc, char **argv) { poptFreeContext(optCon); -// here, we are finally finished reading the options + // here, we are finally finished reading the options #ifdef CONFIG_LIBDAEMON if ((daemonisewith) && (daemonisewithout)) @@ -1224,8 +1195,9 @@ int parse_options(int argc, char **argv) { #else /* Check if we are called with -d or --daemon or -j or justDaemoniseNoPIDFile options*/ if ((daemonisewith != 0) || (daemonisewithout != 0)) { - fprintf(stderr, "%s was built without libdaemon, so does not support daemonisation using the " - "-d, --daemon, -j or --justDaemoniseNoPIDFile options\n", + fprintf(stderr, + "%s was built without libdaemon, so does not support daemonisation using the " + "-d, --daemon, -j or --justDaemoniseNoPIDFile options\n", config.appName); exit(EXIT_FAILURE); } @@ -1248,8 +1220,6 @@ int parse_options(int argc, char **argv) { char hostname[100]; gethostname(hostname, 100); - - char *i0; if (raw_service_name == NULL) i0 = strdup("%H"); // this is the default it the Service Name wasn't specified @@ -1320,25 +1290,24 @@ const char *pid_file_proc(void) { } #endif - void exit_function() { // the following is to ensure that if libdaemon has been included // that most of this code will be skipped when the parent process is exiting // exec #ifdef CONFIG_LIBDAEMON - if (this_is_the_daemon_process) { //this is the daemon that is exiting + if (this_is_the_daemon_process) { // this is the daemon that is exiting #endif - debug(1, "exit function called..."); + debug(1, "exit function called..."); -/* -Actually, there is no terminate_mqtt() function. -#ifdef CONFIG_MQTT - if (config.mqtt_enabled) { - terminate_mqtt(); - } -#endif -*/ + /* + Actually, there is no terminate_mqtt() function. + #ifdef CONFIG_MQTT + if (config.mqtt_enabled) { + terminate_mqtt(); + } + #endif + */ #if defined(CONFIG_DBUS_INTERFACE) || defined(CONFIG_MPRIS_INTERFACE) @@ -1349,62 +1318,61 @@ Actually, there is no stop_mpris_service() function. #endif */ #ifdef CONFIG_DBUS_INTERFACE - stop_dbus_service(); + stop_dbus_service(); #endif - if (g_main_loop) { - debug(2, "Stopping DBUS Loop Thread"); - g_main_loop_quit(g_main_loop); - pthread_join(dbus_thread, NULL); - } + if (g_main_loop) { + debug(2, "Stopping DBUS Loop Thread"); + g_main_loop_quit(g_main_loop); + pthread_join(dbus_thread, NULL); + } #endif #ifdef CONFIG_DACP_CLIENT - debug(2, "Stopping DACP Monitor"); - dacp_monitor_stop(); + debug(2, "Stopping DACP Monitor"); + dacp_monitor_stop(); #endif #ifdef CONFIG_METADATA_HUB - debug(2, "Stopping metadata hub"); - metadata_hub_stop(); + debug(2, "Stopping metadata hub"); + metadata_hub_stop(); #endif #ifdef CONFIG_METADATA - metadata_stop(); // close down the metadata pipe + metadata_stop(); // close down the metadata pipe #endif - activity_monitor_stop(0); + activity_monitor_stop(0); - if ((config.output) && (config.output->deinit)) { - debug(2, "Deinitialise the audio backend."); - config.output->deinit(); - } + if ((config.output) && (config.output->deinit)) { + debug(2, "Deinitialise the audio backend."); + config.output->deinit(); + } #ifdef CONFIG_SOXR - // be careful -- not sure if the thread can be cancelled cleanly, so wait for it to shut down - pthread_join(soxr_time_check_thread, NULL); + // be careful -- not sure if the thread can be cancelled cleanly, so wait for it to shut down + pthread_join(soxr_time_check_thread, NULL); #endif + if (conns) + free(conns); // make sure the connections have been deleted first - if (conns) - free(conns); // make sure the connections have been deleted first - - if (config.service_name) - free(config.service_name); + if (config.service_name) + free(config.service_name); #ifdef CONFIG_CONVOLUTION - if (config.convolution_ir_file) - free(config.convolution_ir_file); + if (config.convolution_ir_file) + free(config.convolution_ir_file); #endif - if (config.regtype) - free(config.regtype); + if (config.regtype) + free(config.regtype); #ifdef CONFIG_LIBDAEMON - daemon_retval_send(0); - daemon_pid_file_remove(); - daemon_signal_done(); - if (config.computed_piddir) - free(config.computed_piddir); + daemon_retval_send(0); + daemon_pid_file_remove(); + daemon_signal_done(); + if (config.computed_piddir) + free(config.computed_piddir); } #endif @@ -1421,7 +1389,8 @@ Actually, there is no stop_mpris_service() function. void handle_sigchld(__attribute__((unused)) int sig) { int saved_errno = errno; - while (waitpid((pid_t)(-1), 0, WNOHANG) > 0) {} + while (waitpid((pid_t)(-1), 0, WNOHANG) > 0) { + } errno = saved_errno; } @@ -1500,7 +1469,7 @@ int main(int argc, char **argv) { // config.userSuppliedLatency = 0; // zero means none supplied config.debugger_show_file_and_line = - 1; // by default, log the file and line of the originating message + 1; // by default, log the file and line of the originating message config.debugger_show_relative_time = 1; // by default, log the time back to the previous debug message config.resyncthreshold = 0.05; // 50 ms @@ -1532,9 +1501,6 @@ int main(int argc, char **argv) { config.output_rate_auto_requested = 1; // default auto select format config.decoders_supported = 1 << decoder_hammerton; // David Hammerton's decoder supported by default - config.lead_in_silence_initial_period = 0.5; // the size of the first block of silence to send to the DAC - config.lead_in_silence_minimum_adjustments_to_make = 3; // make at least this number of checks and timing adjustments - config.lead_in_silence_adjustment_interval = 0.025; // make checks and adjustments at this interval #ifdef CONFIG_APPLE_ALAC config.decoders_supported += 1 << decoder_apple_alac; config.use_apple_decoder = 1; // use the ALAC decoder by default if support has been included @@ -1766,8 +1732,9 @@ int main(int argc, char **argv) { /* Print out options */ debug(1, "disable resend requests is %s.", config.disable_resend_requests ? "on" : "off"); - debug(1, "diagnostic_drop_packet_fraction is %f. A value of 0.0 means no packets will be dropped " - "deliberately.", + debug(1, + "diagnostic_drop_packet_fraction is %f. A value of 0.0 means no packets will be dropped " + "deliberately.", config.diagnostic_drop_packet_fraction); debug(1, "statistics_requester status is %d.", config.statistics_requested); #if CONFIG_LIBDAEMON @@ -1790,8 +1757,8 @@ int main(int argc, char **argv) { debug(1, "mdns backend \"%s\".", config.mdns_name); debug(2, "userSuppliedLatency is %d.", config.userSuppliedLatency); debug(1, "interpolation setting is \"%s\".", - config.packet_stuffing == ST_basic ? "basic" : config.packet_stuffing == ST_soxr ? "soxr" - : "auto"); + config.packet_stuffing == ST_basic ? "basic" + : config.packet_stuffing == ST_soxr ? "soxr" : "auto"); debug(1, "interpolation soxr_delay_threshold is %d.", config.soxr_delay_threshold); debug(1, "resync time is %f seconds.", config.resyncthreshold); debug(1, "allow a session to be interrupted: %d.", config.allow_session_interruption); @@ -1805,8 +1772,9 @@ int main(int argc, char **argv) { debug(1, "volume_max_db is not set"); debug(1, "volume range in dB (zero means use the range specified by the mixer): %u.", config.volume_range_db); - debug(1, "volume_range_combined_hardware_priority (1 means hardware mixer attenuation is used " - "first) is %d.", + debug(1, + "volume_range_combined_hardware_priority (1 means hardware mixer attenuation is used " + "first) is %d.", config.volume_range_hw_priority); debug(1, "playback_mode is %d (0-stereo, 1-mono, 1-reverse_stereo, 2-both_left, 3-both_right).", config.playback_mode); @@ -1828,15 +1796,8 @@ int main(int argc, char **argv) { if (config.audio_backend_silent_lead_in_time_auto == 1) debug(1, "audio backend silence lead-in time is \"auto\"."); else - debug(1, "audio backend silence lead-in time is %f seconds.", - config.audio_backend_silent_lead_in_time); - - debug(1, "audio backend silent lead-in initial period is %f seconds.", - config.lead_in_silence_initial_period); - debug(1, "audio backend silent lead-in minimum adjustments to make is %d.", config.lead_in_silence_minimum_adjustments_to_make); - debug(1, "audio backend silent lead-in adjustment interval is %f seconds.", - config.lead_in_silence_adjustment_interval); - + debug(1, "audio backend silence lead-in time is %f seconds.", + config.audio_backend_silent_lead_in_time); debug(1, "zeroconf regtype is \"%s\".", config.regtype); debug(1, "decoders_supported field is %d.", config.decoders_supported); debug(1, "use_apple_decoder is %d.", config.use_apple_decoder); diff --git a/tinysvcmdns.c b/tinysvcmdns.c index 069a6436..9e605e07 100644 --- a/tinysvcmdns.c +++ b/tinysvcmdns.c @@ -1466,7 +1466,7 @@ int close_pipe(int s) { // main loop to receive, process and send out MDNS replies // also handles MDNS service announces -void* main_loop(struct mdnsd *svr) { +void *main_loop(struct mdnsd *svr) { fd_set sockfd_set; int max_fd = svr->sockfd; char notify_buf[2]; // buffer for reading of notify_pipe @@ -1724,7 +1724,7 @@ struct mdnsd *mdnsd_start() { pthread_attr_init(&attr); pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED); - if (pthread_create(&tid, &attr, (void * (*)(void *))&main_loop, (void *)server) != 0) { + if (pthread_create(&tid, &attr, (void *(*)(void *)) & main_loop, (void *)server) != 0) { pthread_mutex_destroy(&server->data_lock); free(server); return NULL; @@ -1737,7 +1737,8 @@ void mdnsd_stop(struct mdnsd *s) { assert(s != NULL); struct timeval tv = { - .tv_sec = 0, .tv_usec = 500 * 1000, + .tv_sec = 0, + .tv_usec = 500 * 1000, }; s->stop_flag = 1; From 89e99f28a58db9c3baf49e9d8f563db9bc88c678 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 18 May 2020 17:00:56 +0100 Subject: [PATCH 087/186] Improve initial sync calculations, even with short (0.3 sec) silent lead-in times. --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 7f148707..026345ef 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.7d5], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.7d6], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 923a4f0a3cecacc64adf5839a8c7b57ea7615957 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 18 May 2020 17:25:09 +0100 Subject: [PATCH 088/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index a1a91eb7..58a7fb0b 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,9 @@ +Version 3.3.7d6 +==== +**Bug Fixes and Enhancements** +* Clean up and improve the code to synchronise the first frame of audio. This should result in more accurate and reliable initial synchrnoisation, certainly to under a millisecond in normal usage. Sync should improve even when the silent lead-in time is as short as 0.3 seconds or when the `audio_backend_latency_offset_in_seconds` is as much as -1.7 seconds. +* Removed a bug which would affect initial synchronisation if a flush command was received from the player at an inopportune time. + Version 3.3.7d5 ==== **Bug Fixes and Enhancements** From 034dd53af5c3dfd3420fddfb010ea2c5bb6a20bc Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 23 May 2020 10:18:23 +0100 Subject: [PATCH 089/186] Calculate the delay and sync error modulo the output frame numbers. Duh! A long-standing bug. --- player.c | 90 +++++++++++++++++++++++++------------------------------- player.h | 2 +- 2 files changed, 41 insertions(+), 51 deletions(-) diff --git a/player.c b/player.c index 344662c3..94dccd02 100644 --- a/player.c +++ b/player.c @@ -249,6 +249,30 @@ static inline seq_t seq_sum(seq_t a, seq_t b) { return r; } +// This orders u and v by picking the smaller of the two modulo differences +// in unsigned modulo arithmetic and setting the sign of the result accordingly. + +// (Think of a modulo ring starting at 0 and circling around finally to +// modulo-2, modulo-1 to 0 again. The ordering of u and v is always based on +// the shorter way around the ring, taking the 0, 1, 2, ... direction as positive.) + +// The result will always fit in an int64_t, +// Due to the asymmetry of 2's complement representation, however, +// if the difference is equal to modulo / 2, it is ambiguous as to whether +// u is "before" v or v is "before" u. +// and will be returned as - modulo / 2. + +int64_t int64_mod_difference(const uint64_t u, const uint64_t v, const uint64_t modulo) { + int64_t response; + uint64_t diff = (u - v) % modulo; + uint64_t invdiff = (v - u) % modulo; + if (diff < invdiff) + response = diff; + else + response = -invdiff; + return response; +} + void reset_input_flow_metrics(rtsp_conn_info *conn) { conn->play_number_after_flush = 0; conn->packet_count_since_flush = 0; @@ -1793,7 +1817,7 @@ void *player_thread_func(void *arg) { int64_t tsum_of_sync_errors, tsum_of_corrections, tsum_of_insertions_and_deletions, tsum_of_drifts; int64_t previous_sync_error = 0, previous_correction = 0; - int64_t minimum_dac_queue_size = INT64_MAX; + uint64_t minimum_dac_queue_size = UINT64_MAX; int32_t minimum_buffer_occupancy = INT32_MAX; int32_t maximum_buffer_occupancy = INT32_MIN; @@ -1808,7 +1832,7 @@ void *player_thread_func(void *arg) { conn->buffer_occupancy = 0; int play_samples = 0; - int64_t current_delay; + uint64_t current_delay; int play_number = 0; conn->play_number_after_flush = 0; // int last_timestamp = 0; // for debugging only @@ -2136,35 +2160,12 @@ void *player_thread_func(void *arg) { uint64_t reference_timestamp_time, remote_reference_timestamp_time; get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time, &remote_reference_timestamp_time, conn); // types okay - int64_t rt, nt; - rt = reference_timestamp; // uint32_t to int64_t + uint64_t nt; nt = inframe->given_timestamp; // uint32_t to int64_t - rt = rt * conn->output_sample_ratio; nt = nt * conn->output_sample_ratio; uint64_t local_time_now = get_absolute_time_in_ns(); // types okay - int64_t td = 0; // td is the time difference between the reference timestamp time and the - // present time. Only used to calculate td_in_frames - int64_t td_in_frames = 0; // td_in_frames is the number of frames between between the - // reference timestamp time and the present time - - if (local_time_now >= reference_timestamp_time) { - td = local_time_now - reference_timestamp_time; // this is the positive value. - // Conversion is positive uint64_t to - // int64_t, thus okay - td_in_frames = (td * config.output_rate) / 1000000000; - } else { - td = reference_timestamp_time - local_time_now; // this is the absolute value, which - // should be negated. Conversion is - // positive uint64_t to int64_t, thus - // okay. - td_in_frames = (td * config.output_rate) / - 1000000000; // use the absolute td value for the present. Types okay - td_in_frames = -td_in_frames; - td = -td; // should be okay, as the range of values should be very small w.r.t 64 bits - } - // This is the timing error for the next audio frame in the DAC, if applicable int64_t sync_error = 0; @@ -2202,15 +2203,16 @@ void *player_thread_func(void *arg) { // If any of these are false, we don't do any synchronisation stuff int resp = -1; // use this as a flag -- if negative, we can't rely on a real known delay - current_delay = -1; // use this as a failure flag if (config.output->delay) { long l_delay; resp = config.output->delay(&l_delay); if (resp == 0) { // no error current_delay = l_delay; - if (current_delay < 0) { - debug(2, "Underrun of %lld frames reported, but ignored.", current_delay); + if (l_delay >= 0) + current_delay = l_delay; + else { + debug(2, "Underrun of %ld frames reported, but ignored.", l_delay); current_delay = 0; // could get a negative value if there was underrun, but ignore it. } @@ -2244,29 +2246,17 @@ void *player_thread_func(void *arg) { uint32_t should_be_frame_32; local_time_to_frame(local_time_now, &should_be_frame_32, conn); - int64_t should_be_frame = ((int64_t)should_be_frame_32) * conn->output_sample_ratio; + // int64_t should_be_frame = ((int64_t)should_be_frame_32) * conn->output_sample_ratio; - // int64_t absolute_difference_in_frames = td_in_frames + rt - should_be_frame; - // if (absolute_difference_in_frames < 0) - // absolute_difference_in_frames = -absolute_difference_in_frames; + int64_t delay = int64_mod_difference(should_be_frame_32 * conn->output_sample_ratio, nt - current_delay, UINT32_MAX * conn->output_sample_ratio); - // if (absolute_difference_in_frames > 10 * conn->output_sample_ratio) - // debug(1, "Difference between old and new frame number is %" PRId64 " frames.", - // td_in_frames + rt - should_be_frame); - // this is the actual delay, including the latency we actually want, which will - // fluctuate a good bit about a potentially rising or falling trend. + //int64_t delay = should_be_frame - (nt - current_delay); // all int64_t - // int64_t delay = td_in_frames + rt - (nt - current_delay); // all int64_t - // cut over to the new calculation method - int64_t delay = should_be_frame - (nt - current_delay); // all int64_t + // the original frame numbers are unsigned 32-bit integers that roll over modulo 2^32 + // hence these delay figures will be unsigned numbers that roll over modulo 2^32 * conn->output_sample_ratio; + // therefore, calculating the delay must be done in the light of possible rollover - // td_in_frames + rt is the frame number that should be output at local_time_now. - // This is the timing error for the next audio frame in the DAC. - - // if positive, it means that the packet will be late -- the delay is longer than - // requested - // if negative, the packet will be early -- the delay is less than expected. sync_error = delay - ((int64_t)conn->latency * conn->output_sample_ratio + @@ -2724,7 +2714,7 @@ void *player_thread_func(void *arg) { "%*" PRIu64 "," /* late packets */ "%*" PRIu64 "," /* too late packets */ "%*" PRIu64 "," /* resend requests */ - "%*" PRId64 "," /* min DAC queue size */ + "%*" PRIu64 "," /* min DAC queue size */ "%*" PRId32 "," /* min buffer occupancy */ "%*" PRId32 "," /* max buffer occupancy */ "%*.2f," /* source nominal frame rate */ @@ -2757,7 +2747,7 @@ void *player_thread_func(void *arg) { "%*" PRIu64 "," /* late packets */ "%*" PRIu64 "," /* too late packets */ "%*" PRIu64 "," /* resend requests */ - "%*" PRId64 "," /* min DAC queue size */ + "%*" PRIu64 "," /* min DAC queue size */ "%*" PRId32 "," /* min buffer occupancy */ "%*" PRId32 "," /* max buffer occupancy */ "%*.2f," /* source nominal frame rate */ @@ -2797,7 +2787,7 @@ void *player_thread_func(void *arg) { inform("No frames received in the last sampling interval."); } } - minimum_dac_queue_size = INT64_MAX; // hack reset + minimum_dac_queue_size = UINT64_MAX; // hack reset maximum_buffer_occupancy = INT32_MIN; // can't be less than this minimum_buffer_occupancy = INT32_MAX; // can't be more than this at_least_one_frame_seen = 0; diff --git a/player.h b/player.h index 1db4b36f..55b4426b 100644 --- a/player.h +++ b/player.h @@ -258,7 +258,7 @@ typedef struct { void *dapo_private_storage; // this is used for compatibility, if dacp stuff isn't enabled. int enable_dither; // needed for filling silences before play actually starts - int64_t dac_buffer_queue_minimum_length; + uint64_t dac_buffer_queue_minimum_length; } rtsp_conn_info; uint32_t modulo_32_offset(uint32_t from, uint32_t to); From 4f5ff881a5b50630c0ed61bf4536179fd2d5d532 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 23 May 2020 13:54:21 +0100 Subject: [PATCH 090/186] Tidy up some of the modulo arithmetic stuff, including tidying up how to deal with out-of-order packet arrival. --- player.c | 141 +++++++++++++++++-------------------------------------- 1 file changed, 44 insertions(+), 97 deletions(-) diff --git a/player.c b/player.c index 94dccd02..f472959e 100644 --- a/player.c +++ b/player.c @@ -143,6 +143,7 @@ static void ab_resync(rtsp_conn_info *conn) { conn->ab_buffering = 1; } + // given starting and ending points as unsigned 16-bit integers running modulo 2^16, returns the // position of x in the interval in *pos // returns true if x is actually within the buffer @@ -173,94 +174,47 @@ int position_in_modulo_uint16_t_buffer(uint16_t x, uint16_t start, uint16_t end, return response; } -// given starting and ending points as unsigned 32-bit integers running modulo 2^32, returns the -// position of x in the interval in *pos -// returns true if x is actually within the buffer +static inline seq_t SUCCESSOR(seq_t x) { + return x + 1; +} -int position_in_modulo_uint32_t_buffer(uint32_t x, uint32_t start, uint32_t end, uint32_t *pos) { - int response = 0; // not in the buffer - if (start <= end) { - if (x < start) { - if (pos) - *pos = UINT32_MAX - start + 1 + x; - } else { - if (pos) - *pos = x - start; - if (x < end) - response = 1; - } - } else if ((x >= start)) { // && (x <= UINT32_MAX)) { // always true - response = 1; - if (pos) - *pos = x - start; - } else { - if (pos) - *pos = UINT32_MAX - start + 1 + x; - if (x < end) { - response = 1; - } - } +// a minus b +int16_t seq_diff(seq_t a, seq_t b) { + int16_t response; + seq_t diff = a - b; + seq_t invdiff = b - a; + if (diff < invdiff) + response = diff; + else + response = -invdiff; return response; } -// this is used. -static inline seq_t SUCCESSOR(seq_t x) { - uint32_t p = x & 0xffff; - p += 1; - p = p & 0xffff; - return p; -} - -// used in seq_diff and seq_order - -// anything with ORDINATE in it must be protected by the ab_mutex -static inline int32_t ORDINATE(seq_t x, seq_t base) { - int32_t p = x; // int32_t from seq_t, i.e. uint16_t, so okay - int32_t q = base; // int32_t from seq_t, i.e. uint16_t, so okay - int32_t t = (p + 0x10000 - q) & 0xffff; - // we definitely will get a positive number in t at this point, but it might be a - // positive alias of a negative number, i.e. x might actually be "before" ab_read - // So, if the result is greater than 32767, we will assume its an - // alias and subtract 65536 from it - if (t >= 32767) { - // debug(1,"OOB: %u, ab_r: %u, ab_w: %u",x,ab_read,ab_write); - t -= 65536; - } - return t; -} - -// wrapped number between two seq_t. -int32_t seq_diff(seq_t a, seq_t b, seq_t base) { - int32_t diff = ORDINATE(b, base) - ORDINATE(a, base); - return diff; -} - -// the sequence numbers will wrap pretty often. -// this returns true if the second arg is after the first -static inline int seq_order(seq_t a, seq_t b, seq_t base) { - int32_t d = ORDINATE(b, base) - ORDINATE(a, base); - return d > 0; -} - static inline seq_t seq_sum(seq_t a, seq_t b) { - // uint32_t p = a & 0xffff; - // uint32_t q = b & 0x0ffff; - uint32_t r = (a + b) & 0xffff; - return r; + return a + b; } // This orders u and v by picking the smaller of the two modulo differences // in unsigned modulo arithmetic and setting the sign of the result accordingly. -// (Think of a modulo ring starting at 0 and circling around finally to -// modulo-2, modulo-1 to 0 again. The ordering of u and v is always based on -// the shorter way around the ring, taking the 0, 1, 2, ... direction as positive.) +// if u - v gives the smaller modulo difference, then that modulo difference is returned +// otherwise the negative of the v - u modulo difference is returned. -// The result will always fit in an int64_t, +// (Think of a modulo ring starting at 0 and circling around clockwise finally to +// modulo-2, modulo-1 to 0 again. To determine the ordering of the two numbers, +// Find the shorter path between them around the ring. +// If you go from v to u in the positive (0, 1, 2 ... or clockwise) direction, +// then u is greater than (or "after") v.) + +// The result will always fit in an int64_t as it must be less or equal to half way +// around the ring. // Due to the asymmetry of 2's complement representation, however, // if the difference is equal to modulo / 2, it is ambiguous as to whether -// u is "before" v or v is "before" u. -// and will be returned as - modulo / 2. +// u is "after" or "before" (i.e. greater or less than) v. +// and will be returned as -modulo / 2, that is, "before". +// If that ever happens and there is a real ambiguity in the application, +// the modulo chosen is too small. + int64_t int64_mod_difference(const uint64_t u, const uint64_t v, const uint64_t modulo) { int64_t response; @@ -539,8 +493,9 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in conn->ab_read = seqno; conn->ab_synced = 1; } - if (conn->ab_write == - seqno) { // if this is the expected packet (which could be the first packet...) + int16_t write_point_gap = seq_diff(seqno,conn->ab_write); // this is the difference between + // the incoming packet number and the packet number that was expected. + if (write_point_gap == 0) { // if this is the expected packet (which could be the first packet...) if (conn->input_frame_rate_starting_point_is_valid == 0) { if ((conn->packet_count_since_flush >= 500) && (conn->packet_count_since_flush <= 510)) { conn->frames_inward_measurement_start_time = time_now; @@ -552,12 +507,10 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in conn->frames_inward_frames_received_at_measurement_time = actual_timestamp; abuf = conn->audio_buffer + BUFIDX(seqno); conn->ab_write = SUCCESSOR(seqno); // move the write pointer to the next free space - } else if (seq_order(conn->ab_write, seqno, conn->ab_read)) { // newer than expected - int32_t gap = seq_diff(conn->ab_write, seqno, conn->ab_read); - if (gap <= 0) - debug(1, "Unexpected gap size: %d.", gap); + } else if (write_point_gap > 0) { // newer than expected + // initialise the frames in between int i; - for (i = 0; i < gap; i++) { + for (i = 0; i < write_point_gap; i++) { abuf = conn->audio_buffer + BUFIDX(seq_sum(conn->ab_write, i)); abuf->ready = 0; // to be sure, to be sure abuf->resend_request_number = 0; @@ -568,13 +521,9 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in abuf->given_timestamp = 0; abuf->sequence_number = 0; } - // debug(1,"N %d s %u.",seq_diff(ab_write,PREDECESSOR(seqno))+1,ab_write); abuf = conn->audio_buffer + BUFIDX(seqno); - // rtp_request_resend(ab_write, gap); - // resend_requests++; conn->ab_write = SUCCESSOR(seqno); - } else if (seq_order(conn->ab_read, seqno, - conn->ab_read)) { // older than expected but not too late + } else if (seq_diff(seqno,conn->ab_read) > 0) { // older than expected but still not too late conn->late_packets++; abuf = conn->audio_buffer + BUFIDX(seqno); } else { // too late. @@ -1035,11 +984,10 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { if (curframe->sequence_number != conn->ab_read) { // some kind of sync problem has occurred. if (BUFIDX(curframe->sequence_number) == BUFIDX(conn->ab_read)) { - // it looks like some kind of aliasing has happened - if (seq_order(conn->ab_read, curframe->sequence_number, conn->ab_read)) { - conn->ab_read = curframe->sequence_number; - debug(1, "Aliasing of buffer index -- reset."); - } + // it looks like aliasing has happened + // jump to the new incoming stuff... + conn->ab_read = curframe->sequence_number; + debug(1, "Aliasing of buffer index -- reset."); } else { debug(1, "Inconsistent sequence numbers detected"); } @@ -1789,9 +1737,9 @@ void *player_thread_func(void *arg) { debug(3, "Output frame bytes is %d.", conn->output_bytes_per_frame); - conn->dac_buffer_queue_minimum_length = (int64_t)( + conn->dac_buffer_queue_minimum_length = (uint64_t)( config.audio_backend_buffer_interpolation_threshold_in_seconds * config.output_rate); - debug(3, "dac_buffer_queue_minimum_length is %" PRId64 " frames.", + debug(3, "dac_buffer_queue_minimum_length is %" PRIu64 " frames.", conn->dac_buffer_queue_minimum_length); conn->session_corrections = 0; @@ -2033,8 +1981,7 @@ void *player_thread_func(void *arg) { "status 0x%X after %u resend requests.", SUCCESSOR(conn->last_seqno_read), play_number, inframe->status, inframe->resend_request_number); - conn->last_seqno_read = (SUCCESSOR(conn->last_seqno_read) & - 0xffff); // manage the packet out of sequence minder + conn->last_seqno_read = SUCCESSOR(conn->last_seqno_read); // manage the packet out of sequence minder void *silence = malloc(conn->output_bytes_per_frame * conn->max_frames_per_packet * conn->output_sample_ratio); @@ -2189,7 +2136,7 @@ void *player_thread_func(void *arg) { } conn->buffer_occupancy = - seq_diff(conn->ab_read, conn->ab_write, conn->ab_read); // int32_t from int32 + seq_diff(conn->ab_write, conn->ab_read); // int32_t from int16_t if (conn->buffer_occupancy < minimum_buffer_occupancy) minimum_buffer_occupancy = conn->buffer_occupancy; From 1945cd0b7fa31718b8c0530d591ecadaa60ba77e Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 23 May 2020 14:26:52 +0100 Subject: [PATCH 091/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 ++ 1 file changed, 2 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 58a7fb0b..2c5108ed 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,6 +1,8 @@ Version 3.3.7d6 ==== **Bug Fixes and Enhancements** +* Fix a bug calculating the instantaneous synchronisation error. This could occasionally cause Shairport Sync to lose synchronisation and maybe even to mute for a few seconds before regaining it. It was caused by not doing modulo arithmetic properly, and it's been there for a while. +* Clean up some other uses of modulo arithmetic. * Clean up and improve the code to synchronise the first frame of audio. This should result in more accurate and reliable initial synchrnoisation, certainly to under a millisecond in normal usage. Sync should improve even when the silent lead-in time is as short as 0.3 seconds or when the `audio_backend_latency_offset_in_seconds` is as much as -1.7 seconds. * Removed a bug which would affect initial synchronisation if a flush command was received from the player at an inopportune time. From 759b87f2a291939b46a4cd00e7630764d8143927 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 23 May 2020 20:31:54 +0100 Subject: [PATCH 092/186] Move the statistics array from the stack into the heap. --- player.c | 54 +++++++++++++++++++++++++----------------------------- player.h | 8 ++++++++ 2 files changed, 33 insertions(+), 29 deletions(-) diff --git a/player.c b/player.c index f472959e..19fed7b3 100644 --- a/player.c +++ b/player.c @@ -1579,10 +1579,6 @@ int stuff_buffer_soxr_32(int32_t *inptr, int32_t *scratchBuffer, int length, } #endif -typedef struct stats { // statistics for running averages - int64_t sync_error, correction, drift; -} stats_t; - void player_thread_initial_cleanup_handler(__attribute__((unused)) void *arg) { rtsp_conn_info *conn = (rtsp_conn_info *)arg; debug(3, "Connection %d: player thread main loop exit via player_thread_initial_cleanup_handler.", @@ -1649,6 +1645,11 @@ void player_thread_cleanup_handler(void *arg) { free(conn->tbuf); conn->tbuf = NULL; } + + if (conn->statistics) { + free(conn->statistics); + conn->statistics = NULL; + } free_audio_buffers(conn); if (conn->stream.type == ast_apple_lossless) terminate_decoders(conn); @@ -1756,9 +1757,10 @@ void *player_thread_func(void *arg) { conn->connection_state_to_output = get_requested_connection_state_to_output(); // this is about half a minute //#define trend_interval 3758 + +// this is about 8 seconds #define trend_interval 1003 - stats_t statistics[trend_interval]; int number_of_statistics, oldest_statistic, newest_statistic; int at_least_one_frame_seen = 0; int at_least_one_frame_seen_this_session = 0; @@ -1853,9 +1855,6 @@ void *player_thread_func(void *arg) { // we need an intermediate "transition" buffer - // if ((input_rate!=config.output_rate) || (input_bit_depth!=output_bit_depth)) { - // debug(1,"Define tbuf of length - // %d.",output_bytes_per_frame*(max_frames_per_packet*output_sample_ratio+max_frame_size_change)); conn->tbuf = malloc( sizeof(int32_t) * 2 * (conn->max_frames_per_packet * conn->output_sample_ratio + conn->max_frame_size_change)); @@ -1884,6 +1883,10 @@ void *player_thread_func(void *arg) { int sync_error_out_of_bounds = 0; // number of times in a row that there's been a serious sync error + conn->statistics = malloc(sizeof(stats_t)*trend_interval); + if (conn->statistics == NULL) + die("Failed to allocate a statistics buffer"); + conn->framesProcessedInThisEpoch = 0; conn->framesGeneratedInThisEpoch = 0; conn->correctionsRequestedInThisEpoch = 0; @@ -2565,34 +2568,34 @@ void *player_thread_func(void *arg) { // valid samples and the number of times sync wasn't checked due to non availability of a // delay figure. // for the present, stats are only updated when sync has been checked - if (sync_error != -1) { + if (config.output->delay != NULL) { if (number_of_statistics == trend_interval) { // here we remove the oldest statistical data and take it from the summaries as well - tsum_of_sync_errors -= statistics[oldest_statistic].sync_error; - tsum_of_drifts -= statistics[oldest_statistic].drift; - if (statistics[oldest_statistic].correction > 0) - tsum_of_insertions_and_deletions -= statistics[oldest_statistic].correction; + tsum_of_sync_errors -= conn->statistics[oldest_statistic].sync_error; + tsum_of_drifts -= conn->statistics[oldest_statistic].drift; + if (conn->statistics[oldest_statistic].correction > 0) + tsum_of_insertions_and_deletions -= conn->statistics[oldest_statistic].correction; else - tsum_of_insertions_and_deletions += statistics[oldest_statistic].correction; - tsum_of_corrections -= statistics[oldest_statistic].correction; + tsum_of_insertions_and_deletions += conn->statistics[oldest_statistic].correction; + tsum_of_corrections -= conn->statistics[oldest_statistic].correction; oldest_statistic = (oldest_statistic + 1) % trend_interval; number_of_statistics--; } - statistics[newest_statistic].sync_error = sync_error; - statistics[newest_statistic].correction = conn->amountStuffed; + conn->statistics[newest_statistic].sync_error = sync_error; + conn->statistics[newest_statistic].correction = conn->amountStuffed; if (number_of_statistics == 0) - statistics[newest_statistic].drift = 0; + conn->statistics[newest_statistic].drift = 0; else - statistics[newest_statistic].drift = + conn->statistics[newest_statistic].drift = sync_error - previous_sync_error - previous_correction; previous_sync_error = sync_error; previous_correction = conn->amountStuffed; tsum_of_sync_errors += sync_error; - tsum_of_drifts += statistics[newest_statistic].drift; + tsum_of_drifts += conn->statistics[newest_statistic].drift; if (conn->amountStuffed > 0) { tsum_of_insertions_and_deletions += conn->amountStuffed; } else { @@ -3034,17 +3037,10 @@ int player_play(rtsp_conn_info *conn) { if (pt == NULL) die("Couldn't allocate space for pthread_t"); conn->player_thread = pt; - size_t size = (PTHREAD_STACK_MIN + 256 * 1024); - pthread_attr_t tattr; - pthread_attr_init(&tattr); - int rc = pthread_attr_setstacksize(&tattr, size); - if (rc) - debug(1, "Error setting stack size for player_thread: %s", strerror(errno)); - // finished initialising. - rc = pthread_create(pt, &tattr, player_thread_func, (void *)conn); + int rc = pthread_create(pt, NULL, player_thread_func, (void *)conn); if (rc) debug(1, "Error creating player_thread: %s", strerror(errno)); - pthread_attr_destroy(&tattr); + #ifdef CONFIG_METADATA debug(2, "pbeg"); send_ssnc_metadata('pbeg', NULL, 0, 1); // contains cancellation points diff --git a/player.h b/player.h index 55b4426b..f79f8806 100644 --- a/player.h +++ b/player.h @@ -50,6 +50,10 @@ typedef struct audio_buffer_entry { // decoded audio packets int length; // the length of the decoded data } abuf_t; +typedef struct stats { // statistics for running averages + int64_t sync_error, correction, drift; +} stats_t; + // default buffer size // This needs to be a power of 2 because of the way BUFIDX(seqno) works. // 512 is the minimum for normal operation -- it gives 512*352/44100 or just over 4 seconds of @@ -110,6 +114,10 @@ typedef struct { int32_t *sbuf; char *outbuf; + // for generating running statistics... + + stats_t *statistics; + // for holding the output rate information until printed out at the end of a session double frame_rate; int frame_rate_status; From 5b24dddecd8031739c53939aa762c7b8a7a1d224 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 23 May 2020 20:33:50 +0100 Subject: [PATCH 093/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 2c5108ed..0cfb96da 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -2,9 +2,10 @@ Version 3.3.7d6 ==== **Bug Fixes and Enhancements** * Fix a bug calculating the instantaneous synchronisation error. This could occasionally cause Shairport Sync to lose synchronisation and maybe even to mute for a few seconds before regaining it. It was caused by not doing modulo arithmetic properly, and it's been there for a while. -* Clean up some other uses of modulo arithmetic. * Clean up and improve the code to synchronise the first frame of audio. This should result in more accurate and reliable initial synchrnoisation, certainly to under a millisecond in normal usage. Sync should improve even when the silent lead-in time is as short as 0.3 seconds or when the `audio_backend_latency_offset_in_seconds` is as much as -1.7 seconds. * Removed a bug which would affect initial synchronisation if a flush command was received from the player at an inopportune time. +* Clean up some other uses of modulo arithmetic. +* Clean up allocation of memory for gathering running statistics -- use the heap instead of the stack. Version 3.3.7d5 ==== From 9280e1430baee3db257bdb8468fb584d83ffed39 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 24 May 2020 09:45:03 +0100 Subject: [PATCH 094/186] Clean up statistics display for backends with no sync. Flush pipe and stdout backends when play stops. --- audio_pipe.c | 6 ++++-- audio_stdout.c | 5 ++++- player.c | 19 +++++++++---------- 3 files changed, 17 insertions(+), 13 deletions(-) diff --git a/audio_pipe.c b/audio_pipe.c index 4ec9b240..c2e71642 100644 --- a/audio_pipe.c +++ b/audio_pipe.c @@ -86,8 +86,10 @@ static int play(void *buf, int samples) { static void stop(void) { // Don't close the pipe just because a play session has stopped. - // if (fd > 0) - // close(fd); + // Flush it instead + if (fd > 0) + if (fsync(fd) != 0) + debug(1, "Error flushing pipe when play stopped: \"%s\".", strerror(errno)); } static int init(int argc, char **argv) { diff --git a/audio_stdout.c b/audio_stdout.c index 00ded166..aca660b1 100644 --- a/audio_stdout.c +++ b/audio_stdout.c @@ -56,7 +56,10 @@ static int play(void *buf, int samples) { } static void stop(void) { - // don't close stdout + // Flush stdout when play stops instead + if (fd > 0) + if (fsync(fd) != 0) + debug(1, "Error flushing STDOUT when play stopped: \"%s\".", strerror(errno)); } static int init(__attribute__((unused)) int argc, __attribute__((unused)) char **argv) { diff --git a/player.c b/player.c index 19fed7b3..a750415a 100644 --- a/player.c +++ b/player.c @@ -1927,8 +1927,7 @@ void *player_thread_func(void *arg) { "source clock drift sample count"); } } else { - inform("sync error in milliseconds, " - "total packets, " + inform("total packets, " "missing packets, " "late packets, " "too late packets, " @@ -2094,7 +2093,8 @@ void *player_thread_func(void *arg) { die("Shairport Sync only supports 16 bit input"); } - inbuflength *= conn->output_sample_ratio; + + at_least_one_frame_seen = 1; // We have a frame of data. We need to see if we want to add or remove a frame from it to // keep in sync. @@ -2106,6 +2106,7 @@ void *player_thread_func(void *arg) { // now, go back as far as the total latency less, say, 100 ms, and check the presence of // frames from then onwards + inbuflength *= conn->output_sample_ratio; uint32_t reference_timestamp; uint64_t reference_timestamp_time, remote_reference_timestamp_time; get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time, @@ -2147,6 +2148,9 @@ void *player_thread_func(void *arg) { if (conn->buffer_occupancy > maximum_buffer_occupancy) maximum_buffer_occupancy = conn->buffer_occupancy; + + + // here, we want to check (a) if we are meant to do synchronisation, // (b) if we have a delay procedure, (c) if we can get the delay. @@ -2252,10 +2256,6 @@ void *player_thread_func(void *arg) { sync_error = 0; // say the error was fixed! } } - - - at_least_one_frame_seen = 1; - // not too sure if abs() is implemented for int64_t, so we'll do it manually int64_t abs_sync_error = sync_error; if (abs_sync_error < 0) @@ -2713,8 +2713,7 @@ void *player_thread_func(void *arg) { conn->local_to_remote_time_gradient_sample_count); } } else { - inform("%*.2f," /* Sync error in milliseconds */ - "%*d," /* total packets */ + inform("%*d," /* total packets */ "%*" PRIu64 "," /* missing packets */ "%*" PRIu64 "," /* late packets */ "%*" PRIu64 "," /* too late packets */ @@ -2725,7 +2724,7 @@ void *player_thread_func(void *arg) { "%*.2f," /* source actual (average) frame rate */ "%*.2f," /* source clock drift */ "%*d", /* source clock drift sample count */ - 10, 1000 * moving_average_sync_error / config.output_rate, 12, play_number, + 12, play_number, 7, conn->missing_packets, 7, conn->late_packets, 7, conn->too_late_packets, 7, conn->resend_requests, 5, minimum_buffer_occupancy, 5, maximum_buffer_occupancy, 11, conn->remote_frame_rate, 11, From 5b8a8c879eb6cd7faa056f7573ad60da1614ea3e Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 24 May 2020 10:05:42 +0100 Subject: [PATCH 095/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 0cfb96da..df6fb29b 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -2,10 +2,12 @@ Version 3.3.7d6 ==== **Bug Fixes and Enhancements** * Fix a bug calculating the instantaneous synchronisation error. This could occasionally cause Shairport Sync to lose synchronisation and maybe even to mute for a few seconds before regaining it. It was caused by not doing modulo arithmetic properly, and it's been there for a while. -* Clean up and improve the code to synchronise the first frame of audio. This should result in more accurate and reliable initial synchrnoisation, certainly to under a millisecond in normal usage. Sync should improve even when the silent lead-in time is as short as 0.3 seconds or when the `audio_backend_latency_offset_in_seconds` is as much as -1.7 seconds. -* Removed a bug which would affect initial synchronisation if a flush command was received from the player at an inopportune time. -* Clean up some other uses of modulo arithmetic. -* Clean up allocation of memory for gathering running statistics -- use the heap instead of the stack. +* Clean up and improve the code to synchronise the first frame of audio. This should result in more accurate and reliable initial synchronisation, usually to under a millisecond, and often to within 20 or 30 microseconds in normal usage. Sync should improve even when the silent lead-in time is as short as 0.3 seconds or when the `audio_backend_latency_offset_in_seconds` is as great as -1.7 seconds. +* Remove a bug which would affect initial synchronisation if a `FLUSH` command was received from the player at an inopportune time. +* Clean up some confused uses of modulo arithmetic. +* Clean up allocation of memory for gathering running statistics – use the heap instead of the stack. +* Clean up the display of statistics for backends that do not implement active synchronisation, e.g. the `pipe` and `STDOUT` back ends. +* Add code to flush output to the `pipe` or to `STDOUT` when playing stops. Version 3.3.7d5 ==== From 5e65976eed8a861e03240717467f5c7939e92b92 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 24 May 2020 10:06:06 +0100 Subject: [PATCH 096/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index df6fb29b..4861e8fc 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,7 +1,7 @@ Version 3.3.7d6 ==== **Bug Fixes and Enhancements** -* Fix a bug calculating the instantaneous synchronisation error. This could occasionally cause Shairport Sync to lose synchronisation and maybe even to mute for a few seconds before regaining it. It was caused by not doing modulo arithmetic properly, and it's been there for a while. +* Fix a bug calculating the instantaneous synchronisation error. This bug could occasionally cause Shairport Sync to lose synchronisation and maybe even to mute for a few seconds before regaining it. It was caused by not doing modulo arithmetic properly, and it's been there for a while. * Clean up and improve the code to synchronise the first frame of audio. This should result in more accurate and reliable initial synchronisation, usually to under a millisecond, and often to within 20 or 30 microseconds in normal usage. Sync should improve even when the silent lead-in time is as short as 0.3 seconds or when the `audio_backend_latency_offset_in_seconds` is as great as -1.7 seconds. * Remove a bug which would affect initial synchronisation if a `FLUSH` command was received from the player at an inopportune time. * Clean up some confused uses of modulo arithmetic. From d8039d1d88e1f7bad374c8a71e9e4359bab463d3 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 24 May 2020 10:07:04 +0100 Subject: [PATCH 097/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 4861e8fc..6fa28671 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,13 +1,13 @@ Version 3.3.7d6 ==== **Bug Fixes and Enhancements** -* Fix a bug calculating the instantaneous synchronisation error. This bug could occasionally cause Shairport Sync to lose synchronisation and maybe even to mute for a few seconds before regaining it. It was caused by not doing modulo arithmetic properly, and it's been there for a while. -* Clean up and improve the code to synchronise the first frame of audio. This should result in more accurate and reliable initial synchronisation, usually to under a millisecond, and often to within 20 or 30 microseconds in normal usage. Sync should improve even when the silent lead-in time is as short as 0.3 seconds or when the `audio_backend_latency_offset_in_seconds` is as great as -1.7 seconds. -* Remove a bug which would affect initial synchronisation if a `FLUSH` command was received from the player at an inopportune time. -* Clean up some confused uses of modulo arithmetic. -* Clean up allocation of memory for gathering running statistics – use the heap instead of the stack. -* Clean up the display of statistics for backends that do not implement active synchronisation, e.g. the `pipe` and `STDOUT` back ends. -* Add code to flush output to the `pipe` or to `STDOUT` when playing stops. +* Fixed a bug calculating the instantaneous synchronisation error. This bug could occasionally cause Shairport Sync to lose synchronisation and maybe even to mute for a few seconds before regaining it. It was caused by not doing modulo arithmetic properly, and it's been there for a while. +* Cleaned up and improve the code to synchronise the first frame of audio. This should result in more accurate and reliable initial synchronisation, usually to under a millisecond, and often to within 20 or 30 microseconds in normal usage. Sync should improve even when the silent lead-in time is as short as 0.3 seconds or when the `audio_backend_latency_offset_in_seconds` is as great as -1.7 seconds. +* Removed a bug which would affect initial synchronisation if a `FLUSH` command was received from the player at an inopportune time. +* Cleaned up some confused uses of modulo arithmetic. +* Cleaned up allocation of memory for gathering running statistics – use the heap instead of the stack. +* Cleaned up the display of statistics for backends that do not implement active synchronisation, e.g. the `pipe` and `STDOUT` back ends. +* Added code to flush output to the `pipe` or to `STDOUT` when playing stops. Version 3.3.7d5 ==== From b0164820add2810ddd3646e9113fe13d342d7e2a Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 24 May 2020 10:18:15 +0100 Subject: [PATCH 098/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 6fa28671..9ce1afad 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -78,7 +78,7 @@ resulting in: ``` The MPRIS `trackid` corresponding to this would be `/org/gnome/ShairportSync/FD84B4B40FA33A85`. -Thanks to to Scott Simon (https://github.com/zutroy97) for finding a related bug in the [shairport-sync-metadata-reader](https://github.com/mikebrady/shairport-sync-metadata-reader). +Thanks to to [Scott Simon](https://github.com/zutroy97) for finding a related bug in the [shairport-sync-metadata-reader](https://github.com/mikebrady/shairport-sync-metadata-reader). **Other Changes** * Include the `-fno-common` flag in the compilation options to detect incorrect definitions of global variables and multiple definitions of `enum` data types. From c1f973dde7c662e9b871a69a4401c9725e578e5f Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 24 May 2020 19:47:04 +0100 Subject: [PATCH 099/186] Remove the fsync calls from stdout and pipe backends. Slight edits to man file. --- README.md | 30 +- audio_pipe.c | 4 - audio_stdout.c | 5 +- man/shairport-sync.7 | 4 +- man/shairport-sync.7.xml | 607 +++++++++---------- man/shairport-sync.html | 1219 +------------------------------------- shairport.c | 2 +- 7 files changed, 347 insertions(+), 1524 deletions(-) diff --git a/README.md b/README.md index 48bd955c..29acc4ca 100644 --- a/README.md +++ b/README.md @@ -19,7 +19,7 @@ For more about the motivation behind Shairport Sync, please see the wiki at http Synchronisation, Latency, "Stuffing" --------- -The AirPlay protocol uses an agreed *latency* – the time difference, or delay, between the time represented by a sound sample's `timestamp` and the time it is actually played by the audio output device, typically a Digital to Audio Converter (DAC). The latency to be used is specified by the audio source when it negotiates with Shairport Sync. Most sources set a latency of two seconds. Recent versions of iTunes and forkedDaapd use a latency of just over 2.25 seconds. A latency of this length allows AirPlay players to correct for network delays, processing time variations and so on. +The AirPlay protocol uses an agreed *latency* – the time difference, or delay, between the time represented by a sound sample's `timestamp` and the time it is actually played by the audio output device, typically a Digital to Audio Converter (DAC). The latency to be used is specified by the audio source when it negotiates with Shairport Sync. Most sources set a latency of two seconds. Recent versions of iTunes and forkedDaapd use a latency of just over 2.25 seconds. A latency of this length allows AirPlay players to correct for network delays, processing time variations and so on. As mentioned previously, Shairport Sync implements full audio synchronisation when used with `alsa`, `sndio` or PulseAudio systems. This is done by monitoring the timestamps present in data coming from the audio source and the timing information from the audio system, e.g. `alsa`. To maintain the latency required for exact synchronisation, if the output device is running slow relative to the source, Shairport Sync will delete frames of audio to allow the device to keep up. If the output device is running fast, Shairport Sync will insert frames to keep time. The number of frames inserted or deleted is so small as to be almost inaudible on normal audio material. Frames are inserted or deleted as necessary at pseudorandom intervals. Alternatively, with `libsoxr` support, Shairport Sync can resample the audio feed to ensure the output device can keep up. This is less obtrusive than insertion and deletion but requires a good deal of processing power — most embedded devices probably can't support it. The process of insertion/deletion or resampling is rather inelegantly called “stuffing”. @@ -42,7 +42,7 @@ What else? * An [MPRIS](https://specifications.freedesktop.org/mpris-spec/2.2/) interface, partially complete and very functional, including access to metadata and artwork, and some limited remote control. * An interface to [MQTT](https://en.wikipedia.org/wiki/MQTT), an often-used protocol in home automation projects. * A native D-Bus interface, including access to metadata and artwork, some limited remote control and some system settings. - + Heritage ------- Shairport Sync is a substantial rewrite of the fantastic work done in Shairport 1.0 by James Laird and others — please see https://github.com/abrasive/shairport/blob/master/README.md#contributors-to-version-1x for a list of the contributors to Shairport 1.x and Shairport 0.x. From a "heritage" point of view, Shairport Sync is a fork of Shairport 1.0. @@ -65,7 +65,7 @@ See [here](https://github.com/mikebrady/shairport-sync/blob/master/INSTALL.md) f Shairport Sync may already be available as a package in your Linux distribution (search for `shairport-sync` – the package named `shairport` is a different program). Packages are available on recent versions of Debian, Ubuntu, Arch, OpenWrt and possibly more: -**Ubuntu:** A `shairport-sync` installer package is available for Ubuntu. Additionally, a Personal Package Archives for Shairport Sync master and development branches are available at https://launchpad.net/~dantheperson. +**Ubuntu:** A `shairport-sync` installer package is available for Ubuntu. Additionally, a Personal Package Archives for Shairport Sync master and development branches are available at https://launchpad.net/~dantheperson. **Debian:** shairport-sync is in the Debian archive. @@ -75,7 +75,7 @@ Shairport Sync may already be available as a package in your Linux distribution To build and install the latest version of Shairport Sync, an [Arch Linux build and installation guide](https://github.com/mikebrady/shairport-sync-for-arch-linux) is available, based on original work by [Elia Cereda](https://github.com/EliaCereda). -**Mac OS X:** A HomeBrew package exists for Shairport Sync. With HomeBrew installed, Shairport Sync can be installed using the command: +**Mac OS X:** A HomeBrew package exists for Shairport Sync. With HomeBrew installed, Shairport Sync can be installed using the command: ``` $brew install shairport-sync ``` @@ -91,7 +91,7 @@ If you wish to build and install the latest version of Shairport Sync on Debian, **Remove Old Versions of Shairport Sync and its Startup Scripts** -You should check to see if `shairport-sync` is already installed – you can use the command `$ which shairport-sync` to find where it is located, if installed. If it is installed you should delete it – you may need superuser privileges. After deleting, check again in case further copies are installed elsewhere. +You should check to see if `shairport-sync` is already installed – you can use the command `$ which shairport-sync` to find where it is located, if installed. If it is installed you should delete it – you may need superuser privileges. After deleting, check again in case further copies are installed elsewhere. You should also remove the startup script files `/etc/systemd/system/shairport-sync.service` and `/etc/init.d/shairport-sync` if they exist – new ones will be installed in necessary. @@ -123,13 +123,13 @@ If PulseAudio in not installed, you'll get something like this: ``` $ pactl info -bash: pactl: command not found -$ +$ ``` If your system does not use PulseAudio, then it is likely that it uses the Advanced Linux Sound Architecture (ALSA), so you should build Shairport Sync with the ALSA backend. By the way, many systems with PulseAudio also have ALSA (in fact, PulseAudio is effectively a client of ALSA); in those cases you should choose the PulseAudio backend. If PulseAudio is not installed, there is no necessity to install it for Shairport Sync. In fact, Shairport Sync works better without it. -**Building** +**Building** To build Shairport Sync from sources on Debian, Ubuntu, Raspbian, etc. follow these instructions. @@ -179,7 +179,7 @@ $ autoreconf -i -f (Don't worry -- there's a recommended set of configuration options further down.) -- `--with-alsa` include the ALSA backend module to audio to be output through the Advanced Linux Sound Architecture (ALSA) system directly. This is recommended for highest quality. +- `--with-alsa` include the ALSA backend module to audio to be output through the Advanced Linux Sound Architecture (ALSA) system directly. This is recommended for highest quality. - `--with-pa` include the PulseAudio audio back end. This is recommended if your Linux installation already has PulseAudio installed. Although ALSA would be better, it requires direct and exclusive access to to a real (hardware) soundcard, and this is often impractical if PulseAudio is installed. - `--with-stdout` include an optional backend module to enable raw audio to be output through standard output (stdout). - `--with-pipe` include an optional backend module to enable raw audio to be output through a unix pipe. @@ -199,9 +199,9 @@ $ autoreconf -i -f **Determine if it's a `systemd` or a "System V" installation:** -If you wish to have Shairport Sync start automatically when your system boots, you need to figure out what so-called "init system" your system is using. (If you are using Shairport Sync with PulseAudio, as installed in many desktop systems, this section doesn't apply.) +If you wish to have Shairport Sync start automatically when your system boots, you need to figure out what so-called "init system" your system is using. (If you are using Shairport Sync with PulseAudio, as installed in many desktop systems, this section doesn't apply.) -There are a number of init systems in use: `systemd`, `upstart` and "System V" among others, and it's actually difficult to be certain which one your system is using. Fortunately, for Shairport Sync, all you have to do is figure out if it's a `systemd` init system or not. If it is not a `systemd` init system, you can assume that it is either a System V init system or else it is compatible with a System V init system. Recent systems tend to use `systemd`, whereas older systems use `upstart` or the earlier System V init system. +There are a number of init systems in use: `systemd`, `upstart` and "System V" among others, and it's actually difficult to be certain which one your system is using. Fortunately, for Shairport Sync, all you have to do is figure out if it's a `systemd` init system or not. If it is not a `systemd` init system, you can assume that it is either a System V init system or else it is compatible with a System V init system. Recent systems tend to use `systemd`, whereas older systems use `upstart` or the earlier System V init system. The easiest way is to look at the first few lines of the `init` manual. Enter the command: @@ -363,7 +363,7 @@ Note: Shairport Sync can take configuration settings from command line options. **Raspberry Pi** -The Raspberry Pi Models A and B have a built-in audio DAC that is connected to the device's headphone jack. Apart from a loud click when used for the first time after power-up, it is now quite adequate for casual listening. +The Raspberry Pi Models A and B have a built-in audio DAC that is connected to the device's headphone jack. Apart from a loud click when used for the first time after power-up, it is now quite adequate for casual listening. To get the benefits of improvements in the Pi's software and firmware, you should update to the Raspian release of October 2018 or later, as a number of improvements have been made to the built-in DAC. @@ -403,7 +403,7 @@ The System V init script at `/etc/init.d/shairport-sync` has a bare minimum : Examples -------- -Here are some examples of complete configuration files. +Here are some examples of complete configuration files. ``` general = { @@ -516,7 +516,7 @@ The UDP metadata format is very simple - the first four bytes are the metadata * Latency ------- -Latency is the exact time from a sound signal's original timestamp until that signal actually "appears" on the output of the audio output device, usually a Digital to Audio Converter (DAC), irrespective of any internal delays, processing times, etc. in the computer. +Latency is the exact time from a sound signal's original timestamp until that signal actually "appears" on the output of the audio output device, usually a Digital to Audio Converter (DAC), irrespective of any internal delays, processing times, etc. in the computer. Shairport Sync uses latencies supplied by the source, typically either 2 seconds or just over 2.25 seconds. You shouldn't need to change them. @@ -524,7 +524,7 @@ Problems can arise when you are trying to synchronise with speaker systems — t Resynchronisation ------------- -Shairport Sync actively maintains synchronisation with the source. +Shairport Sync actively maintains synchronisation with the source. If synchronisation is lost — say due to a busy source or a congested network — Shairport Sync will mute its output and resynchronise. The loss-of-sync threshold is a very conservative 0.050 seconds — i.e. the actual time and the expected time must differ by more than 50 ms to trigger a resynchronisation. Smaller disparities are corrected by insertions or deletions, as described above. * You can vary the resync threshold, or turn resync off completely, with the `general` `resync_threshold_in_seconds` setting. @@ -535,7 +535,7 @@ Playback synchronisation is allowed to wander — to "drift" — a small amount Some Statistics --------------- -If you turn on the `general` `statistics` setting, a heading like this will be output to the console or log file: +If you turn on the `general` `statistics` setting, a heading like this will be output to the log file (or to `STDERR` if the `-u` command line option is chosen): ``` sync error in milliseconds, net correction in ppm, corrections in ppm, total packets, missing packets, late packets, too late packets, resend requests, min DAC queue size, min buffer occupancy, max buffer occupancy, source nominal frames per second, source actual frames per second, output frames per second, source clock drift in ppm, source clock drift sample count, rough calculated correction in ppm ``` diff --git a/audio_pipe.c b/audio_pipe.c index c2e71642..2abef94b 100644 --- a/audio_pipe.c +++ b/audio_pipe.c @@ -86,10 +86,6 @@ static int play(void *buf, int samples) { static void stop(void) { // Don't close the pipe just because a play session has stopped. - // Flush it instead - if (fd > 0) - if (fsync(fd) != 0) - debug(1, "Error flushing pipe when play stopped: \"%s\".", strerror(errno)); } static int init(int argc, char **argv) { diff --git a/audio_stdout.c b/audio_stdout.c index aca660b1..8b37883e 100644 --- a/audio_stdout.c +++ b/audio_stdout.c @@ -56,10 +56,7 @@ static int play(void *buf, int samples) { } static void stop(void) { - // Flush stdout when play stops instead - if (fd > 0) - if (fsync(fd) != 0) - debug(1, "Error flushing STDOUT when play stopped: \"%s\".", strerror(errno)); + // Do nothing when play stops } static int init(__attribute__((unused)) int argc, __attribute__((unused)) char **argv) { diff --git a/man/shairport-sync.7 b/man/shairport-sync.7 index 9048b551..f85bce3a 100644 --- a/man/shairport-sync.7 +++ b/man/shairport-sync.7 @@ -326,10 +326,10 @@ The maximum packet size for any UDP metadata. This must be between 500 or 65000. \fB"DIAGNOSTICS" SETTINGS\f1 .TP \fBstatistics=\f1\fI"setting"\f1\fB;\f1 -Use this \fIsetting\f1 to enable ("yes") or disable ("no") the output of some statistical information on the console or in the log. The default is to disable statistics. +Use this \fIsetting\f1 to enable ("yes") or disable ("no") the output of some statistical information to the system log (or to \fISTDERR\f1 if the \fB-u\f1 command line option is chosen). The default is to disable statistics. .TP \fBlog_verbosity=\f1\fI0\f1\fB;\f1 -Use this to specify how much debugging information should be output or logged. The value \fI0\f1 means no debug information, \fI3\f1 means most debug information. The default is \fI0\f1. +Use this to specify how much debugging information should sent to the system log (or to \fISTDERR\f1 if the \fB-u\f1 command line option is chosen). The value \fI0\f1 means no debug information, \fI3\f1 means most debug information. The default is \fI0\f1. .SH OPTIONS This section is about the command-line options available in shairport-sync. diff --git a/man/shairport-sync.7.xml b/man/shairport-sync.7.xml index 93b24146..39311a86 100644 --- a/man/shairport-sync.7.xml +++ b/man/shairport-sync.7.xml @@ -7,7 +7,7 @@ Copyright (c) Mike Brady 2014-2019 All rights reserved. - + Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files (the "Software"), to deal in the Software without @@ -15,10 +15,10 @@ copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Software, and to permit persons to whom the Software is furnished to do so, subject to the following conditions: - + The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software. - + THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND @@ -61,61 +61,61 @@ -

Shairport Sync plays - audio streamed from iTunes - or from an AirPlay device to an ALSA-compatible audio output device (available on - Linux and FreeBSD), to a "sndio" output device (available on OpenBSD, FreeBSD and +

Shairport Sync plays + audio streamed from iTunes + or from an AirPlay device to an ALSA-compatible audio output device (available on + Linux and FreeBSD), to a "sndio" output device (available on OpenBSD, FreeBSD and Linux), to a PulseAudio output stream or to Jack Audio.

- +

Shairport Sync offers full audio synchronisation. - Full audio synchronisation means that audio is played on the output device at exactly + Full audio synchronisation means that audio is played on the output device at exactly the time specified by the audio source. This means that if many devices are playing the same stream at the same time, all the outputs will stay in synchrony with one another. - This allows multiple devices to play the same source without getting out of step with + This allows multiple devices to play the same source without getting out of step with one another, enabling, for example, simultaneous multi-room operation.

- -

Shairport Sync can stream synchronised audio to a unix - pipe or to standard output, or to audio systems that do not provide timing - information. This could perhaps be described as partial audio synchronisation, where - synchronised audio is provided by Shairport Sync, but what happens to it in the - subsequent processing chain, before it reaches the listener's ear, is outside the + +

Shairport Sync can stream synchronised audio to a unix + pipe or to standard output, or to audio systems that do not provide timing + information. This could perhaps be described as partial audio synchronisation, where + synchronised audio is provided by Shairport Sync, but what happens to it in the + subsequent processing chain, before it reaches the listener's ear, is outside the control of shairport-sync.

-

Shairport Sync can be compiled to stream metadata, including cover art, to a pipe +

Shairport Sync can be compiled to stream metadata, including cover art, to a pipe or socket.

-

Shairport Sync can be compiled to offer a standard MPRIS interface, a "native" - D-Bus interface and an MQTT client interface. Through these interfaces, it can provide +

Shairport Sync can be compiled to offer a standard MPRIS interface, a "native" + D-Bus interface and an MQTT client interface. Through these interfaces, it can provide metadata, including cover art, and can offer remote control of the audio source.

- -

Settings can be made using the configuration file (recommended for all new + +

Settings can be made using the configuration file (recommended for all new installations) or by using command-line options.

- +

The name of the Shairport Sync executable is shairport-sync. Both names are used in these man pages.

- + - - + +

You should use the configuration file for setting up Shairport Sync. This file is usually shairport-sync.conf and is generally located in the - System Configuration Directory, which is normally the /etc directory in + System Configuration Directory, which is normally the /etc directory in Linux or the /usr/local/etc directory in BSD unixes. You may need to have root privileges to modify it.

- -

(Note: Shairport Sync may have been compiled to use a different configuration - directory. You can determine which by performing the command $ shairport-sync - -V. One of the items in the output string is the value of the + +

(Note: Shairport Sync may have been compiled to use a different configuration + directory. You can determine which by performing the command $ shairport-sync + -V. One of the items in the output string is the value of the sysconfdir, i.e. the System Configuration Directory.)

- - + +

Within the configuration file, settings are organised into groups, for example, there is a "general" group of standard settings, and there is an "alsa" group with settings that pertain to the ALSA back end. Here is an example of a typical configuration file:

- +

general = {

name = "Mike's Boombox";

interpolation = "soxr";

@@ -127,20 +127,20 @@

output_device = "hw:0";

mixer_control_name = "PCM";

};

- -

Most settings have sensible default values, so -- as in the example above -- users + +

Most settings have sensible default values, so -- as in the example above -- users generally only need to set (1) the service name, (2) a password (if desired) and - (3) the output device. If the output device has a mixer that can be used for volume - control, then (4) the volume control's name should be specified. It is highly - desirable to use the output device's mixer for volume control, if available -- - response time is reduced to zero and the processor load is reduced. In the example + (3) the output device. If the output device has a mixer that can be used for volume + control, then (4) the volume control's name should be specified. It is highly + desirable to use the output device's mixer for volume control, if available -- + response time is reduced to zero and the processor load is reduced. In the example above, "soxr" interpolation was also enabled.

- -

A sample configuration file with all possible settings, but with all of them - commented out, is installed at shairport-sync.conf.sample, within the - System Configuration Directory -- /etc in Linux, + +

A sample configuration file with all possible settings, but with all of them + commented out, is installed at shairport-sync.conf.sample, within the + System Configuration Directory -- /etc in Linux, /usr/local/etc in BSD unixes.

- +

To retain backwards compatibility with previous versions of shairport-sync you can use still use command line options, but any new features, etc. will be available only via configuration file settings.

@@ -157,25 +157,25 @@

Use this service_name to identify this player in iTunes, etc.

The following substitutions are allowed: %h for the computer's hostname, - %H for the computer's hostname with the first letter capitalised (ASCII + %H for the computer's hostname with the first letter capitalised (ASCII only), %v for the shairport-sync version number, e.g. "3.0.1" and - %V for the shairport-sync version string, e.g. + %V for the shairport-sync version string, e.g. "3.0.1-OpenSSL-Avahi-ALSA-soxr-metadata-sysconfdir:/etc".

-

The default is "%H", which is replaced by the hostname with the first letter +

The default is "%H", which is replaced by the hostname with the first letter capitalised.

- + - + @@ -256,71 +256,71 @@ - + - + - - + - + - + @@ -401,25 +401,25 @@ @@ -427,7 +427,7 @@ @@ -435,207 +435,207 @@ - + - + -

These settings are for the ALSA back end, used to communicate with audio output - devices in the ALSA system. (By the way, you can use tools such as +

These settings are for the ALSA back end, used to communicate with audio output + devices in the ALSA system. (By the way, you can use tools such as alsamixer or aplay to discover what devices are available.) - Use these settings to select the output device and the mixer control to be used to + Use these settings to select the output device and the mixer control to be used to control the output volume. - You can additionally set the desired size of the output buffer and you can adjust + You can additionally set the desired size of the output buffer and you can adjust overall latency. Here are the alsa group settings:

- + - + -

These settings are for the SNDIO back end, used to communicate with audio output +

These settings are for the SNDIO back end, used to communicate with audio output devices in the SNDIO system.

@@ -712,113 +712,114 @@ -

These settings are for the PIPE backend, used to route audio to a named unix pipe. - The audio is in raw CD audio format: PCM 16 bit little endian, 44,100 samples per +

These settings are for the PIPE backend, used to route audio to a named unix pipe. + The audio is in raw CD audio format: PCM 16 bit little endian, 44,100 samples per second, interleaved stereo.

- +

There are no settings for the STDOUT backend.

- +

There are no configuration file settings for the AO backend.

-

shairport-sync can process metadata provided by the source, such as Track Number, +

shairport-sync can process metadata provided by the source, such as Track Number, Album Name, cover art, etc. and can provide additional metadata such as volume level, - pause/resume, etc. It sends the metadata to a pipe, by default + pause/resume, etc. It sends the metadata to a pipe, by default /tmp/shairport-sync-metadata. - To process metadata, shairport-sync must have been compiled with metadata support + To process metadata, shairport-sync must have been compiled with metadata support included. - You can check that this is so by running the command $ shairport-sync -V; + You can check that this is so by running the command $ shairport-sync -V; the identification string will contain the word metadata.

-

Please note that different sources provide different levels of metadata. Some +

Please note that different sources provide different levels of metadata. Some provide a lot; some provide almost none.

-

The metadata group of settings allow you to enable metadata handling and +

The metadata group of settings allow you to enable metadata handling and to control certain aspects of it:

- + - +

This section is about the command-line options available in shairport-sync.

-

Note: if you are setting up shairport-sync for the first time or are updating an - existing installation, you are encouraged to use the configuration file settings +

Note: if you are setting up shairport-sync for the first time or are updating an + existing installation, you are encouraged to use the configuration file settings described above. Most of the command-line options described below - simply replicate the configuration settings and are retained to provide backward + simply replicate the configuration settings and are retained to provide backward compatibility with older installations of shairport-sync.

- +

Many command-line options take sensible default values, so you can normally ignore most of them. See the EXAMPLES section for typical usages.

@@ -827,25 +828,25 @@ Program options are always listed first, followed by any audio backend options, preceded by a -- symbol.

- +

These command-line options are used by shairport-sync itself.

- + @@ -853,11 +854,11 @@ @@ -889,7 +890,7 @@ @@ -1010,7 +1011,7 @@ If the output timing differs from the source timing by more than the threshold, output will be muted and a full resynchronisation will occur. The default threshold is 2,205 frames, i.e. 50 - milliseconds. Specify 0 to disable resynchronisation. This setting is + milliseconds. Specify 0 to disable resynchronisation. This setting is deprecated and will be removed in a future version of shairport-sync.

@@ -1056,13 +1057,13 @@ @@ -1098,23 +1099,23 @@
-

These command-line options are passed to the chosen audio backend. The audio +

These command-line options are passed to the chosen audio backend. The audio backend options are preceded by a -- symbol to introduce them and to separate them from any program options. In this way, option letters can be used as program options and also as audio backend options without ambiguity.

- +

In the ALSA backend, audio is sent to an output device which you can specify using the -d option. - The output level (the "volume") is controlled using a level control associated with a + The output level (the "volume") is controlled using a level control associated with a mixer. By default, the mixer is implemented in shairport-sync itself in software. - To use a hardware level control on a mixer on the sound card, specify the name of the + To use a hardware level control on a mixer on the sound card, specify the name of the mixer control with the -c option. - If the mixer is not associated with the output device, then you need to specify where + If the mixer is not associated with the output device, then you need to specify where the mixer is to be found with the -m option.

- + @@ -1168,7 +1169,7 @@ -- -d hw:1,0 -m hw:1 - -c PCM + -c PCM

The program will run in daemon mode ( -d ), will be visible as "Joe's Stereo" ( -a "Joe's Stereo" ) and will use the SoX Resampler @@ -1178,17 +1179,17 @@ ( -d hw:1,0 ) and will take advantage of the same sound card's mixer ( -m hw:1 ) using the level control named "PCM" ( -c "PCM" ).

-

The example above is slightly contrived in order to show the use of the -m - option. Typically, output 0 is the default output of a card, so the output device could +

The example above is slightly contrived in order to show the use of the -m + option. Typically, output 0 is the default output of a card, so the output device could be written -d hw:1 and then the mixer option would be unnecessary, giving the following, simpler, command:

shairport-sync -d -a "Joe's Stereo" -S soxr -- -d hw:1 - -c PCM + -c PCM - +
diff --git a/man/shairport-sync.html b/man/shairport-sync.html index c16a56ea..8f0d3468 100644 --- a/man/shairport-sync.html +++ b/man/shairport-sync.html @@ -1,1199 +1,28 @@

shairport-sync

Synchronised Audio Player for iTunes / AirPlay

- - -

Synopsis

- - - shairport-sync [-djvuw] - [-a name] - [-A latency] - [-B command] - [-c configurationfile] - [-E command] - [--get-cover-art] - [--logOutputLevel] - [-L latency] - [-m backend] - [--meta-dir=directory] - [-o backend] - [--password=secret] - [-r threshold] - [--statistics] - [-S mode] - [-t timeout] - [--tolerance=frames] - [-- audio_backend_options] -
- - shairport-sync -k
- - shairport-sync -h
- - shairport-sync -V
- - - - -

Description

- -

Shairport Sync plays - audio streamed from iTunes - or from an AirPlay device to an ALSA-compatible audio output device (available on - Linux and FreeBSD), to a "sndio" output device (available on OpenBSD, FreeBSD and - Linux), to a PulseAudio output stream or to Jack Audio.

- -

Shairport Sync offers full audio synchronisation. - Full audio synchronisation means that audio is played on the output device at exactly - the time specified by the audio source. - This means that if many devices are playing the same stream at the same - time, all the outputs will stay in synchrony with one another. - This allows multiple devices to play the same source without getting out of step with - one another, enabling, for example, simultaneous multi-room operation. -

- -

Shairport Sync can stream synchronised audio to a unix - pipe or to standard output, or to audio systems that do not provide timing - information. This could perhaps be described as partial audio synchronisation, where - synchronised audio is provided by Shairport Sync, but what happens to it in the - subsequent processing chain, before it reaches the listener's ear, is outside the - control of shairport-sync.

-

Shairport Sync can be compiled to stream metadata, including cover art, to a pipe - or socket.

-

Shairport Sync can be compiled to offer a standard MPRIS interface, a "native" - D-Bus interface and an MQTT client interface. Through these interfaces, it can provide - metadata, including cover art, and can offer remote control of the audio source.

- -

Settings can be made using the configuration file (recommended for all new - installations) or by using command-line options.

- -

The name of the Shairport Sync executable is shairport-sync. - Both names are used in these man pages.

- - - - - -

Configuration File Settings

- -

You should use the configuration file for setting up Shairport Sync. - This file is usually shairport-sync.conf and is generally located in the - System Configuration Directory, which is normally the /etc directory in - Linux or the /usr/local/etc directory in BSD unixes. - You may need to have root privileges to modify it.

- -

(Note: Shairport Sync may have been compiled to use a different configuration - directory. You can determine which by performing the command $ shairport-sync - -V. One of the items in the output string is the value of the - sysconfdir, - i.e. the System Configuration Directory.)

- - -

Within the configuraton file, settings are organised into groups, for - example, there is a "general" group of - standard settings, and there is an "alsa" group with settings that pertain to the ALSA - back end. Here is an example of a typical configuration file:

- -

general = {

-

name = "Mike's Boombox";

-

interpolation = "soxr";

-

password = "secret";

-

output_backend = "alsa";

-

};

-

-

alsa = {

-

output_device = "hw:0";

-

mixer_control_name = "PCM";

-

};

- -

Most settings have sensible default values, so -- as in the example above -- users - generally only need to set (1) the service name, (2) a password (if desired) and - (3) the output device. If the output device has a mixer that can be used for volume - control, then (4) the volume control's name should be specified. It is highly - desirable to use the output device's mixer for volume control, if available -- - response time is reduced to zero and the processor load is reduced. In the example - above, "soxr" interpolation was also enabled.

- -

A sample configuration file with all possible settings, but with all of them - commented out, is installed at shairport-sync.conf.sample, within the - System Configuration Directory -- /etc in Linux, - /usr/local/etc in BSD unixes.

- -

To retain backwards compatibility with previous versions of shairport-sync - you can use still use command line options, but any new features, etc. will - be available only via configuration file settings.

- -

The configuration file is processed using the libconfig library - -- see http://www.hyperrealm.com/libconfig/libconfig_manual.html.

- -

"GENERAL" SETTINGS

-

These are the settings available within the general group:

- - -

name="service_name";

- -

Use this service_name to identify this player in iTunes, etc.

-

The following substitutions are allowed: - %h for the computer's hostname, - %H for the computer's hostname with the first letter capitalised (ASCII - only), - %v for the shairport-sync version number, e.g. "3.0.1" and - %V for the shairport-sync version string, e.g. - "3.0.1-OpenSSL-Avahi-ALSA-soxr-metadata-sysconfdir:/etc".

-

The default is "%H", which is replaced by the hostname with the first letter - capitalised.

- - - - -

password="password";

-

Require the password password to connect to the service. If you - leave this setting commented out, no password is needed.

- - - -

interpolation="mode";

-

Interpolate, or "stuff", the audio stream using the mode. - Interpolation here refers to the - process of adding or removing frames of audio to or from the - stream sent to the output device to keep it exactly in synchrony - with the player. - The default mode, "basic", is normally almost completely inaudible. - The alternative mode, "soxr", is even less obtrusive but - requires much more processing power. For this mode, support for - libsoxr, the SoX Resampler Library, must be selected when - shairport-sync is compiled. -

- - - -

output_backend="backend";

-

shairport-sync has a number of modules of code ("backends") through which - audio is output. Normally, the first audio backend that works is selected. This - setting forces the selection of the specific audio backend. Perform the - command shairport-sync -h to get a list of available audio backends -- the - default is the first on this list. Only the "alsa", "sndio" and "pa" backends support - synchronisation.

- - - -

mdns_backend="backend";

-

shairport-sync has a number of modules of code ("backends") for - interacting with the mDNS service to be used to advertise itself. Normally, the first - mDNS backend that works is selected. This setting forces the selection of the specific - mDNS backend. The default is "avahi". Perform the command - shairport-sync -h to get a list of available mDNS modules.

- - - -

port=portnumber;

-

Use this to specify the portnumber shairport-sync uses to - listen for service requests from iTunes, etc. The default is port 5000.

- - - -

udp_port_base=portnumber;

-

When shairport-sync starts to play audio, it establises three UDP - connections to the audio source. Use this setting to specify the starting - portnumber for these three ports. It will pick the first three unused ports - starting from portnumber. The default is port 6001.

- - - -

udp_port_range=range;

-

Use this in conjunction with the prevous setting to specify the - range of ports that can be checked for availability. Only three ports are - needed. - The default is 100, thus 100 ports will be checked from port 6001 upwards until three - are found.

- - - -

drift_tolerance_in_seconds=seconds;

-

Allow playback to drift up to seconds out of exact - synchronization before attempting to correct it. - The default is 0.002 seconds, i.e. 2 milliseconds. The smaller the tolerance, the more - likely it is that overcorrection will occur. - Overcorrection is when more corrections (insertions and deletions) are made than are - strictly necessary to keep the stream in sync. Use the statistics setting - to monitor correction levels. Corrections should not greatly exceed net corrections. - This setting replaces the deprecated drift setting. -

- - - -

resync_threshold_in_seconds=threshold;

-

Resynchronise if timings differ by more than threshold seconds. - If the output timing differs from the source timing by more than - the threshold, output will be muted and a full resynchronisation - will occur. The default threshold is 0.050 seconds, i.e. 50 - milliseconds. Specify 0.0 to disable resynchronisation. - This setting replaces the deprecated resync_threshold setting. -

- - - -

ignore_volume_control="choice";

-

Set this choice to "yes" if you want the volume to - be at 100% no matter what the source's volume control is set to. - This might be useful if you want to set the volume on the output device, independently - of the setting at the source. The default is "no".

- - - -

volume_range_db=dBvalue;

-

Use this dBvalue to reduce or increase the attenuation range, - in decibels, between the minimum and maximum volume.

-

For example, if a mixer has a minimum volume of -80 dB and a maximum of +20 dB, you - might wish to use only 60 dB of the 100 dB available. - This might be because the sound becomes inaudible at the lowest setting and unbearably - loud at the highest setting -- - indeed, many domestic HiFi systems have a volume control range of just 60 to 80dB.

-

Another potential use might be where the range specified by the mixer does not - match the capabilities of the device. - For example, the Raspberry Pi's DAC that feeds the built-in audio jack claims a range - of 106 dB but has a useful range of only about 30 dB. - The setting allows you to specify the maximum range from highest to lowest. - The range suggested for the Raspberry Pi's built-in audio DAC, which feeds the - headphone jack, is 30. - Using it in this case gives the volume control a much more useful range of - settings.

-

As a third example, you can actually extend the range provided by a mixer. - Many cheaper DACs have hardware mixers that offer a restricted attenuation range. - If you specify a volume range greater than the range of the mixer, software - attenuation and hardware attenuation will be combined to give the specified range.

-

If you omit this setting, the native range of the mixer is used.

- - - -

volume_max_db=dBvalue;

-

Specify the maximum output level to be used with the hardware mixer, if - used. If no hardware mixed is used, this setting specifies the maximum setting - permissible in the software mixer, which has an attenuation range from 0.0 dB down to - -96.3 dB. -

- - - -

volume_control_profile="choice";

-

Use this advanced setting to specify how the airplay volume is transferred - to the mixer volume. The "standard" profile, which is the default, makes - the volume change more quickly at lower volumes and slower at higher volumes. Choose - the "flat" profile to makes the volume change at the same rate at all - volume levels. -

- - - -

volume_range_combined_hardware_priority= - "choice";

-

Use this advanced setting to specify how to combine the hardware - attenuator with software attenuation to provide a greater attenuation range than the - hardware attenuator alone can provide. Choosing "yes" means that when - attenuation is required, the hardware attenuator will be used in preference. - If more attenuation than it can provide is needed, the hardware attenuator is set to - its greatest attenuation and software attenuation is added.

-

For example, if 40 dB of attenuation is required and the hardware attenuator - offers a maximum of 30 dB, then the hardware attenuator will be set to give 30 dB - attenuation and 10 dB of software attenuation will be added.

-

Unfortunately, certain hardware attenuators will mute at their greatest - attenuation, so can't be combined with software attenuation in this way. Choosing - "no" means that software attenuation is used to bring the remaining - attenuation required into the range offered by the hardware attenuator. - This is the default. -

- - - -

run_this_when_volume_is_set= - "/full/path/to/application/and/args";

-

Here you can specify a program and its arguments that will be run when the - volume is set or changed. Be careful to include the full path to the application. - The application must be marked as executable and, if it is a script, its first line - must begin with the standard shebang #!/bin/... as appropriate.

-

The desired AirPlay volume is appended to the end of the command line -- leave a - space at the end of the command line you specify here if you want it treated as an - extra argument. - AirPlay volume goes from 0.0 to -30.0 and -144.0 means "mute".

- - - -

regtype="regTypeString";

-

Use this advanced setting to set the service type and transport to be - advertised by Zeroconf/Bonjour. Default is "_raop._tcp".

- - - -

playback_mode="mode";

-

The mode can be "stereo", "mono", "reverse stereo", "both left" - or "both right". Default is "stereo". Note that dither will be added to the signal in - the mono mode.

- - - -

alac_decoder="decodername";

-

This can be "hammerton" or "apple". This advanced setting allows you to - choose the original Shairport decoder by David Hammerton or the Apple Lossless Audio - Codec (ALAC) decoder written by Apple. Shairport Sync must have been compiled with the - configuration setting "--with-apple-alac" and the Apple ALAC decoder library must be - present for this to work.

- - - -

interface="name";

-

Use this advanced setting if you want to confine Shairport Sync to the - named interface. Leave it commented out to get the default bahaviour.

- - - -

audio_backend_latency_offset_in_seconds= - offset_in_seconds;

-

Set this offset_in_seconds to compensate for a fixed delay in - the audio back end. For example, if the output device delays by 100 ms, set this to - -0.1.

- - - -

audio_backend_buffer_desired_length_in_seconds= - length_in_seconds;

-

Use this length_in_seconds to set the desired length of the - queue of audio frames in the backend's output buffer.

-

The default is 0.15 seconds for the ALSA backend, 0.35 seconds for the PA backend - and one second for all other backends.

-

If this value is set too small, underflow may occur on low-powered machines. - If set too large, the response times to the volume control may become excessive, or it - may exceed the backend's buffer size. - It may need to be larger on low-powered machines that are also performing other tasks, - such as processing metadata.

- - - -

audio_backend_buffer_interpolation_threshold_in_seconds= - time_in_seconds;

-

This is an advanced feature. If the length of the audio backend buffer - size drops below this, it's a sign that shairport sync can not process frames of audio - quickly enough. It this threshold is reached, shairport sync will stop using - time-consuming interpolation like soxr to avoid underruns.

- - - -

audio_backend_silent_lead_in_time= - lead_in_time_in_seconds;

-

This is an advanced setting. Use the lead_in_time_in_seconds to - set the desired length of the period of silence (a "silent lead-in") played before a - play session begins.

-

The purpose of this silent lead-in is to give the backend sufficient time to - prepare for operation and to make an estimate (and, importantly, to correct the - estimate) of the exact time at which to begin playing audio to achieve initial - synchronisation. The value can be from 0.0 up to a maximum of either 4.0 seconds. The - actual duration will be close to the setting but can not exceed the latency set by the - client, usually 2 seconds or a little more.

-

If the value chosen is too short for synchronised backends such as the ALSA, sndio - or PA backends, then audio will not be synchronised correctly at the start of play. - The default is to have a silent lead-in of approximately the same time as the latency - set by the client.

- - - -

dbus_service_bus= - "bus_name";

-

If shairport sync is compiled with the D-Bus interface, it can offer it on - the "system" or the "session" D-Bus "bus". - Use this to specify which. The default is to use the "system" bus.

- - - -

mpris_service_bus= - "bus_name";

-

If shairport sync is compiled with the MPRIS interface, it can offer the - service on the "system" or the "session" D-Bus "bus". - Use this to specify which. The default is to use the "system" bus.

- - -

"SESSIONCONTROL" SETTINGS

- - -

run_this_before_play_begins="/path/to/application and - args";

-

Here you can specify a program and its arguments that will be run just - before a play session begins. Be careful to include the full path to the application. - The application must be marked as executable and, if it is a script, its first line - must begin with the standard shebang #!/bin/... as - appropriate.

- - - -

run_this_after_play_ends="/path/to/application and - args";

-

Here you can specify a program and its arguments that will be run just - after a play session ends. Be careful to include the full path to the application. - The application must be marked as executable and, if it is a script, its first line - must begin with the standard shebang #!/bin/... as - appropriate.

- - - -

run_this_before_entering_active_state="/path/to/application and - args";

-

Here you can specify a program and its arguments that will be run just - before shairport-sync goes active.

- -

Shairport Sync goes "active" when a play session starts. When the play - session ends, the system will stay active until the time - specified in the active_state_timeout setting elapses. - If a new play session starts before that, the system will remain active. Otherwise, - the system will go inactive. -

- -

Be careful to include the full path to the application. - The application must be marked as executable and, if it is a script, its first line - must begin with the standard shebang #!/bin/... as - appropriate.

- - - -

run_this_after_exiting_active_state="/path/to/application and - args";

-

Here you can specify a program and its arguments that will be run just - after shairport-sync goes inactive (see the previous entry for an explanation - of the idea). - Be careful to include the full path to the application. - The application must be marked as executable and, if it is a script, its first line - must begin with the standard shebang #!/bin/... as - appropriate.

- - - -

active_state_timeout=seconds;

-

After a play session has ended, the system will remain active for - seconds seconds. If a new play session starts before this time has elapsed, - the system will remain active. However, if no new session starts in the interval, the - system will go inactive at the end of it. The default is 10 seconds.

- - - -

run_this_if_an_unfixable_error_is_detected="/path/to/application - and args";

-

Here you can specify a program and its arguments that will be run if the - system detects an unfixable error. At present, there are two types of - unfixable errors. One is where a play session cannot be terminated. - The second is if an output device has "stalled" -- that is, if an output device - refuses to accept any more output frames.

-

Although the first problem could, in principle, be fixed by restarting - Shairport Sync, it is usually caused by a malfunctioning output device. - Typically, the most reliable way to recover from either of these errors - is to reboot the entire machine.

-

Be careful to include the full path to the application. - The application must be marked as executable and, if it is a script, its first line - must begin with the standard shebang #!/bin/... as - appropriate.

- - - -

wait_for_completion="choice";

-

Set choice to "yes" to make shairport-sync wait until the - programs specified in the run_this_... settings have - completed execution before continuing. The default is "no".

- - - -

allow_session_interruption="choice";

-

If choice is set to "yes", then another source will be able to - interrupt an existing play session and start a new one. - When set to "no" (the default), other devices attempting to interrupt a session will - fail, receiving a busy signal.

- - - -

session_timeout=seconds;

-

If a play session has been established and the source disappears without - warning (such as a device going out of range of a network) - then wait for the number of seconds specified before ending the session. - Once the session has terminated, other devices can use it. The default is 120 - seconds.

- - - -

"ALSA" SETTINGS

-

These settings are for the ALSA back end, used to communicate with audio output - devices in the ALSA system. (By the way, you can use tools such as - alsamixer or aplay to discover what devices are available.) - Use these settings to select the output device and the mixer control to be used to - control the output volume. - You can additionally set the desired size of the output buffer and you can adjust - overall latency. Here are the alsa group settings:

- - -

output_device="output_device";

-

Use the output device called output_device. The default is the - device called "default".

- - - -

mixer_control_name="name";

-

Specify the name of the mixer control to be used by - shairport-sync to control the volume. - The mixer control must be on the mixer device, which by default is the output device. - If you do not specify a mixer control name, shairport-sync will adjust the volume in - software.

- - - -

mixer_device="mixer_device";

-

By default, the mixer is assumed to be output_device. Use this setting to - specify a device other than the output device.

- - - -

output_rate=frame rate;

-

Use this setting to specify the frame rate to output to the ALSA device. - Allowable values are 44100 (default), 88200, 176400 and 352800. The device must have - the capability to accept the format you specify. There is no particular reason to use - anything other than 44100 if it is available. -

- - - -

output_format="format";

-

Use this setting to specify the format that should be used to send data to - the ALSA device. Allowable values are "U8", "S8", "S16", "S24", "S24_3LE", "S24_3BE" - or "S32". The device must have the capability to accept the format you - specify.

"S" means signed; "U" means unsigned; BE means big-endian and LE means - little-endian. Except where stated (using *LE or *BE), endianness matches that of the - processor. The default is "S16".

If you are using a hardware mixer, the best - setting is S16, as audio will pass through Shairport Sync unmodifed except for - interpolation. If you are using the software mixer, use 32- or 24-bit, if your device - is capable of it, to get the lowest possible levels of dither. -

- - - -

disable_synchronization="no";

-

This is an advanced setting and is for debugging only. Set to - "yes" to disable synchronization. Default is "no". - If you use it to disable synchronisation, then sooner or later you'll experience audio - glitches due to audio buffer overflow or underflow. -

- - - -

period_size=number;

-

Use this optional advanced setting to set the alsa period size near to - this value.

- - - -

buffer_size=number;

-

Use this optional advanced setting to set the alsa buffer size near to - this value.

- - - -

use_mmap_if_available="yes";

-

Use this optional advanced setting to control whether MMAP-based output - is used to communicate with the DAC. Default is "yes".

- - - -

mute_using_playback_switch="no";

- -

This is an advanced setting and the default is "no". If it is set to - "yes", hardware mute will be used where it is available. - Set it to "no" to prevent the hardware mute being used.

-

If Shairport Sync is sharing the output device with other applications, it is best - to leave this set to "no" for compatibility with those applications.

-

Another motivation for this is to allow the ALSA function call - "snd_mixer_selem_set_playback_switch_all" to be avoided. It is incorrectly implemented - on certain soundcards, including the emulated card in VMWare Fusion 8.5.

- - - - -

maximum_stall_time=seconds;

-

If an output device fails to accept any audio frames for more than the - time, in seconds, specified here (0.2 seconds by default), - it is considered to have malfunctioned. It will result in the - run_this_if_an_unfixable_error_is_detected program, - if any, being called.

-

Implemented for the ALSA back end only.

- - - - -

disable_standby_mode="never";

- -

Shairport Sync has a "Disable Standby" feature to eliminate certain - faint-but-annoying audible pops and clicks. When activsted, it prevents - an output device from entering standby mode and thus it minimises standby/busy - transitions, which can sometimes be heard. Use this setting to control when the - Disable Standby feature is active: "never" means it will never be activated, "always" - means it will be active as soon as shairport-sync starts running, and "auto" - means it will be active while shairport-sync is in the "active" state.

-

Shairport Sync goes "active" when a play session starts. When the play - session ends, the system will stay active until the time - specified in the active_state_timeout setting elapses. - If a new play session starts before that, the system will remain active. Otherwise, - the system will go inactive. -

- - - -

"SNDIO" SETTINGS

-

These settings are for the SNDIO back end, used to communicate with audio output - devices in the SNDIO system.

- - -

device="snd/0";

-

Use this optional setting to specify the name of the output device, e.g. - "snd/0". The default is to use the SNDIO system's default.

- - - -

rate=44100;

-

Use this optional setting to specify the output rate in frames per second. - Valid rates are 44100, 88200, 176400 or 352800. - The output device must have the capability to accept data at the specified rate. The - default is 44100.

- - - -

format="S16";

-

Use this optional setting to specify the output format. Allowable values - are "U8", "S8", "S16", "S24", "S24_3LE", "S24_3BE" or "S32". - The device must have the capability to accept the format you specify.

"S" means - signed; "U" means unsigned; BE means big-endian and LE means little-endian. - Except where stated (using *LE or *BE), endianness matches that of the processor. The - default is "S16".

- Since the SNDIO backend does not use a hardware mixer for volume control, dither will - be introduced into the output if it is less than full volume. - Thus, (unless you are ignoring the volume control setting), - consider using 32- or 24-bit output if your device is capable of it, to get the lowest - possible levels of dither.

-

Please note that 32- or 24-bit has not been extensively tested on - SNDIO.

- - - -

round=value;

-

Use this optional advanced setting to specify the period size of the SNDIO - channel. If omitted, a SNDIO system default value will be used.

- - - -

bufsiz=value;

-

Use this optional advanced setting to specify the buffer size of the SNDIO - channel. If omitted, a SNDIO system default value will be used.

- - -

"PA" SETTINGS

-

These settings are for the new PulseAudio backend.

- - -

application_name="Shairport Sync";

-

Use this to set the name to appear in the Sounds "Applications" tab when - Shairport Sync is active. The default is the name "Shairport Sync".

- - -

"PIPE" SETTINGS

-

These settings are for the PIPE backend, used to route audio to a named unix pipe. - The audio is in raw CD audio format: PCM 16 bit little endian, 44,100 samples per - second, interleaved stereo.

- - -

name="/path/to/pipe";

-

Use this to specify the name and location of the pipe. The pipe will be - created and opened when shairport-sync starts up and will be closed upon shutdown. - Frames of audio will be sent to the pipe in packets of 352 frames and will be - discarded if the pipe has not have a reader attached. - The sender will wait for up to five seconds for a packet to be written before - discarding it.

- - -

"STDOUT" SETTINGS

-

There are no settings for the STDOUT backend.

- -

"AO" SETTINGS

-

There are no configuration file settings for the AO backend.

- -

"METADATA" SETTINGS

-

shairport-sync can process metadata provided by the source, such as Track Number, - Album Name, cover art, etc. and can provide additional metadata such as volume level, - pause/resume, etc. It sends the metadata to a pipe, by default - /tmp/shairport-sync-metadata. - To process metadata, shairport-sync must have been compiled with metadata support - included. - You can check that this is so by running the command $ shairport-sync -V; - the identification string will contain the word metadata.

-

Please note that different sources provide different levels of metadata. Some - provide a lot; some provide almost none.

-

The metadata group of settings allow you to enable metadata handling and - to control certain aspects of it:

- - -

enabled="choice";

-

Set the choice to "yes" to enable shairport-sync to look for - metadata from the audio source and to forward it, along with metadata generated by - shairport-sync itself, to the metadata pipe. The default is "no".

- - - -

include_cover_art="choice";

-

Set the choice to "yes" to enable shairport-sync to look for - cover art from the audio source and to include it in the feed to the metadata pipe. - You must also enable metadata (see above). - One reason for not including cover art is that the images can sometimes be very large - and may delay transmission of subsequent metadata through the pipe. - The default is "no".

- - - -

pipe_name="filepathname";

-

Specify the absolute path name of the pipe through which metadata should - be sent The default is /tmp/shairport-sync-metadata.

- - - -

socket_address="hostnameOrIP";

-

If hostnameOrIP is set to a host name or and IP address, UDP - packets containing metadata will be sent to this address. - May be a multicast address. Additionally, socket-port must be non-zero and - enabled must be set to "yes".

- - - -

socket_port=port;

-

If socket_address is set, use port to specify the - port to send UDP packets to. Must not be zero.

- - - -

socket_msglength=65000;

-

The maximum packet size for any UDP metadata. This must be between 500 or - 65000. The default is 500.

- - -

"DIAGNOSTICS" SETTINGS

- -

statistics="setting";

-

Use this setting to enable ("yes") or disable ("no") the output - of some statistical information on the console or in the log. The default is to - disable statistics.

- - - -

log_verbosity=0;

-

Use this to specify how much debugging information should be output or - logged. The value 0 means no debug information, 3 means most - debug information. The default is 0.

- - - - - -

Options

- -

This section is about the command-line options available in shairport-sync.

-

Note: if you are setting up shairport-sync for the first time or are updating an - existing installation, you are encouraged to use the configuration file settings - described above. Most of the command-line options described below - simply replicate the configuration settings and are retained to provide backward - compatibility with older installations of shairport-sync.

- -

Many command-line options take sensible default values, so you can normally - ignore most of them. See the EXAMPLES section for typical usages.

- -

There are two kinds of command-line options for shairport-sync: - regular program options and audio backend options. - Program options are - always listed first, followed by any audio backend options, preceded by - a -- symbol.

- -

Program Options

- -

These command-line options are used by shairport-sync itself.

- - - - -

-a service name | --name=service - name

-

- Use this service name to identify this player in iTunes, etc.

- -

The following substitutions are allowed: - %h for the computer's hostname, - %H for the computer's hostname with the first letter capitalised (ASCII - only), - %v for the shairport-sync version number, e.g. "3.0.1" and - %V for the shairport-sync version string, e.g. - "3.0.1-OpenSSL-Avahi-ALSA-soxr-metadata-sysconfdir:/etc".

-

The default is "%H", which is replaced by the hostname with the first letter - capitalised.

- - - - -

-B program | --on-start=program

-

- Execute program when playback is about to begin. Specify the - full path to the program, e.g. /usr/bin/logger. - Executable scripts can be used, but they must have the appropriate shebang - (#!/bin/sh in the headline.

- -

If you want shairport-sync to wait until the command has - completed before starting to play, select the -w option as well. -

- - - -

-c filename | --configfile=filename

-

- Read configuration settings from filename. The default is to read them from - the shairport-sync.conf in the System Configuration Directory -- - /etc in Linux, /usr/local/etc in BSD unixes. - For information about configuration settings, see the "Configuration File Settings" - section above. -

- - - -

-d | --daemon

-

- Instruct shairport-sync to demonise itself. It will write its - Process ID (PID) to a file, usually at - /var/run/shairport-sync/shairport-sync.pid, which is used by the - -k, -D and -R options to locate - the daemon at a later time. See also the -j option. Only available if - shaiport-sync has been compiled with libdaemon support. -

- - - -

-E program | --on-stop=program

-

- Execute program when playback has ended. Specify the - full path to the program, e.g. /usr/bin/logger. - Executable scripts can be used, but they must have the appropriate shebang - (#!/bin/sh in the headline.

-

If you want shairport-sync to wait until the command has - completed before continuing, select the -w option as well. -

- - - -

--get-coverart

-

- This option requires the --meta-dir option to be set, and enables - shairport-sync to request cover art from the source and to transmit it through - the metadata pipe.

-

Please note that cover art data may be very large, and may place too great a - burden on your network. -

- - - -

-h | --help

-

- Print brief help message and exit. -

- - - -

-j

-

- Instruct shairport-sync to demonise itself. Unlike the -d option, it will - not write a Process ID (PID) to a file -- it will just (hence the "j") demonise - itself. Only available if shaiport-sync has been compiled with libdaemon support. -

- - - -

-k | --kill

-

- Kill the shairport-sync daemon and exit. (Requires that the daemon has - written its PID to an agreed file -- see the -d option. Only available if - shaiport-sync has been compiled with libdaemon support.) -

- - - -

--logOutputLevel

-

- Use this to log the volume level when the volume is changed. It may be useful if you - are trying to determine a suitable value for the maximum volume level. Not available - as a configuration file setting. -

- - - - -

-L | --latency=latency

-

- Use this to set the default latency, in frames, for audio coming from an - unidentified source or from an iTunes Version 9 or earlier source. The standard value - for the default latency is 88,200 frames, where there are 44,100 - frames to the second. -

-

Please note that this feature is deprecated and will be removed in a future version - of shairport-sync.

- - - - -

--meta-dir=directory

-

- Listen for metadata coming from the source and send it, along with metadata from - shairport-sync itself, to a pipe called shairport-sync-metadata - in the directory you specify. If you add the --get-cover-art - then cover art will be sent through the pipe too. See https://github.com/mikebrady/shairport-sync-metadata-reader - for a sample metadata reader. -

- - - -

-m mdnsbackend | --mdns=mdnsbackend

-

- Force the use of the specified mDNS backend to advertise the - player on the network. The default is to try all mDNS backends until one - works. -

- - - -

-o outputbackend | - --output=outputbackend

-

- Force the use of the specified output backend to play the audio. - The default is to try the first one. -

- - - -

-p port | --port=port

-

- Listen for play requests on port. The default is to use port - 5000. -

- - - -

--password=secret

-

- Require the password secret to be able to connect and stream to the - service. -

- - - -

-r threshold | --resync=threshold

-

- Resynchronise if timings differ by more than threshold frames. - If the output timing differs from the source timing by more than - the threshold, output will be muted and a full resynchronisation - will occur. The default threshold is 2,205 frames, i.e. 50 - milliseconds. Specify 0 to disable resynchronisation. This setting is - deprecated and will be removed in a future version of shairport-sync. -

- - - -

--statistics

-

- Print some statistics in the standard output, or in the logfile if in daemon mode. -

- - - -

-S mode | --stuffing=mode

-

- Stuff the audio stream using the mode. "Stuffing" refers to the - process of adding or removing frames of audio to or from the - stream sent to the output device to keep it exactly in synchrony - with the player. - The default mode, basic, is normally almost completely inaudible. - The alternative mode, soxr, is even less obtrusive but - requires much more processing power. For this mode, support for - libsoxr, the SoX Resampler Library, must be selected when - shairport-sync is compiled. -

- - - -

-t timeout | --timeout=timeout

-

- Exit play mode if the stream disappears for more than timeout - seconds.

-

When shairport-sync plays an audio stream, it starts a play - session and will return a busy signal to any other sources that - attempt to use it. If the audio stream disappears for longer - than timeout seconds, the play session will be terminated. - If you specify a timeout time of 0, - shairport-sync will never signal that - it is busy and will not prevent other sources from "barging in" - on an existing play session. The default value is 120 seconds. -

- - - -

--tolerance=frames

-

- Allow playback to be up to frames out of exact synchronization before - attempting to correct it. - The default is 88 frames, i.e. 2 ms. The smaller the tolerance, the more likely it is - that overcorrection will occur. - Overcorrection is when more corrections (insertions and deletions) are made than are - strictly necessary to keep the stream in sync. Use the --statistics option - to monitor correction levels. Corrections should not greatly exceed net corrections. - This setting is deprecated and will be removed in a future version of shairport-sync. -

- - - -

-u

-

- If you are running shairport-sync from the command line and want logs to appear there, - use this option. Otherwise, logs may go to the system log. -

- - - -

-V | --version

-

- Print version information and exit. -

- - - -

-v | --verbose

-

- Print debug information. Repeat up to three times to get more detail. -

- - - -

-w | --wait-cmd

-

- Wait for commands specified using -B or -E to complete before - continuing execution. -

- - -

Audio Backend Options

- -

These command-line options are passed to the chosen audio backend. The audio - backend options are - preceded by a -- symbol to introduce them and to separate them from any - program options. In this way, option letters can be used as program - options and also as audio backend options without ambiguity.

- -

In the ALSA backend, audio is sent to an output device - which you can specify using the -d option. - The output level (the "volume") is controlled using a level control associated with a - mixer. - By default, the mixer is implemented in shairport-sync itself in software. - To use a hardware level control on a mixer on the sound card, specify the name of the - mixer control with the -c option. - If the mixer is not associated with the output device, then you need to specify where - the mixer is to be found with the -m option.

- - - - -

-c controlname

-

- Use the level control called controlname on the hardware mixer for - controlling volume. - This is needed if the mixer type is specified, using the -t option, - to be hardware. There is no default. -

- - - -

-d device

-

- Use the specified output device. You may specify a card, e.g. - hw:0, in which case the default output device on the card will be chosen. - Alternatively, you can specify a specific device on a card, e.g. hw:0,0. - The default is the device named default. -

- - - -

-m mixer

-

- Use the specified hardware mixer for volume control. Use this to specify - where the mixer is to be found. For example, if the mixer is associated with a card, - as is often the case, specify the card, e.g. hw:0. - If (unusually) the mixer is associated with a specific device on a card, - specify the device, e.g. hw:0,1. - The default is the device named in the -d option, - if given, or the device named default. -

- - - -

-t devicetype

- -

- This option is deprecated and is ignored. For your information, its functionality has - been automatically incorporated in the -c option - -- if you specify a mixer name with the -c option, it is assumed that the mixer is - implemented in hardware. -

- - - -

Examples

- -

Here is a slightly contrived example:

- shairport-sync -d - -a "Joe's Stereo" - -S soxr - -- - -d hw:1,0 - -m hw:1 - -c PCM -
- -

The program will run in daemon mode ( -d ), will be visible as - "Joe's Stereo" ( -a "Joe's Stereo" ) and will use the SoX Resampler - Library-based stuffing ( -S soxr ). - The audio backend options following the -- separator specify - that the audio will be output on output 0 of soundcard 1 - ( -d hw:1,0 ) and will take advantage of the same sound card's mixer - ( -m hw:1 ) using the level control named "PCM" ( -c "PCM" ). -

-

The example above is slightly contrived in order to show the use of the -m - option. Typically, output 0 is the default output of a card, so the output device could - be written -d hw:1 and then the mixer option would be unnecessary, giving the following, simpler, command:

- shairport-sync -d - -a "Joe's Stereo" - -S soxr - -- - -d hw:1 - -c PCM -
- - - - - -

Credits

- -

Mike Brady developed shairport-sync from the original Shairport by James Laird.

-

shairport-sync can be found at - https://github.com/mikebrady/shairport-sync.

-

Shairport can be found at - https://github.com/abrasive/shairport.

- - - -

Comments

- -

This man page was written using xml2man by Oliver Kurth.

- - - -
+

Synopsis

+
+
+
+
+ +

Description

+

+

Configuration File Settings

+

http://www.hyperrealm.com/libconfig/libconfig_manual.html

+

Options

+

Program Options

+

+

https://github.com/mikebrady/shairport-sync-metadata-reader

Audio Backend Options

+

+

Examples

+


+


+ +

Credits

+

https://github.com/mikebrady/shairport-sync.

https://github.com/abrasive/shairport.

+

Comments

+

xml2man

+ diff --git a/shairport.c b/shairport.c index e9cd70d3..510413e4 100644 --- a/shairport.c +++ b/shairport.c @@ -268,7 +268,7 @@ void usage(char *progname) { printf(" The default is /tmp/shairport-sync-metadata.\n"); printf(" --get-coverart send cover art through the metadata pipe.\n"); #endif - printf(" -u, --use-stderr log messages through STDERR rather than syslog.\n"); + printf(" -u, --use-stderr log messages through STDERR rather than the system log.\n"); printf("\n"); mdns_ls_backends(); printf("\n"); From 58ae9454c5abfe751c8b460f181c22422a6cd090 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 25 May 2020 08:44:16 +0100 Subject: [PATCH 100/186] Slight edits to the man file. --- man/shairport-sync.7 | 2 +- man/shairport-sync.7.xml | 3 ++- man/shairport-sync.html | 2 +- 3 files changed, 4 insertions(+), 3 deletions(-) diff --git a/man/shairport-sync.7 b/man/shairport-sync.7 index f85bce3a..f37d85a2 100644 --- a/man/shairport-sync.7 +++ b/man/shairport-sync.7 @@ -425,7 +425,7 @@ If you are running shairport-sync from the command line and want logs to appear Print version information and exit. .TP \fB-v | --verbose\f1 -Print debug information. Repeat up to three times to get more detail. +Print debug information to the system log, or or to \fISTDERR\f1 if the \fB-u\f1 command line option is also chosen. Repeat up to three times to get more detail. .TP \fB-w | --wait-cmd\f1 Wait for commands specified using \fB-B\f1 or \fB-E\f1 to complete before continuing execution. diff --git a/man/shairport-sync.7.xml b/man/shairport-sync.7.xml index 39311a86..b86bdbad 100644 --- a/man/shairport-sync.7.xml +++ b/man/shairport-sync.7.xml @@ -1086,7 +1086,8 @@ diff --git a/man/shairport-sync.html b/man/shairport-sync.html index 8f0d3468..94723e9b 100644 --- a/man/shairport-sync.html +++ b/man/shairport-sync.html @@ -14,7 +14,7 @@

Options

Program Options

-

https://github.com/mikebrady/shairport-sync-metadata-reader

Audio Backend Options

+

https://github.com/mikebrady/shairport-sync-metadata-reader

Audio Backend Options

Examples


From e8ef0e937111f38a726305322662c284207676f2 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 25 May 2020 12:48:39 +0100 Subject: [PATCH 101/186] rewrite the non_blocking_write to not use poll() and to return immediately if there is no reader at the end of the pipe. --- common.c | 45 ++++++++++++++++++++++++++++++++++++++++++--- common.h | 3 --- shairport.c | 3 +++ 3 files changed, 45 insertions(+), 6 deletions(-) diff --git a/common.c b/common.c index 4aba7e1a..f99c7752 100644 --- a/common.c +++ b/common.c @@ -1049,7 +1049,45 @@ uint64_t get_absolute_time_in_ns() { return time_now_ns; } +ssize_t non_blocking_write_with_timeout(int fd, const void *buf, size_t count, uint64_t timeout) { + // try to write to this non-blocking file descriptor + // wait for up to the timeout for it to be possible to write something. + // exit immediately if there is no connection at the other end + // timeout is in nanoseconds + void *ibuf = (void *)buf; + size_t bytes_remaining = count; + ssize_t rc = 0; + uint64_t end_time = get_absolute_time_in_ns() + timeout; + int timed_out = 0; + while ((bytes_remaining) && (rc == 0)) { + rc = write(fd, ibuf, bytes_remaining); + if (rc >= 0) { + bytes_remaining = bytes_remaining - rc; + ibuf += rc; + rc = 0; // not an error + } else if (errno == EAGAIN) { + // delay for a little while if it couldn't get everything out... + uint64_t time_remaining = end_time - get_absolute_time_in_ns(); + useconds_t t = time_remaining / 1000; // ns to us + if (t > 50000) + t = 50000; // wait for 50 milliseconds at most + if (t != 0) { + rc = 0; // not an error + usleep(t); + } + } else { + debug(2,"Error %d in non_blocking_write: \"%s\".",errno,strerror(errno)); + } + } + if (rc > 0) + return count - bytes_remaining; // this is just to mimic a normal write/3. + else + return rc; +} + +/* ssize_t non_blocking_write_with_timeout(int fd, const void *buf, size_t count, int timeout) { + // // timeout is in milliseconds void *ibuf = (void *)buf; size_t bytes_remaining = count; @@ -1083,9 +1121,10 @@ ssize_t non_blocking_write_with_timeout(int fd, const void *buf, size_t count, i return rc; // return write(fd,buf,count); } +*/ ssize_t non_blocking_write(int fd, const void *buf, size_t count) { - return non_blocking_write_with_timeout(fd, buf, count, 5000); // default is 5 seconds. + return non_blocking_write_with_timeout(fd, buf, count, 500000000); // default is 0.5 seconds. } /* from @@ -1239,9 +1278,9 @@ int sps_pthread_mutex_timedlock(pthread_mutex_t *mutex, useconds_t dally_time, timeoutTime.tv_sec = time_then; timeoutTime.tv_nsec = time_then_nsec; - int64_t start_time = get_absolute_time_in_ns(); + uint64_t start_time = get_absolute_time_in_ns(); int r = pthread_mutex_timedlock(mutex, &timeoutTime); - int64_t et = get_absolute_time_in_ns() - start_time; + uint64_t et = get_absolute_time_in_ns() - start_time; if ((debuglevel != 0) && (r != 0) && (debugmessage != NULL)) { char errstr[1000]; diff --git a/common.h b/common.h index e7efc72e..e07cb7c9 100644 --- a/common.h +++ b/common.h @@ -303,9 +303,6 @@ int get_requested_connection_state_to_output(); void set_requested_connection_state_to_output(int v); -ssize_t non_blocking_write_with_timeout(int fd, const void *buf, size_t count, - int timeout); // timeout in milliseconds - ssize_t non_blocking_write(int fd, const void *buf, size_t count); // used in a few places /* from diff --git a/shairport.c b/shairport.c index 510413e4..0ad40ad8 100644 --- a/shairport.c +++ b/shairport.c @@ -1655,6 +1655,9 @@ int main(int argc, char **argv) { #endif debug(1, "Started!"); + // stop a pipe signal from killing the program + signal(SIGPIPE, SIG_IGN); + // install a zombie process reaper // see: http://www.microhowto.info/howto/reap_zombie_processes_using_a_sigchld_handler.html struct sigaction sa; From 76c4947d640bf549054adf72435be57719440da4 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 25 May 2020 12:50:48 +0100 Subject: [PATCH 102/186] rewrite the non_blocking_write to not use poll() and to return immediately if there is no reader at the end of the pipe. --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 026345ef..a28dd4db 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.7d6], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.7d7], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From ba4b6b66c6c2090c973096854e89611cac559f29 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 25 May 2020 14:56:33 +0100 Subject: [PATCH 103/186] Brown bag --- common.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/common.c b/common.c index f99c7752..43bb41af 100644 --- a/common.c +++ b/common.c @@ -1058,7 +1058,6 @@ ssize_t non_blocking_write_with_timeout(int fd, const void *buf, size_t count, u size_t bytes_remaining = count; ssize_t rc = 0; uint64_t end_time = get_absolute_time_in_ns() + timeout; - int timed_out = 0; while ((bytes_remaining) && (rc == 0)) { rc = write(fd, ibuf, bytes_remaining); if (rc >= 0) { @@ -1075,8 +1074,6 @@ ssize_t non_blocking_write_with_timeout(int fd, const void *buf, size_t count, u rc = 0; // not an error usleep(t); } - } else { - debug(2,"Error %d in non_blocking_write: \"%s\".",errno,strerror(errno)); } } if (rc > 0) @@ -1124,7 +1121,8 @@ ssize_t non_blocking_write_with_timeout(int fd, const void *buf, size_t count, i */ ssize_t non_blocking_write(int fd, const void *buf, size_t count) { - return non_blocking_write_with_timeout(fd, buf, count, 500000000); // default is 0.5 seconds. + // if it would block, wait up to 0.5 seconds for it to unblock. + return non_blocking_write_with_timeout(fd, buf, count, 500000000); } /* from From e0faaeaefac621e82e64b179b0f586e1629af0ad Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 25 May 2020 15:30:07 +0100 Subject: [PATCH 104/186] Edit some of the man page --- man/shairport-sync.7 | 14 ++++++-------- man/shairport-sync.7.xml | 40 +++++++++++++++++++++++----------------- man/shairport-sync.html | 2 +- 3 files changed, 30 insertions(+), 26 deletions(-) diff --git a/man/shairport-sync.7 b/man/shairport-sync.7 index f37d85a2..7446662c 100644 --- a/man/shairport-sync.7 +++ b/man/shairport-sync.7 @@ -35,8 +35,6 @@ Within the configuration file, settings are organised into groups, for example, \fBname = "Mike's Boombox";\f1 -\fBinterpolation = "soxr";\f1 - \fBpassword = "secret";\f1 \fBoutput_backend = "alsa";\f1 @@ -53,7 +51,7 @@ Within the configuration file, settings are organised into groups, for example, \fB};\f1 -Most settings have sensible default values, so -- as in the example above -- users generally only need to set (1) the service name, (2) a password (if desired) and (3) the output device. If the output device has a mixer that can be used for volume control, then (4) the volume control's name should be specified. It is highly desirable to use the output device's mixer for volume control, if available -- response time is reduced to zero and the processor load is reduced. In the example above, "soxr" interpolation was also enabled. +Most settings have sensible default values, so -- as in the example above -- users generally only need to set (1) the service name, (2) a password (if desired) and (3) the output device. If the output device has a mixer that can be used for volume control, then (4) the mixer name should be specified. It is important to do this if the mixer exists. Otherwise, the maximum output from the output device will be whatever setting the mixer happens to have, which will be a matter of chance and which could be very low or even silent. A sample configuration file with all possible settings, but with all of them commented out, is installed at \fIshairport-sync.conf.sample\f1, within the System Configuration Directory -- \fI/etc\f1 in Linux, \fI/usr/local/etc\f1 in BSD unixes. @@ -67,7 +65,7 @@ These are the settings available within the \fBgeneral\f1 group: \fBname=\f1\fI"service_name"\f1\fB;\f1 Use this \fIservice_name\f1 to identify this player in iTunes, etc. -The following substitutions are allowed: \fB%h\f1 for the computer's hostname, \fB%H\f1 for the computer's hostname with the first letter capitalised (ASCII only), \fB%v\f1 for the shairport-sync version number, e.g. "3.0.1" and \fB%V\f1 for the shairport-sync version string, e.g. "3.0.1-OpenSSL-Avahi-ALSA-soxr-metadata-sysconfdir:/etc". +The following substitutions are allowed: \fB%h\f1 for the computer's hostname, \fB%H\f1 for the computer's hostname with the first letter capitalised (ASCII only), \fB%v\f1 for the shairport-sync version number, e.g. "3.3.6" and \fB%V\f1 for the shairport-sync version string, e.g. "3.3.6-OpenSSL-Avahi-ALSA-soxr-metadata-sysconfdir:/etc". The default is "%H", which is replaced by the hostname with the first letter capitalised. .TP @@ -75,7 +73,7 @@ The default is "%H", which is replaced by the hostname with the first letter cap Require the password \fIpassword\f1 to connect to the service. If you leave this setting commented out, no password is needed. .TP \fBinterpolation=\f1\fI"mode"\f1\fB;\f1 -Interpolate, or "stuff", the audio stream using the \fImode\f1. Interpolation here refers to the process of adding or removing frames of audio to or from the stream sent to the output device to keep it exactly in synchrony with the player. The default mode, "basic", is normally almost completely inaudible. The alternative mode, "soxr", is even less obtrusive but requires much more processing power. For this mode, support for libsoxr, the SoX Resampler Library, must be selected when shairport-sync is compiled. +Interpolate, or "stuff", the audio stream using the \fImode\f1. Interpolation here refers to the process of adding or removing frames of audio to or from the stream sent to the output device to keep it exactly synchronised with the player. The "basic" mode is normally almost completely inaudible. The alternative mode, "soxr", is even less obtrusive but requires much more processing power. For this mode, support for libsoxr, the SoX Resampler Library, must be selected when shairport-sync is compiled. The default setting is "auto", which will choose "soxr" if support for it has been compiled into the build of Shairport Synce and if the CPU is fast enough. Otherwise, "basic" stuffing will be chosen. .TP \fBoutput_backend=\f1\fI"backend"\f1\fB;\f1 shairport-sync has a number of modules of code ("backends") through which audio is output. Normally, the first audio backend that works is selected. This setting forces the selection of the specific audio \fIbackend\f1. Perform the command \fBshairport-sync -h\f1 to get a list of available audio backends -- the default is the first on this list. Only the "alsa", "sndio" and "pa" backends support synchronisation. @@ -218,14 +216,14 @@ Specify the \fIname\f1 of the mixer control to be used by shairport-sync to cont By default, the mixer is assumed to be output_device. Use this setting to specify a device other than the output device. .TP \fBoutput_rate=\f1\fIframe rate\f1\fB;\f1 -Use this setting to specify the frame rate to output to the ALSA device. Allowable values are 44100 (default), 88200, 176400 and 352800. The device must have the capability to accept the format you specify. There is no particular reason to use anything other than 44100 if it is available. +Use this setting to specify the frame rate to output to the ALSA device. Allowable values are "auto" (default), 44100, 88200, 176400 and 352800. The device must have the capability to accept the rate you specify. There is no particular reason to use anything other than 44100 if it is available, and if "auto" is selected, the lowest of these rates available, starting at 44100, will be selected. .TP \fBoutput_format=\f1\fI"format"\f1\fB;\f1 -Use this setting to specify the format that should be used to send data to the ALSA device. Allowable values are "U8", "S8", "S16", "S24", "S24_3LE", "S24_3BE" or "S32". The device must have the capability to accept the format you specify. +Use this setting to specify the format that should be used to send data to the ALSA device. Allowable values are "auto" (default), "U8", "S8", "S16", "S24", "S24_3LE", "S24_3BE" or "S32". The device must have the capability to accept the format you specify. "S" means signed; "U" means unsigned; BE means big-endian and LE means little-endian. Except where stated (using *LE or *BE), endianness matches that of the processor. The default is "S16". -If you are using a hardware mixer, the best setting is S16, as audio will pass through Shairport Sync unmodified except for interpolation. If you are using the software mixer, use 32- or 24-bit, if your device is capable of it, to get the lowest possible levels of dither. +If you are using a hardware mixer, S16 is fine, as audio will pass through Shairport Sync unmodified except for interpolation, but any of the higher-resolution formats are okay too. If you are using the software mixer, use 32- or 24-bit, if your device is capable of it, in order to get the lowest possible levels of dither. The "auto" setting will cause Shairport Sync to choose the highest resolution available. .TP \fBdisable_synchronization=\f1\fI"no"\f1\fB;\f1 This is an advanced setting and is for debugging only. Set to \fI"yes"\f1 to disable synchronization. Default is \fI"no"\f1. If you use it to disable synchronisation, then sooner or later you'll experience audio glitches due to audio buffer overflow or underflow. diff --git a/man/shairport-sync.7.xml b/man/shairport-sync.7.xml index b86bdbad..e5dc1c56 100644 --- a/man/shairport-sync.7.xml +++ b/man/shairport-sync.7.xml @@ -118,7 +118,6 @@

general = {

name = "Mike's Boombox";

-

interpolation = "soxr";

password = "secret";

output_backend = "alsa";

};

@@ -131,10 +130,10 @@

Most settings have sensible default values, so -- as in the example above -- users generally only need to set (1) the service name, (2) a password (if desired) and (3) the output device. If the output device has a mixer that can be used for volume - control, then (4) the volume control's name should be specified. It is highly - desirable to use the output device's mixer for volume control, if available -- - response time is reduced to zero and the processor load is reduced. In the example - above, "soxr" interpolation was also enabled.

+ control, then (4) the mixer name should be specified. It is important + to do this if the mixer exists. Otherwise, the + maximum output from the output device will be whatever setting the mixer happens to + have, which will be a matter of chance and which could be very low or even silent.

A sample configuration file with all possible settings, but with all of them commented out, is installed at shairport-sync.conf.sample, within the @@ -159,9 +158,9 @@ %h for the computer's hostname, %H for the computer's hostname with the first letter capitalised (ASCII only), - %v for the shairport-sync version number, e.g. "3.0.1" and + %v for the shairport-sync version number, e.g. "3.3.6" and %V for the shairport-sync version string, e.g. - "3.0.1-OpenSSL-Avahi-ALSA-soxr-metadata-sysconfdir:/etc".

+ "3.3.6-OpenSSL-Avahi-ALSA-soxr-metadata-sysconfdir:/etc".

The default is "%H", which is replaced by the hostname with the first letter capitalised.

@@ -178,13 +177,17 @@

Interpolate, or "stuff", the audio stream using the mode. Interpolation here refers to the process of adding or removing frames of audio to or from the - stream sent to the output device to keep it exactly in synchrony + stream sent to the output device to keep it exactly synchronised with the player. - The default mode, "basic", is normally almost completely inaudible. + The "basic" mode is normally almost completely inaudible. The alternative mode, "soxr", is even less obtrusive but requires much more processing power. For this mode, support for libsoxr, the SoX Resampler Library, must be selected when shairport-sync is compiled. + The default setting is "auto", which will choose "soxr" if support for it has been + compiled into the build of Shairport Synce and if the CPU is fast enough. Otherwise, + "basic" stuffing will be chosen. +

@@ -569,23 +572,26 @@ diff --git a/man/shairport-sync.html b/man/shairport-sync.html index 94723e9b..5d51b3c0 100644 --- a/man/shairport-sync.html +++ b/man/shairport-sync.html @@ -10,7 +10,7 @@

Description

Configuration File Settings

-

http://www.hyperrealm.com/libconfig/libconfig_manual.html

+

http://www.hyperrealm.com/libconfig/libconfig_manual.html

Options

Program Options

From 9e34b1334b93d89a02bae43b319569be81a81618 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 27 May 2020 15:02:49 +0100 Subject: [PATCH 105/186] Stop using home-brew non-blocking write. Instead, having opened the pipe non-blocking, change to blocking. --- audio_pipe.c | 32 +++++----------- common.c | 104 ++++++++++++++++---------------------------------- common.h | 2 +- rtsp.c | 105 ++++++++++++++++++++++++--------------------------- 4 files changed, 93 insertions(+), 150 deletions(-) diff --git a/audio_pipe.c b/audio_pipe.c index 2abef94b..a2b33349 100644 --- a/audio_pipe.c +++ b/audio_pipe.c @@ -42,7 +42,6 @@ static int fd = -1; char *pipename = NULL; -int warned = 0; static void start(__attribute__((unused)) int sample_rate, __attribute__((unused)) int sample_format) { @@ -51,37 +50,26 @@ static void start(__attribute__((unused)) int sample_rate, // we check that it's not a "real" error though. From the "man 2 open" page: // "ENXIO O_NONBLOCK | O_WRONLY is set, the named file is a FIFO, and no process has the FIFO // open for reading." - fd = open(pipename, O_WRONLY | O_NONBLOCK); - if ((fd == -1) && (errno != ENXIO) && (warned == 0)) { - char errorstring[1024]; - strerror_r(errno, (char *)errorstring, sizeof(errorstring)); - debug(1, "pipe: start -- error %d (\"%s\") opening the pipe named \"%s\".", errno, - (char *)errorstring, pipename); - warn("Error %d opening the pipe named \"%s\".", errno, pipename); - warned = 1; - } -} + + fd = try_to_open_pipe_for_writing(pipename); + } static int play(void *buf, int samples) { // if the file is not open, try to open it. char errorstring[1024]; if (fd == -1) { - fd = open(pipename, O_WRONLY | O_NONBLOCK); + fd = try_to_open_pipe_for_writing(pipename); } // if it's got a reader, write to it. if (fd > 0) { - int rc = non_blocking_write(fd, buf, samples * 4); - if ((rc < 0) && (warned == 0)) { + //int rc = non_blocking_write(fd, buf, samples * 4); + int rc = write(fd, buf, samples * 4); + if ((rc < 0) && (errno != EPIPE)) { strerror_r(errno, (char *)errorstring, 1024); - warn("Error %d writing to the pipe named \"%s\": \"%s\".", errno, pipename, errorstring); - warned = 1; + debug(1, "audio_pip play: error %d writing to the pipe named \"%s\": \"%s\".", errno, pipename, errorstring); } - } else if ((fd == -1) && (errno != ENXIO) && (warned == 0)) { - strerror_r(errno, (char *)errorstring, 1024); - warn("Error %d opening the pipe named \"%s\": \"%s\".", errno, pipename, errorstring); - warned = 1; } - return warned; + return 0; } static void stop(void) { @@ -89,7 +77,7 @@ static void stop(void) { } static int init(int argc, char **argv) { - debug(1, "pipe init"); + // debug(1, "pipe init"); // const char *str; // int value; // double dvalue; diff --git a/common.c b/common.c index 43bb41af..74f89869 100644 --- a/common.c +++ b/common.c @@ -40,6 +40,7 @@ #include #include #include +#include #ifdef COMPILE_FOR_OSX #include @@ -1049,80 +1050,39 @@ uint64_t get_absolute_time_in_ns() { return time_now_ns; } -ssize_t non_blocking_write_with_timeout(int fd, const void *buf, size_t count, uint64_t timeout) { - // try to write to this non-blocking file descriptor - // wait for up to the timeout for it to be possible to write something. - // exit immediately if there is no connection at the other end - // timeout is in nanoseconds - void *ibuf = (void *)buf; - size_t bytes_remaining = count; - ssize_t rc = 0; - uint64_t end_time = get_absolute_time_in_ns() + timeout; - while ((bytes_remaining) && (rc == 0)) { - rc = write(fd, ibuf, bytes_remaining); - if (rc >= 0) { - bytes_remaining = bytes_remaining - rc; - ibuf += rc; - rc = 0; // not an error - } else if (errno == EAGAIN) { - // delay for a little while if it couldn't get everything out... - uint64_t time_remaining = end_time - get_absolute_time_in_ns(); - useconds_t t = time_remaining / 1000; // ns to us - if (t > 50000) - t = 50000; // wait for 50 milliseconds at most - if (t != 0) { - rc = 0; // not an error - usleep(t); - } - } - } - if (rc > 0) - return count - bytes_remaining; // this is just to mimic a normal write/3. - else - return rc; -} +int try_to_open_pipe_for_writing(const char* pathname) { + // tries to open the pipe in non-blocking mode first. + // if it succeeds, it sets it to blocking. + // if not, it returns -1. -/* -ssize_t non_blocking_write_with_timeout(int fd, const void *buf, size_t count, int timeout) { - // - // timeout is in milliseconds - void *ibuf = (void *)buf; - size_t bytes_remaining = count; - int rc = 1; - struct pollfd ufds[1]; - while ((bytes_remaining > 0) && (rc > 0)) { - // check that we can do some writing - ufds[0].fd = fd; - ufds[0].events = POLLOUT; - rc = poll(ufds, 1, timeout); - if (rc < 0) { - // debug(1, "non-blocking write error waiting for pipe to become ready for writing..."); - } else if (rc == 0) { - // warn("non-blocking write timeout waiting for pipe to become ready for writing"); - rc = -1; - errno = -ETIMEDOUT; - } else { // rc > 0, implying it might be ready - ssize_t bytes_written = write(fd, ibuf, bytes_remaining); - if (bytes_written == -1) { - // debug(1,"Error %d in non_blocking_write: \"%s\".",errno,strerror(errno)); - rc = bytes_written; // to imitate the return from write() - } else { - ibuf += bytes_written; - bytes_remaining -= bytes_written; - } - } - } - if (rc > 0) - return count - bytes_remaining; // this is just to mimic a normal write/3. - else - return rc; - // return write(fd,buf,count); -} -*/ + char errorstring[1024]; + int fdis = open(pathname, O_WRONLY | O_NONBLOCK); // open it in non blocking mode first + // we check that it's not a "real" error. From the "man 2 open" page: + // "ENXIO O_NONBLOCK | O_WRONLY is set, the named file is a FIFO, and no process has the FIFO + // open for reading." Which is okay. -ssize_t non_blocking_write(int fd, const void *buf, size_t count) { - // if it would block, wait up to 0.5 seconds for it to unblock. - return non_blocking_write_with_timeout(fd, buf, count, 500000000); + if ((fdis == -1) && (errno != ENXIO)) { + strerror_r(errno, (char *)errorstring, sizeof(errorstring)); + debug(1, "try_to_open_pipe -- error %d (\"%s\") opening pipe: \"%s\".", errno, + (char *)errorstring, pathname); + } else + if (fdis >= 0) { + // now we switch to blocking mode + int flags = fcntl(fdis, F_GETFL); + if ((flags == -1)) { + strerror_r(errno, (char *)errorstring, sizeof(errorstring)); + debug(1, "try_to_open_pipe -- error %d (\"%s\") getting flags of pipe: \"%s\".", errno, + (char *)errorstring, pathname); + } else { + flags = fcntl(fdis, F_SETFL,flags & ~O_NONBLOCK); + if (flags == -1) { + strerror_r(errno, (char *)errorstring, sizeof(errorstring)); + debug(1, "try_to_open_pipe -- error %d (\"%s\") unsetting NONBLOCK of pipe: \"%s\".", errno, + (char *)errorstring, pathname); + } + } + } + return fdis; } /* from diff --git a/common.h b/common.h index e07cb7c9..b964e051 100644 --- a/common.h +++ b/common.h @@ -303,7 +303,7 @@ int get_requested_connection_state_to_output(); void set_requested_connection_state_to_output(int v); -ssize_t non_blocking_write(int fd, const void *buf, size_t count); // used in a few places +int try_to_open_pipe_for_writing(const char* pathname); // open it without blocking if it's not hooked up /* from * http://coding.debuntu.org/c-implementing-str_replace-replace-all-occurrences-substring#comment-722 diff --git a/rtsp.c b/rtsp.c index a1252133..0a9c0859 100644 --- a/rtsp.c +++ b/rtsp.c @@ -213,6 +213,7 @@ int pc_queue_add_item(pc_queue *the_queue, const void *the_stuff, int block) { i = 0; the_queue->eoq = i; the_queue->count++; + debug(2,"metadata queue+ %d/%d.", the_queue->count, the_queue->capacity); if (the_queue->count == the_queue->capacity) debug(1, "pc_queue is full with %d items in it!", the_queue->count); rc = pthread_cond_signal(&the_queue->pc_queue_item_added_signal); @@ -249,6 +250,7 @@ int pc_queue_get_item(pc_queue *the_queue, void *the_stuff) { i = 0; the_queue->toq = i; the_queue->count--; + debug(2,"metadata queue- %d/%d.", the_queue->count, the_queue->capacity); rc = pthread_cond_signal(&the_queue->pc_queue_item_removed_signal); if (rc) debug(1, "Error signalling after pc_queue_removed_item"); @@ -286,17 +288,15 @@ void *player_watchdog_thread_code(void *arg) { uint64_t last_watchdog_bark_time = conn->watchdog_bark_time; debug_mutex_unlock(&conn->watchdog_mutex, 0); if (last_watchdog_bark_time != 0) { - uint64_t time_since_last_bark = - (get_absolute_time_in_ns() - last_watchdog_bark_time) / 1000000000; + uint64_t time_since_last_bark = (get_absolute_time_in_ns() - last_watchdog_bark_time) / 1000000000; uint64_t ct = config.timeout; // go from int to 64-bit int if (time_since_last_bark >= ct) { conn->watchdog_barks++; if (conn->watchdog_barks == 1) { // debuglev = 3; // tell us everything. - debug(1, - "Connection %d: As Yeats almost said, \"Too long a silence / can make a stone " - "of the heart\".", + debug(1, "Connection %d: As Yeats almost said, \"Too long a silence / can make a stone " + "of the heart\".", conn->connection_number); conn->stop = 1; pthread_cancel(conn->thread); @@ -573,7 +573,7 @@ int msg_handle_line(rtsp_message **pmsg, char *line) { } fail: - debug(3, "msg_handle_line fail"); + debug(3,"msg_handle_line fail"); msg_free(pmsg); *pmsg = NULL; return 0; @@ -616,8 +616,7 @@ enum rtsp_read_request_response rtsp_read_request(rtsp_conn_info *conn, rtsp_mes if (errno == EINTR) continue; if (errno == EAGAIN) { - debug(1, "Connection %d: getting Error 11 -- EAGAIN from a blocking read!", - conn->connection_number); + debug(1, "Connection %d: getting Error 11 -- EAGAIN from a blocking read!", conn->connection_number); continue; } if (errno != ECONNRESET) { @@ -630,15 +629,15 @@ enum rtsp_read_request_response rtsp_read_request(rtsp_conn_info *conn, rtsp_mes goto shutdown; } - /* // this outputs the message received - { - void *pt = malloc(nread+1); - memset(pt, 0, nread+1); - memcpy(pt, buf + inbuf, nread); - debug(1, "Incoming string on port: \"%s\"",pt); - free(pt); - } - */ +/* // this outputs the message received + { + void *pt = malloc(nread+1); + memset(pt, 0, nread+1); + memcpy(pt, buf + inbuf, nread); + debug(1, "Incoming string on port: \"%s\"",pt); + free(pt); + } +*/ inbuf += nread; @@ -647,8 +646,7 @@ enum rtsp_read_request_response rtsp_read_request(rtsp_conn_info *conn, rtsp_mes msg_size = msg_handle_line(the_packet, buf); if (!(*the_packet)) { - debug(1, "Connection %d: rtsp_read_request can't find an RTSP header.", - conn->connection_number); + debug(1,"Connection %d: rtsp_read_request can't find an RTSP header.", conn->connection_number); reply = rtsp_read_request_response_bad_packet; goto shutdown; } @@ -855,10 +853,9 @@ void handle_options(rtsp_conn_info *conn, __attribute__((unused)) rtsp_message * rtsp_message *resp) { debug(3, "Connection %d: OPTIONS", conn->connection_number); resp->respcode = 200; - msg_add_header(resp, "Public", - "ANNOUNCE, SETUP, RECORD, " - "PAUSE, FLUSH, TEARDOWN, " - "OPTIONS, GET_PARAMETER, SET_PARAMETER"); + msg_add_header(resp, "Public", "ANNOUNCE, SETUP, RECORD, " + "PAUSE, FLUSH, TEARDOWN, " + "OPTIONS, GET_PARAMETER, SET_PARAMETER"); } void handle_teardown(rtsp_conn_info *conn, __attribute__((unused)) rtsp_message *req, @@ -999,9 +996,8 @@ void handle_setup(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp) { msg_add_header(resp, "Session", "1"); resp->respcode = 200; // it all worked out okay - debug(1, - "Connection %d: SETUP DACP-ID \"%s\" from %s to %s with UDP ports Control: " - "%d, Timing: %d and Audio: %d.", + debug(1, "Connection %d: SETUP DACP-ID \"%s\" from %s to %s with UDP ports Control: " + "%d, Timing: %d and Audio: %d.", conn->connection_number, conn->dacp_id, &conn->client_ip_string, &conn->self_ip_string, conn->local_control_port, conn->local_timing_port, conn->local_audio_port); @@ -1291,11 +1287,7 @@ void metadata_open(void) { char *path = malloc(pl + 1); snprintf(path, pl + 1, "%s", config.metadata_pipename); - fd = open(path, O_WRONLY | O_NONBLOCK); - // if (fd < 0) - // debug(1, "Could not open metadata FIFO %s. Will try again later.", - // path); - + fd = try_to_open_pipe_for_writing(path); free(path); } @@ -1307,8 +1299,8 @@ static void metadata_close(void) { } void metadata_process(uint32_t type, uint32_t code, char *data, uint32_t length) { - // debug(2, "Process metadata with type %x, code %x and length %u.", type, code, length); - int ret; + // debug(1, "Process metadata with type %x, code %x and length %u.", type, code, length); + int ret = 0; if (metadata_sock >= 0 && length < config.metadata_sockmsglength - 8) { char *ptr = metadata_sockmsg; @@ -1366,6 +1358,7 @@ void metadata_process(uint32_t type, uint32_t code, char *data, uint32_t length) } // readers may go away and come back + if (fd < 0) metadata_open(); if (fd < 0) @@ -1373,14 +1366,16 @@ void metadata_process(uint32_t type, uint32_t code, char *data, uint32_t length) char thestring[1024]; snprintf(thestring, 1024, "%x%x%u", type, code, length); - ret = non_blocking_write(fd, thestring, strlen(thestring)); + // ret = non_blocking_write(fd, thestring, strlen(thestring)); + ret = write(fd, thestring, strlen(thestring)); if (ret < 0) { // debug(1,"metadata_process error %d exit 1",ret); return; } if ((data != NULL) && (length > 0)) { snprintf(thestring, 1024, "\n\n"); - ret = non_blocking_write(fd, thestring, strlen(thestring)); + // ret = non_blocking_write(fd, thestring, strlen(thestring)); + ret = write(fd, thestring, strlen(thestring)); if (ret < 0) { // debug(1,"metadata_process error %d exit 2",ret); return; @@ -1403,7 +1398,8 @@ void metadata_process(uint32_t type, uint32_t code, char *data, uint32_t length) debug(1, "Error encoding base64 data."); // debug(1,"Remaining count: %d ret: %d, outbuf_size: // %d.",remaining_count,ret,outbuf_size); - ret = non_blocking_write(fd, outbuf, outbuf_size); + //ret = non_blocking_write(fd, outbuf, outbuf_size); + ret = write(fd, outbuf, outbuf_size); if (ret < 0) { // debug(1,"metadata_process error %d exit 3",ret); return; @@ -1412,14 +1408,16 @@ void metadata_process(uint32_t type, uint32_t code, char *data, uint32_t length) remaining_count -= towrite_count; } snprintf(thestring, 1024, ""); - ret = non_blocking_write(fd, thestring, strlen(thestring)); + // ret = non_blocking_write(fd, thestring, strlen(thestring)); + ret = write(fd, thestring, strlen(thestring)); if (ret < 0) { // debug(1,"metadata_process error %d exit 4",ret); return; } } snprintf(thestring, 1024, "\n"); - ret = non_blocking_write(fd, thestring, strlen(thestring)); + //ret = non_blocking_write(fd, thestring, strlen(thestring)); + ret = write(fd, thestring, strlen(thestring)); if (ret < 0) { // debug(1,"metadata_process error %d exit 5",ret); return; @@ -1451,6 +1449,7 @@ void *metadata_thread_function(__attribute__((unused)) void *ignore) { pthread_cleanup_push(metadata_thread_cleanup_function, NULL); while (1) { pc_queue_get_item(&metadata_queue, &pack); + // debug(1,"pc_queue get item."); pthread_cleanup_push(metadata_pack_cleanup_function, (void *)&pack); if (config.metadata_enabled) { metadata_process(pack.type, pack.code, pack.data, pack.length); @@ -1596,7 +1595,7 @@ static void handle_set_parameter(rtsp_conn_info *conn, rtsp_message *req, rtsp_m char *ct = msg_get_header(req, "Content-Type"); if (ct) { - // debug(2, "SET_PARAMETER Content-Type:\"%s\".", ct); +// debug(2, "SET_PARAMETER Content-Type:\"%s\".", ct); #ifdef CONFIG_METADATA // It seems that the rtptime of the message is used as a kind of an ID that @@ -1894,7 +1893,7 @@ static void handle_announce(rtsp_conn_info *conn, rtsp_message *req, rtsp_messag unsigned int i = 0; unsigned int max_param = sizeof(conn->stream.fmtp) / sizeof(conn->stream.fmtp[0]); - char *found; + char* found; while ((found = strsep(&pfmtp, " \t")) != NULL && i < max_param) { conn->stream.fmtp[i++] = atoi(found); } @@ -2341,7 +2340,6 @@ static void *rtsp_conversation_thread_func(void *pconn) { pthread_condattr_init(&attr); pthread_condattr_setclock(&attr, CLOCK_MONOTONIC); // can't do this in OS X, and don't need it. rc = pthread_cond_init(&conn->flowcontrol, &attr); - pthread_condattr_destroy(&attr); #endif #ifdef COMPILE_FOR_OSX rc = pthread_cond_init(&conn->flowcontrol, NULL); @@ -2376,9 +2374,8 @@ static void *rtsp_conversation_thread_func(void *pconn) { if (strcmp(req->method, "OPTIONS") != 0) // the options message is very common, so don't log it until level 3 debug_level = 2; - debug(debug_level, - "Connection %d: Received an RTSP Packet of type \"%s\":", conn->connection_number, - req->method), + debug(debug_level, "Connection %d: Received an RTSP Packet of type \"%s\":", + conn->connection_number, req->method), debug_print_msg_headers(debug_level, req); apple_challenge(conn->fd, req, resp); @@ -2427,9 +2424,8 @@ static void *rtsp_conversation_thread_func(void *pconn) { if (conn->stop == 0) { int err = msg_write_response(conn->fd, resp); if (err) { - debug(1, - "Connection %d: Unable to write an RTSP message response. Terminating the " - "connection.", + debug(1, "Connection %d: Unable to write an RTSP message response. Terminating the " + "connection.", conn->connection_number); struct linger so_linger; so_linger.l_onoff = 1; // "true" @@ -2482,11 +2478,10 @@ static void *rtsp_conversation_thread_func(void *pconn) { if (reply == -1) { char errorstring[1024]; strerror_r(errno, (char *)errorstring, sizeof(errorstring)); - debug(1, "rtsp_read_request_response_bad_packet write response error %d: \"%s\".", errno, - (char *)errorstring); + debug(1, "rtsp_read_request_response_bad_packet write response error %d: \"%s\".", errno, (char *)errorstring); } else if (reply != (ssize_t)strlen(response_text)) { - debug(1, "rtsp_read_request_response_bad_packet write %d bytes requested but %d written.", - strlen(response_text), reply); + debug(1, "rtsp_read_request_response_bad_packet write %d bytes requested but %d written.", strlen(response_text), + reply); } } else { debug(1, "Connection %d: rtsp_read_request error %d, packet ignored.", @@ -2699,7 +2694,7 @@ void rtsp_listen_loop(void) { debug(2, "Connection %d: new connection from %s:%u to self at %s:%u.", conn->connection_number, remote_ip4, rport, ip4, tport); } -#ifdef AF_INET6 + #ifdef AF_INET6 if (local_info->SAFAMILY == AF_INET6) { // IPv6: @@ -2716,7 +2711,7 @@ void rtsp_listen_loop(void) { debug(2, "Connection %d: new connection from [%s]:%u to self at [%s]:%u.", conn->connection_number, remote_ip6, rport, ip6, tport); } -#endif + #endif } else { debug(1, "Error figuring out Shairport Sync's own IP number."); @@ -2744,8 +2739,8 @@ void rtsp_listen_loop(void) { pthread_cleanup_pop(1); // should never happen } else { warn("could not establish a service on port %d -- program terminating. Is another instance of " - "Shairport Sync running on this device?", - config.port); + "Shairport Sync running on this device?", + config.port); } // debug(1, "Oops -- fell out of the RTSP select loop"); } From 2d7ae4bce902b9ec890efbf82587c9f934148045 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 27 May 2020 15:26:08 +0100 Subject: [PATCH 106/186] Add the beggings of a hacky check on the metadata modification lock. --- metadata_hub.c | 33 ++++++++++++++++----------------- metadata_hub.h | 14 ++++++++++---- 2 files changed, 26 insertions(+), 21 deletions(-) diff --git a/metadata_hub.c b/metadata_hub.c index b2f370ac..5994a46c 100644 --- a/metadata_hub.c +++ b/metadata_hub.c @@ -92,13 +92,6 @@ void add_metadata_watcher(metadata_watcher fn, void *userdata) { } } -/* -void metadata_hub_unlock_hub_mutex_cleanup(__attribute__((unused)) void *arg) { - // debug(1, "metadata_hub_unlock_hub_mutex_cleanup called."); - pthread_rwlock_unlock(&metadata_hub_re_lock); -} -*/ - void run_metadata_watchers(void) { int i; for (i = 0; i < number_of_watchers; i++) { @@ -129,19 +122,24 @@ void run_metadata_watchers(void) { metadata_store.songtime_in_milliseconds_changed = 0; } -void metadata_hub_modify_prolog(void) { +void metadata_hub_unlock_hub_mutex_cleanup(__attribute__((unused)) void *arg) { + // debug(1, "metadata_hub_unlock_hub_mutex_cleanup called."); + pthread_rwlock_unlock(&metadata_hub_re_lock); +} + +void _metadata_hub_modify_prolog(const char *filename, const int linenumber) { // always run this before changing an entry or a sequence of entries in the metadata_hub // debug(1, "locking metadata hub for writing"); if (pthread_rwlock_trywrlock(&metadata_hub_re_lock) != 0) { - debug(2, "Metadata_hub write lock is already taken -- must wait."); + debug(1, "Metadata_hub write lock is already taken -- must wait."); pthread_rwlock_wrlock(&metadata_hub_re_lock); - debug(2, "Okay -- acquired the metadata_hub write lock."); + debug(1, "Okay -- acquired the metadata_hub write lock."); } else { debug(3, "Metadata_hub write lock acquired."); } } -void metadata_hub_modify_epilog(int modified) { +void _metadata_hub_modify_epilog(int modified, const char *filename, const int linenumber) { metadata_store.dacp_server_has_been_active = metadata_store.dacp_server_active; // set the scanner_has_been_active now. if (modified) { @@ -151,17 +149,17 @@ void metadata_hub_modify_epilog(int modified) { debug(3, "Metadata_hub write lock unlocked."); } -void metadata_hub_read_prolog(void) { +void _metadata_hub_read_prolog(const char *filename, const int linenumber) { // always run this before reading an entry or a sequence of entries in the metadata_hub // debug(1, "locking metadata hub for reading"); if (pthread_rwlock_tryrdlock(&metadata_hub_re_lock) != 0) { - debug(2, "Metadata_hub read lock is already taken -- must wait."); + debug(1, "Metadata_hub read lock is already taken -- must wait."); pthread_rwlock_rdlock(&metadata_hub_re_lock); - debug(2, "Okay -- acquired the metadata_hub read lock."); + debug(1, "Okay -- acquired the metadata_hub read lock."); } } -void metadata_hub_read_epilog(void) { +void _metadata_hub_read_epilog(const char *filename, const int linenumber) { // always run this after reading an entry or a sequence of entries in the metadata_hub // debug(1, "unlocking metadata hub for reading"); pthread_rwlock_unlock(&metadata_hub_re_lock); @@ -463,10 +461,10 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin break; case 'PICT': metadata_hub_modify_prolog(); + pthread_cleanup_push(metadata_hub_unlock_hub_mutex_cleanup, NULL); debug(2, "MH Picture received, length %u bytes.", length); char uri[2048]; - if ((length > 16) && - (strcmp(config.cover_art_cache_dir, "") != 0)) { // if it's okay to write the file + if ((length > 16) && (strcmp(config.cover_art_cache_dir,"")!=0)) { // if it's okay to write the file char *pathname = metadata_write_image_file(data, length); snprintf(uri, sizeof(uri), "file://%s", pathname); free(pathname); @@ -479,6 +477,7 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin metadata_hub_modify_epilog(1); else metadata_hub_modify_epilog(0); + pthread_cleanup_pop(0); // don't remove the lock -- it'll have been done break; case 'clip': metadata_hub_modify_prolog(); diff --git a/metadata_hub.h b/metadata_hub.h index 75d2f852..8ade9b86 100644 --- a/metadata_hub.h +++ b/metadata_hub.h @@ -148,9 +148,15 @@ void metadata_hub_release_track_artwork(void); // these functions lock and unlock the read-write mutex on the metadata hub and run the watchers // afterwards -void metadata_hub_modify_prolog(void); -void metadata_hub_modify_epilog(int modified); // set to true if modifications occurred, 0 otherwise +void _metadata_hub_modify_prolog(const char *filename, const int linenumber); +void _metadata_hub_modify_epilog(int modified, const char *filename, const int linenumber); // set to true if modifications occurred, 0 otherwise // these are for safe reading -void metadata_hub_read_prolog(void); -void metadata_hub_read_epilog(void); +void _metadata_hub_read_prolog(const char *filename, const int linenumber); +void _metadata_hub_read_epilog(const char *filename, const int linenumber); + +#define metadata_hub_modify_prolog(void) _metadata_hub_modify_prolog(__FILE__, __LINE__) +#define metadata_hub_modify_epilog(modified) _metadata_hub_modify_epilog(modified, __FILE__, __LINE__) + +#define metadata_hub_read_prolog(void) _metadata_hub_read_prolog(__FILE__, __LINE__) +#define metadata_hub_read_epilog(void) _metadata_hub_modify_epilog(__FILE__, __LINE__) From 6406c296f587218d9ec53a57da30be882aef8fd9 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 27 May 2020 16:21:35 +0100 Subject: [PATCH 107/186] Add setting to direct logs to syslog, STDERR or STDOUT --- common.c | 21 +++++++++++++++++++-- common.h | 5 ++++- scripts/shairport-sync.conf | 3 ++- shairport.c | 13 +++++++++++++ 4 files changed, 38 insertions(+), 4 deletions(-) diff --git a/common.c b/common.c index 74f89869..bb824ca6 100644 --- a/common.c +++ b/common.c @@ -121,7 +121,7 @@ static void (*sps_log)(int prio, const char *t, ...) = daemon_log; static void (*sps_log)(int prio, const char *t, ...) = syslog; #endif -void do_sps_log(__attribute__((unused)) int prio, const char *t, ...) { +void do_sps_log_to_stderr(__attribute__((unused)) int prio, const char *t, ...) { char s[1024]; va_list args; va_start(args, t); @@ -130,7 +130,24 @@ void do_sps_log(__attribute__((unused)) int prio, const char *t, ...) { fprintf(stderr, "%s\n", s); } -void log_to_stderr() { sps_log = do_sps_log; } +void do_sps_log_to_stdout(__attribute__((unused)) int prio, const char *t, ...) { + char s[1024]; + va_list args; + va_start(args, t); + vsnprintf(s, sizeof(s), t, args); + va_end(args); + fprintf(stdout, "%s\n", s); +} + +void log_to_stderr() { sps_log = do_sps_log_to_stderr; } +void log_to_stdout() { sps_log = do_sps_log_to_stdout; } +void log_to_syslog() { +#ifdef CONFIG_LIBDAEMON + sps_log = daemon_log; +#else + sps_log = syslog; +#endif +} shairport_cfg config; diff --git a/common.h b/common.h index b964e051..df4243a9 100644 --- a/common.h +++ b/common.h @@ -295,7 +295,10 @@ uint16_t nctohs(const uint8_t *p); // read 2 characters from *p and do ntohs on void memory_barrier(); -void log_to_stderr(); // call this to director logging to stderr; +void log_to_stderr(); // call this to direct logging to stderr; +void log_to_stdout(); // call this to direct logging to stdout; +void log_to_syslog(); // call this to direct logging to the system log; + // true if Shairport Sync is supposed to be sending output to the output device, false otherwise diff --git a/scripts/shairport-sync.conf b/scripts/shairport-sync.conf index 09098fbc..1b6105cb 100644 --- a/scripts/shairport-sync.conf +++ b/scripts/shairport-sync.conf @@ -102,7 +102,7 @@ alsa = // output_rate = "auto"; // can be "auto", 44100, 88200, 176400 or 352800, but the device must have the capability. // output_format = "auto"; // can be "auto", "U8", "S8", "S16", "S16_LE", "S16_BE", "S24", "S24_LE", "S24_BE", "S24_3LE", "S24_3BE", "S32", "S32_LE" or "S32_BE" but the device must have the capability. Except where stated using (*LE or *BE), endianness matches that of the processor. -// disable_synchronization = "no"; // Set to "yes" to disable synchronization. Default is "no" This is really meant for troubleshooting. +// disable_synchronization = "no"; // Set to "yes" to disable synchronization. Default is "no" This is really meant for troubleshootingG. // period_size = ; // Use this optional advanced setting to set the alsa period size near to this value // buffer_size = ; // Use this optional advanced setting to set the alsa buffer size near to this value @@ -245,6 +245,7 @@ mqtt = diagnostics = { // disable_resend_requests = "no"; // set this to yes to stop Shairport Sync from requesting the retransmission of missing packets. Default is "no". +// log_output_to = "syslog"; // set this to "syslog" (default), "STDERR" or "STDOUT" to specify were all logs, statistics and diagnostic messages are written to. // statistics = "no"; // set to "yes" to print statistics in the log // log_verbosity = 0; // "0" means no debug verbosity, "3" is most verbose. // log_show_file_and_line = "yes"; // set this to yes if you want the file and line number of the message source in the log file diff --git a/shairport.c b/shairport.c index 0ad40ad8..fc1643ab 100644 --- a/shairport.c +++ b/shairport.c @@ -662,6 +662,19 @@ int parse_options(int argc, char **argv) { dvalue); } + /* Get the diagnostics output default. */ + if (config_lookup_string(config.cfg, "diagnostics.log_output_to", &str)) { + if (strcasecmp(str, "syslog") == 0) + log_to_syslog(); + else if (strcasecmp(str, "stdout") == 0) { + log_to_stdout(); + } else if (strcasecmp(str, "stderr") == 0) { + log_to_stderr(); + } else { + log_to_syslog(); + die("Invalid diagnostics log_output_to setting \"%s\". It should be \"syslog\", \"STDERR\" or \"STDOUT\". It is set to syslog."); + } + } /* Get the ignore_volume_control setting. */ if (config_lookup_string(config.cfg, "general.ignore_volume_control", &str)) { if (strcasecmp(str, "no") == 0) From c0e5d0473ecdff35386af98abd091bff4a8f56e2 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 28 May 2020 12:19:59 +0100 Subject: [PATCH 108/186] Add the ability to direct logs etc. to a file or pipe. --- common.c | 63 +++++++++++++++++++++++++++++++++++++ common.h | 4 +++ scripts/shairport-sync.conf | 2 +- shairport.c | 6 ++-- 4 files changed, 72 insertions(+), 3 deletions(-) diff --git a/common.c b/common.c index bb824ca6..f10ef7bf 100644 --- a/common.c +++ b/common.c @@ -41,6 +41,7 @@ #include #include #include +#include #ifdef COMPILE_FOR_OSX #include @@ -139,8 +140,70 @@ void do_sps_log_to_stdout(__attribute__((unused)) int prio, const char *t, ...) fprintf(stdout, "%s\n", s); } +int create_log_file(const char* path) { + int fd = -1; + if (path != NULL) { + char *dirc = strdup(path); + if (dirc) { + char *dname = dirname(dirc); + // create the directory, if necessary + int result = 0; + if (dname) { + char *pdir = realpath(dname, NULL); // will return a NULL if the directory doesn't exist + if (pdir == NULL) { + mode_t oldumask = umask(000); + result = mkpath(dname, 0777); + umask(oldumask); + } else { + free(pdir); + } + if ((result == 0) || (result == -EEXIST)) { + // now open the file + fd = open(path, O_WRONLY | O_NONBLOCK | O_CREAT | O_EXCL, S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH); + if ((fd == -1) && (errno == EEXIST)) + fd = open(path, O_WRONLY | O_APPEND | O_NONBLOCK); + + if (fd >= 0) { + // now we switch to blocking mode + int flags = fcntl(fd, F_GETFL); + if ((flags == -1)) { +// strerror_r(errno, (char *)errorstring, sizeof(errorstring)); +// debug(1, "create_log_file -- error %d (\"%s\") getting flags of pipe: \"%s\".", errno, +// (char *)errorstring, pathname); + } else { + flags = fcntl(fd, F_SETFL,flags & ~O_NONBLOCK); +// if (flags == -1) { +// strerror_r(errno, (char *)errorstring, sizeof(errorstring)); +// debug(1, "create_log_file -- error %d (\"%s\") unsetting NONBLOCK of pipe: \"%s\".", errno, +// (char *)errorstring, pathname); + } + } + } + } + free(dirc); + } + } + return fd; +} + +void do_sps_log_to_fd(__attribute__((unused)) int prio, const char *t, ...) { + char s[1024]; + va_list args; + va_start(args, t); + vsnprintf(s, sizeof(s), t, args); + va_end(args); + if (config.log_fd == -1) + config.log_fd = create_log_file(config.log_file_path); + if (config.log_fd >= 0) { + dprintf(config.log_fd, "%s\n", s); + } else if (errno != ENXIO) { // maybe there is a pipe there but not hooked up + fprintf(stderr, "%s\n", s); + } +} + void log_to_stderr() { sps_log = do_sps_log_to_stderr; } void log_to_stdout() { sps_log = do_sps_log_to_stdout; } +void log_to_file() { sps_log = do_sps_log_to_fd; } void log_to_syslog() { #ifdef CONFIG_LIBDAEMON sps_log = daemon_log; diff --git a/common.h b/common.h index df4243a9..3540bcd0 100644 --- a/common.h +++ b/common.h @@ -182,6 +182,8 @@ typedef struct { char *pidfile; #endif + int log_fd; // file descriptor of the file or pipe to log stuff to. + char *log_file_path; // path to file or pipe to log to, if any int logOutputLevel; // log output level int debugger_show_elapsed_time; // in the debug message, display the time since startup int debugger_show_relative_time; // in the debug message, display the time since the last one @@ -298,6 +300,8 @@ void memory_barrier(); void log_to_stderr(); // call this to direct logging to stderr; void log_to_stdout(); // call this to direct logging to stdout; void log_to_syslog(); // call this to direct logging to the system log; +void log_to_file(); // call this to direct logging to a file or (pre-existing) pipe; + // true if Shairport Sync is supposed to be sending output to the output device, false otherwise diff --git a/scripts/shairport-sync.conf b/scripts/shairport-sync.conf index 1b6105cb..efa34996 100644 --- a/scripts/shairport-sync.conf +++ b/scripts/shairport-sync.conf @@ -245,7 +245,7 @@ mqtt = diagnostics = { // disable_resend_requests = "no"; // set this to yes to stop Shairport Sync from requesting the retransmission of missing packets. Default is "no". -// log_output_to = "syslog"; // set this to "syslog" (default), "STDERR" or "STDOUT" to specify were all logs, statistics and diagnostic messages are written to. +// log_output_to = "syslog"; // set this to "syslog" (default), "stderr" or "stdout" or a file or pipe path to specify were all logs, statistics and diagnostic messages are written to. If there's anything wrong with the file spec, output will be to "stderr". // statistics = "no"; // set to "yes" to print statistics in the log // log_verbosity = 0; // "0" means no debug verbosity, "3" is most verbose. // log_show_file_and_line = "yes"; // set this to yes if you want the file and line number of the message source in the log file diff --git a/shairport.c b/shairport.c index fc1643ab..af11ade2 100644 --- a/shairport.c +++ b/shairport.c @@ -671,8 +671,9 @@ int parse_options(int argc, char **argv) { } else if (strcasecmp(str, "stderr") == 0) { log_to_stderr(); } else { - log_to_syslog(); - die("Invalid diagnostics log_output_to setting \"%s\". It should be \"syslog\", \"STDERR\" or \"STDOUT\". It is set to syslog."); + config.log_file_path = (char *)str; + config.log_fd = -1; + log_to_file(); } } /* Get the ignore_volume_control setting. */ @@ -1423,6 +1424,7 @@ int main(int argc, char **argv) { #ifdef CONFIG_LIBDAEMON pid = getpid(); #endif + config.log_fd = -1; conns = NULL; // no connections active memset((void *)&main_thread_id, 0, sizeof(main_thread_id)); memset(&config, 0, sizeof(config)); // also clears all strings, BTW From e681f47c8c30ff7b04cd7ff0d1f2a7ecd653cfd2 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 28 May 2020 12:20:41 +0100 Subject: [PATCH 109/186] Add the ability to direct logs etc. to a file or pipe. --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index a28dd4db..f20b0a4a 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.7d7], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.7d8], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 9ae649fa9a0b6038882e4fa8adb94d75c5452df5 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 28 May 2020 15:39:10 +0100 Subject: [PATCH 110/186] change some debugging messages --- metadata_hub.c | 1 + mqtt.c | 4 ++-- player.c | 2 +- 3 files changed, 4 insertions(+), 3 deletions(-) diff --git a/metadata_hub.c b/metadata_hub.c index 5994a46c..59130823 100644 --- a/metadata_hub.c +++ b/metadata_hub.c @@ -301,6 +301,7 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin // all the following items of metadata are contained in one metadata packet // they are preceded by an 'ssnc' 'mdst' item and followed by an 'ssnc 'mden' item. + debug(1, "metadata_hub_process_metadata type %x, code %x and length %u.", type, code, length); char *cs; int changed = 0; diff --git a/mqtt.c b/mqtt.c index f1726f21..e2b86c5d 100644 --- a/mqtt.c +++ b/mqtt.c @@ -24,10 +24,10 @@ void _cb_log(__attribute__((unused)) struct mosquitto *mosq, __attribute__((unus int level, const char *str) { switch (level) { case MOSQ_LOG_DEBUG: - debug(1, str); + debug(3, str); break; case MOSQ_LOG_INFO: - debug(2, str); + debug(3, str); break; case MOSQ_LOG_NOTICE: debug(3, str); diff --git a/player.c b/player.c index a750415a..8847b8b5 100644 --- a/player.c +++ b/player.c @@ -636,7 +636,7 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in if (((check_buf->ready) || (x == conn->ab_write)) && (missing_frame_run_count > 0)) { // send a resend request if (missing_frame_run_count > 1) - debug(2, "request resend of %d packets starting at seqno %u.", + debug(3, "request resend of %d packets starting at seqno %u.", missing_frame_run_count, start_of_missing_frame_run); if (config.disable_resend_requests == 0) { debug_mutex_unlock(&conn->ab_mutex, 3); From c099105a576358f93c352105feceb9ea6b19f3e0 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 28 May 2020 16:15:15 +0100 Subject: [PATCH 111/186] Add a really wacky debug message --- metadata_hub.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/metadata_hub.c b/metadata_hub.c index 59130823..118caf06 100644 --- a/metadata_hub.c +++ b/metadata_hub.c @@ -127,14 +127,26 @@ void metadata_hub_unlock_hub_mutex_cleanup(__attribute__((unused)) void *arg) { pthread_rwlock_unlock(&metadata_hub_re_lock); } + +char *last_metadata_hub_modify_prolog_file = NULL; +int last_metadata_hub_modify_prolog_line; + void _metadata_hub_modify_prolog(const char *filename, const int linenumber) { // always run this before changing an entry or a sequence of entries in the metadata_hub // debug(1, "locking metadata hub for writing"); if (pthread_rwlock_trywrlock(&metadata_hub_re_lock) != 0) { - debug(1, "Metadata_hub write lock is already taken -- must wait."); + if (last_metadata_hub_modify_prolog_file) + debug(1, "Metadata_hub write lock is already taken at \"%s:%d\" -- must wait.", last_metadata_hub_modify_prolog_file, last_metadata_hub_modify_prolog_line); + else + debug(1, "Metadata_hub write lock is already taken by unknown -- must wait."); pthread_rwlock_wrlock(&metadata_hub_re_lock); debug(1, "Okay -- acquired the metadata_hub write lock."); } else { + if (last_metadata_hub_modify_prolog_file) { + free(last_metadata_hub_modify_prolog_file); + } + last_metadata_hub_modify_prolog_file = strdup(filename); + last_metadata_hub_modify_prolog_line = linenumber; debug(3, "Metadata_hub write lock acquired."); } } @@ -145,6 +157,9 @@ void _metadata_hub_modify_epilog(int modified, const char *filename, const int l if (modified) { run_metadata_watchers(); } + if (last_metadata_hub_modify_prolog_file) { + free(last_metadata_hub_modify_prolog_file); + } pthread_rwlock_unlock(&metadata_hub_re_lock); debug(3, "Metadata_hub write lock unlocked."); } From 0988ce536cb6c2bd5640f6e50628f73f2f1dbc8b Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 28 May 2020 16:38:48 +0100 Subject: [PATCH 112/186] debug mods --- metadata_hub.c | 17 +---------------- 1 file changed, 1 insertion(+), 16 deletions(-) diff --git a/metadata_hub.c b/metadata_hub.c index 118caf06..59130823 100644 --- a/metadata_hub.c +++ b/metadata_hub.c @@ -127,26 +127,14 @@ void metadata_hub_unlock_hub_mutex_cleanup(__attribute__((unused)) void *arg) { pthread_rwlock_unlock(&metadata_hub_re_lock); } - -char *last_metadata_hub_modify_prolog_file = NULL; -int last_metadata_hub_modify_prolog_line; - void _metadata_hub_modify_prolog(const char *filename, const int linenumber) { // always run this before changing an entry or a sequence of entries in the metadata_hub // debug(1, "locking metadata hub for writing"); if (pthread_rwlock_trywrlock(&metadata_hub_re_lock) != 0) { - if (last_metadata_hub_modify_prolog_file) - debug(1, "Metadata_hub write lock is already taken at \"%s:%d\" -- must wait.", last_metadata_hub_modify_prolog_file, last_metadata_hub_modify_prolog_line); - else - debug(1, "Metadata_hub write lock is already taken by unknown -- must wait."); + debug(1, "Metadata_hub write lock is already taken -- must wait."); pthread_rwlock_wrlock(&metadata_hub_re_lock); debug(1, "Okay -- acquired the metadata_hub write lock."); } else { - if (last_metadata_hub_modify_prolog_file) { - free(last_metadata_hub_modify_prolog_file); - } - last_metadata_hub_modify_prolog_file = strdup(filename); - last_metadata_hub_modify_prolog_line = linenumber; debug(3, "Metadata_hub write lock acquired."); } } @@ -157,9 +145,6 @@ void _metadata_hub_modify_epilog(int modified, const char *filename, const int l if (modified) { run_metadata_watchers(); } - if (last_metadata_hub_modify_prolog_file) { - free(last_metadata_hub_modify_prolog_file); - } pthread_rwlock_unlock(&metadata_hub_re_lock); debug(3, "Metadata_hub write lock unlocked."); } From f27c5aebffae3c20e8662de1d64ef2abda677b1e Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 28 May 2020 17:01:26 +0100 Subject: [PATCH 113/186] debug mods --- metadata_hub.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/metadata_hub.c b/metadata_hub.c index 59130823..591ab02f 100644 --- a/metadata_hub.c +++ b/metadata_hub.c @@ -127,14 +127,25 @@ void metadata_hub_unlock_hub_mutex_cleanup(__attribute__((unused)) void *arg) { pthread_rwlock_unlock(&metadata_hub_re_lock); } +char *last_metadata_hub_modify_prolog_file = NULL; +int last_metadata_hub_modify_prolog_line; + void _metadata_hub_modify_prolog(const char *filename, const int linenumber) { // always run this before changing an entry or a sequence of entries in the metadata_hub // debug(1, "locking metadata hub for writing"); if (pthread_rwlock_trywrlock(&metadata_hub_re_lock) != 0) { - debug(1, "Metadata_hub write lock is already taken -- must wait."); + if (last_metadata_hub_modify_prolog_file) + debug(1, "Metadata_hub write lock is already taken at \"%s:%d\" -- must wait.", last_metadata_hub_modify_prolog_file, last_metadata_hub_modify_prolog_line); + else + debug(1, "Metadata_hub write lock is already taken by unknown -- must wait."); pthread_rwlock_wrlock(&metadata_hub_re_lock); debug(1, "Okay -- acquired the metadata_hub write lock."); } else { + if (last_metadata_hub_modify_prolog_file) { + free(last_metadata_hub_modify_prolog_file); + } + last_metadata_hub_modify_prolog_file = strdup(filename); + last_metadata_hub_modify_prolog_line = linenumber; debug(3, "Metadata_hub write lock acquired."); } } @@ -145,6 +156,10 @@ void _metadata_hub_modify_epilog(int modified, const char *filename, const int l if (modified) { run_metadata_watchers(); } + if (last_metadata_hub_modify_prolog_file) { + free(last_metadata_hub_modify_prolog_file); + last_metadata_hub_modify_prolog_file = NULL; + } pthread_rwlock_unlock(&metadata_hub_re_lock); debug(3, "Metadata_hub write lock unlocked."); } From 79c16c3236b55979358cfe5f32704edc94eb5384 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 28 May 2020 17:14:34 +0100 Subject: [PATCH 114/186] debug mods --- metadata_hub.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/metadata_hub.c b/metadata_hub.c index 591ab02f..eb63d2df 100644 --- a/metadata_hub.c +++ b/metadata_hub.c @@ -135,7 +135,7 @@ void _metadata_hub_modify_prolog(const char *filename, const int linenumber) { // debug(1, "locking metadata hub for writing"); if (pthread_rwlock_trywrlock(&metadata_hub_re_lock) != 0) { if (last_metadata_hub_modify_prolog_file) - debug(1, "Metadata_hub write lock is already taken at \"%s:%d\" -- must wait.", last_metadata_hub_modify_prolog_file, last_metadata_hub_modify_prolog_line); + debug(1, "Metadata_hub write lock at \"%s:%d\" is already taken at \"%s:%d\" -- must wait.", filename, linenumber, last_metadata_hub_modify_prolog_file, last_metadata_hub_modify_prolog_line); else debug(1, "Metadata_hub write lock is already taken by unknown -- must wait."); pthread_rwlock_wrlock(&metadata_hub_re_lock); From 16ecb5f37830d64d1473e00875cad610876278f0 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 28 May 2020 17:41:49 +0100 Subject: [PATCH 115/186] debug mods --- rtsp.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/rtsp.c b/rtsp.c index 0a9c0859..70d5dcc8 100644 --- a/rtsp.c +++ b/rtsp.c @@ -1452,16 +1452,20 @@ void *metadata_thread_function(__attribute__((unused)) void *ignore) { // debug(1,"pc_queue get item."); pthread_cleanup_push(metadata_pack_cleanup_function, (void *)&pack); if (config.metadata_enabled) { + debug(1, "metadata_process type %x, code %x and length %u.", pack.type, pack.code, pack.length); metadata_process(pack.type, pack.code, pack.data, pack.length); #ifdef CONFIG_METADATA_HUB + debug(1, "metadata_hub_process."); metadata_hub_process_metadata(pack.type, pack.code, pack.data, pack.length); #endif #ifdef CONFIG_MQTT if (config.mqtt_enabled) { + debug(1, "mqtt_process_metadata."); mqtt_process_metadata(pack.type, pack.code, pack.data, pack.length); } #endif + debug(1, "done."); } pthread_cleanup_pop(1); } From b915c2f527cf37e059b49d2bb5cd08569366240c Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 28 May 2020 19:50:03 +0100 Subject: [PATCH 116/186] temporarily disable the metadata pipe feed. --- rtsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/rtsp.c b/rtsp.c index 70d5dcc8..14807f74 100644 --- a/rtsp.c +++ b/rtsp.c @@ -1452,8 +1452,8 @@ void *metadata_thread_function(__attribute__((unused)) void *ignore) { // debug(1,"pc_queue get item."); pthread_cleanup_push(metadata_pack_cleanup_function, (void *)&pack); if (config.metadata_enabled) { - debug(1, "metadata_process type %x, code %x and length %u.", pack.type, pack.code, pack.length); - metadata_process(pack.type, pack.code, pack.data, pack.length); + // debug(1, "metadata_process type %x, code %x and length %u.", pack.type, pack.code, pack.length); + // metadata_process(pack.type, pack.code, pack.data, pack.length); #ifdef CONFIG_METADATA_HUB debug(1, "metadata_hub_process."); metadata_hub_process_metadata(pack.type, pack.code, pack.data, pack.length); From 5decc1bb060d408427bdfd6483dd5ffd4b727026 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 30 May 2020 15:49:00 +0100 Subject: [PATCH 117/186] Make the pipe, hub and mqtt separate threads with their own metadata queues. If one blocks, no new metadata is sent to it and the others are unaffected. --- common.c | 10 ++ common.h | 3 + metadata_hub.c | 84 ++++++++------ player.c | 36 +++--- rtp.c | 4 +- rtsp.c | 294 +++++++++++++++++++++++++++++++++++++------------ 6 files changed, 301 insertions(+), 130 deletions(-) diff --git a/common.c b/common.c index f10ef7bf..9bf0dd14 100644 --- a/common.c +++ b/common.c @@ -1700,3 +1700,13 @@ int string_update_with_size(char **str, int *flag, char *s, size_t len) { } return *flag; } + +// from https://stackoverflow.com/questions/13663617/memdup-function-in-c, with thanks +void* memdup(const void* mem, size_t size) { + void* out = malloc(size); + + if(out != NULL) + memcpy(out, mem, size); + + return out; +} diff --git a/common.h b/common.h index 3540bcd0..93bb9c36 100644 --- a/common.h +++ b/common.h @@ -441,4 +441,7 @@ void malloc_cleanup(void *arg); int string_update_with_size(char **str, int *flag, char *s, size_t len); +// from https://stackoverflow.com/questions/13663617/memdup-function-in-c, with thanks +void* memdup(const void* mem, size_t size); + #endif // _COMMON_H diff --git a/metadata_hub.c b/metadata_hub.c index eb63d2df..2ea442a5 100644 --- a/metadata_hub.c +++ b/metadata_hub.c @@ -123,12 +123,13 @@ void run_metadata_watchers(void) { } void metadata_hub_unlock_hub_mutex_cleanup(__attribute__((unused)) void *arg) { - // debug(1, "metadata_hub_unlock_hub_mutex_cleanup called."); - pthread_rwlock_unlock(&metadata_hub_re_lock); + debug(1, "metadata_hub_unlock_hub_mutex_cleanup called."); + metadata_hub_modify_epilog(0); } char *last_metadata_hub_modify_prolog_file = NULL; int last_metadata_hub_modify_prolog_line; +int metadata_hub_re_lock_access_is_delayed; void _metadata_hub_modify_prolog(const char *filename, const int linenumber) { // always run this before changing an entry or a sequence of entries in the metadata_hub @@ -138,8 +139,9 @@ void _metadata_hub_modify_prolog(const char *filename, const int linenumber) { debug(1, "Metadata_hub write lock at \"%s:%d\" is already taken at \"%s:%d\" -- must wait.", filename, linenumber, last_metadata_hub_modify_prolog_file, last_metadata_hub_modify_prolog_line); else debug(1, "Metadata_hub write lock is already taken by unknown -- must wait."); + metadata_hub_re_lock_access_is_delayed = 0; pthread_rwlock_wrlock(&metadata_hub_re_lock); - debug(1, "Okay -- acquired the metadata_hub write lock."); + debug(1, "Okay -- acquired the metadata_hub write lock at \"%s:%d\".", filename, linenumber); } else { if (last_metadata_hub_modify_prolog_file) { free(last_metadata_hub_modify_prolog_file); @@ -148,6 +150,7 @@ void _metadata_hub_modify_prolog(const char *filename, const int linenumber) { last_metadata_hub_modify_prolog_line = linenumber; debug(3, "Metadata_hub write lock acquired."); } + metadata_hub_re_lock_access_is_delayed = 0; } void _metadata_hub_modify_epilog(int modified, const char *filename, const int linenumber) { @@ -156,9 +159,14 @@ void _metadata_hub_modify_epilog(int modified, const char *filename, const int l if (modified) { run_metadata_watchers(); } - if (last_metadata_hub_modify_prolog_file) { - free(last_metadata_hub_modify_prolog_file); - last_metadata_hub_modify_prolog_file = NULL; + if (metadata_hub_re_lock_access_is_delayed) { + if (last_metadata_hub_modify_prolog_file) { + debug(1, "Metadata_hub write lock taken at \"%s:%d\" is freed at \"%s:%d\".", last_metadata_hub_modify_prolog_file, last_metadata_hub_modify_prolog_line, filename, linenumber); + free(last_metadata_hub_modify_prolog_file); + last_metadata_hub_modify_prolog_file = NULL; + } else { + debug(1, "Metadata_hub write lock taken at an unknown place is freed at \"%s:%d\".", filename, linenumber); + } } pthread_rwlock_unlock(&metadata_hub_re_lock); debug(3, "Metadata_hub write lock unlocked."); @@ -316,10 +324,15 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin // all the following items of metadata are contained in one metadata packet // they are preceded by an 'ssnc' 'mdst' item and followed by an 'ssnc 'mden' item. - debug(1, "metadata_hub_process_metadata type %x, code %x and length %u.", type, code, length); + // we don't set "changed" for them individually; instead we set it when the 'mden' token + // comes in if the metadata_packet_item_changed is set. + + int changed = 0; + int metadata_packet_item_changed; + metadata_hub_modify_prolog(); + pthread_cleanup_push(metadata_hub_unlock_hub_mutex_cleanup, NULL); char *cs; - int changed = 0; if (type == 'core') { switch (code) { case 'mper': { @@ -334,6 +347,7 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin metadata_store.item_id_changed = 1; metadata_store.item_id_received = 1; debug(2, "MH Item ID set to: \"%" PRIx64 "\"", metadata_store.item_id); + metadata_packet_item_changed = 1; } } break; case 'astm': { @@ -343,12 +357,14 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin metadata_store.songtime_in_milliseconds = ui; metadata_store.songtime_in_milliseconds_changed = 1; debug(2, "MH Song Time set to: \"%u\"", metadata_store.songtime_in_milliseconds); + metadata_packet_item_changed = 1; } } break; case 'asal': cs = strndup(data, length); if (string_update(&metadata_store.album_name, &metadata_store.album_name_changed, cs)) { debug(2, "MH Album name set to: \"%s\"", metadata_store.album_name); + metadata_packet_item_changed = 1; } free(cs); break; @@ -356,6 +372,7 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin cs = strndup(data, length); if (string_update(&metadata_store.artist_name, &metadata_store.artist_name_changed, cs)) { debug(2, "MH Artist name set to: \"%s\"", metadata_store.artist_name); + metadata_packet_item_changed = 1; } free(cs); break; @@ -364,6 +381,7 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin if (string_update(&metadata_store.album_artist_name, &metadata_store.album_artist_name_changed, cs)) { debug(2, "MH Album Artist name set to: \"%s\"", metadata_store.album_artist_name); + metadata_packet_item_changed = 1; } free(cs); break; @@ -371,6 +389,7 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin cs = strndup(data, length); if (string_update(&metadata_store.comment, &metadata_store.comment_changed, cs)) { debug(2, "MH Comment set to: \"%s\"", metadata_store.comment); + metadata_packet_item_changed = 1; } free(cs); break; @@ -378,6 +397,7 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin cs = strndup(data, length); if (string_update(&metadata_store.genre, &metadata_store.genre_changed, cs)) { debug(2, "MH Genre set to: \"%s\"", metadata_store.genre); + metadata_packet_item_changed = 1; } free(cs); break; @@ -385,6 +405,7 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin cs = strndup(data, length); if (string_update(&metadata_store.track_name, &metadata_store.track_name_changed, cs)) { debug(2, "MH Track Name set to: \"%s\"", metadata_store.track_name); + metadata_packet_item_changed = 1; } free(cs); break; @@ -392,6 +413,7 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin cs = strndup(data, length); if (string_update(&metadata_store.composer, &metadata_store.composer_changed, cs)) { debug(2, "MH Composer set to: \"%s\"", metadata_store.composer); + metadata_packet_item_changed = 1; } free(cs); break; @@ -408,6 +430,7 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin if (string_update(&metadata_store.song_album_artist, &metadata_store.song_album_artist_changed, cs)) { debug(2, "MH Song Album Artist set to: \"%s\"", metadata_store.song_album_artist); + metadata_packet_item_changed = 1; } free(cs); break; @@ -415,6 +438,7 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin cs = strndup(data, length); if (string_update(&metadata_store.sort_name, &metadata_store.sort_name_changed, cs)) { debug(2, "MH Sort Name set to: \"%s\"", metadata_store.sort_name); + metadata_packet_item_changed = 1; } free(cs); break; @@ -422,6 +446,7 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin cs = strndup(data, length); if (string_update(&metadata_store.sort_artist, &metadata_store.sort_artist_changed, cs)) { debug(2, "MH Sort Artist set to: \"%s\"", metadata_store.sort_artist); + metadata_packet_item_changed = 1; } free(cs); break; @@ -429,6 +454,7 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin cs = strndup(data, length); if (string_update(&metadata_store.sort_album, &metadata_store.sort_album_changed, cs)) { debug(2, "MH Sort Album set to: \"%s\"", metadata_store.sort_album); + metadata_packet_item_changed = 1; } free(cs); break; @@ -436,6 +462,7 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin cs = strndup(data, length); if (string_update(&metadata_store.sort_composer, &metadata_store.sort_composer_changed, cs)) { debug(2, "MH Sort Composer set to: \"%s\"", metadata_store.sort_composer); + metadata_packet_item_changed = 1; } free(cs); default: @@ -468,45 +495,45 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin break; case 'mdst': debug(2, "MH Metadata stream processing start."); - metadata_hub_modify_prolog(); + metadata_packet_item_changed = 0; break; case 'mden': debug(2, "MH Metadata stream processing end."); - metadata_hub_modify_epilog(1); - debug(2, "MH Metadata stream processing epilog complete."); + changed = metadata_packet_item_changed; break; case 'PICT': - metadata_hub_modify_prolog(); - pthread_cleanup_push(metadata_hub_unlock_hub_mutex_cleanup, NULL); debug(2, "MH Picture received, length %u bytes.", length); + char uri[2048]; if ((length > 16) && (strcmp(config.cover_art_cache_dir,"")!=0)) { // if it's okay to write the file + // make this uncancellable + int oldState; + pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); // make this un-cancellable char *pathname = metadata_write_image_file(data, length); snprintf(uri, sizeof(uri), "file://%s", pathname); free(pathname); + pthread_setcancelstate(oldState, NULL); + } else { uri[0] = '\0'; } if (string_update(&metadata_store.cover_art_pathname, &metadata_store.cover_art_pathname_changed, uri)) // if the picture's file path is different from the stored one... - metadata_hub_modify_epilog(1); + changed = 1; else - metadata_hub_modify_epilog(0); - pthread_cleanup_pop(0); // don't remove the lock -- it'll have been done + changed = 0; +// pthread_cleanup_pop(0); // don't remove the lock -- it'll have been done break; case 'clip': - metadata_hub_modify_prolog(); cs = strndup(data, length); if (string_update(&metadata_store.client_ip, &metadata_store.client_ip_changed, cs)) { changed = 1; debug(2, "MH Client IP set to: \"%s\"", metadata_store.client_ip); } free(cs); - metadata_hub_modify_epilog(changed); break; case 'prgr': - metadata_hub_modify_prolog(); cs = strndup(data, length); if (string_update(&metadata_store.progress_string, &metadata_store.progress_string_changed, cs)) { @@ -514,61 +541,45 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin debug(2, "MH Progress String set to: \"%s\"", metadata_store.progress_string); } free(cs); - metadata_hub_modify_epilog(changed); break; case 'svip': - metadata_hub_modify_prolog(); cs = strndup(data, length); if (string_update(&metadata_store.server_ip, &metadata_store.server_ip_changed, cs)) { changed = 1; debug(2, "MH Server IP set to: \"%s\"", metadata_store.server_ip); } free(cs); - metadata_hub_modify_epilog(changed); break; case 'abeg': - metadata_hub_modify_prolog(); changed = (metadata_store.active_state != AM_ACTIVE); metadata_store.active_state = AM_ACTIVE; - metadata_hub_modify_epilog(changed); break; case 'aend': - metadata_hub_modify_prolog(); changed = (metadata_store.active_state != AM_INACTIVE); metadata_store.active_state = AM_INACTIVE; - metadata_hub_modify_epilog(changed); break; case 'pbeg': - metadata_hub_modify_prolog(); changed = ((metadata_store.player_state != PS_PLAYING) || (metadata_store.player_thread_active == 0)); metadata_store.player_state = PS_PLAYING; metadata_store.player_thread_active = 1; - metadata_hub_modify_epilog(changed); break; case 'pend': - metadata_hub_modify_prolog(); changed = ((metadata_store.player_state != PS_STOPPED) || (metadata_store.player_thread_active == 1)); metadata_store.player_state = PS_STOPPED; metadata_store.player_thread_active = 0; - metadata_hub_modify_epilog(changed); break; case 'pfls': - metadata_hub_modify_prolog(); changed = (metadata_store.player_state != PS_PAUSED); metadata_store.player_state = PS_PAUSED; - metadata_hub_modify_epilog(changed); break; case 'pffr': // this is sent when the first frame has been received case 'prsm': - metadata_hub_modify_prolog(); changed = (metadata_store.player_state != PS_PLAYING); metadata_store.player_state = PS_PLAYING; - metadata_hub_modify_epilog(changed); break; case 'pvol': { - metadata_hub_modify_prolog(); // Note: it's assumed that the config.airplay volume has already been correctly set. // int32_t actual_volume; // int gv = dacp_get_volume(&actual_volume); @@ -581,7 +592,6 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin metadata_store.airplay_volume = config.airplay_volume; changed = 1; } - metadata_hub_modify_epilog(changed); // change } break; default: { @@ -604,4 +614,6 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin } } } + pthread_cleanup_pop(0); // don't remove the lock + metadata_hub_modify_epilog(changed); } diff --git a/player.c b/player.c index 8847b8b5..c7020fc7 100644 --- a/player.c +++ b/player.c @@ -2941,20 +2941,18 @@ void player_volume_without_notification(double airplay_volume, rtsp_conn_info *c #ifdef CONFIG_METADATA // here, send the 'pvol' metadata message when the airplay volume information // is being used by shairport sync to control the output volume - char *dv = malloc(128); // will be freed in the metadata thread - if (dv) { - memset(dv, 0, 128); - if (volume_mode == vol_both) { - // normalise the maximum output to the hardware device's max output - snprintf(dv, 127, "%.2f,%.2f,%.2f,%.2f", airplay_volume, - (scaled_attenuation - max_db + hw_max_db) / 100.0, - (min_db - max_db + hw_max_db) / 100.0, (max_db - max_db + hw_max_db) / 100.0); - } else { - snprintf(dv, 127, "%.2f,%.2f,%.2f,%.2f", airplay_volume, scaled_attenuation / 100.0, - min_db / 100.0, max_db / 100.0); - } - send_ssnc_metadata('pvol', dv, strlen(dv), 1); - } + char dv[128]; + memset(dv, 0, 128); + if (volume_mode == vol_both) { + // normalise the maximum output to the hardware device's max output + snprintf(dv, 127, "%.2f,%.2f,%.2f,%.2f", airplay_volume, + (scaled_attenuation - max_db + hw_max_db) / 100.0, + (min_db - max_db + hw_max_db) / 100.0, (max_db - max_db + hw_max_db) / 100.0); + } else { + snprintf(dv, 127, "%.2f,%.2f,%.2f,%.2f", airplay_volume, scaled_attenuation / 100.0, + min_db / 100.0, max_db / 100.0); + } + send_ssnc_metadata('pvol', dv, strlen(dv), 1); #endif if (config.output->mute) @@ -2973,12 +2971,10 @@ void player_volume_without_notification(double airplay_volume, rtsp_conn_info *c else { // here, send the 'pvol' metadata message when the airplay volume information // is being used by shairport sync to control the output volume - char *dv = malloc(128); // will be freed in the metadata thread - if (dv) { - memset(dv, 0, 128); - snprintf(dv, 127, "%.2f,%.2f,%.2f,%.2f", airplay_volume, 0.0, 0.0, 0.0); - send_ssnc_metadata('pvol', dv, strlen(dv), 1); - } + char dv[128]; + memset(dv, 0, 128); + snprintf(dv, 127, "%.2f,%.2f,%.2f,%.2f", airplay_volume, 0.0, 0.0, 0.0); + send_ssnc_metadata('pvol', dv, strlen(dv), 1); } #endif diff --git a/rtp.c b/rtp.c index f7e12c50..71d6df0e 100644 --- a/rtp.c +++ b/rtp.c @@ -998,8 +998,8 @@ void rtp_setup(SOCKADDR *local, SOCKADDR *remote, uint16_t cport, uint16_t tport conn->rtp_running = 1; #ifdef CONFIG_METADATA - send_ssnc_metadata('clip', strdup(conn->client_ip_string), strlen(conn->client_ip_string), 1); - send_ssnc_metadata('svip', strdup(conn->self_ip_string), strlen(conn->self_ip_string), 1); + send_ssnc_metadata('clip', conn->client_ip_string, strlen(conn->client_ip_string), 1); + send_ssnc_metadata('svip', conn->self_ip_string, strlen(conn->self_ip_string), 1); #endif } } diff --git a/rtsp.c b/rtsp.c index 14807f74..d039a161 100644 --- a/rtsp.c +++ b/rtsp.c @@ -114,7 +114,8 @@ typedef struct { pthread_mutex_t pc_queue_lock; pthread_cond_t pc_queue_item_added_signal; pthread_cond_t pc_queue_item_removed_signal; - size_t item_size; // number of bytes in each item + char *name; + size_t item_size; // number of bytes in each item uint32_t count; // number of items in the queue uint32_t capacity; // maximum number of items uint32_t toq; // first item to take @@ -151,7 +152,11 @@ typedef struct { rtsp_message *carrier; } metadata_package; -void pc_queue_init(pc_queue *the_queue, char *items, size_t item_size, uint32_t number_of_items) { +void pc_queue_init(pc_queue *the_queue, char *items, size_t item_size, uint32_t number_of_items, const char* name) { + if (name) + debug(1, "Creating metadata queue \"%s\".", name); + else + debug(1, "Creating an unnamed metadata queue."); pthread_mutex_init(&the_queue->pc_queue_lock, NULL); pthread_cond_init(&the_queue->pc_queue_item_added_signal, NULL); pthread_cond_init(&the_queue->pc_queue_item_removed_signal, NULL); @@ -161,9 +166,19 @@ void pc_queue_init(pc_queue *the_queue, char *items, size_t item_size, uint32_t the_queue->capacity = number_of_items; the_queue->toq = 0; the_queue->eoq = 0; + if (name == NULL) + the_queue->name = NULL; + else + the_queue->name = strdup(name); } void pc_queue_delete(pc_queue *the_queue) { + if (the_queue->name) + debug(1, "Deleting metadata queue \"%s\".", the_queue->name); + else + debug(1, "Deleting an unnamed metadata queue."); + if (the_queue->name != NULL) + free(the_queue->name); pthread_cond_destroy(&the_queue->pc_queue_item_removed_signal); pthread_cond_destroy(&the_queue->pc_queue_item_added_signal); pthread_mutex_destroy(&the_queue->pc_queue_lock); @@ -185,6 +200,7 @@ void pc_queue_cleanup_handler(void *arg) { } int pc_queue_add_item(pc_queue *the_queue, const void *the_stuff, int block) { + int response = 0; int rc; if (the_queue) { if (block == 0) { @@ -196,34 +212,43 @@ int pc_queue_add_item(pc_queue *the_queue, const void *the_stuff, int block) { if (rc) debug(1, "Error locking for pc_queue_add_item"); pthread_cleanup_push(pc_queue_cleanup_handler, (void *)the_queue); - while (the_queue->count == the_queue->capacity) { - rc = pthread_cond_wait(&the_queue->pc_queue_item_removed_signal, &the_queue->pc_queue_lock); - if (rc) - debug(1, "Error waiting for item to be removed"); - } - uint32_t i = the_queue->eoq; - void *p = the_queue->items + the_queue->item_size * i; - // void * p = &the_queue->qbase + the_queue->item_size*the_queue->eoq; - memcpy(p, the_stuff, the_queue->item_size); + // leave this out if you want this to return if the queue is already full + // irrespective of the block flag. + /* + while (the_queue->count == the_queue->capacity) { + rc = pthread_cond_wait(&the_queue->pc_queue_item_removed_signal, &the_queue->pc_queue_lock); + if (rc) + debug(1, "Error waiting for item to be removed"); + } + */ + if (the_queue->count < the_queue->capacity) { + uint32_t i = the_queue->eoq; + void *p = the_queue->items + the_queue->item_size * i; + // void * p = &the_queue->qbase + the_queue->item_size*the_queue->eoq; + memcpy(p, the_stuff, the_queue->item_size); - // update the pointer - i++; - if (i == the_queue->capacity) - // fold pointer if necessary - i = 0; - the_queue->eoq = i; - the_queue->count++; - debug(2,"metadata queue+ %d/%d.", the_queue->count, the_queue->capacity); - if (the_queue->count == the_queue->capacity) - debug(1, "pc_queue is full with %d items in it!", the_queue->count); - rc = pthread_cond_signal(&the_queue->pc_queue_item_added_signal); - if (rc) - debug(1, "Error signalling after pc_queue_add_item"); + // update the pointer + i++; + if (i == the_queue->capacity) + // fold pointer if necessary + i = 0; + the_queue->eoq = i; + the_queue->count++; + //debug(2,"metadata queue+ \"%s\" %d/%d.", the_queue->name, the_queue->count, the_queue->capacity); + if (the_queue->count == the_queue->capacity) + debug(3, "metadata queue \"%s\": is now full with %d items in it!", the_queue->name, the_queue->count); + rc = pthread_cond_signal(&the_queue->pc_queue_item_added_signal); + if (rc) + debug(1, "metadata queue \"%s\": error signalling after pc_queue_add_item", the_queue->name); + } else { + response = EWOULDBLOCK; // a bit arbitrary, this. + debug(3,"metadata queue \"%s\": is already full with %d items in it. Not adding this item to the queue.", the_queue->name, the_queue->count); + } pthread_cleanup_pop(1); // unlock the queue lock. } else { debug(1, "Adding an item to a NULL queue"); } - return 0; + return response; } int pc_queue_get_item(pc_queue *the_queue, void *the_stuff) { @@ -231,12 +256,12 @@ int pc_queue_get_item(pc_queue *the_queue, void *the_stuff) { if (the_queue) { rc = pthread_mutex_lock(&the_queue->pc_queue_lock); if (rc) - debug(1, "Error locking for pc_queue_get_item"); + debug(1, "metadata queue \"%s\": error locking for pc_queue_get_item", the_queue->name); pthread_cleanup_push(pc_queue_cleanup_handler, (void *)the_queue); while (the_queue->count == 0) { rc = pthread_cond_wait(&the_queue->pc_queue_item_added_signal, &the_queue->pc_queue_lock); if (rc) - debug(1, "Error waiting for item to be added"); + debug(1, "metadata queue \"%s\": error waiting for item to be added", the_queue->name); } uint32_t i = the_queue->toq; // void * p = &the_queue->qbase + the_queue->item_size*the_queue->toq; @@ -250,10 +275,10 @@ int pc_queue_get_item(pc_queue *the_queue, void *the_stuff) { i = 0; the_queue->toq = i; the_queue->count--; - debug(2,"metadata queue- %d/%d.", the_queue->count, the_queue->capacity); + debug(3,"metadata queue- \"%s\" %d/%d.", the_queue->name, the_queue->count, the_queue->capacity); rc = pthread_cond_signal(&the_queue->pc_queue_item_removed_signal); if (rc) - debug(1, "Error signalling after pc_queue_removed_item"); + debug(1, "metadata queue \"%s\": error signalling after pc_queue_get_item", the_queue->name); pthread_cleanup_pop(1); // unlock the queue lock. } else { debug(1, "Removing an item from a NULL queue"); @@ -417,6 +442,7 @@ void msg_retain(rtsp_message *msg) { debug(1, "Error %d locking reference counter lock"); if (msg > (rtsp_message *)0x00010000) { msg->referenceCount++; + debug(3,"msg_free increment reference counter message %d to %d.", msg->index_number, msg->referenceCount); // debug(1,"msg_retain -- item %d reference count %d.", msg->index_number, msg->referenceCount); rc = pthread_mutex_unlock(&reference_counter_lock); if (rc) @@ -432,6 +458,7 @@ rtsp_message *msg_init(void) { memset(msg, 0, sizeof(rtsp_message)); msg->referenceCount = 1; // from now on, any access to this must be protected with the lock msg->index_number = msg_indexes++; + debug(3,"msg_init message %d", msg->index_number); } else { die("msg_init -- can not allocate memory for rtsp_message %d.", msg_indexes); } @@ -495,6 +522,8 @@ void msg_free(rtsp_message **msgh) { if (*msgh > (rtsp_message *)0x00010000) { rtsp_message *msg = *msgh; msg->referenceCount--; + if (msg->referenceCount) + debug(3,"msg_free decrement reference counter message %d to %d", msg->index_number, msg->referenceCount); if (msg->referenceCount == 0) { unsigned int i; for (i = 0; i < msg->nheaders; i++) { @@ -509,6 +538,7 @@ void msg_free(rtsp_message **msgh) { index = 0x10000; // ensure it doesn't fold to zero. *msgh = (rtsp_message *)(index); // put a version of the index number of the freed message in here + debug(3,"msg_free freed message %d", msg->index_number); free(msg); } else { // debug(1,"msg_free item %d -- decrement reference to @@ -1072,7 +1102,7 @@ void handle_set_parameter_parameter(rtsp_conn_info *conn, rtsp_message *req, char *progress = cp + strlen("progress: "); // debug(2, "progress: \"%s\"",progress); // rtpstampstart/rtpstampnow/rtpstampend 44100 per // second - send_ssnc_metadata('prgr', strdup(progress), strlen(progress), 1); + send_ssnc_metadata('prgr', progress, strlen(progress), 1); } else #endif @@ -1220,14 +1250,28 @@ char *base64_encode_so(const unsigned char *data, size_t input_length, char *enc static int fd = -1; // static int dirty = 0; pc_queue metadata_queue; + static int metadata_sock = -1; static struct sockaddr_in metadata_sockaddr; static char *metadata_sockmsg; #define metadata_queue_size 500 metadata_package metadata_queue_items[metadata_queue_size]; - pthread_t metadata_thread; +#ifdef CONFIG_METADATA_HUB +pc_queue metadata_hub_queue; +#define metadata_hub_queue_size 500 +metadata_package metadata_hub_queue_items[metadata_hub_queue_size]; +pthread_t metadata_hub_thread; +#endif + +#ifdef CONFIG_MQTT +pc_queue metadata_mqtt_queue; +#define metadata_mqtt_queue_size 500 +metadata_package metadata_mqtt_queue_items[metadata_mqtt_queue_size]; +pthread_t metadata_mqtt_thread; +#endif + void metadata_create_multicast_socket(void) { if (config.metadata_enabled == 0) return; @@ -1424,13 +1468,6 @@ void metadata_process(uint32_t type, uint32_t code, char *data, uint32_t length) } } -void metadata_thread_cleanup_function(__attribute__((unused)) void *arg) { - debug(2, "metadata_thread_cleanup_function called"); - metadata_delete_multicast_socket(); - metadata_close(); - pc_queue_delete(&metadata_queue); -} - void metadata_pack_cleanup_function(void *arg) { // debug(1, "metadata_pack_cleanup_function called"); metadata_package *pack = (metadata_package *)arg; @@ -1440,32 +1477,31 @@ void metadata_pack_cleanup_function(void *arg) { free(pack->data); } +void metadata_thread_cleanup_function(__attribute__((unused)) void *arg) { + debug(2, "metadata_thread_cleanup_function called"); + metadata_delete_multicast_socket(); + metadata_close(); + pc_queue_delete(&metadata_queue); +} + void *metadata_thread_function(__attribute__((unused)) void *ignore) { // create a pc_queue for passing information to a threaded metadata handler pc_queue_init(&metadata_queue, (char *)&metadata_queue_items, sizeof(metadata_package), - metadata_queue_size); + metadata_queue_size, "pipe"); metadata_create_multicast_socket(); metadata_package pack; pthread_cleanup_push(metadata_thread_cleanup_function, NULL); while (1) { pc_queue_get_item(&metadata_queue, &pack); - // debug(1,"pc_queue get item."); pthread_cleanup_push(metadata_pack_cleanup_function, (void *)&pack); if (config.metadata_enabled) { - // debug(1, "metadata_process type %x, code %x and length %u.", pack.type, pack.code, pack.length); - // metadata_process(pack.type, pack.code, pack.data, pack.length); -#ifdef CONFIG_METADATA_HUB - debug(1, "metadata_hub_process."); - metadata_hub_process_metadata(pack.type, pack.code, pack.data, pack.length); -#endif - -#ifdef CONFIG_MQTT - if (config.mqtt_enabled) { - debug(1, "mqtt_process_metadata."); - mqtt_process_metadata(pack.type, pack.code, pack.data, pack.length); - } -#endif - debug(1, "done."); + if (pack.carrier) { + debug(3, " pipe: type %x, code %x, length %u, message %d.", pack.type, pack.code, pack.length, pack.carrier->index_number); + } else { + debug(3, " pipe: type %x, code %x, length %u.", pack.type, pack.code, pack.length); + } + metadata_process(pack.type, pack.code, pack.data, pack.length); + debug(3, " pipe: done."); } pthread_cleanup_pop(1); } @@ -1473,22 +1509,109 @@ void *metadata_thread_function(__attribute__((unused)) void *ignore) { pthread_exit(NULL); } +#ifdef CONFIG_METADATA_HUB +void metadata_hub_close(void) { +} + +void metadata_hub_thread_cleanup_function(__attribute__((unused)) void *arg) { + debug(2, "metadata_hub_thread_cleanup_function called"); + metadata_hub_close(); + pc_queue_delete(&metadata_hub_queue); +} + +void *metadata_hub_thread_function(__attribute__((unused)) void *ignore) { + // create a pc_queue for passing information to a threaded metadata handler + pc_queue_init(&metadata_hub_queue, (char *)&metadata_hub_queue_items, sizeof(metadata_package), + metadata_hub_queue_size, "hub"); + metadata_package pack; + pthread_cleanup_push(metadata_hub_thread_cleanup_function, NULL); + while (1) { + pc_queue_get_item(&metadata_hub_queue, &pack); + pthread_cleanup_push(metadata_pack_cleanup_function, (void *)&pack); + if (pack.carrier) { + debug(3, " hub: type %x, code %x, length %u, message %d.", pack.type, pack.code, pack.length, pack.carrier->index_number); + } else { + debug(3, " hub: type %x, code %x, length %u.", pack.type, pack.code, pack.length); + } + metadata_hub_process_metadata(pack.type, pack.code, pack.data, pack.length); + debug(3, " hub: done."); + pthread_cleanup_pop(1); + } + pthread_cleanup_pop(1); // will never happen + pthread_exit(NULL); +} +#endif + +#ifdef CONFIG_MQTT +void metadata_mqtt_close(void) { +} + +void metadata_mqtt_thread_cleanup_function(__attribute__((unused)) void *arg) { + debug(2, "metadata_mqtt_thread_cleanup_function called"); + metadata_mqtt_close(); + pc_queue_delete(&metadata_hub_queue); +} + +void *metadata_mqtt_thread_function(__attribute__((unused)) void *ignore) { + // create a pc_queue for passing information to a threaded metadata handler + pc_queue_init(&metadata_mqtt_queue, (char *)&metadata_mqtt_queue_items, sizeof(metadata_package), + metadata_mqtt_queue_size, "mqtt"); + metadata_package pack; + pthread_cleanup_push(metadata_mqtt_thread_cleanup_function, NULL); + while (1) { + pc_queue_get_item(&metadata_mqtt_queue, &pack); + pthread_cleanup_push(metadata_pack_cleanup_function, (void *)&pack); + if (config.mqtt_enabled) { + if (pack.carrier) { + debug(3, " mqtt: type %x, code %x, length %u, message %d.", pack.type, pack.code, pack.length, pack.carrier->index_number); + } else { + debug(3, " mqtt: type %x, code %x, length %u.", pack.type, pack.code, pack.length); + } + mqtt_process_metadata(pack.type, pack.code, pack.data, pack.length); + debug(3, " mqtt: done."); + } + + pthread_cleanup_pop(1); + } + pthread_cleanup_pop(1); // will never happen + pthread_exit(NULL); +} +#endif + void metadata_init(void) { int ret = pthread_create(&metadata_thread, NULL, metadata_thread_function, NULL); if (ret) debug(1, "Failed to create metadata thread!"); +#ifdef CONFIG_METADATA_HUB + ret = pthread_create(&metadata_hub_thread, NULL, metadata_hub_thread_function, NULL); + if (ret) + debug(1, "Failed to create metadata hub thread!"); +#endif +#ifdef CONFIG_MQTT + ret = pthread_create(&metadata_mqtt_thread, NULL, metadata_mqtt_thread_function, NULL); + if (ret) + debug(1, "Failed to create metadata mqtt thread!"); +#endif metadata_running = 1; } void metadata_stop(void) { if (metadata_running) { debug(2, "metadata_stop called."); +#ifdef CONFIG_MQTT + pthread_join(metadata_mqtt_thread, NULL); + pthread_cancel(metadata_mqtt_thread); +#endif +#ifdef CONFIG_METADATA_HUB + pthread_join(metadata_hub_thread, NULL); + pthread_cancel(metadata_hub_thread); +#endif pthread_cancel(metadata_thread); pthread_join(metadata_thread, NULL); } } -int send_metadata(uint32_t type, uint32_t code, char *data, uint32_t length, rtsp_message *carrier, +int send_metadata_to_queue(pc_queue* queue, uint32_t type, uint32_t code, char *data, uint32_t length, rtsp_message *carrier, int block) { // parameters: type, code, pointer to data or NULL, length of data or NULL, @@ -1505,37 +1628,64 @@ int send_metadata(uint32_t type, uint32_t code, char *data, uint32_t length, rts // The reading of the parameters is a bit complex // If the rtsp_message field is non-null, then it represents an rtsp_message - // which should be freed - // in the thread handler when the parameter pointed to by the pointer and - // specified by the length - // is finished with - // If the rtsp_message is NULL, then if the pointer is non-null, it points to - // a malloc'ed block - // and should be freed when the thread is finished with it. The length of the - // data in the block is - // given in length + // and the data pointer is assumed to point to something within it. + // The reference counter of the rtsp_message is incremented here and + // should be decremented by the metadata handler when finished. + // If the reference count reduces to zero, the message will be freed. + + // If the rtsp_message is NULL, then if the pointer is non-null then the data it + // points to, of the length specified, is memcpy'd and passed to the metadata + // handler. The handler should free it when done. // If the rtsp_message is NULL and the pointer is also NULL, nothing further // is done. metadata_package pack; pack.type = type; pack.code = code; - pack.data = data; pack.length = length; pack.carrier = carrier; - if (pack.carrier) + pack.data = data; + if (pack.carrier) { msg_retain(pack.carrier); - int rc = pc_queue_add_item(&metadata_queue, &pack, block); - if (rc == EBUSY) { - if (pack.carrier) + } else { + if (data) + pack.data = memdup(data,length); // only if it's not a null + } + int rc = pc_queue_add_item(queue, &pack, block); + if (rc != 0) { + if (pack.carrier) { + if (rc == EWOULDBLOCK) + debug(2, "metadata queue \"%s\" full, dropping message item: type %x, code %x, data %x, length %u, message %d.", queue->name, pack.type, pack.code, pack.data, pack.length, pack.carrier->index_number); msg_free(&pack.carrier); - else if (data) - free(data); - warn("Metadata queue is busy, discarding message of type 0x%08X, code 0x%08X.", type, code); + } else { + if (rc == EWOULDBLOCK) + debug(2, "metadata queue \"%s\" full, dropping data item: type %x, code %x, data %x, length %u.", queue->name, pack.type, pack.code, pack.data, pack.length); + if (pack.data) + free(pack.data); + } } return rc; } +int send_metadata(uint32_t type, uint32_t code, char *data, uint32_t length, rtsp_message *carrier, + int block) { + int rc; + rc = send_metadata_to_queue(&metadata_queue, type, code, data, length, carrier, block); + +#ifdef CONFIG_METADATA_HUB + rc = send_metadata_to_queue(&metadata_hub_queue, type, code, data, length, carrier, block); +#endif + +#ifdef CONFIG_MQTT + rc = send_metadata_to_queue(&metadata_mqtt_queue, type, code, data, length, carrier, block); +#endif + + return rc; +} + + + + static void handle_set_parameter_metadata(__attribute__((unused)) rtsp_conn_info *conn, rtsp_message *req, __attribute__((unused)) rtsp_message *resp) { From 3db2b05654445f90b15e48ed6bdb18a5aba3ec0d Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 30 May 2020 17:41:16 +0100 Subject: [PATCH 118/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 9ce1afad..72e0470b 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,18 @@ +Version 3.3.7d7 +==== +**Enhancements** +* Logs and statistics can now be directed to the system log (default), `stdout`, `stderr` or to a file or pipe of your choice using a new setting, `log_output_to` in the `diagnostics` section of the configuration file. This is very useful when the system log is disabled or diverted. +* Audio data from the `pipe` back end, and metadata from the metadata pipe are now written using standard blocking `write` commands rather than a slightly complex non-blocking write function. Pipes are opend in non-blocking mode and chaned to blocking mode when successfully opened. +* Separate threads for the metadata subsystems. Until now, all metadata was processed on a single thread. This included supplying the metadata pipe and the multicast stream and supplying metadata for the mqtt interface and for the `dbus` and `MPRIS` interfaces. Unfortunately, that meant that a problem with any one of these subsystems could propagate into the others. Now they all run on separate threads (almost -- the metadata pipe and multicast are still on one thread). If a subsystem blocks, it will only block itself and will not interfere with the other subsystems. + +**Bug Fixes and Enhancements** +* Fixed a bug calculating the instantaneous synchronisation error. This bug could occasionally cause Shairport Sync to lose synchronisation and maybe even to mute for a few seconds before regaining it. It was caused by not doing modulo arithmetic properly, and it's been there for a while. +* Cleaned up and improve the code to synchronise the first frame of audio. This should result in more accurate and reliable initial synchronisation, usually to under a millisecond, and often to within 20 or 30 microseconds in normal usage. Sync should improve even when the silent lead-in time is as short as 0.3 seconds or when the `audio_backend_latency_offset_in_seconds` is as great as -1.7 seconds. +* Removed a bug which would affect initial synchronisation if a `FLUSH` command was received from the player at an inopportune time. +* Cleaned up some confused uses of modulo arithmetic. +* Cleaned up allocation of memory for gathering running statistics – use the heap instead of the stack. +* Cleaned up the display of statistics for backends that do not implement active synchronisation, e.g. the `pipe` and `STDOUT` back ends. + Version 3.3.7d6 ==== **Bug Fixes and Enhancements** From 18b9116d97554363d5fa3b261932b5d68a1db177 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 2 Jun 2020 19:37:05 +0100 Subject: [PATCH 119/186] Separate metadata multicast from metadata pipe. --- metadata_hub.c | 3 ++- metadata_hub.h | 2 ++ rtsp.c | 69 +++++++++++++++++++++++++++++++++++++++++++------- 3 files changed, 64 insertions(+), 10 deletions(-) diff --git a/metadata_hub.c b/metadata_hub.c index 2ea442a5..3dc42c5a 100644 --- a/metadata_hub.c +++ b/metadata_hub.c @@ -172,6 +172,7 @@ void _metadata_hub_modify_epilog(int modified, const char *filename, const int l debug(3, "Metadata_hub write lock unlocked."); } +/* void _metadata_hub_read_prolog(const char *filename, const int linenumber) { // always run this before reading an entry or a sequence of entries in the metadata_hub // debug(1, "locking metadata hub for reading"); @@ -187,7 +188,7 @@ void _metadata_hub_read_epilog(const char *filename, const int linenumber) { // debug(1, "unlocking metadata hub for reading"); pthread_rwlock_unlock(&metadata_hub_re_lock); } - +*/ char *metadata_write_image_file(const char *buf, int len) { // warning -- this removes all files from the directory apart from this one, if it exists diff --git a/metadata_hub.h b/metadata_hub.h index 8ade9b86..760a848b 100644 --- a/metadata_hub.h +++ b/metadata_hub.h @@ -151,9 +151,11 @@ void metadata_hub_release_track_artwork(void); void _metadata_hub_modify_prolog(const char *filename, const int linenumber); void _metadata_hub_modify_epilog(int modified, const char *filename, const int linenumber); // set to true if modifications occurred, 0 otherwise +/* // these are for safe reading void _metadata_hub_read_prolog(const char *filename, const int linenumber); void _metadata_hub_read_epilog(const char *filename, const int linenumber); +*/ #define metadata_hub_modify_prolog(void) _metadata_hub_modify_prolog(__FILE__, __LINE__) #define metadata_hub_modify_epilog(modified) _metadata_hub_modify_epilog(modified, __FILE__, __LINE__) diff --git a/rtsp.c b/rtsp.c index d039a161..55789a12 100644 --- a/rtsp.c +++ b/rtsp.c @@ -1249,11 +1249,10 @@ char *base64_encode_so(const unsigned char *data, size_t input_length, char *enc static int fd = -1; // static int dirty = 0; -pc_queue metadata_queue; -static int metadata_sock = -1; -static struct sockaddr_in metadata_sockaddr; -static char *metadata_sockmsg; + + +pc_queue metadata_queue; #define metadata_queue_size 500 metadata_package metadata_queue_items[metadata_queue_size]; pthread_t metadata_thread; @@ -1272,6 +1271,15 @@ metadata_package metadata_mqtt_queue_items[metadata_mqtt_queue_size]; pthread_t metadata_mqtt_thread; #endif +static int metadata_sock = -1; +static struct sockaddr_in metadata_sockaddr; +static char *metadata_sockmsg; +pc_queue metadata_multicast_queue; +#define metadata_multicast_queue_size 500 +metadata_package metadata_multicast_queue_items[metadata_queue_size]; +pthread_t metadata_multicast_thread; + + void metadata_create_multicast_socket(void) { if (config.metadata_enabled == 0) return; @@ -1322,6 +1330,8 @@ void metadata_delete_multicast_socket(void) { free(metadata_sockmsg); } + + void metadata_open(void) { if (config.metadata_enabled == 0) return; @@ -1342,10 +1352,8 @@ static void metadata_close(void) { fd = -1; } -void metadata_process(uint32_t type, uint32_t code, char *data, uint32_t length) { - // debug(1, "Process metadata with type %x, code %x and length %u.", type, code, length); - int ret = 0; - +void metadata_multicast_process(uint32_t type, uint32_t code, char *data, uint32_t length) { + // debug(1, "Process multicast metadata with type %x, code %x and length %u.", type, code, length); if (metadata_sock >= 0 && length < config.metadata_sockmsglength - 8) { char *ptr = metadata_sockmsg; uint32_t v; @@ -1400,7 +1408,11 @@ void metadata_process(uint32_t type, uint32_t code, char *data, uint32_t length) break; } while (1); } +} +void metadata_process(uint32_t type, uint32_t code, char *data, uint32_t length) { + // debug(1, "Process metadata with type %x, code %x and length %u.", type, code, length); + int ret = 0; // readers may go away and come back if (fd < 0) @@ -1479,7 +1491,6 @@ void metadata_pack_cleanup_function(void *arg) { void metadata_thread_cleanup_function(__attribute__((unused)) void *arg) { debug(2, "metadata_thread_cleanup_function called"); - metadata_delete_multicast_socket(); metadata_close(); pc_queue_delete(&metadata_queue); } @@ -1509,6 +1520,38 @@ void *metadata_thread_function(__attribute__((unused)) void *ignore) { pthread_exit(NULL); } +void metadata_multicast_thread_cleanup_function(__attribute__((unused)) void *arg) { + debug(2, "metadata_multicast_thread_cleanup_function called"); + metadata_delete_multicast_socket(); + pc_queue_delete(&metadata_multicast_queue); +} + +void *metadata_multicast_thread_function(__attribute__((unused)) void *ignore) { + // create a pc_queue for passing information to a threaded metadata handler + pc_queue_init(&metadata_multicast_queue, (char *)&metadata_multicast_queue_items, sizeof(metadata_package), + metadata_multicast_queue_size, "multicast"); + metadata_create_multicast_socket(); + metadata_package pack; + pthread_cleanup_push(metadata_multicast_thread_cleanup_function, NULL); + while (1) { + pc_queue_get_item(&metadata_multicast_queue, &pack); + pthread_cleanup_push(metadata_pack_cleanup_function, (void *)&pack); + if (config.metadata_enabled) { + if (pack.carrier) { + debug(3, " multicast: type %x, code %x, length %u, message %d.", pack.type, pack.code, pack.length, pack.carrier->index_number); + } else { + debug(3, " multicast: type %x, code %x, length %u.", pack.type, pack.code, pack.length); + } + metadata_multicast_process(pack.type, pack.code, pack.data, pack.length); + debug(3, " multicast: done."); + } + pthread_cleanup_pop(1); + } + pthread_cleanup_pop(1); // will never happen + pthread_exit(NULL); +} + + #ifdef CONFIG_METADATA_HUB void metadata_hub_close(void) { } @@ -1582,6 +1625,11 @@ void metadata_init(void) { int ret = pthread_create(&metadata_thread, NULL, metadata_thread_function, NULL); if (ret) debug(1, "Failed to create metadata thread!"); + + ret = pthread_create(&metadata_multicast_thread, NULL, metadata_multicast_thread_function, NULL); + if (ret) + debug(1, "Failed to create metadata multicast thread!"); + #ifdef CONFIG_METADATA_HUB ret = pthread_create(&metadata_hub_thread, NULL, metadata_hub_thread_function, NULL); if (ret) @@ -1606,6 +1654,9 @@ void metadata_stop(void) { pthread_join(metadata_hub_thread, NULL); pthread_cancel(metadata_hub_thread); #endif + pthread_cancel(metadata_multicast_thread); + pthread_join(metadata_multicast_thread, NULL); + pthread_cancel(metadata_thread); pthread_join(metadata_thread, NULL); } From 89ed826514d6e011a3db65c1bdec567f19cdd90a Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 2 Jun 2020 19:38:55 +0100 Subject: [PATCH 120/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 72e0470b..46be45a3 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -3,7 +3,7 @@ Version 3.3.7d7 **Enhancements** * Logs and statistics can now be directed to the system log (default), `stdout`, `stderr` or to a file or pipe of your choice using a new setting, `log_output_to` in the `diagnostics` section of the configuration file. This is very useful when the system log is disabled or diverted. * Audio data from the `pipe` back end, and metadata from the metadata pipe are now written using standard blocking `write` commands rather than a slightly complex non-blocking write function. Pipes are opend in non-blocking mode and chaned to blocking mode when successfully opened. -* Separate threads for the metadata subsystems. Until now, all metadata was processed on a single thread. This included supplying the metadata pipe and the multicast stream and supplying metadata for the mqtt interface and for the `dbus` and `MPRIS` interfaces. Unfortunately, that meant that a problem with any one of these subsystems could propagate into the others. Now they all run on separate threads (almost -- the metadata pipe and multicast are still on one thread). If a subsystem blocks, it will only block itself and will not interfere with the other subsystems. +* Separate threads for the metadata subsystems. Until now, all metadata was processed on a single thread. This included supplying the metadata pipe and the multicast stream and supplying metadata for the mqtt interface and for the `dbus` and `MPRIS` interfaces. Unfortunately, that meant that a problem with any one of these subsystems could propagate into the others. Now they all run on separate threads. If one thread blocks, it will not interfere with the other subsystems. **Bug Fixes and Enhancements** * Fixed a bug calculating the instantaneous synchronisation error. This bug could occasionally cause Shairport Sync to lose synchronisation and maybe even to mute for a few seconds before regaining it. It was caused by not doing modulo arithmetic properly, and it's been there for a while. From dfe48c28feeef27fb5a46815b2d584e539d45f7b Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 2 Jun 2020 19:41:53 +0100 Subject: [PATCH 121/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 46be45a3..02730215 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -2,11 +2,11 @@ Version 3.3.7d7 ==== **Enhancements** * Logs and statistics can now be directed to the system log (default), `stdout`, `stderr` or to a file or pipe of your choice using a new setting, `log_output_to` in the `diagnostics` section of the configuration file. This is very useful when the system log is disabled or diverted. -* Audio data from the `pipe` back end, and metadata from the metadata pipe are now written using standard blocking `write` commands rather than a slightly complex non-blocking write function. Pipes are opend in non-blocking mode and chaned to blocking mode when successfully opened. +* Audio data from the `pipe` back end and metadata from the metadata pipe are now written using standard blocking `write` commands rather than a slightly complex non-blocking write function. Pipes are now opened in non-blocking mode and changed to blocking mode when successfully opened. * Separate threads for the metadata subsystems. Until now, all metadata was processed on a single thread. This included supplying the metadata pipe and the multicast stream and supplying metadata for the mqtt interface and for the `dbus` and `MPRIS` interfaces. Unfortunately, that meant that a problem with any one of these subsystems could propagate into the others. Now they all run on separate threads. If one thread blocks, it will not interfere with the other subsystems. **Bug Fixes and Enhancements** -* Fixed a bug calculating the instantaneous synchronisation error. This bug could occasionally cause Shairport Sync to lose synchronisation and maybe even to mute for a few seconds before regaining it. It was caused by not doing modulo arithmetic properly, and it's been there for a while. +* Fixed a bug calculating the instantaneous synchronisation error. This bug could occasionally cause Shairport Sync to lose synchronisation and maybe even to mute for a few seconds before resynchronising. It was caused doing modulo arithmetic incorrectly and it's been there for a while. * Cleaned up and improve the code to synchronise the first frame of audio. This should result in more accurate and reliable initial synchronisation, usually to under a millisecond, and often to within 20 or 30 microseconds in normal usage. Sync should improve even when the silent lead-in time is as short as 0.3 seconds or when the `audio_backend_latency_offset_in_seconds` is as great as -1.7 seconds. * Removed a bug which would affect initial synchronisation if a `FLUSH` command was received from the player at an inopportune time. * Cleaned up some confused uses of modulo arithmetic. From 72e5274fca4f6a215596b0743b29ecdea33e6e40 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 2 Jun 2020 19:43:01 +0100 Subject: [PATCH 122/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 02730215..abc9cb07 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -3,7 +3,7 @@ Version 3.3.7d7 **Enhancements** * Logs and statistics can now be directed to the system log (default), `stdout`, `stderr` or to a file or pipe of your choice using a new setting, `log_output_to` in the `diagnostics` section of the configuration file. This is very useful when the system log is disabled or diverted. * Audio data from the `pipe` back end and metadata from the metadata pipe are now written using standard blocking `write` commands rather than a slightly complex non-blocking write function. Pipes are now opened in non-blocking mode and changed to blocking mode when successfully opened. -* Separate threads for the metadata subsystems. Until now, all metadata was processed on a single thread. This included supplying the metadata pipe and the multicast stream and supplying metadata for the mqtt interface and for the `dbus` and `MPRIS` interfaces. Unfortunately, that meant that a problem with any one of these subsystems could propagate into the others. Now they all run on separate threads. If one thread blocks, it will not interfere with the other subsystems. +* Separate threads for the metadata subsystems. Until now, all metadata was processed on a single thread. This included supplying the metadata pipe and the multicast stream and supplying metadata for the `mqtt` interface and for the `dbus` and `MPRIS` interfaces. Unfortunately, that meant that a problem with any one of these subsystems could propagate into the others. Now they all run on separate threads. If one thread blocks, it will not interfere with the other subsystems. **Bug Fixes and Enhancements** * Fixed a bug calculating the instantaneous synchronisation error. This bug could occasionally cause Shairport Sync to lose synchronisation and maybe even to mute for a few seconds before resynchronising. It was caused doing modulo arithmetic incorrectly and it's been there for a while. From 36eba7079a06e1d758fa23da6edb88e18b4cc386 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 2 Jun 2020 19:43:37 +0100 Subject: [PATCH 123/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index abc9cb07..3b491b5f 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -7,7 +7,7 @@ Version 3.3.7d7 **Bug Fixes and Enhancements** * Fixed a bug calculating the instantaneous synchronisation error. This bug could occasionally cause Shairport Sync to lose synchronisation and maybe even to mute for a few seconds before resynchronising. It was caused doing modulo arithmetic incorrectly and it's been there for a while. -* Cleaned up and improve the code to synchronise the first frame of audio. This should result in more accurate and reliable initial synchronisation, usually to under a millisecond, and often to within 20 or 30 microseconds in normal usage. Sync should improve even when the silent lead-in time is as short as 0.3 seconds or when the `audio_backend_latency_offset_in_seconds` is as great as -1.7 seconds. +* Cleaned up and improve the code to synchronise the first frame of audio. This should result in more accurate and reliable initial synchronisation, usually to under a millisecond, and often to within 20 or 30 microseconds in normal usage. Sync should improve even when the silent lead-in time is as short as 0.3 seconds or when the `audio_backend_latency_offset_in_seconds` is as much as -1.7 seconds. * Removed a bug which would affect initial synchronisation if a `FLUSH` command was received from the player at an inopportune time. * Cleaned up some confused uses of modulo arithmetic. * Cleaned up allocation of memory for gathering running statistics – use the heap instead of the stack. From fb60c8fbf5ef617578ab925d2f1c2cbe6112331f Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 2 Jun 2020 19:45:11 +0100 Subject: [PATCH 124/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 3b491b5f..fa731e9a 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -5,6 +5,8 @@ Version 3.3.7d7 * Audio data from the `pipe` back end and metadata from the metadata pipe are now written using standard blocking `write` commands rather than a slightly complex non-blocking write function. Pipes are now opened in non-blocking mode and changed to blocking mode when successfully opened. * Separate threads for the metadata subsystems. Until now, all metadata was processed on a single thread. This included supplying the metadata pipe and the multicast stream and supplying metadata for the `mqtt` interface and for the `dbus` and `MPRIS` interfaces. Unfortunately, that meant that a problem with any one of these subsystems could propagate into the others. Now they all run on separate threads. If one thread blocks, it will not interfere with the other subsystems. +Version 3.3.7d6 +==== **Bug Fixes and Enhancements** * Fixed a bug calculating the instantaneous synchronisation error. This bug could occasionally cause Shairport Sync to lose synchronisation and maybe even to mute for a few seconds before resynchronising. It was caused doing modulo arithmetic incorrectly and it's been there for a while. * Cleaned up and improve the code to synchronise the first frame of audio. This should result in more accurate and reliable initial synchronisation, usually to under a millisecond, and often to within 20 or 30 microseconds in normal usage. Sync should improve even when the silent lead-in time is as short as 0.3 seconds or when the `audio_backend_latency_offset_in_seconds` is as much as -1.7 seconds. @@ -13,17 +15,6 @@ Version 3.3.7d7 * Cleaned up allocation of memory for gathering running statistics – use the heap instead of the stack. * Cleaned up the display of statistics for backends that do not implement active synchronisation, e.g. the `pipe` and `STDOUT` back ends. -Version 3.3.7d6 -==== -**Bug Fixes and Enhancements** -* Fixed a bug calculating the instantaneous synchronisation error. This bug could occasionally cause Shairport Sync to lose synchronisation and maybe even to mute for a few seconds before regaining it. It was caused by not doing modulo arithmetic properly, and it's been there for a while. -* Cleaned up and improve the code to synchronise the first frame of audio. This should result in more accurate and reliable initial synchronisation, usually to under a millisecond, and often to within 20 or 30 microseconds in normal usage. Sync should improve even when the silent lead-in time is as short as 0.3 seconds or when the `audio_backend_latency_offset_in_seconds` is as great as -1.7 seconds. -* Removed a bug which would affect initial synchronisation if a `FLUSH` command was received from the player at an inopportune time. -* Cleaned up some confused uses of modulo arithmetic. -* Cleaned up allocation of memory for gathering running statistics – use the heap instead of the stack. -* Cleaned up the display of statistics for backends that do not implement active synchronisation, e.g. the `pipe` and `STDOUT` back ends. -* Added code to flush output to the `pipe` or to `STDOUT` when playing stops. - Version 3.3.7d5 ==== **Bug Fixes and Enhancements** From a4d59275272c287989501ede6e9267d975264067 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 2 Jun 2020 19:45:57 +0100 Subject: [PATCH 125/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index fa731e9a..81b72eff 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -3,7 +3,7 @@ Version 3.3.7d7 **Enhancements** * Logs and statistics can now be directed to the system log (default), `stdout`, `stderr` or to a file or pipe of your choice using a new setting, `log_output_to` in the `diagnostics` section of the configuration file. This is very useful when the system log is disabled or diverted. * Audio data from the `pipe` back end and metadata from the metadata pipe are now written using standard blocking `write` commands rather than a slightly complex non-blocking write function. Pipes are now opened in non-blocking mode and changed to blocking mode when successfully opened. -* Separate threads for the metadata subsystems. Until now, all metadata was processed on a single thread. This included supplying the metadata pipe and the multicast stream and supplying metadata for the `mqtt` interface and for the `dbus` and `MPRIS` interfaces. Unfortunately, that meant that a problem with any one of these subsystems could propagate into the others. Now they all run on separate threads. If one thread blocks, it will not interfere with the other subsystems. +* Separate threads for the metadata subsystems. Until now, all metadata was processed on a single thread. This included writing to the metadata pipe and the multicast stream and supplying metadata for the `mqtt` interface and for the `dbus` and `MPRIS` interfaces. Unfortunately, that meant that a problem with any one of these subsystems could propagate into the others. Now they all run on separate threads. If one thread blocks, it will not interfere with the other subsystems. Version 3.3.7d6 ==== From dfd90bac6bd5528c08a32dbfbf7be6ac598cc5d5 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 5 Jun 2020 17:22:53 +0100 Subject: [PATCH 126/186] Clean up code for the silent lead-in on a non-syncing back end e.g. a pipe. Clean up pipe creation and initial opening and associated error messages. Fix error with warning/die messages. Tidy up some debug messages. --- audio_pipe.c | 16 ++- common.c | 18 +-- dbus-service.c | 4 +- mpris-service.c | 2 +- mqtt.c | 10 +- player.c | 340 ++++++++++++++++++++++++++---------------------- rtsp.c | 41 ++++-- 7 files changed, 243 insertions(+), 188 deletions(-) diff --git a/audio_pipe.c b/audio_pipe.c index a2b33349..7953dc23 100644 --- a/audio_pipe.c +++ b/audio_pipe.c @@ -51,8 +51,19 @@ static void start(__attribute__((unused)) int sample_rate, // "ENXIO O_NONBLOCK | O_WRONLY is set, the named file is a FIFO, and no process has the FIFO // open for reading." - fd = try_to_open_pipe_for_writing(pipename); - } + fd = try_to_open_pipe_for_writing(pipename); + // we check that it's not a "real" error. From the "man 2 open" page: + // "ENXIO O_NONBLOCK | O_WRONLY is set, the named file is a FIFO, and no process has the FIFO + // open for reading." Which is okay. + if ((fd == -1) && (errno != ENXIO)) { + char errorstring[1024]; + strerror_r(errno, (char *)errorstring, sizeof(errorstring)); + debug(1, "audio_pipe start -- error %d (\"%s\") opening pipe: \"%s\".", errno, + (char *)errorstring, pipename); + warn("can not open audio pipe -- error %d (\"%s\") opening pipe: \"%s\".", errno, + (char *)errorstring, pipename); + } +} static int play(void *buf, int samples) { // if the file is not open, try to open it. @@ -74,6 +85,7 @@ static int play(void *buf, int samples) { static void stop(void) { // Don't close the pipe just because a play session has stopped. + } static int init(int argc, char **argv) { diff --git a/common.c b/common.c index 9bf0dd14..480410c1 100644 --- a/common.c +++ b/common.c @@ -283,10 +283,10 @@ char *generate_preliminary_string(char *buffer, size_t buffer_length, double tss insertion_point = insertion_point + strlen(insertion_point); space_remaining = space_remaining - strlen(insertion_point); } - if (prefix) { snprintf(insertion_point, space_remaining, "%s", prefix); insertion_point = insertion_point + strlen(insertion_point); + space_remaining = space_remaining - strlen(insertion_point); } return insertion_point; } @@ -308,7 +308,8 @@ void _die(const char *filename, const int linenumber, const char *format, ...) { 1.0 * time_since_last_debug_message / 1000000000, filename, linenumber, " *fatal error: "); } else { - s = b; + strncpy(b, "fatal error: ", sizeof(b)); + s = b+strlen(b); } va_list args; va_start(args, format); @@ -336,7 +337,8 @@ void _warn(const char *filename, const int linenumber, const char *format, ...) 1.0 * time_since_last_debug_message / 1000000000, filename, linenumber, " *warning: "); } else { - s = b; + strncpy(b, "warning: ", sizeof(b)); + s = b+strlen(b); } va_list args; va_start(args, format); @@ -1135,27 +1137,25 @@ int try_to_open_pipe_for_writing(const char* pathname) { // if it succeeds, it sets it to blocking. // if not, it returns -1. - char errorstring[1024]; int fdis = open(pathname, O_WRONLY | O_NONBLOCK); // open it in non blocking mode first + // we check that it's not a "real" error. From the "man 2 open" page: // "ENXIO O_NONBLOCK | O_WRONLY is set, the named file is a FIFO, and no process has the FIFO // open for reading." Which is okay. + // This is checked by the caller. - if ((fdis == -1) && (errno != ENXIO)) { - strerror_r(errno, (char *)errorstring, sizeof(errorstring)); - debug(1, "try_to_open_pipe -- error %d (\"%s\") opening pipe: \"%s\".", errno, - (char *)errorstring, pathname); - } else if (fdis >= 0) { // now we switch to blocking mode int flags = fcntl(fdis, F_GETFL); if ((flags == -1)) { + char errorstring[1024]; strerror_r(errno, (char *)errorstring, sizeof(errorstring)); debug(1, "try_to_open_pipe -- error %d (\"%s\") getting flags of pipe: \"%s\".", errno, (char *)errorstring, pathname); } else { flags = fcntl(fdis, F_SETFL,flags & ~O_NONBLOCK); if (flags == -1) { + char errorstring[1024]; strerror_r(errno, (char *)errorstring, sizeof(errorstring)); debug(1, "try_to_open_pipe -- error %d (\"%s\") unsetting NONBLOCK of pipe: \"%s\".", errno, (char *)errorstring, pathname); diff --git a/dbus-service.c b/dbus-service.c index 3c4e6daf..bbd5dcd4 100644 --- a/dbus-service.c +++ b/dbus-service.c @@ -1050,14 +1050,14 @@ static void on_dbus_name_acquired(GDBusConnection *connection, const gchar *name static void on_dbus_name_lost_again(__attribute__((unused)) GDBusConnection *connection, __attribute__((unused)) const gchar *name, __attribute__((unused)) gpointer user_data) { - warn("Could not acquire a Shairport Sync native D-Bus interface \"%s\" on the %s bus.", name, + warn("could not acquire a Shairport Sync native D-Bus interface \"%s\" on the %s bus.", name, (config.dbus_service_bus_type == DBT_session) ? "session" : "system"); } static void on_dbus_name_lost(__attribute__((unused)) GDBusConnection *connection, __attribute__((unused)) const gchar *name, __attribute__((unused)) gpointer user_data) { - // debug(1, "Could not acquire a Shairport Sync native D-Bus interface \"%s\" on the %s bus -- + // debug(1, "could not acquire a Shairport Sync native D-Bus interface \"%s\" on the %s bus -- // will try adding the process " // "number to the end of it.", // name, (config.dbus_service_bus_type == DBT_session) ? "session" : "system"); diff --git a/mpris-service.c b/mpris-service.c index 8258baf6..9851f0b7 100644 --- a/mpris-service.c +++ b/mpris-service.c @@ -345,7 +345,7 @@ static void on_mpris_name_acquired(GDBusConnection *connection, const gchar *nam static void on_mpris_name_lost_again(__attribute__((unused)) GDBusConnection *connection, const gchar *name, __attribute__((unused)) gpointer user_data) { - warn("Could not acquire an MPRIS interface named \"%s\" on the %s bus.", name, + warn("could not acquire an MPRIS interface named \"%s\" on the %s bus.", name, (config.mpris_service_bus_type == DBT_session) ? "session" : "system"); } diff --git a/mqtt.c b/mqtt.c index e2b86c5d..ad5a4c40 100644 --- a/mqtt.c +++ b/mqtt.c @@ -50,7 +50,7 @@ void on_message(__attribute__((unused)) struct mosquitto *mosq, memcpy(payload, msg->payload, msg->payloadlen); payload[msg->payloadlen] = 0; - debug(1, "[MQTT]: received Message on topic %s: %s\n", msg->topic, payload); + debug(2, "[MQTT]: received Message on topic %s: %s\n", msg->topic, payload); // All recognized commands char *commands[] = {"command", "beginff", "beginrew", "mutetoggle", "nextitem", @@ -63,7 +63,7 @@ void on_message(__attribute__((unused)) struct mosquitto *mosq, while (commands[it] != NULL) { if ((size_t)msg->payloadlen >= strlen(commands[it]) && strncmp(msg->payload, commands[it], strlen(commands[it])) == 0) { - debug(1, "[MQTT]: DACP Command: %s\n", commands[it]); + debug(2, "[MQTT]: DACP Command: %s\n", commands[it]); send_simple_dacp_command(commands[it]); break; } @@ -74,13 +74,13 @@ void on_message(__attribute__((unused)) struct mosquitto *mosq, void on_disconnect(__attribute__((unused)) struct mosquitto *mosq, __attribute__((unused)) void *userdata, __attribute__((unused)) int rc) { connected = 0; - debug(1, "[MQTT]: disconnected"); + debug(2, "[MQTT]: disconnected"); } void on_connect(struct mosquitto *mosq, __attribute__((unused)) void *userdata, __attribute__((unused)) int rc) { connected = 1; - debug(1, "[MQTT]: connected"); + debug(2, "[MQTT]: connected"); // subscribe if requested if (config.mqtt_enable_remote) { @@ -95,7 +95,7 @@ void mqtt_publish(char *topic, char *data, uint32_t length) { char fulltopic[strlen(config.mqtt_topic) + strlen(topic) + 3]; snprintf(fulltopic, strlen(config.mqtt_topic) + strlen(topic) + 2, "%s/%s", config.mqtt_topic, topic); - debug(1, "[MQTT]: publishing under %s", fulltopic); + debug(2, "[MQTT]: publishing under %s", fulltopic); int rc; if ((rc = mosquitto_publish(global_mosq, NULL, fulltopic, length, data, 0, 0)) != diff --git a/player.c b/player.c index c7020fc7..6f4978b5 100644 --- a/player.c +++ b/player.c @@ -1040,8 +1040,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { #endif // Here, calculate when we should start playing. We need to know when to allow the // packets to be sent to the player. - // We will send packets of silence from now until that time and then we will send the - // first packet, which will be followed by the subsequent packets. + // every second or so, we get a reference on when a particular packet should be // played. @@ -1087,175 +1086,170 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { } } + + if (conn->first_packet_time_to_play != 0) { - // recalculate conn->first_packet_time_to_play -- the latency might change + // Now that we know the timing of the first packet... + if (config.output->delay) { + // and that the output device is capable of synchronization... - uint64_t should_be_time; - uint32_t effective_latency = conn->latency; + // We may send packets of + // silence from now until the time the first audio packet should be sent + // and then we will send the first packet, which will be followed by + // the subsequent packets. + // here, we figure out whether and what silence to send. - switch (get_and_check_effective_latency(conn, &effective_latency, - config.audio_backend_latency_offset)) { - case -1: - if (conn->unachievable_audio_backend_latency_offset_notified == 0) { - warn("Negative latency! A latency of %d frames requested by the player, when " - "combined with an audio_backend_latency_offset of %f seconds, would make the " - "overall latency negative. The audio_backend_latency_offset setting is " - "ignored.", - conn->latency, config.audio_backend_latency_offset); - config.audio_backend_latency_offset = 0; // set it to zero - conn->unachievable_audio_backend_latency_offset_notified = 1; - }; - break; - case 1: - if (conn->unachievable_audio_backend_latency_offset_notified == 0) { - warn("An audio_backend_latency_offset of %f seconds may exceed the frame buffering " - "capacity -- the setting is ignored.", - config.audio_backend_latency_offset); - config.audio_backend_latency_offset = 0; // set it to zero; - conn->unachievable_audio_backend_latency_offset_notified = 1; - }; - break; - default: - break; - } + uint64_t should_be_time; + uint32_t effective_latency = conn->latency; - frame_to_local_time(conn->first_packet_timestamp + - effective_latency, // this will go modulo 2^32 - &should_be_time, conn); + switch (get_and_check_effective_latency(conn, &effective_latency, + config.audio_backend_latency_offset)) { + case -1: + if (conn->unachievable_audio_backend_latency_offset_notified == 0) { + warn("Negative latency! A latency of %d frames requested by the player, when " + "combined with an audio_backend_latency_offset of %f seconds, would make the " + "overall latency negative. The audio_backend_latency_offset setting is " + "ignored.", + conn->latency, config.audio_backend_latency_offset); + config.audio_backend_latency_offset = 0; // set it to zero + conn->unachievable_audio_backend_latency_offset_notified = 1; + }; + break; + case 1: + if (conn->unachievable_audio_backend_latency_offset_notified == 0) { + warn("An audio_backend_latency_offset of %f seconds may exceed the frame buffering " + "capacity -- the setting is ignored.", + config.audio_backend_latency_offset); + config.audio_backend_latency_offset = 0; // set it to zero; + conn->unachievable_audio_backend_latency_offset_notified = 1; + }; + break; + default: + break; + } - conn->first_packet_time_to_play = should_be_time; + // readjust first packet time to play + frame_to_local_time(conn->first_packet_timestamp + + effective_latency, // this will go modulo 2^32 + &should_be_time, conn); - if (local_time_now > conn->first_packet_time_to_play) { - uint64_t lateness = local_time_now - conn->first_packet_time_to_play; - debug(2, "Gone past starting time by %" PRIu64 " nanoseconds.", lateness); - // have_sent_prefiller_silence = 1; - conn->ab_buffering = 0; - } else { - // do some calculations - int64_t lead_time = conn->first_packet_time_to_play - local_time_now; - if ((config.audio_backend_silent_lead_in_time_auto == 1) || - (lead_time <= - (int64_t)(config.audio_backend_silent_lead_in_time * (int64_t)1000000000))) { - debug(3, "Lead time: %" PRId64 ".", lead_time); - if (config.output->delay) { // if the output device has a delay function - debug(3, "Checking"); - int resp = 0; - dac_delay = 0; - if (have_sent_prefiller_silence != 0) - resp = config.output->delay(&dac_delay); - if (resp == 0) { - int64_t gross_frame_gap = - ((conn->first_packet_time_to_play - local_time_now) * config.output_rate) / - 1000000000; - int64_t exact_frame_gap = gross_frame_gap - dac_delay; - int64_t frames_needed_to_maintain_desired_buffer = - (int64_t)(config.audio_backend_buffer_desired_length * config.output_rate) - - dac_delay; - // below, remember that exact_frame_gap and - // frames_needed_to_maintain_desired_buffer could both be negative - int64_t fs = frames_needed_to_maintain_desired_buffer; + conn->first_packet_time_to_play = should_be_time; - // if there isn't enough time to have the desired buffer size - if (exact_frame_gap <= frames_needed_to_maintain_desired_buffer) { - fs = conn->max_frames_per_packet * 2; - } + if (local_time_now > conn->first_packet_time_to_play) { + uint64_t lateness = local_time_now - conn->first_packet_time_to_play; + debug(2, "Gone past starting time by %" PRIu64 " nanoseconds.", lateness); + conn->ab_buffering = 0; + } else { + // do some calculations + int64_t lead_time = conn->first_packet_time_to_play - local_time_now; + if ((config.audio_backend_silent_lead_in_time_auto == 1) || + (lead_time <= + (int64_t)(config.audio_backend_silent_lead_in_time * (int64_t)1000000000))) { + debug(1, "Lead time: %" PRId64 " nanoseconds.", lead_time); + int resp = 0; + dac_delay = 0; + if (have_sent_prefiller_silence != 0) + resp = config.output->delay(&dac_delay); // we know the output device must have a delay function + if (resp == 0) { + int64_t gross_frame_gap = + ((conn->first_packet_time_to_play - local_time_now) * config.output_rate) / + 1000000000; + int64_t exact_frame_gap = gross_frame_gap - dac_delay; + int64_t frames_needed_to_maintain_desired_buffer = + (int64_t)(config.audio_backend_buffer_desired_length * config.output_rate) - + dac_delay; + // below, remember that exact_frame_gap and + // frames_needed_to_maintain_desired_buffer could both be negative + int64_t fs = frames_needed_to_maintain_desired_buffer; - // if we are very close to the end of buffering, i.e. within to frame-lengths, - // add the remaining silence needed and end buffering - if (exact_frame_gap <= conn->max_frames_per_packet * 2) { - fs = exact_frame_gap; - if (fs > first_frame_early_bias) - fs = fs - first_frame_early_bias; // deliberately make the first packet a tiny bit early so that the player may compensate for it at the last minute - conn->ab_buffering = 0; - } - void *silence; - if (fs > 0) { - silence = malloc(conn->output_bytes_per_frame * fs); - if (silence == NULL) - debug(1, "Failed to allocate %d byte silence buffer.", fs); - else { - // generate frames of silence with dither if necessary - conn->previous_random_number = - generate_zero_frames(silence, fs, config.output_format, - conn->enable_dither, conn->previous_random_number); - config.output->play(silence, fs); - debug(3, "Sent %" PRId64 " frames of silence", fs); - free(silence); - have_sent_prefiller_silence = 1; - } - } - } else { + // if there isn't enough time to have the desired buffer size + if (exact_frame_gap <= frames_needed_to_maintain_desired_buffer) { + fs = conn->max_frames_per_packet * 2; + } - if (resp == sps_extra_code_output_stalled) { - if (conn->unfixable_error_reported == 0) { - conn->unfixable_error_reported = 1; - if (config.cmd_unfixable) { - command_execute(config.cmd_unfixable, "output_device_stalled", 1); - } else { - warn("an unrecoverable error, \"output_device_stalled\", has been " - "detected.", - conn->connection_number); - } - } - } else { - debug(2, "Unexpected response to getting dac delay: %d.", resp); - } - } - } else { - // no delay function on back end -- just send the prefiller silence - // debug(2,"Back end has no delay function."); - // send the appropriate prefiller here... + // if we are very close to the end of buffering, i.e. within two frame-lengths, + // add the remaining silence needed and end buffering + if (exact_frame_gap <= conn->max_frames_per_packet * 2) { + fs = exact_frame_gap; + if (fs > first_frame_early_bias) + fs = fs - first_frame_early_bias; // deliberately make the first packet a tiny bit early so that the player may compensate for it at the last minute + conn->ab_buffering = 0; + } + void *silence; + if (fs > 0) { + silence = malloc(conn->output_bytes_per_frame * fs); + if (silence == NULL) + debug(1, "Failed to allocate %d byte silence buffer.", fs); + else { + // generate frames of silence with dither if necessary + conn->previous_random_number = + generate_zero_frames(silence, fs, config.output_format, + conn->enable_dither, conn->previous_random_number); + config.output->play(silence, fs); + debug(1, "Sent %" PRId64 " frames of silence", fs); + free(silence); + have_sent_prefiller_silence = 1; + } + } + } else { - void *silence; - if (lead_time != 0) { - int64_t frame_gap = (lead_time * config.output_rate) / 1000000000; - // debug(1,"%d frames needed.",frame_gap); - while (frame_gap > 0) { - ssize_t fs = config.output_rate / 10; - if (fs > frame_gap) - fs = frame_gap; + if (resp == sps_extra_code_output_stalled) { + if (conn->unfixable_error_reported == 0) { + conn->unfixable_error_reported = 1; + if (config.cmd_unfixable) { + command_execute(config.cmd_unfixable, "output_device_stalled", 1); + } else { + warn("an unrecoverable error, \"output_device_stalled\", has been " + "detected.", + conn->connection_number); + } + } + } else { + debug(2, "Unexpected response to getting dac delay: %d.", resp); + } + } + } + } + } else { + // if the output device doesn't have a delay, we simply send the lead-in + int64_t lead_time = conn->first_packet_time_to_play - local_time_now; // negative if we are late + void *silence; + int64_t frame_gap = (lead_time * config.output_rate) / 1000000000; + // debug(1,"%d frames needed.",frame_gap); + while (frame_gap > 0) { + ssize_t fs = config.output_rate / 10; + if (fs > frame_gap) + fs = frame_gap; - silence = malloc(conn->output_bytes_per_frame * fs); - if (silence == NULL) - debug(1, "Failed to allocate %d frame silence buffer.", fs); - else { - // debug(1, "No delay function -- outputting %d frames of silence.", fs); - conn->previous_random_number = - generate_zero_frames(silence, fs, config.output_format, - conn->enable_dither, conn->previous_random_number); - config.output->play(silence, fs); - free(silence); - } - frame_gap -= fs; - } - } - have_sent_prefiller_silence = 1; - conn->ab_buffering = 0; - } - } - } - } - if (conn->ab_buffering == 0) { -/* - // note the time of the playing of the first frame - uint64_t reference_timestamp_time; // don't need this... - get_reference_timestamp_stuff(&conn->play_segment_reference_frame, - &reference_timestamp_time, - &conn->play_segment_reference_frame_remote_time, conn); - conn->play_segment_reference_frame *= conn->output_sample_ratio; -*/ + silence = malloc(conn->output_bytes_per_frame * fs); + if (silence == NULL) + debug(1, "Failed to allocate %d frame silence buffer.", fs); + else { + // debug(1, "No delay function -- outputting %d frames of silence.", fs); + conn->previous_random_number = + generate_zero_frames(silence, fs, config.output_format, + conn->enable_dither, conn->previous_random_number); + config.output->play(silence, fs); + free(silence); + } + frame_gap -= fs; + } + conn->ab_buffering = 0; + } + } #ifdef CONFIG_METADATA - debug(2, "prsm"); - send_ssnc_metadata('prsm', NULL, 0, - 0); // "resume", but don't wait if the queue is locked + if (conn->ab_buffering == 0) { + debug(2, "prsm"); + send_ssnc_metadata('prsm', NULL, 0, + 0); // "resume", but don't wait if the queue is locked + } #endif } } } - } // Here, we work out whether to release a packet or wait - // We release a buffer when the time is right. + // We release a packet when the time is right. // To work out when the time is right, we need to take account of (1) the actual time the packet // should be released, @@ -2107,10 +2101,18 @@ void *player_thread_func(void *arg) { // frames from then onwards inbuflength *= conn->output_sample_ratio; + /* uint32_t reference_timestamp; uint64_t reference_timestamp_time, remote_reference_timestamp_time; get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time, &remote_reference_timestamp_time, conn); // types okay + */ + + // nt is the rtp timestamp of the first frame of the current packet of + // frames from the source. + // multiply it by the output frame ratio to get, effectively, the rtp timestamp + // of the first frame of the corresponding packet of output frames. + uint64_t nt; nt = inframe->given_timestamp; // uint32_t to int64_t nt = nt * conn->output_sample_ratio; @@ -2529,8 +2531,36 @@ void *player_thread_func(void *arg) { */ } } else { - // if there is no delay procedure, or it's not working or not allowed, there can be no - // synchronising + + // if there is no delay procedure, then we should be sending the packet + // to the output at the time determined by + // the packet's time to play + requested latency + requested offset. +/* + // This is just for checking during development + + uint32_t should_be_frame_32; + local_time_to_frame(local_time_now, &should_be_frame_32, conn); + // int64_t should_be_frame = ((int64_t)should_be_frame_32) * conn->output_sample_ratio; + + int32_t ilatency = (int32_t)((config.audio_backend_latency_offset - config.audio_backend_buffer_desired_length) * conn->input_rate) + conn->latency; + if (ilatency < 0) + debug(1,"incorrect latency %d.", ilatency); + + int32_t idelay = (int32_t)(should_be_frame_32 - inframe->given_timestamp); + + idelay = idelay - ilatency; + + debug(2,"delay is %d input frames.", idelay); +*/ + + // if this is the first frame, see if it's close to when it's supposed to be + // release, which will be its time plus latency and any offset_time + if (at_least_one_frame_seen_this_session == 0) { + at_least_one_frame_seen_this_session = 1; + + + } + play_samples = stuff_buffer_basic_32((int32_t *)conn->tbuf, inbuflength, config.output_format, conn->outbuf, 0, conn->enable_dither, conn); diff --git a/rtsp.c b/rtsp.c index 55789a12..9057b983 100644 --- a/rtsp.c +++ b/rtsp.c @@ -1306,19 +1306,6 @@ void metadata_create_multicast_socket(void) { } } } - - size_t pl = strlen(config.metadata_pipename) + 1; - - char *path = malloc(pl + 1); - snprintf(path, pl + 1, "%s", config.metadata_pipename); - - mode_t oldumask = umask(000); - - if (mkfifo(path, 0666) && errno != EEXIST) - die("Could not create metadata FIFO %s", path); - - free(path); - umask(oldumask); } void metadata_delete_multicast_socket(void) { @@ -1331,7 +1318,6 @@ void metadata_delete_multicast_socket(void) { } - void metadata_open(void) { if (config.metadata_enabled == 0) return; @@ -1563,6 +1549,33 @@ void metadata_hub_thread_cleanup_function(__attribute__((unused)) void *arg) { } void *metadata_hub_thread_function(__attribute__((unused)) void *ignore) { + + // create the fifo, if necessary + size_t pl = strlen(config.metadata_pipename) + 1; + char *path = malloc(pl + 1); + snprintf(path, pl + 1, "%s", config.metadata_pipename); + + mode_t oldumask = umask(000); + if (mkfifo(path, 0644) && errno != EEXIST) + die("Could not create metadata pipe \"%s\".", path); + umask(oldumask); + debug(1, "metadata pipe name is \"%s\".", path); + + // try to open it + fd = try_to_open_pipe_for_writing(path); + // we check that it's not a "real" error. From the "man 2 open" page: + // "ENXIO O_NONBLOCK | O_WRONLY is set, the named file is a FIFO, and no process has the FIFO + // open for reading." Which is okay. + if ((fd == -1) && (errno != ENXIO)) { + char errorstring[1024]; + strerror_r(errno, (char *)errorstring, sizeof(errorstring)); + debug(1, "metadata_hub_thread_function -- error %d (\"%s\") opening pipe: \"%s\".", errno, + (char *)errorstring, path); + warn("can not open metadata pipe -- error %d (\"%s\") opening pipe: \"%s\".", errno, + (char *)errorstring, path); + } + free(path); + // create a pc_queue for passing information to a threaded metadata handler pc_queue_init(&metadata_hub_queue, (char *)&metadata_hub_queue_items, sizeof(metadata_package), metadata_hub_queue_size, "hub"); From 98f526063f1a49b340824ac03be39cdb83dfb9d5 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 5 Jun 2020 17:23:10 +0100 Subject: [PATCH 127/186] Clean up code for the silent lead-in on a non-syncing back end e.g. a pipe. Clean up pipe creation and initial opening and associated error messages. Fix error with warning/die messages. Tidy up some debug messages. --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index f20b0a4a..22facc97 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.7d8], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.7d9], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 2d7d8e161e761135f7b8e01afd4f39be1462003e Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 5 Jun 2020 17:25:12 +0100 Subject: [PATCH 128/186] BB 3.3.7d8 --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 22facc97..f20b0a4a 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.7d9], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.7d8], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 426b4e6c4cc402edab7c95c5e23c5cbf1b155bac Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 5 Jun 2020 17:32:13 +0100 Subject: [PATCH 129/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 81b72eff..8eefa7ac 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,11 @@ +Version 3.3.7d7 +==== +**Enhancements** +* Tidy up the creation and initials opening of pipes. Suppress multiple pipe-opening error messages. +* Tidy up warnings and fatal error messages when log verbosity is zero. +* Clean up the code that provides a silent lead-in to play on a back end without synchronisation, e.g. a pipe or `stdout`. +* Added in commented-out code to check the timeliness of the release of audio to a back end without synchronisation, e.g. a pipe or `stdout`. TL;DR – so long as the back end does not block, frames will be released to it not more than one packet (352 frames) late. + Version 3.3.7d7 ==== **Enhancements** From c0abd35ce9bbde5b47f0373b29390f12bd91dd5c Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 5 Jun 2020 17:32:23 +0100 Subject: [PATCH 130/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 8eefa7ac..79e45593 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,4 +1,4 @@ -Version 3.3.7d7 +Version 3.3.7d8 ==== **Enhancements** * Tidy up the creation and initials opening of pipes. Suppress multiple pipe-opening error messages. From 002d5451bc99fed6bafe1d5793f7f897a94d2cc6 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 5 Jun 2020 17:38:31 +0100 Subject: [PATCH 131/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 79e45593..beb61dcf 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,9 +1,9 @@ Version 3.3.7d8 ==== **Enhancements** -* Tidy up the creation and initials opening of pipes. Suppress multiple pipe-opening error messages. -* Tidy up warnings and fatal error messages when log verbosity is zero. -* Clean up the code that provides a silent lead-in to play on a back end without synchronisation, e.g. a pipe or `stdout`. +* Tidied up the creation and initial opening of pipes. Suppress repeated pipe-opening error messages. +* Tidied up warnings and fatal error messages when log verbosity is zero. +* Cleaned up the code that provides a silent lead-in to play on a back end without synchronisation, e.g. a pipe or `stdout`. * Added in commented-out code to check the timeliness of the release of audio to a back end without synchronisation, e.g. a pipe or `stdout`. TL;DR – so long as the back end does not block, frames will be released to it not more than one packet (352 frames) late. Version 3.3.7d7 @@ -11,16 +11,16 @@ Version 3.3.7d7 **Enhancements** * Logs and statistics can now be directed to the system log (default), `stdout`, `stderr` or to a file or pipe of your choice using a new setting, `log_output_to` in the `diagnostics` section of the configuration file. This is very useful when the system log is disabled or diverted. * Audio data from the `pipe` back end and metadata from the metadata pipe are now written using standard blocking `write` commands rather than a slightly complex non-blocking write function. Pipes are now opened in non-blocking mode and changed to blocking mode when successfully opened. -* Separate threads for the metadata subsystems. Until now, all metadata was processed on a single thread. This included writing to the metadata pipe and the multicast stream and supplying metadata for the `mqtt` interface and for the `dbus` and `MPRIS` interfaces. Unfortunately, that meant that a problem with any one of these subsystems could propagate into the others. Now they all run on separate threads. If one thread blocks, it will not interfere with the other subsystems. +* Separate threads are now used for each metadata subsystem. Until now, all metadata was processed on a single thread. This included writing to the metadata pipe and the multicast stream and supplying metadata for the `mqtt` interface and for the `dbus` and `MPRIS` interfaces. Unfortunately, that meant that a problem with any one of these subsystems could propagate into the others. Now they all run on separate threads. If one thread blocks, it will not interfere with the other subsystems. Version 3.3.7d6 ==== **Bug Fixes and Enhancements** * Fixed a bug calculating the instantaneous synchronisation error. This bug could occasionally cause Shairport Sync to lose synchronisation and maybe even to mute for a few seconds before resynchronising. It was caused doing modulo arithmetic incorrectly and it's been there for a while. -* Cleaned up and improve the code to synchronise the first frame of audio. This should result in more accurate and reliable initial synchronisation, usually to under a millisecond, and often to within 20 or 30 microseconds in normal usage. Sync should improve even when the silent lead-in time is as short as 0.3 seconds or when the `audio_backend_latency_offset_in_seconds` is as much as -1.7 seconds. +* Cleaned up and improved the code to synchronise the first frame of audio. This should result in more accurate and reliable initial synchronisation, usually to under a millisecond, and often to within 20 or 30 microseconds. Syncronisation should improve even when the silent lead-in time is as short as 0.3 seconds or when the `audio_backend_latency_offset_in_seconds` is as much as -1.7 seconds, i.e. when only 0.3 seconds of latency are left when the latency would normally be 2.0 seconds. * Removed a bug which would affect initial synchronisation if a `FLUSH` command was received from the player at an inopportune time. * Cleaned up some confused uses of modulo arithmetic. -* Cleaned up allocation of memory for gathering running statistics – use the heap instead of the stack. +* Cleaned up the allocation of memory for gathering running statistics – the heap is now used instead of the stack. * Cleaned up the display of statistics for backends that do not implement active synchronisation, e.g. the `pipe` and `STDOUT` back ends. Version 3.3.7d5 @@ -31,7 +31,7 @@ Version 3.3.7d5 * Improved synchronisation accuracy with short silence lead-ins. **New Feature** -* For the PulseAudio backend `pa`, add a new `server` entry to `pa` section of the configuration file, allowing you to specify a connection to a remote or a local system PulseAudio instance instead of letting PulseAudio choose. Thanks to [Guillaume Revaillot](https://github.com/grevaillot) for this new feature. +* For the PulseAudio backend `pa`, added a new `server` entry to the `pa` section of the configuration file, allowing you to specify a connection to a remote or a local system PulseAudio instance instead of letting PulseAudio choose. Thanks to [Guillaume Revaillot](https://github.com/grevaillot) for this new feature. Version 3.3.7d4 ==== @@ -46,7 +46,7 @@ Version 3.3.7d3 Version 3.3.7d2 ==== **Pesky Changes** -* The underlying timing system has been moved from 64-bit fixed-point time representation (a là NTP) to a 64-bit unsigned nanoseconds representation. This should make precisely no difference to the functionality of Shairport Sync but the transition might inadvertently have introduced bugs. Problem reports gratefully received. +* The underlying timing system has been moved from 64-bit fixed-point time representation (like NTP) to a 64-bit unsigned nanoseconds representation. This should make precisely no difference to the functionality of Shairport Sync but the transition might inadvertently have introduced bugs. Problem reports gratefully received. **Enhancement** * The timing software in the `sndio` backend does some extra sanity checking on certain time estimates, it may help a little when running on virtual machines. From 346595386b3873a7e61668ac9182a273d81cbeb5 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 6 Jun 2020 10:53:08 +0100 Subject: [PATCH 132/186] Small mod to creating the audio pipe --- audio_pipe.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/audio_pipe.c b/audio_pipe.c index 7953dc23..6dedc464 100644 --- a/audio_pipe.c +++ b/audio_pipe.c @@ -120,10 +120,12 @@ static int init(int argc, char **argv) { pipename = strdup(argv[0]); // here, create the pipe + mode_t oldumask = umask(000); if (mkfifo(pipename, 0644) && errno != EEXIST) - die("Could not create output pipe \"%s\"", pipename); + die("Could not create audio pipe \"%s\"", pipename); + umask(oldumask); - debug(1, "Pipename is \"%s\"", pipename); + debug(1, "audio pipe name is \"%s\"", pipename); return 0; } From 74894261f5c5245a30f021b1fbaf89821adfa744 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 9 Jun 2020 09:33:10 +0100 Subject: [PATCH 133/186] quieten some debug messages --- player.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/player.c b/player.c index 6f4978b5..bc9ed329 100644 --- a/player.c +++ b/player.c @@ -1145,7 +1145,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { if ((config.audio_backend_silent_lead_in_time_auto == 1) || (lead_time <= (int64_t)(config.audio_backend_silent_lead_in_time * (int64_t)1000000000))) { - debug(1, "Lead time: %" PRId64 " nanoseconds.", lead_time); + // debug(1, "Lead time: %" PRId64 " nanoseconds.", lead_time); int resp = 0; dac_delay = 0; if (have_sent_prefiller_silence != 0) @@ -1186,7 +1186,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { generate_zero_frames(silence, fs, config.output_format, conn->enable_dither, conn->previous_random_number); config.output->play(silence, fs); - debug(1, "Sent %" PRId64 " frames of silence", fs); + // debug(1, "Sent %" PRId64 " frames of silence", fs); free(silence); have_sent_prefiller_silence = 1; } From efec3cc7e5201e149bcbb2da9bb129526cfab422 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 9 Jun 2020 09:36:12 +0100 Subject: [PATCH 134/186] Make the list of remote to local timing values a power of 2 -- necessary later for allowing headroom for calculations. --- player.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/player.h b/player.h index f79f8806..c6315aae 100644 --- a/player.h +++ b/player.h @@ -24,7 +24,8 @@ #include "alac.h" #include "audio.h" -#define time_ping_history 128 // at 1 per three seconds, approximately six minutes of records +#define time_ping_history_power_of_two 7 +#define time_ping_history (1 << time_ping_history_power_of_two) // 2^7 is 128. At 1 per three seconds, approximately six minutes of records typedef struct time_ping_record { uint64_t local_to_remote_difference; From 1312772bc8f3b07e6b4aec047946918cea197b07 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 9 Jun 2020 11:33:32 +0100 Subject: [PATCH 135/186] Remove a redundant field from a time ping record --- player.h | 1 - 1 file changed, 1 deletion(-) diff --git a/player.h b/player.h index c6315aae..c5713de1 100644 --- a/player.h +++ b/player.h @@ -28,7 +28,6 @@ #define time_ping_history (1 << time_ping_history_power_of_two) // 2^7 is 128. At 1 per three seconds, approximately six minutes of records typedef struct time_ping_record { - uint64_t local_to_remote_difference; uint64_t dispersion; uint64_t local_time; uint64_t remote_time; From 9f9b36c01fd6e921b4df0d3223e19acaf9daeb92 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 9 Jun 2020 11:37:24 +0100 Subject: [PATCH 136/186] Clean up and bug-fix the code for estimating remote-to-local clock drift using least squares linear regression. Use the estimated remote-to-local difference and the calculated drift instead of just using the difference estimated from the last-exchanged timing message. --- rtp.c | 85 +++++++++++++++++++---------------------------------------- 1 file changed, 27 insertions(+), 58 deletions(-) diff --git a/rtp.c b/rtp.c index 71d6df0e..41d3cfe9 100644 --- a/rtp.c +++ b/rtp.c @@ -602,17 +602,13 @@ void *rtp_timing_receiver(void *arg) { // debug(1,"Return trip time: %" PRIu64 " nS, remote processing time: %" PRIu64 " // nS.",return_time, remote_processing_time); - uint64_t local_time_by_remote_clock = distant_transmit_time + return_time / 2; - - // remove the remote processing time from the record of the return time, as long at the - // processing time looks sensible. - if (remote_processing_time < return_time) return_time -= remote_processing_time; else debug(1, "Remote processing time greater than return time -- ignored."); int cc; + // debug(1, "time ping history is %d entries.", time_ping_history); for (cc = time_ping_history - 1; cc > 0; cc--) { conn->time_pings[cc] = conn->time_pings[cc - 1]; // if ((conn->time_ping_count) && (conn->time_ping_count < 10)) @@ -625,12 +621,9 @@ void *rtp_timing_receiver(void *arg) { } // these are used for doing a least squares calculation to get the drift conn->time_pings[0].local_time = arrival_time; - conn->time_pings[0].remote_time = distant_transmit_time; + conn->time_pings[0].remote_time = distant_transmit_time + return_time / 2; conn->time_pings[0].sequence_number = sequence_number++; conn->time_pings[0].chosen = 0; - - conn->time_pings[0].local_to_remote_difference = - local_time_by_remote_clock - arrival_time; conn->time_pings[0].dispersion = return_time; if (conn->time_ping_count < time_ping_history) conn->time_ping_count++; @@ -653,14 +646,14 @@ void *rtp_timing_receiver(void *arg) { // uint64_t local_time_chosen = arrival_time; // uint64_t remote_time_chosen = distant_transmit_time; // now pick the timestamp with the lowest dispersion - uint64_t l2rtd = conn->time_pings[0].local_to_remote_difference; + uint64_t rt = conn->time_pings[0].remote_time; uint64_t lt = conn->time_pings[0].local_time; uint64_t tld = conn->time_pings[0].dispersion; int chosen = 0; for (cc = 1; cc < conn->time_ping_count; cc++) if (conn->time_pings[cc].dispersion < tld) { chosen = cc; - l2rtd = conn->time_pings[cc].local_to_remote_difference; + rt = conn->time_pings[cc].remote_time; lt = conn->time_pings[cc].local_time; tld = conn->time_pings[cc].dispersion; // local_time_chosen = conn->time_pings[cc].local_time; @@ -670,44 +663,8 @@ void *rtp_timing_receiver(void *arg) { // dispersion.",chosen,1.0*((tld * 1000000) >> 32)); conn->time_pings[chosen].chosen = 1; // record the fact that it has been used for timing - /* - // calculate the jitter -- the absolute time between the current - local_to_remote_time_difference and the new one and add it to the total jitter count - int64_t ji; - int64_t ltd =0; // local time difference for the jitter - - if (conn->time_ping_count > 1) { - if (l2rtd > conn->local_to_remote_time_difference) { - local_to_remote_time_jitter = - local_to_remote_time_jitter + l2rtd - conn->local_to_remote_time_difference; - ji = l2rtd - conn->local_to_remote_time_difference; // this is the difference - between the present local-to-remote-time-difference and the new one, i.e. the jitter - step - } else { - local_to_remote_time_jitter = - local_to_remote_time_jitter + conn->local_to_remote_time_difference - l2rtd; - ji = -(conn->local_to_remote_time_difference - l2rtd); - } - local_to_remote_time_jitter_count += 1; - } - if (conn->local_to_remote_time_difference_measurement_time < lt) - ltd = lt-conn->local_to_remote_time_difference_measurement_time; - else - ltd = -(conn->local_to_remote_time_difference_measurement_time-lt); - - if (ltd) { - debug(1,"Jitter: %" PRId64 " nanoseconds in %" PRId64 " nanoseconds.", ji, ltd); - debug(1,"Source clock to local clock drift: %.2f ppm.",((1.0*ji)/ltd)*1000000.0); - } - // uncomment below to print jitter between client's clock and our clock - - if (ji) { - debug(1,"Choosing time difference[%d] with dispersion of %" PRId64 " ns with an - adjustment of %" PRId64 " ns",chosen, tld, ji); - } - */ conn->local_to_remote_time_difference = - l2rtd; // make this the new local-to-remote-time-difference + rt - lt; // make this the new local-to-remote-time-difference conn->local_to_remote_time_difference_measurement_time = lt; // done at this time. if (first_local_to_remote_time_difference == 0) { @@ -730,57 +687,69 @@ void *rtp_timing_receiver(void *arg) { int sample_count = 0; // approximate time in seconds to let the system settle down - const int settling_time = 60; + const int settling_time = 30; // number of points to have for calculating a valid drift const int sample_point_minimum = 8; for (cc = 0; cc < conn->time_ping_count; cc++) if ((conn->time_pings[cc].chosen) && (conn->time_pings[cc].sequence_number > (settling_time / 3))) { // wait for a approximate settling time - y_bar += (conn->time_pings[cc].remote_time >> - 12); // precision is down to 1/4th of a microsecond - x_bar += (conn->time_pings[cc].local_time >> 12); + // have to scale them down so that the sum, possibly over every term in the array, doesn't overflow + y_bar += (conn->time_pings[cc].remote_time >> time_ping_history_power_of_two); + x_bar += (conn->time_pings[cc].local_time >> time_ping_history_power_of_two); sample_count++; } if (sample_count > sample_point_minimum) { y_bar = y_bar / sample_count; x_bar = x_bar / sample_count; + + int64_t xid, yid; - int64_t mtl, mbl; + double mtl, mbl; mtl = 0; mbl = 0; for (cc = 0; cc < conn->time_ping_count; cc++) if ((conn->time_pings[cc].chosen) && (conn->time_pings[cc].sequence_number > (settling_time / 3))) { - uint64_t slt = conn->time_pings[cc].local_time >> 12; + uint64_t slt = conn->time_pings[cc].local_time >> time_ping_history_power_of_two; if (slt > x_bar) xid = slt - x_bar; else xid = -(x_bar - slt); - uint64_t srt = conn->time_pings[cc].remote_time >> 12; + uint64_t srt = conn->time_pings[cc].remote_time >> time_ping_history_power_of_two; if (srt > y_bar) yid = srt - y_bar; else yid = -(y_bar - srt); - mtl = mtl + xid * yid; - mbl = mbl + xid * xid; + mtl = mtl + (1.0 * xid) * yid; + mbl = mbl + (1.0 * xid) * xid; } conn->local_to_remote_time_gradient_sample_count = sample_count; if (mbl) - conn->local_to_remote_time_gradient = (1.0 * mtl) / mbl; + conn->local_to_remote_time_gradient = mtl / mbl; else { conn->local_to_remote_time_gradient = 1.0; debug(1, "rtp_timing_receiver: mbl is 0"); } + + // scale the numbers back up + uint64_t ybf = y_bar << time_ping_history_power_of_two; + uint64_t xbf = x_bar << time_ping_history_power_of_two; + + conn->local_to_remote_time_difference = + ybf - xbf; // make this the new local-to-remote-time-difference + conn->local_to_remote_time_difference_measurement_time = xbf; + } else { conn->local_to_remote_time_gradient = 1.0; } // debug(1,"local to remote time gradient is %12.2f ppm, based on %d // samples.",conn->local_to_remote_time_gradient*1000000,sample_count); + } else { debug(2, "Time ping turnaround time: %" PRIu64 From f63abd2f09ba85a3b1a4fe77712d748ddef1018e Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 9 Jun 2020 11:40:50 +0100 Subject: [PATCH 137/186] Start using timing based on a linear least squares error-based estimate of remote to local time difference and drift. --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index f20b0a4a..22facc97 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.7d8], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.7d9], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From fcaf25f9ca9533c9c326e2a15307662138ac8786 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Tue, 9 Jun 2020 12:24:09 +0100 Subject: [PATCH 138/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index beb61dcf..a64fb123 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,10 @@ +Version 3.3.7d9 +==== +**Enhancements** +* Improved timing estimation. Shairport Sync has been using linear least-squares regression to estimate timing drift between the (remote) source clock and the local clock. This technique is now extended to provide an estimate of the remote-to-local clock difference, used to remap the timing of audio frames from the remote device's clock to the local clock. + + In practice, the timing technique in use up to now has been very accurate, but this should result in slightly smoother rates of correction. + Version 3.3.7d8 ==== **Enhancements** From 0517a233c163f9dff58c5fd2f8970eaa85fdd9bb Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 10 Jun 2020 14:21:29 +0100 Subject: [PATCH 139/186] Fix the recording of the sample count if it's less than the threshold. Use the last gradient / drift if the number of samples is less than the threshold. --- rtp.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/rtp.c b/rtp.c index 41d3cfe9..939f89f4 100644 --- a/rtp.c +++ b/rtp.c @@ -541,6 +541,12 @@ void *rtp_timing_receiver(void *arg) { // uint64_t first_local_time = 0; uint64_t first_local_to_remote_time_difference = 0; + + // maybe this initial value could be retrieved from a temporary store + // keyed to the source. + + conn->local_to_remote_time_gradient = 1.0; // initial value. + // uint64_t first_local_to_remote_time_difference_time; // uint64_t l2rtd = 0; int sequence_number = 0; @@ -699,6 +705,7 @@ void *rtp_timing_receiver(void *arg) { x_bar += (conn->time_pings[cc].local_time >> time_ping_history_power_of_two); sample_count++; } + conn->local_to_remote_time_gradient_sample_count = sample_count; if (sample_count > sample_point_minimum) { y_bar = y_bar / sample_count; x_bar = x_bar / sample_count; @@ -728,12 +735,11 @@ void *rtp_timing_receiver(void *arg) { mtl = mtl + (1.0 * xid) * yid; mbl = mbl + (1.0 * xid) * xid; } - conn->local_to_remote_time_gradient_sample_count = sample_count; if (mbl) conn->local_to_remote_time_gradient = mtl / mbl; else { - conn->local_to_remote_time_gradient = 1.0; - debug(1, "rtp_timing_receiver: mbl is 0"); + // conn->local_to_remote_time_gradient = 1.0; + debug(1,"mbl is zero. Drift remains at %.2f ppm.", (conn->local_to_remote_time_gradient - 1.0)*1000000); } // scale the numbers back up @@ -745,7 +751,8 @@ void *rtp_timing_receiver(void *arg) { conn->local_to_remote_time_difference_measurement_time = xbf; } else { - conn->local_to_remote_time_gradient = 1.0; + debug(3,"not enough samples to estimate drift -- remaining at %.2f ppm.", (conn->local_to_remote_time_gradient - 1.0)*1000000); + // conn->local_to_remote_time_gradient = 1.0; } // debug(1,"local to remote time gradient is %12.2f ppm, based on %d // samples.",conn->local_to_remote_time_gradient*1000000,sample_count); From 92074c2b96ffddf8e376e06b06a3a84c25aabd9c Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 10 Jun 2020 20:04:03 +0100 Subject: [PATCH 140/186] add code to calculate the diffusion multiplier based on the ending diffusion multiplier and the number of entries. --- rtp.c | 26 ++++++++++++++++++++++---- 1 file changed, 22 insertions(+), 4 deletions(-) diff --git a/rtp.c b/rtp.c index 939f89f4..eb476c99 100644 --- a/rtp.c +++ b/rtp.c @@ -547,6 +547,21 @@ void *rtp_timing_receiver(void *arg) { conn->local_to_remote_time_gradient = 1.0; // initial value. + // calculate diffusion factor + + // at the end of the array of time pings, the diffusion factor + // must be diffusion_expansion_factor + // this, at each step, the diffusion multiplication constant must + // be the nth root of diffusion_expansion_factor + // where n is the number of elements in the array + + const double diffusion_expansion_factor = 10; + double log_of_multiplier = log10(diffusion_expansion_factor)/time_ping_history; + double multiplier = pow(10,log_of_multiplier); + uint64_t dispersion_factor = (uint64_t)(multiplier * 100); + debug(1,"dispersion factor is %" PRIu64 ".", dispersion_factor); + + // uint64_t first_local_to_remote_time_difference_time; // uint64_t l2rtd = 0; int sequence_number = 0; @@ -573,7 +588,7 @@ void *rtp_timing_receiver(void *arg) { if (return_time < 300000000) { // must be less than 0.3 seconds - // debug(1,"Synchronisation ping return time is %" PRIu64 " nanoseconds.",return_time); + //debug(1,"Ping return time is %8.3f milliseconds.",0.000001*return_time); // distant_receive_time = // ((uint64_t)ntohl(*((uint32_t*)&packet[16])))<<32+ntohl(*((uint32_t*)&packet[20])); @@ -621,8 +636,11 @@ void *rtp_timing_receiver(void *arg) { // conn->time_pings[cc].dispersion = // conn->time_pings[cc].dispersion * pow(2.14, // 1.0/conn->time_ping_count); - conn->time_pings[cc].dispersion = - (conn->time_pings[cc].dispersion * 110) / + if (conn->time_pings[cc].dispersion > UINT64_MAX / dispersion_factor) + debug(1,"dispersion factor is too large at %" PRIu64 "."); + else + conn->time_pings[cc].dispersion = + (conn->time_pings[cc].dispersion * dispersion_factor) / 100; // make the dispersions 'age' by this rational factor } // these are used for doing a least squares calculation to get the drift @@ -693,7 +711,7 @@ void *rtp_timing_receiver(void *arg) { int sample_count = 0; // approximate time in seconds to let the system settle down - const int settling_time = 30; + const int settling_time = 60; // number of points to have for calculating a valid drift const int sample_point_minimum = 8; for (cc = 0; cc < conn->time_ping_count; cc++) From f5eceff6f7a0e37a5547fe5b45786092bcad1efd Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 11 Jun 2020 12:09:55 +0100 Subject: [PATCH 141/186] Store clock drift information for each client based on its IP. This is stored in RAM, not on disk. Used at the start of a new session until new clock drift data is ready. --- common.h | 5 ++++- rtp.c | 46 +++++++++++++++++++++++++++++++++++++++++----- 2 files changed, 45 insertions(+), 6 deletions(-) diff --git a/common.h b/common.h index 93bb9c36..dce7a00a 100644 --- a/common.h +++ b/common.h @@ -285,7 +285,10 @@ typedef struct { int jack_soxr_resample_quality; #endif #endif - + void *gradients; // a linked list of the clock gradients discovered for all DACP IDs + // can't use IP numbers as they might be given to different devices + // can't get hold of MAC addresses. + // can't define the nvll linked list struct here } shairport_cfg; // accessors to config for multi-thread access diff --git a/rtp.c b/rtp.c index eb476c99..5775fb5a 100644 --- a/rtp.c +++ b/rtp.c @@ -45,6 +45,14 @@ #include #include +struct Nvll { + char* name; + double value; + struct Nvll *next; +}; + +typedef struct Nvll nvll; + uint64_t local_to_remote_time_jitter; uint64_t local_to_remote_time_jitter_count; @@ -514,8 +522,29 @@ void *rtp_timing_sender(void *arg) { } void rtp_timing_receiver_cleanup_handler(void *arg) { - debug(3, "Timing Receiver Cleanup."); rtsp_conn_info *conn = (rtsp_conn_info *)arg; + debug(3, "Timing Receiver Cleanup."); + // walk down the list of DACP / gradient pairs, if any + nvll *gradients = config.gradients; + if (conn->dacp_id) + while ((gradients) && (strcasecmp((const char *)&conn->client_ip_string,gradients->name) != 0)) + gradients = gradients->next; + + // if gradients comes out of this non-null, it is pointing to the DACP and it's last-known gradient + if (gradients) { + gradients->value = conn->local_to_remote_time_gradient; + debug(1,"Updating a drift of %.2f ppm for \"%s\".", (conn->local_to_remote_time_gradient - 1.0)*1000000, gradients->name); + } else { + nvll *new_entry = (nvll*)malloc(sizeof(nvll)); + if (new_entry) { + new_entry->name = strdup((const char *)&conn->client_ip_string); + new_entry->value = conn->local_to_remote_time_gradient; + new_entry->next = config.gradients; + config.gradients = new_entry; + debug(1,"Setting a new drift of %.2f ppm for \"%s\".", (conn->local_to_remote_time_gradient - 1.0)*1000000, new_entry->name); + } + } + debug(3, "Cancel Timing Requester."); pthread_cancel(conn->timer_requester); int oldState; @@ -542,10 +571,17 @@ void *rtp_timing_receiver(void *arg) { uint64_t first_local_to_remote_time_difference = 0; - // maybe this initial value could be retrieved from a temporary store - // keyed to the source. - conn->local_to_remote_time_gradient = 1.0; // initial value. + // walk down the list of DACP / gradient pairs, if any + nvll *gradients = config.gradients; + while ((gradients) && (strcasecmp((const char *)&conn->client_ip_string,gradients->name) != 0)) + gradients = gradients->next; + + // if gradients comes out of this non-null, it is pointing to the IP and it's last-known gradient + if (gradients) { + conn->local_to_remote_time_gradient = gradients->value; + debug(1,"Using a stored drift of %.2f ppm for \"%s\".", (conn->local_to_remote_time_gradient - 1.0)*1000000, gradients->name); + } // calculate diffusion factor @@ -559,7 +595,7 @@ void *rtp_timing_receiver(void *arg) { double log_of_multiplier = log10(diffusion_expansion_factor)/time_ping_history; double multiplier = pow(10,log_of_multiplier); uint64_t dispersion_factor = (uint64_t)(multiplier * 100); - debug(1,"dispersion factor is %" PRIu64 ".", dispersion_factor); + // debug(1,"dispersion factor is %" PRIu64 ".", dispersion_factor); // uint64_t first_local_to_remote_time_difference_time; From 23f07f625735a32b501c26052614ae3bc80a4b4c Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 11 Jun 2020 14:31:52 +0100 Subject: [PATCH 142/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index a64fb123..07af0aa6 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,9 +1,11 @@ Version 3.3.7d9 ==== **Enhancements** -* Improved timing estimation. Shairport Sync has been using linear least-squares regression to estimate timing drift between the (remote) source clock and the local clock. This technique is now extended to provide an estimate of the remote-to-local clock difference, used to remap the timing of audio frames from the remote device's clock to the local clock. +* Improved timing estimation. Shairport Sync has been using linear least-squares regression to estimate timing drift between the (remote) source clock and the local clock. This technique is now extended to provide an estimate of the remote-to-local clock difference itself. The remote-to-local clock difference is used to remap the timing of audio frames from the remote device's clock to the local clock. - In practice, the timing technique in use up to now has been very accurate, but this should result in slightly smoother rates of correction. + Timing drift estimates are now saved when a sessions ends, keyed to the client IP number. When a new session starts from that IP number, the stored estimate is used until a new estimate can be generated from the new session. + + In practice, the timing techniques in use up to now have been very accurate, but this should result in slightly smoother rates of correction. Version 3.3.7d8 ==== From 54b82884aba3da189d89ae24b4ebbdbd69aa586a Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Thu, 11 Jun 2020 14:48:42 +0100 Subject: [PATCH 143/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 07af0aa6..b941a530 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,7 +1,7 @@ Version 3.3.7d9 ==== **Enhancements** -* Improved timing estimation. Shairport Sync has been using linear least-squares regression to estimate timing drift between the (remote) source clock and the local clock. This technique is now extended to provide an estimate of the remote-to-local clock difference itself. The remote-to-local clock difference is used to remap the timing of audio frames from the remote device's clock to the local clock. +* Improved timing estimation. Shairport Sync has been using linear least-squares regression to estimate timing drift between the (remote) source clock and the local clock. This technique is now extended to provide an estimate of the remote-to-local clock difference itself. (The remote-to-local clock difference is used to remap the timing of audio frames from the remote device's clock to the local clock.) Timing drift estimates are now saved when a sessions ends, keyed to the client IP number. When a new session starts from that IP number, the stored estimate is used until a new estimate can be generated from the new session. From 19a7bd496975a61fdc43003bc35ac1e65f7ed1d6 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 12 Jun 2020 10:10:26 +0100 Subject: [PATCH 144/186] quieten some debug messages --- rtp.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/rtp.c b/rtp.c index 5775fb5a..a8f03f6a 100644 --- a/rtp.c +++ b/rtp.c @@ -533,7 +533,7 @@ void rtp_timing_receiver_cleanup_handler(void *arg) { // if gradients comes out of this non-null, it is pointing to the DACP and it's last-known gradient if (gradients) { gradients->value = conn->local_to_remote_time_gradient; - debug(1,"Updating a drift of %.2f ppm for \"%s\".", (conn->local_to_remote_time_gradient - 1.0)*1000000, gradients->name); + // debug(1,"Updating a drift of %.2f ppm for \"%s\".", (conn->local_to_remote_time_gradient - 1.0)*1000000, gradients->name); } else { nvll *new_entry = (nvll*)malloc(sizeof(nvll)); if (new_entry) { @@ -541,7 +541,7 @@ void rtp_timing_receiver_cleanup_handler(void *arg) { new_entry->value = conn->local_to_remote_time_gradient; new_entry->next = config.gradients; config.gradients = new_entry; - debug(1,"Setting a new drift of %.2f ppm for \"%s\".", (conn->local_to_remote_time_gradient - 1.0)*1000000, new_entry->name); + // debug(1,"Setting a new drift of %.2f ppm for \"%s\".", (conn->local_to_remote_time_gradient - 1.0)*1000000, new_entry->name); } } @@ -580,7 +580,7 @@ void *rtp_timing_receiver(void *arg) { // if gradients comes out of this non-null, it is pointing to the IP and it's last-known gradient if (gradients) { conn->local_to_remote_time_gradient = gradients->value; - debug(1,"Using a stored drift of %.2f ppm for \"%s\".", (conn->local_to_remote_time_gradient - 1.0)*1000000, gradients->name); + // debug(1,"Using a stored drift of %.2f ppm for \"%s\".", (conn->local_to_remote_time_gradient - 1.0)*1000000, gradients->name); } // calculate diffusion factor From 70f60fcd8a20c1f0046fae0c83f1ce50d78c951a Mon Sep 17 00:00:00 2001 From: minix1234 <48591487+minix1234@users.noreply.github.com> Date: Fri, 12 Jun 2020 09:03:30 -0400 Subject: [PATCH 145/186] Updated activity_monitor.c - metadata publishing wrong codes Found that MQTT activity_start: abeg and activity_end: aend topics where never being published. instead immediately after a play session started a play_end: pend topic was sent. Also after the activity_timout a duplicate play_end was received. This commit fixes those bugs. --- activity_monitor.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/activity_monitor.c b/activity_monitor.c index ce7edc5f..735fd03a 100644 --- a/activity_monitor.c +++ b/activity_monitor.c @@ -62,7 +62,7 @@ void going_active(int block) { command_execute(config.cmd_active_start, "", block); #ifdef CONFIG_METADATA debug(2, "abeg"); // active mode begin - send_ssnc_metadata('pend', NULL, 0, 1); // contains cancellation points + send_ssnc_metadata('abeg', NULL, 0, 1); // contains cancellation points #endif #ifdef CONFIG_DBUS_INTERFACE @@ -89,7 +89,7 @@ void going_inactive(int block) { command_execute(config.cmd_active_stop, "", block); #ifdef CONFIG_METADATA debug(2, "aend"); // active mode end - send_ssnc_metadata('pend', NULL, 0, 1); // contains cancellation points + send_ssnc_metadata('aend', NULL, 0, 1); // contains cancellation points #endif #ifdef CONFIG_DBUS_INTERFACE From f8c24c5f2b44508d07cce07a368224945fa2b6a4 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 12 Jun 2020 17:12:50 +0100 Subject: [PATCH 146/186] clean up flush detection code --- player.c | 140 ++++++++++++++++++++++++++++++++++--------------------- player.h | 1 + 2 files changed, 88 insertions(+), 53 deletions(-) diff --git a/player.c b/player.c index bc9ed329..f2fef46d 100644 --- a/player.c +++ b/player.c @@ -460,14 +460,6 @@ int first_possibly_missing_frame = -1; void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, int len, rtsp_conn_info *conn) { - // ignore a request to flush that has been made before the first packet... - if (conn->packet_count == 0) { - debug_mutex_lock(&conn->flush_mutex, 1000, 1); - conn->flush_requested = 0; - conn->flush_rtp_timestamp = 0; - debug_mutex_unlock(&conn->flush_mutex, 3); - } - debug_mutex_lock(&conn->ab_mutex, 30000, 0); uint64_t time_now = get_absolute_time_in_ns(); conn->packet_count++; @@ -477,7 +469,7 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in if ((conn->flush_rtp_timestamp != 0) && (actual_timestamp != conn->flush_rtp_timestamp) && (modulo_32_offset(actual_timestamp, conn->flush_rtp_timestamp) < conn->input_rate * 10)) { // if it's less than 10 seconds - debug(3, + debug(1, "Dropping flushed packet in player_put_packet, seqno %u, timestamp %" PRIu32 ", flushing to " "timestamp: %" PRIu32 ".", @@ -945,34 +937,97 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { conn->connection_state_to_output = rco; // change happening if (conn->connection_state_to_output == 0) { // going off + debug(1, "request flush because connection_state_to_output is off"); debug_mutex_lock(&conn->flush_mutex, 1000, 1); conn->flush_requested = 1; + conn->flush_rtp_timestamp = 0; debug_mutex_unlock(&conn->flush_mutex, 3); } } if (config.output->is_running) if (config.output->is_running() != 0) { // if the back end isn't running for any reason - debug(3, "not running"); + debug(1, "request flush because back end is not running"); debug_mutex_lock(&conn->flush_mutex, 1000, 0); conn->flush_requested = 1; + conn->flush_rtp_timestamp = 0; debug_mutex_unlock(&conn->flush_mutex, 0); } debug_mutex_lock(&conn->flush_mutex, 1000, 0); if (conn->flush_requested == 1) { - if (config.output->flush) - config.output->flush(); // no cancellation points - ab_resync(conn); // no cancellation points - conn->first_packet_timestamp = 0; - conn->first_packet_time_to_play = 0; - conn->time_since_play_started = 0; - conn->flush_requested = 0; - have_sent_prefiller_silence = 0; - dac_delay = 0; + if (conn->flush_output_flushed == 0) + if (config.output->flush) { + config.output->flush(); // no cancellation points + debug(1, "flush request: flush output device."); + } + conn->flush_output_flushed = 1; + // now check to see it the flush request is for frames in the buffer or not + // if the first_packet_timestamp is zero, don't check + int flush_needed = 0; + int drop_request = 0; + if (conn->flush_rtp_timestamp == 0) { + debug(1, "flush request: flush frame 0 -- flush assumed to be needed."); + flush_needed = 1; + drop_request = 1; + } else { + if ((conn->ab_synced) && ((conn->ab_write - conn->ab_read) > 0)) { + abuf_t *firstPacket = conn->audio_buffer + BUFIDX(conn->ab_read); + abuf_t *lastPacket = conn->audio_buffer + BUFIDX(conn->ab_write - 1); + if ((firstPacket != NULL) && (firstPacket->ready)) { + // discard flushes more than 10 seconds into the future -- they are probably bogus + uint32_t first_frame_in_buffer = firstPacket->given_timestamp; + int32_t offset_from_first_frame = (int32_t)(conn->flush_rtp_timestamp - first_frame_in_buffer); + if (offset_from_first_frame > (int)conn->input_rate * 10) { + debug(1, "flush request: sanity check -- flush frame %u is too far into the future from the first frame %u -- discarded.", conn->flush_rtp_timestamp, first_frame_in_buffer); + drop_request = 1; + } else { + if ((lastPacket != NULL) && (lastPacket->ready)) { + // we have enough information to check if the flush is needed or can be discarded + uint32_t last_frame_in_buffer = lastPacket->given_timestamp + lastPacket->length - 1; + // now we have to work out if the flush frame is in the buffer + // if it is later than the end of the buffer, flush everything and keep the request active. + // if it is in the buffer, we need to flush part of the buffer. Actually we flush the entire buffer and drop the request. + // if it is before the buffer, no flush is needed. Drop the request. + if (offset_from_first_frame > 0) { + int32_t offset_to_last_frame = (int32_t)(last_frame_in_buffer - conn->flush_rtp_timestamp); + if (offset_to_last_frame >= 0) { + debug(1,"flush request: flush frame %u active -- buffer contains %u frames, from %u to %u", conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer, first_frame_in_buffer, last_frame_in_buffer); + drop_request = 1; + flush_needed = 1; + } else { + debug(1,"flush request: flush frame %u pending -- buffer contains %u frames, from %u to %u", conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer, first_frame_in_buffer, last_frame_in_buffer); + flush_needed = 1; + } + } else { + debug(1,"flush request: flush frame %u expired -- buffer contains %u frames, from %u to %u", conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer, first_frame_in_buffer, last_frame_in_buffer); + drop_request = 1; + } + } + } + } + } else { + debug(2, "flush request: flush frame %u -- buffer not synced or empty: synced: %d, ab_read: %u, ab_write: %u", conn->flush_rtp_timestamp, conn->ab_synced, conn->ab_read, conn->ab_write); + // leave flush request pending and don't do a buffer flush + } + } + if (flush_needed) { + debug(1, "flush request: flush done."); + ab_resync(conn); // no cancellation points + conn->first_packet_timestamp = 0; + conn->first_packet_time_to_play = 0; + conn->time_since_play_started = 0; + have_sent_prefiller_silence = 0; + dac_delay = 0; + } + if (drop_request) { + debug(1, "flush request: request dropped."); + conn->flush_requested = 0; + conn->flush_rtp_timestamp = 0; + conn->flush_output_flushed = 0; + } } debug_mutex_unlock(&conn->flush_mutex, 0); - if (conn->ab_synced) { curframe = conn->audio_buffer + BUFIDX(conn->ab_read); @@ -992,35 +1047,9 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { debug(1, "Inconsistent sequence numbers detected"); } } - - // if (conn->flush_rtp_timestamp != 0) - // debug(2,"flush_rtp_timestamp is %" PRIx32 " and curframe->given_timestamp is %" PRIx32 - // ".", conn->flush_rtp_timestamp , curframe->given_timestamp); - - if ((conn->flush_rtp_timestamp != 0) && - (curframe->given_timestamp != conn->flush_rtp_timestamp) && - (modulo_32_offset(curframe->given_timestamp, conn->flush_rtp_timestamp) < - conn->input_rate * 10)) { // if it's less than ten seconds - debug(3, - "Dropping flushed packet in buffer_get_frame, seqno %u, timestamp %" PRIu32 - ", flushing to " - "timestamp: %" PRIu32 ".", - curframe->sequence_number, curframe->given_timestamp, conn->flush_rtp_timestamp); - curframe->ready = 0; - curframe->resend_request_number = 0; - curframe = NULL; // this will be returned and will cause the loop to go around again - conn->initial_reference_time = 0; - conn->initial_reference_timestamp = 0; - } else if ((conn->flush_rtp_timestamp != 0) && - (modulo_32_offset(conn->flush_rtp_timestamp, curframe->given_timestamp) > - conn->input_rate / 5) && - (modulo_32_offset(conn->flush_rtp_timestamp, curframe->given_timestamp) < - conn->input_rate * 10)) { - debug(3, "Dropping flush request in buffer_get_frame"); - conn->flush_rtp_timestamp = 0; - } } + if ((curframe) && (curframe->ready)) { notified_buffer_empty = 0; // at least one buffer now -- diagnostic only. if (conn->ab_buffering) { // if we are getting packets but not yet forwarding them to the @@ -1076,6 +1105,8 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { &should_be_time, conn); conn->first_packet_time_to_play = should_be_time; + debug(1,"first_packet_time set for frame %u.", conn->first_packet_timestamp); + if (local_time_now > conn->first_packet_time_to_play) { uint64_t lateness = local_time_now - conn->first_packet_time_to_play; @@ -1133,6 +1164,11 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { effective_latency, // this will go modulo 2^32 &should_be_time, conn); + int64_t change_in_should_be_time = (int64_t)(should_be_time - conn->first_packet_time_to_play); + + if (fabs(0.000001*change_in_should_be_time) > 0.001) // the clock drift estimation might be nudging the estimate, and we can ignore this unless if's more than a microsecond + debug(1,"Change in estimated first_packet_time: %8.4f milliseconds.", 0.000001*change_in_should_be_time); + conn->first_packet_time_to_play = should_be_time; if (local_time_now > conn->first_packet_time_to_play) { @@ -1665,6 +1701,9 @@ void *player_thread_func(void *arg) { conn->ab_synced = 0; conn->first_packet_timestamp = 0; conn->flush_requested = 0; + conn->flush_output_flushed = 0; // only send a flush command to the output device once + conn->flush_rtp_timestamp = 0; // it seems this number has a special significance -- it seems to + // be used as a null operand, so we'll use it like that too conn->fix_volume = 0x10000; if (conn->latency == 0) { @@ -1872,8 +1911,6 @@ void *player_thread_func(void *arg) { die("Failed to allocate memory for an output buffer."); conn->first_packet_timestamp = 0; conn->missing_packets = conn->late_packets = conn->too_late_packets = conn->resend_requests = 0; - conn->flush_rtp_timestamp = 0; // it seems this number has a special significance -- it seems to - // be used as a null operand, so we'll use it like that too int sync_error_out_of_bounds = 0; // number of times in a row that there's been a serious sync error @@ -3020,15 +3057,12 @@ void player_volume(double airplay_volume, rtsp_conn_info *conn) { void do_flush(uint32_t timestamp, rtsp_conn_info *conn) { - debug(3, "Flush requested up to %u. It seems as if 0 is special.", timestamp); + debug(1, "do_flush: flush to %u.", timestamp); debug_mutex_lock(&conn->flush_mutex, 1000, 1); conn->flush_requested = 1; - // if (timestamp!=0) conn->flush_rtp_timestamp = timestamp; // flush all packets up to (and including?) this - // conn->play_segment_reference_frame = 0; reset_input_flow_metrics(conn); debug_mutex_unlock(&conn->flush_mutex, 3); - debug(3, "Flush request made."); } void player_flush(uint32_t timestamp, rtsp_conn_info *conn) { diff --git a/player.h b/player.h index c5713de1..f6b86cd4 100644 --- a/player.h +++ b/player.h @@ -161,6 +161,7 @@ typedef struct { int ab_buffering, ab_synced; int64_t first_packet_timestamp; int flush_requested; + int flush_output_flushed; // true if the output device has been flushed. uint32_t flush_rtp_timestamp; uint64_t time_of_last_audio_packet; seq_t ab_read, ab_write; From e39f86dc10ea8a85f0ab70fd14f6b52205a8eacb Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 12 Jun 2020 17:54:02 +0100 Subject: [PATCH 147/186] remove some unnecessary code for deleting flushed frames --- player.c | 328 +++++++++++++++++++++++++++---------------------------- 1 file changed, 158 insertions(+), 170 deletions(-) diff --git a/player.c b/player.c index f2fef46d..dcfa1811 100644 --- a/player.c +++ b/player.c @@ -466,184 +466,172 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in conn->packet_count_since_flush++; conn->time_of_last_audio_packet = time_now; if (conn->connection_state_to_output) { // if we are supposed to be processing these packets - if ((conn->flush_rtp_timestamp != 0) && (actual_timestamp != conn->flush_rtp_timestamp) && - (modulo_32_offset(actual_timestamp, conn->flush_rtp_timestamp) < - conn->input_rate * 10)) { // if it's less than 10 seconds - debug(1, - "Dropping flushed packet in player_put_packet, seqno %u, timestamp %" PRIu32 - ", flushing to " - "timestamp: %" PRIu32 ".", - seqno, actual_timestamp, conn->flush_rtp_timestamp); - conn->initial_reference_time = 0; - conn->initial_reference_timestamp = 0; - } else { - abuf_t *abuf = 0; - if (!conn->ab_synced) { - // if this is the first packet... - debug(3, "syncing to seqno %u.", seqno); - conn->ab_write = seqno; - conn->ab_read = seqno; - conn->ab_synced = 1; - } - int16_t write_point_gap = seq_diff(seqno,conn->ab_write); // this is the difference between - // the incoming packet number and the packet number that was expected. - if (write_point_gap == 0) { // if this is the expected packet (which could be the first packet...) - if (conn->input_frame_rate_starting_point_is_valid == 0) { - if ((conn->packet_count_since_flush >= 500) && (conn->packet_count_since_flush <= 510)) { - conn->frames_inward_measurement_start_time = time_now; - conn->frames_inward_frames_received_at_measurement_start_time = actual_timestamp; - conn->input_frame_rate_starting_point_is_valid = 1; // valid now - } - } - conn->frames_inward_measurement_time = time_now; - conn->frames_inward_frames_received_at_measurement_time = actual_timestamp; - abuf = conn->audio_buffer + BUFIDX(seqno); - conn->ab_write = SUCCESSOR(seqno); // move the write pointer to the next free space - } else if (write_point_gap > 0) { // newer than expected - // initialise the frames in between - int i; - for (i = 0; i < write_point_gap; i++) { - abuf = conn->audio_buffer + BUFIDX(seq_sum(conn->ab_write, i)); - abuf->ready = 0; // to be sure, to be sure - abuf->resend_request_number = 0; - abuf->initialisation_time = - time_now; // this represents when the packet was noticed to be missing - abuf->status = 1 << 0; // signifying missing - abuf->resend_time = 0; - abuf->given_timestamp = 0; - abuf->sequence_number = 0; - } - abuf = conn->audio_buffer + BUFIDX(seqno); - conn->ab_write = SUCCESSOR(seqno); - } else if (seq_diff(seqno,conn->ab_read) > 0) { // older than expected but still not too late - conn->late_packets++; - abuf = conn->audio_buffer + BUFIDX(seqno); - } else { // too late. - conn->too_late_packets++; - } + abuf_t *abuf = 0; + if (!conn->ab_synced) { + // if this is the first packet... + debug(3, "syncing to seqno %u.", seqno); + conn->ab_write = seqno; + conn->ab_read = seqno; + conn->ab_synced = 1; + } + int16_t write_point_gap = seq_diff(seqno,conn->ab_write); // this is the difference between + // the incoming packet number and the packet number that was expected. + if (write_point_gap == 0) { // if this is the expected packet (which could be the first packet...) + if (conn->input_frame_rate_starting_point_is_valid == 0) { + if ((conn->packet_count_since_flush >= 500) && (conn->packet_count_since_flush <= 510)) { + conn->frames_inward_measurement_start_time = time_now; + conn->frames_inward_frames_received_at_measurement_start_time = actual_timestamp; + conn->input_frame_rate_starting_point_is_valid = 1; // valid now + } + } + conn->frames_inward_measurement_time = time_now; + conn->frames_inward_frames_received_at_measurement_time = actual_timestamp; + abuf = conn->audio_buffer + BUFIDX(seqno); + conn->ab_write = SUCCESSOR(seqno); // move the write pointer to the next free space + } else if (write_point_gap > 0) { // newer than expected + // initialise the frames in between + int i; + for (i = 0; i < write_point_gap; i++) { + abuf = conn->audio_buffer + BUFIDX(seq_sum(conn->ab_write, i)); + abuf->ready = 0; // to be sure, to be sure + abuf->resend_request_number = 0; + abuf->initialisation_time = + time_now; // this represents when the packet was noticed to be missing + abuf->status = 1 << 0; // signifying missing + abuf->resend_time = 0; + abuf->given_timestamp = 0; + abuf->sequence_number = 0; + } + abuf = conn->audio_buffer + BUFIDX(seqno); + conn->ab_write = SUCCESSOR(seqno); + } else if (seq_diff(seqno,conn->ab_read) > 0) { // older than expected but still not too late + conn->late_packets++; + abuf = conn->audio_buffer + BUFIDX(seqno); + } else { // too late. + conn->too_late_packets++; + } - if (abuf) { - int datalen = conn->max_frames_per_packet; - abuf->initialisation_time = time_now; - abuf->resend_time = 0; - if (audio_packet_decode(abuf->data, &datalen, data, len, conn) == 0) { - abuf->ready = 1; - abuf->status = 0; // signifying that it was received - abuf->length = datalen; - abuf->given_timestamp = actual_timestamp; - abuf->sequence_number = seqno; - } else { - debug(1, "Bad audio packet detected and discarded."); - abuf->ready = 0; - abuf->status = 1 << 1; // bad packet, discarded - abuf->resend_request_number = 0; - abuf->given_timestamp = 0; - abuf->sequence_number = 0; - } - } + if (abuf) { + int datalen = conn->max_frames_per_packet; + abuf->initialisation_time = time_now; + abuf->resend_time = 0; + if (audio_packet_decode(abuf->data, &datalen, data, len, conn) == 0) { + abuf->ready = 1; + abuf->status = 0; // signifying that it was received + abuf->length = datalen; + abuf->given_timestamp = actual_timestamp; + abuf->sequence_number = seqno; + } else { + debug(1, "Bad audio packet detected and discarded."); + abuf->ready = 0; + abuf->status = 1 << 1; // bad packet, discarded + abuf->resend_request_number = 0; + abuf->given_timestamp = 0; + abuf->sequence_number = 0; + } + } - int rc = pthread_cond_signal(&conn->flowcontrol); - if (rc) - debug(1, "Error signalling flowcontrol."); + int rc = pthread_cond_signal(&conn->flowcontrol); + if (rc) + debug(1, "Error signalling flowcontrol."); - // resend checks - { - uint64_t minimum_wait_time = - (uint64_t)(config.resend_control_first_check_time * (uint64_t)1000000000); - uint64_t resend_repeat_interval = - (uint64_t)(config.resend_control_check_interval_time * (uint64_t)1000000000); - uint64_t minimum_remaining_time = (uint64_t)( - (config.resend_control_last_check_time + config.audio_backend_buffer_desired_length) * - (uint64_t)1000000000); - uint64_t latency_time = (uint64_t)(conn->latency * (uint64_t)1000000000); - latency_time = latency_time / (uint64_t)conn->input_rate; + // resend checks + { + uint64_t minimum_wait_time = + (uint64_t)(config.resend_control_first_check_time * (uint64_t)1000000000); + uint64_t resend_repeat_interval = + (uint64_t)(config.resend_control_check_interval_time * (uint64_t)1000000000); + uint64_t minimum_remaining_time = (uint64_t)( + (config.resend_control_last_check_time + config.audio_backend_buffer_desired_length) * + (uint64_t)1000000000); + uint64_t latency_time = (uint64_t)(conn->latency * (uint64_t)1000000000); + latency_time = latency_time / (uint64_t)conn->input_rate; - int x; // this is the first frame to be checked - // if we detected a first empty frame before and if it's still in the buffer! - if ((first_possibly_missing_frame >= 0) && - (position_in_modulo_uint16_t_buffer(first_possibly_missing_frame, conn->ab_read, - conn->ab_write, NULL))) { - x = first_possibly_missing_frame; - } else { - x = conn->ab_read; - } + int x; // this is the first frame to be checked + // if we detected a first empty frame before and if it's still in the buffer! + if ((first_possibly_missing_frame >= 0) && + (position_in_modulo_uint16_t_buffer(first_possibly_missing_frame, conn->ab_read, + conn->ab_write, NULL))) { + x = first_possibly_missing_frame; + } else { + x = conn->ab_read; + } - first_possibly_missing_frame = -1; // has not been set + first_possibly_missing_frame = -1; // has not been set - int missing_frame_run_count = 0; - int start_of_missing_frame_run = -1; - int number_of_missing_frames = 0; - while (x != conn->ab_write) { - abuf_t *check_buf = conn->audio_buffer + BUFIDX(x); - if (!check_buf->ready) { - if (first_possibly_missing_frame < 0) - first_possibly_missing_frame = x; - number_of_missing_frames++; - // debug(1, "frame %u's initialisation_time is 0x%" PRIx64 ", latency_time is 0x%" - // PRIx64 ", time_now is 0x%" PRIx64 ", minimum_remaining_time is 0x%" PRIx64 ".", x, - // check_buf->initialisation_time, latency_time, time_now, minimum_remaining_time); - int too_late = ((check_buf->initialisation_time < (time_now - latency_time)) || - ((check_buf->initialisation_time - (time_now - latency_time)) < - minimum_remaining_time)); - int too_early = ((time_now - check_buf->initialisation_time) < minimum_wait_time); - int too_soon_after_last_request = - ((check_buf->resend_time != 0) && - ((time_now - check_buf->resend_time) < - resend_repeat_interval)); // time_now can never be less than the time_tag + int missing_frame_run_count = 0; + int start_of_missing_frame_run = -1; + int number_of_missing_frames = 0; + while (x != conn->ab_write) { + abuf_t *check_buf = conn->audio_buffer + BUFIDX(x); + if (!check_buf->ready) { + if (first_possibly_missing_frame < 0) + first_possibly_missing_frame = x; + number_of_missing_frames++; + // debug(1, "frame %u's initialisation_time is 0x%" PRIx64 ", latency_time is 0x%" + // PRIx64 ", time_now is 0x%" PRIx64 ", minimum_remaining_time is 0x%" PRIx64 ".", x, + // check_buf->initialisation_time, latency_time, time_now, minimum_remaining_time); + int too_late = ((check_buf->initialisation_time < (time_now - latency_time)) || + ((check_buf->initialisation_time - (time_now - latency_time)) < + minimum_remaining_time)); + int too_early = ((time_now - check_buf->initialisation_time) < minimum_wait_time); + int too_soon_after_last_request = + ((check_buf->resend_time != 0) && + ((time_now - check_buf->resend_time) < + resend_repeat_interval)); // time_now can never be less than the time_tag - if (too_late) - check_buf->status |= 1 << 2; // too late - else - check_buf->status &= 0xFF - (1 << 2); // not too late - if (too_early) - check_buf->status |= 1 << 3; // too early - else - check_buf->status &= 0xFF - (1 << 3); // not too early - if (too_soon_after_last_request) - check_buf->status |= 1 << 4; // too soon after last request - else - check_buf->status &= 0xFF - (1 << 4); // not too soon after last request + if (too_late) + check_buf->status |= 1 << 2; // too late + else + check_buf->status &= 0xFF - (1 << 2); // not too late + if (too_early) + check_buf->status |= 1 << 3; // too early + else + check_buf->status &= 0xFF - (1 << 3); // not too early + if (too_soon_after_last_request) + check_buf->status |= 1 << 4; // too soon after last request + else + check_buf->status &= 0xFF - (1 << 4); // not too soon after last request - if ((!too_soon_after_last_request) && (!too_late) && (!too_early)) { - if (start_of_missing_frame_run == -1) { - start_of_missing_frame_run = x; - missing_frame_run_count = 1; - } else { - missing_frame_run_count++; - } - check_buf->resend_time = time_now; // setting the time to now because we are - // definitely going to take action - check_buf->resend_request_number++; - debug(3, "Frame %d is missing with ab_read of %u and ab_write of %u.", x, - conn->ab_read, conn->ab_write); - } - // if (too_late) { - // debug(1,"too late to get missing frame %u.", x); - // } - } - // if (number_of_missing_frames != 0) - // debug(1,"check with x = %u, ab_read = %u, ab_write = %u, first_possibly_missing_frame - // = %d.", x, conn->ab_read, conn->ab_write, first_possibly_missing_frame); - x = (x + 1) & 0xffff; - if (((check_buf->ready) || (x == conn->ab_write)) && (missing_frame_run_count > 0)) { - // send a resend request - if (missing_frame_run_count > 1) - debug(3, "request resend of %d packets starting at seqno %u.", - missing_frame_run_count, start_of_missing_frame_run); - if (config.disable_resend_requests == 0) { - debug_mutex_unlock(&conn->ab_mutex, 3); - rtp_request_resend(start_of_missing_frame_run, missing_frame_run_count, conn); - debug_mutex_lock(&conn->ab_mutex, 20000, 1); - conn->resend_requests++; - } - start_of_missing_frame_run = -1; - missing_frame_run_count = 0; - } - } - if (number_of_missing_frames == 0) - first_possibly_missing_frame = conn->ab_write; - } - } + if ((!too_soon_after_last_request) && (!too_late) && (!too_early)) { + if (start_of_missing_frame_run == -1) { + start_of_missing_frame_run = x; + missing_frame_run_count = 1; + } else { + missing_frame_run_count++; + } + check_buf->resend_time = time_now; // setting the time to now because we are + // definitely going to take action + check_buf->resend_request_number++; + debug(3, "Frame %d is missing with ab_read of %u and ab_write of %u.", x, + conn->ab_read, conn->ab_write); + } + // if (too_late) { + // debug(1,"too late to get missing frame %u.", x); + // } + } + // if (number_of_missing_frames != 0) + // debug(1,"check with x = %u, ab_read = %u, ab_write = %u, first_possibly_missing_frame + // = %d.", x, conn->ab_read, conn->ab_write, first_possibly_missing_frame); + x = (x + 1) & 0xffff; + if (((check_buf->ready) || (x == conn->ab_write)) && (missing_frame_run_count > 0)) { + // send a resend request + if (missing_frame_run_count > 1) + debug(3, "request resend of %d packets starting at seqno %u.", + missing_frame_run_count, start_of_missing_frame_run); + if (config.disable_resend_requests == 0) { + debug_mutex_unlock(&conn->ab_mutex, 3); + rtp_request_resend(start_of_missing_frame_run, missing_frame_run_count, conn); + debug_mutex_lock(&conn->ab_mutex, 20000, 1); + conn->resend_requests++; + } + start_of_missing_frame_run = -1; + missing_frame_run_count = 0; + } + } + if (number_of_missing_frames == 0) + first_possibly_missing_frame = conn->ab_write; + } } debug_mutex_unlock(&conn->ab_mutex, 0); } From 34d0a4df8b9affe15b4f626e78dd137731d00c66 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 12 Jun 2020 18:17:44 +0100 Subject: [PATCH 148/186] Fix a debug message --- player.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/player.c b/player.c index dcfa1811..32e4e532 100644 --- a/player.c +++ b/player.c @@ -980,15 +980,15 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { if (offset_from_first_frame > 0) { int32_t offset_to_last_frame = (int32_t)(last_frame_in_buffer - conn->flush_rtp_timestamp); if (offset_to_last_frame >= 0) { - debug(1,"flush request: flush frame %u active -- buffer contains %u frames, from %u to %u", conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer, first_frame_in_buffer, last_frame_in_buffer); + debug(1,"flush request: flush frame %u active -- buffer contains %u frames, from %u to %u", conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer + 1, first_frame_in_buffer, last_frame_in_buffer); drop_request = 1; flush_needed = 1; } else { - debug(1,"flush request: flush frame %u pending -- buffer contains %u frames, from %u to %u", conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer, first_frame_in_buffer, last_frame_in_buffer); + debug(1,"flush request: flush frame %u pending -- buffer contains %u frames, from %u to %u", conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer + 1, first_frame_in_buffer, last_frame_in_buffer); flush_needed = 1; } } else { - debug(1,"flush request: flush frame %u expired -- buffer contains %u frames, from %u to %u", conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer, first_frame_in_buffer, last_frame_in_buffer); + debug(1,"flush request: flush frame %u expired -- buffer contains %u frames, from %u to %u", conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer + 1, first_frame_in_buffer, last_frame_in_buffer); drop_request = 1; } } From 8dfaaf8d8d56fae403e6226306dfd041c7433314 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 12 Jun 2020 19:20:58 +0100 Subject: [PATCH 149/186] Add some detail to a debug message. --- player.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/player.c b/player.c index 32e4e532..cfa58f32 100644 --- a/player.c +++ b/player.c @@ -2259,7 +2259,7 @@ void *player_thread_func(void *arg) { // remove the bias when reporting the error to make it the true error - debug(2,"first frame sync error (positive --> late): %" PRId64 " frames, %.3f mS at %d frames per second output.", sync_error+first_frame_early_bias, (1000.0*(sync_error+first_frame_early_bias))/config.output_rate, config.output_rate); + debug(1,"first frame sync error (positive --> late): %" PRId64 " frames, %.3f mS at %d frames per second output.", sync_error+first_frame_early_bias, (1000.0*(sync_error+first_frame_early_bias))/config.output_rate, config.output_rate); // if the packet is early, add the frames needed to put it in sync. if (sync_error < 0) { @@ -2272,7 +2272,7 @@ void *player_thread_func(void *arg) { conn->enable_dither, conn->previous_random_number); int final_adjustment = -sync_error; final_adjustment = final_adjustment - first_frame_early_bias; - debug(2, "final sync adjustment: %d frames.", final_adjustment); + debug(1, "final sync adjustment: %" PRId64 " silent frames added with a bias of %" PRId64 " frames.", -sync_error, first_frame_early_bias); config.output->play(final_adjustment_silence, final_adjustment_length_sized); free(final_adjustment_silence); } else { From 493260b04a40d757d0e5fd558044d5a6797b494a Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 13 Jun 2020 09:37:03 +0100 Subject: [PATCH 150/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index b941a530..3c3fee99 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,11 @@ +Version 3.3.7d10 +==== +**Bug Fix** +* Fixed a bug whereby the start and end of active mode tokens `abeg` and `aend` where not generated or published -- `pend` tokens were being generated instead. Thanks to [minix1234](https://github.com/minix1234) for the bug report and [fix](https://github.com/mikebrady/shairport-sync/pull/1023). + +**Enhancements** +* Cleaned up and simplified the code that handles `FLUSH` requests coming from the player. (Debug messages are still a little verbose.) + Version 3.3.7d9 ==== **Enhancements** From f85c853c14b657385b4bd3aab7fc57047dd82024 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 13 Jun 2020 09:38:12 +0100 Subject: [PATCH 151/186] Fix metadata `abeg` and `aend` generation bug. --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 22facc97..d8bd7dc6 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.7d9], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.7d10], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 4f9f3331e35234bba09a58ecc2e1f2cac226ef31 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 13 Jun 2020 10:41:48 +0100 Subject: [PATCH 152/186] Remove the flsr flush request token -- sometimes it's bogus -- add the frame number to the pfls token. Add a little documentation. --- player.c | 7 +++++-- rtsp.c | 13 ++++++++++++- 2 files changed, 17 insertions(+), 3 deletions(-) diff --git a/player.c b/player.c index cfa58f32..2fd6bc88 100644 --- a/player.c +++ b/player.c @@ -3059,9 +3059,12 @@ void player_flush(uint32_t timestamp, rtsp_conn_info *conn) { #ifdef CONFIG_METADATA // only send a flush metadata message if the first packet has been seen -- it's a bogus message // otherwise + if (conn->first_packet_timestamp) { - debug(2, "pfls"); - send_ssnc_metadata('pfls', NULL, 0, 1); // contains cancellation points + debug(2, "pfls"); + char numbuf[32]; + snprintf(numbuf, sizeof(numbuf),"%u",timestamp); + send_ssnc_metadata('pfls', numbuf, strlen(numbuf), 1); // contains cancellation points } #endif } diff --git a/rtsp.c b/rtsp.c index 9057b983..4296bc19 100644 --- a/rtsp.c +++ b/rtsp.c @@ -926,12 +926,17 @@ void handle_flush(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp) { } } // debug(1,"RTSP Flush Requested: %u.",rtptime); + +// the following is now done better by the player_flush routine as a 'pfls' +/* #ifdef CONFIG_METADATA if (p) send_metadata('ssnc', 'flsr', p + 1, strlen(p + 1), req, 1); else send_metadata('ssnc', 'flsr', NULL, 0, NULL, 0); #endif +*/ + player_flush(rtptime, conn); // will not crash even it there is no player thread. resp->respcode = 200; @@ -1145,7 +1150,10 @@ void handle_set_parameter_parameter(rtsp_conn_info *conn, rtsp_message *req, // 'aend' -- active mode exited. No arguments // 'pbeg' -- play stream begin. No arguments // 'pend' -- play stream end. No arguments -// 'pfls' -- play stream flush. No arguments +// 'pfls' -- play stream flush. The argument is an unsigned 32-bit +// frame number. It seems that all frames up to but not +// including this frame are to be flushed. +// // 'prsm' -- play stream resume. No arguments // `pffr` -- the first frame of a play session has been received and has been validly // timed. @@ -1185,8 +1193,11 @@ void handle_set_parameter_parameter(rtsp_conn_info *conn, rtsp_message *req, // to send commands to the source's remote control (if it has one). // `clip` -- the payload is the IP number of the client, i.e. the sender of audio. // Can be an IPv4 or an IPv6 number. +// `svip` -- the payload is the IP number of the server, i.e. the player itself. +// Can be an IPv4 or an IPv6 number. // `dapo` -- the payload is the port number (as text) on the server to which remote // control commands should be sent. It is 3689 for iTunes but varies for iOS devices. + // A special sub-protocol is used for sending large data items over UDP // If the payload exceeded 4 MB, it is chunked using the following format: // "ssnc", "chnk", packet_ix, packet_counts, packet_tag, packet_type, chunked_data. From 31486a36f323fc6e691a5039cfae9e6f11e72589 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sat, 13 Jun 2020 14:38:27 +0100 Subject: [PATCH 153/186] don't suppress the sending of a pfls token until some frames have been seen. --- player.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/player.c b/player.c index 2fd6bc88..c4f15310 100644 --- a/player.c +++ b/player.c @@ -3057,15 +3057,10 @@ void player_flush(uint32_t timestamp, rtsp_conn_info *conn) { debug(3, "player_flush"); do_flush(timestamp, conn); #ifdef CONFIG_METADATA - // only send a flush metadata message if the first packet has been seen -- it's a bogus message - // otherwise - - if (conn->first_packet_timestamp) { - debug(2, "pfls"); - char numbuf[32]; - snprintf(numbuf, sizeof(numbuf),"%u",timestamp); - send_ssnc_metadata('pfls', numbuf, strlen(numbuf), 1); // contains cancellation points - } + debug(2, "pfls"); + char numbuf[32]; + snprintf(numbuf, sizeof(numbuf),"%u",timestamp); + send_ssnc_metadata('pfls', numbuf, strlen(numbuf), 1); // contains cancellation points #endif } From ead6cfc4add2226f74fdfe9244b35d7239f2c02e Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 14 Jun 2020 09:49:10 +0100 Subject: [PATCH 154/186] Emphasise one debug message --- rtp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/rtp.c b/rtp.c index a8f03f6a..4e6828e3 100644 --- a/rtp.c +++ b/rtp.c @@ -812,7 +812,7 @@ void *rtp_timing_receiver(void *arg) { // samples.",conn->local_to_remote_time_gradient*1000000,sample_count); } else { - debug(2, + debug(1, "Time ping turnaround time: %" PRIu64 " ns -- it looks like a timing ping was lost.", return_time); From f89830eb2c70c54dc8892391f204be65494e5692 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 14 Jun 2020 14:26:22 +0100 Subject: [PATCH 155/186] Send more timing sync requests at the start. The debug messages are a bit noisy. --- rtp.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/rtp.c b/rtp.c index 4e6828e3..4613d627 100644 --- a/rtp.c +++ b/rtp.c @@ -511,8 +511,8 @@ void *rtp_timing_sender(void *arg) { request_number++; - if (request_number <= 4) - usleep(500000); // these are thread cancellation points + if (request_number <= 6) + usleep(300000); // these are thread cancellation points else usleep(3000000); } @@ -621,10 +621,11 @@ void *rtp_timing_receiver(void *arg) { if (packet[1] == 0xd3) { // timing reply return_time = arrival_time - conn->departure_time; + debug(1,"clock synchronisation request: return time is %8.3f milliseconds.",0.000001*return_time); - if (return_time < 300000000) { // must be less than 0.3 seconds + if (return_time < 200000000) { // must be less than 0.2 seconds - //debug(1,"Ping return time is %8.3f milliseconds.",0.000001*return_time); + debug(1,"Ping return time is %8.3f milliseconds.",0.000001*return_time); // distant_receive_time = // ((uint64_t)ntohl(*((uint32_t*)&packet[16])))<<32+ntohl(*((uint32_t*)&packet[20])); From 9d6adc6f1f1f75c4a16203b25b797c5c53a3ad82 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 14 Jun 2020 14:29:55 +0100 Subject: [PATCH 156/186] one less debug message. --- rtp.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/rtp.c b/rtp.c index 4613d627..ba181947 100644 --- a/rtp.c +++ b/rtp.c @@ -624,9 +624,6 @@ void *rtp_timing_receiver(void *arg) { debug(1,"clock synchronisation request: return time is %8.3f milliseconds.",0.000001*return_time); if (return_time < 200000000) { // must be less than 0.2 seconds - - debug(1,"Ping return time is %8.3f milliseconds.",0.000001*return_time); - // distant_receive_time = // ((uint64_t)ntohl(*((uint32_t*)&packet[16])))<<32+ntohl(*((uint32_t*)&packet[20])); From 526a86b73224319325662611e906e29a759973cc Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 14 Jun 2020 15:58:06 +0100 Subject: [PATCH 157/186] remove an insignificant FreeBSD compiler warning. --- common.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/common.c b/common.c index 480410c1..a129792a 100644 --- a/common.c +++ b/common.c @@ -166,7 +166,7 @@ int create_log_file(const char* path) { if (fd >= 0) { // now we switch to blocking mode int flags = fcntl(fd, F_GETFL); - if ((flags == -1)) { + if (flags == -1) { // strerror_r(errno, (char *)errorstring, sizeof(errorstring)); // debug(1, "create_log_file -- error %d (\"%s\") getting flags of pipe: \"%s\".", errno, // (char *)errorstring, pathname); @@ -1147,7 +1147,7 @@ int try_to_open_pipe_for_writing(const char* pathname) { if (fdis >= 0) { // now we switch to blocking mode int flags = fcntl(fdis, F_GETFL); - if ((flags == -1)) { + if (flags == -1) { char errorstring[1024]; strerror_r(errno, (char *)errorstring, sizeof(errorstring)); debug(1, "try_to_open_pipe -- error %d (\"%s\") getting flags of pipe: \"%s\".", errno, From 320baa4db916948933fec5a0859b47e4b285fc24 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 15 Jun 2020 12:16:59 +0100 Subject: [PATCH 158/186] quieten some debug messages --- player.c | 30 +++++++++++++++--------------- rtp.c | 2 +- 2 files changed, 16 insertions(+), 16 deletions(-) diff --git a/player.c b/player.c index c4f15310..0ce0df85 100644 --- a/player.c +++ b/player.c @@ -925,7 +925,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { conn->connection_state_to_output = rco; // change happening if (conn->connection_state_to_output == 0) { // going off - debug(1, "request flush because connection_state_to_output is off"); + debug(2, "request flush because connection_state_to_output is off"); debug_mutex_lock(&conn->flush_mutex, 1000, 1); conn->flush_requested = 1; conn->flush_rtp_timestamp = 0; @@ -935,7 +935,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { if (config.output->is_running) if (config.output->is_running() != 0) { // if the back end isn't running for any reason - debug(1, "request flush because back end is not running"); + debug(2, "request flush because back end is not running"); debug_mutex_lock(&conn->flush_mutex, 1000, 0); conn->flush_requested = 1; conn->flush_rtp_timestamp = 0; @@ -947,7 +947,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { if (conn->flush_output_flushed == 0) if (config.output->flush) { config.output->flush(); // no cancellation points - debug(1, "flush request: flush output device."); + debug(2, "flush request: flush output device."); } conn->flush_output_flushed = 1; // now check to see it the flush request is for frames in the buffer or not @@ -980,15 +980,15 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { if (offset_from_first_frame > 0) { int32_t offset_to_last_frame = (int32_t)(last_frame_in_buffer - conn->flush_rtp_timestamp); if (offset_to_last_frame >= 0) { - debug(1,"flush request: flush frame %u active -- buffer contains %u frames, from %u to %u", conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer + 1, first_frame_in_buffer, last_frame_in_buffer); + debug(2,"flush request: flush frame %u active -- buffer contains %u frames, from %u to %u", conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer + 1, first_frame_in_buffer, last_frame_in_buffer); drop_request = 1; flush_needed = 1; } else { - debug(1,"flush request: flush frame %u pending -- buffer contains %u frames, from %u to %u", conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer + 1, first_frame_in_buffer, last_frame_in_buffer); + debug(2,"flush request: flush frame %u pending -- buffer contains %u frames, from %u to %u", conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer + 1, first_frame_in_buffer, last_frame_in_buffer); flush_needed = 1; } } else { - debug(1,"flush request: flush frame %u expired -- buffer contains %u frames, from %u to %u", conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer + 1, first_frame_in_buffer, last_frame_in_buffer); + debug(2,"flush request: flush frame %u expired -- buffer contains %u frames, from %u to %u", conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer + 1, first_frame_in_buffer, last_frame_in_buffer); drop_request = 1; } } @@ -996,11 +996,11 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { } } else { debug(2, "flush request: flush frame %u -- buffer not synced or empty: synced: %d, ab_read: %u, ab_write: %u", conn->flush_rtp_timestamp, conn->ab_synced, conn->ab_read, conn->ab_write); - // leave flush request pending and don't do a buffer flush + // leave flush request pending and don't do a buffer flush, because there isn't one } } if (flush_needed) { - debug(1, "flush request: flush done."); + debug(2, "flush request: flush done."); ab_resync(conn); // no cancellation points conn->first_packet_timestamp = 0; conn->first_packet_time_to_play = 0; @@ -1009,7 +1009,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { dac_delay = 0; } if (drop_request) { - debug(1, "flush request: request dropped."); + debug(2, "flush request: request dropped."); conn->flush_requested = 0; conn->flush_rtp_timestamp = 0; conn->flush_output_flushed = 0; @@ -1093,7 +1093,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { &should_be_time, conn); conn->first_packet_time_to_play = should_be_time; - debug(1,"first_packet_time set for frame %u.", conn->first_packet_timestamp); + debug(2,"first_packet_time set for frame %u.", conn->first_packet_timestamp); if (local_time_now > conn->first_packet_time_to_play) { @@ -1155,7 +1155,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { int64_t change_in_should_be_time = (int64_t)(should_be_time - conn->first_packet_time_to_play); if (fabs(0.000001*change_in_should_be_time) > 0.001) // the clock drift estimation might be nudging the estimate, and we can ignore this unless if's more than a microsecond - debug(1,"Change in estimated first_packet_time: %8.4f milliseconds.", 0.000001*change_in_should_be_time); + debug(2,"Change in estimated first_packet_time: %8.4f milliseconds.", 0.000001*change_in_should_be_time); conn->first_packet_time_to_play = should_be_time; @@ -2259,7 +2259,7 @@ void *player_thread_func(void *arg) { // remove the bias when reporting the error to make it the true error - debug(1,"first frame sync error (positive --> late): %" PRId64 " frames, %.3f mS at %d frames per second output.", sync_error+first_frame_early_bias, (1000.0*(sync_error+first_frame_early_bias))/config.output_rate, config.output_rate); + debug(2,"first frame sync error (positive --> late): %" PRId64 " frames, %.3f mS at %d frames per second output.", sync_error+first_frame_early_bias, (1000.0*(sync_error+first_frame_early_bias))/config.output_rate, config.output_rate); // if the packet is early, add the frames needed to put it in sync. if (sync_error < 0) { @@ -2272,7 +2272,7 @@ void *player_thread_func(void *arg) { conn->enable_dither, conn->previous_random_number); int final_adjustment = -sync_error; final_adjustment = final_adjustment - first_frame_early_bias; - debug(1, "final sync adjustment: %" PRId64 " silent frames added with a bias of %" PRId64 " frames.", -sync_error, first_frame_early_bias); + debug(2, "final sync adjustment: %" PRId64 " silent frames added with a bias of %" PRId64 " frames.", -sync_error, first_frame_early_bias); config.output->play(final_adjustment_silence, final_adjustment_length_sized); free(final_adjustment_silence); } else { @@ -3045,10 +3045,10 @@ void player_volume(double airplay_volume, rtsp_conn_info *conn) { void do_flush(uint32_t timestamp, rtsp_conn_info *conn) { - debug(1, "do_flush: flush to %u.", timestamp); + debug(2, "do_flush: flush to %u.", timestamp); debug_mutex_lock(&conn->flush_mutex, 1000, 1); conn->flush_requested = 1; - conn->flush_rtp_timestamp = timestamp; // flush all packets up to (and including?) this + conn->flush_rtp_timestamp = timestamp; // flush all packets up to, but not including, this one. reset_input_flow_metrics(conn); debug_mutex_unlock(&conn->flush_mutex, 3); } diff --git a/rtp.c b/rtp.c index ba181947..e56c5454 100644 --- a/rtp.c +++ b/rtp.c @@ -621,7 +621,7 @@ void *rtp_timing_receiver(void *arg) { if (packet[1] == 0xd3) { // timing reply return_time = arrival_time - conn->departure_time; - debug(1,"clock synchronisation request: return time is %8.3f milliseconds.",0.000001*return_time); + debug(3,"clock synchronisation request: return time is %8.3f milliseconds.",0.000001*return_time); if (return_time < 200000000) { // must be less than 0.2 seconds // distant_receive_time = From 4f0794d6cfcae6a0c95d8595e1dc8a91996de8c8 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Fri, 19 Jun 2020 16:53:15 +0100 Subject: [PATCH 159/186] stop an uninitialised variable warning --- metadata_hub.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/metadata_hub.c b/metadata_hub.c index 3dc42c5a..f1a02a22 100644 --- a/metadata_hub.c +++ b/metadata_hub.c @@ -329,7 +329,7 @@ void metadata_hub_process_metadata(uint32_t type, uint32_t code, char *data, uin // comes in if the metadata_packet_item_changed is set. int changed = 0; - int metadata_packet_item_changed; + int metadata_packet_item_changed = 0; metadata_hub_modify_prolog(); pthread_cleanup_push(metadata_hub_unlock_hub_mutex_cleanup, NULL); From f266e4c08bb57263ec77330aa4528bf850e8c245 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 21 Jun 2020 09:57:05 +0100 Subject: [PATCH 160/186] Allow Shairport Sync native D-Bus and MPRIS services to be owned by root as well as user "shairport-sync" --- scripts/shairport-sync-dbus-policy.conf | 6 +++++- scripts/shairport-sync-mpris-policy.conf | 5 ++++- 2 files changed, 9 insertions(+), 2 deletions(-) diff --git a/scripts/shairport-sync-dbus-policy.conf b/scripts/shairport-sync-dbus-policy.conf index 497dd885..2d42619c 100644 --- a/scripts/shairport-sync-dbus-policy.conf +++ b/scripts/shairport-sync-dbus-policy.conf @@ -4,11 +4,15 @@ "http://www.freedesktop.org/standards/dbus/1.0/busconfig.dtd"> - + + + + + diff --git a/scripts/shairport-sync-mpris-policy.conf b/scripts/shairport-sync-mpris-policy.conf index 1efb0072..17ab847a 100644 --- a/scripts/shairport-sync-mpris-policy.conf +++ b/scripts/shairport-sync-mpris-policy.conf @@ -4,10 +4,13 @@ "http://www.freedesktop.org/standards/dbus/1.0/busconfig.dtd"> - + + + + From 4e1f74db87f191b5bb6d489c7b822ba2b5b2155b Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 21 Jun 2020 13:16:08 +0100 Subject: [PATCH 161/186] Quieten some debug messages. --- dacp.c | 2 +- mdns_avahi.c | 10 +++++----- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/dacp.c b/dacp.c index 82fc0dbb..773ba065 100644 --- a/dacp.c +++ b/dacp.c @@ -996,7 +996,7 @@ void dacp_monitor_stop() { debug(3, "DACP Conversation Lock Mutex Destroyed"); pthread_mutex_destroy(&dacp_conversation_lock); pthread_cond_destroy(&dacp_server_information_cv); - debug(1, "DACP Server Information Condition Variable destroyed."); + debug(3, "DACP Server Information Condition Variable destroyed."); } } diff --git a/mdns_avahi.c b/mdns_avahi.c index 65bd6477..f3c1f1c9 100644 --- a/mdns_avahi.c +++ b/mdns_avahi.c @@ -342,17 +342,17 @@ static int avahi_register(char *srvname, int srvport) { static void avahi_unregister(void) { // debug(1, "avahi_unregister."); if (tpoll) { - debug(1, "avahi: stop the threaded poll."); + debug(2, "avahi: stop the threaded poll."); avahi_threaded_poll_stop(tpoll); if (client) { - debug(1, "avahi: free the client."); + debug(2, "avahi: free the client."); avahi_client_free(client); client = NULL; } else { debug(1, "avahi attempting to unregister a NULL client"); } - debug(1, "avahi: free the threaded poll."); + debug(2, "avahi: free the threaded poll."); avahi_threaded_poll_free(tpoll); tpoll = NULL; } else { @@ -360,7 +360,7 @@ static void avahi_unregister(void) { } if (service_name) { - debug(1, "avahi: free the service name."); + debug(2, "avahi: free the service name."); free(service_name); } else debug(1, "avahi attempt to free NULL service name"); @@ -414,7 +414,7 @@ void avahi_dacp_monitor_stop() { } avahi_threaded_poll_unlock(tpoll); free(dbs->dacp_id); - debug(1, "avahi_dacp_monitor_stop Avahi DACP monitor successfully stopped"); + debug(2, "avahi_dacp_monitor_stop Avahi DACP monitor successfully stopped"); } mdns_backend mdns_avahi = {.name = "avahi", From dd0d24b2558330addee6a6fe1f93da0dadcb5349 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 21 Jun 2020 13:17:57 +0100 Subject: [PATCH 162/186] The metadata threads were not terminating properly due to multiple errors. Quietened a few debug messages too. --- rtsp.c | 34 ++++++++++++++++++++++++---------- 1 file changed, 24 insertions(+), 10 deletions(-) diff --git a/rtsp.c b/rtsp.c index 4296bc19..301a6b87 100644 --- a/rtsp.c +++ b/rtsp.c @@ -174,14 +174,18 @@ void pc_queue_init(pc_queue *the_queue, char *items, size_t item_size, uint32_t void pc_queue_delete(pc_queue *the_queue) { if (the_queue->name) - debug(1, "Deleting metadata queue \"%s\".", the_queue->name); + debug(2, "Deleting metadata queue \"%s\".", the_queue->name); else debug(1, "Deleting an unnamed metadata queue."); if (the_queue->name != NULL) free(the_queue->name); + // debug(2, "destroying pc_queue_item_removed_signal"); pthread_cond_destroy(&the_queue->pc_queue_item_removed_signal); + // debug(2, "destroying pc_queue_item_added_signal"); pthread_cond_destroy(&the_queue->pc_queue_item_added_signal); + // debug(2, "destroying pc_queue_lock"); pthread_mutex_destroy(&the_queue->pc_queue_lock); + // debug(2, "destroying signals and locks done"); } int send_metadata(uint32_t type, uint32_t code, char *data, uint32_t length, rtsp_message *carrier, @@ -1478,16 +1482,17 @@ void metadata_process(uint32_t type, uint32_t code, char *data, uint32_t length) } void metadata_pack_cleanup_function(void *arg) { - // debug(1, "metadata_pack_cleanup_function called"); + debug(1, "metadata_pack_cleanup_function called"); metadata_package *pack = (metadata_package *)arg; if (pack->carrier) msg_free(&pack->carrier); // release the message else if (pack->data) free(pack->data); + debug(1, "metadata_pack_cleanup_function exit"); } void metadata_thread_cleanup_function(__attribute__((unused)) void *arg) { - debug(2, "metadata_thread_cleanup_function called"); + // debug(2, "metadata_thread_cleanup_function called"); metadata_close(); pc_queue_delete(&metadata_queue); } @@ -1518,7 +1523,7 @@ void *metadata_thread_function(__attribute__((unused)) void *ignore) { } void metadata_multicast_thread_cleanup_function(__attribute__((unused)) void *arg) { - debug(2, "metadata_multicast_thread_cleanup_function called"); + // debug(2, "metadata_multicast_thread_cleanup_function called"); metadata_delete_multicast_socket(); pc_queue_delete(&metadata_multicast_queue); } @@ -1554,7 +1559,7 @@ void metadata_hub_close(void) { } void metadata_hub_thread_cleanup_function(__attribute__((unused)) void *arg) { - debug(2, "metadata_hub_thread_cleanup_function called"); + // debug(2, "metadata_hub_thread_cleanup_function called"); metadata_hub_close(); pc_queue_delete(&metadata_hub_queue); } @@ -1614,9 +1619,10 @@ void metadata_mqtt_close(void) { } void metadata_mqtt_thread_cleanup_function(__attribute__((unused)) void *arg) { - debug(2, "metadata_mqtt_thread_cleanup_function called"); + // debug(2, "metadata_mqtt_thread_cleanup_function called"); metadata_mqtt_close(); - pc_queue_delete(&metadata_hub_queue); + pc_queue_delete(&metadata_mqtt_queue); + // debug(2, "metadata_mqtt_thread_cleanup_function done"); } void *metadata_mqtt_thread_function(__attribute__((unused)) void *ignore) { @@ -1671,18 +1677,26 @@ void metadata_stop(void) { if (metadata_running) { debug(2, "metadata_stop called."); #ifdef CONFIG_MQTT - pthread_join(metadata_mqtt_thread, NULL); + // debug(2, "metadata stop mqtt thread."); pthread_cancel(metadata_mqtt_thread); + pthread_join(metadata_mqtt_thread, NULL); + // debug(2, "metadata stop mqtt done."); #endif #ifdef CONFIG_METADATA_HUB - pthread_join(metadata_hub_thread, NULL); + // debug(2, "metadata stop hub thread."); pthread_cancel(metadata_hub_thread); + pthread_join(metadata_hub_thread, NULL); + // debug(2, "metadata stop hub done."); #endif + // debug(2, "metadata stop multicast thread."); pthread_cancel(metadata_multicast_thread); pthread_join(metadata_multicast_thread, NULL); + // debug(2, "metadata stop multicast done."); + // debug(2, "metadata stop metadata_thread thread."); pthread_cancel(metadata_thread); pthread_join(metadata_thread, NULL); + // debug(2, "metadata_stop finished successfully."); } } @@ -2748,7 +2762,7 @@ static const char *format_address(struct sockaddr *fsa) { void rtsp_listen_loop_cleanup_handler(__attribute__((unused)) void *arg) { int oldState; pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState); - debug(1, "rtsp_listen_loop_cleanup_handler called."); + debug(2, "rtsp_listen_loop_cleanup_handler called."); cancel_all_RTSP_threads(); int *sockfd = (int *)arg; mdns_unregister(); From 827504a3ffc019308172566a0f5f62978c0782d6 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 21 Jun 2020 13:20:08 +0100 Subject: [PATCH 163/186] Reinstated the main thread cancellation cleanup and hooked it up the the exit function. Made the exit function work properly if libdaemon was included and either used or not used. CLeaned up some debug messages and warnings. --- shairport.c | 67 ++++++++++++++++++++++++++++++++++++++--------------- 1 file changed, 48 insertions(+), 19 deletions(-) diff --git a/shairport.c b/shairport.c index af11ade2..1553bf23 100644 --- a/shairport.c +++ b/shairport.c @@ -443,9 +443,9 @@ int parse_options(int argc, char **argv) { char *config_file_real_path = realpath(config.configfile, NULL); if (config_file_real_path == NULL) { - debug(2, "Can't resolve the configuration file \"%s\".", config.configfile); + debug(2, "can't resolve the configuration file \"%s\".", config.configfile); } else { - debug(2, "Looking for configuration file at full path \"%s\"", config_file_real_path); + debug(2, "looking for configuration file at full path \"%s\"", config_file_real_path); /* Read the file. If there is an error, report it and exit. */ if (config_read_file(&config_file_stuff, config_file_real_path)) { free(config_file_real_path); @@ -1310,9 +1310,9 @@ void exit_function() { // that most of this code will be skipped when the parent process is exiting // exec #ifdef CONFIG_LIBDAEMON - if (this_is_the_daemon_process) { // this is the daemon that is exiting + if ((this_is_the_daemon_process) || (config.daemonise == 0)) { // if this is the daemon process that is exiting or it's not actually deamonised at all #endif - debug(1, "exit function called..."); + debug(2, "exit function called..."); /* Actually, there is no terminate_mqtt() function. @@ -1382,11 +1382,13 @@ Actually, there is no stop_mpris_service() function. free(config.regtype); #ifdef CONFIG_LIBDAEMON - daemon_retval_send(0); - daemon_pid_file_remove(); - daemon_signal_done(); - if (config.computed_piddir) - free(config.computed_piddir); + if (this_is_the_daemon_process) { + daemon_retval_send(0); + daemon_pid_file_remove(); + daemon_signal_done(); + if (config.computed_piddir) + free(config.computed_piddir); + } } #endif @@ -1395,6 +1397,19 @@ Actually, there is no stop_mpris_service() function. if (config.appName) free(config.appName); // probably should be freeing malloc'ed memory here, including strdup-created strings... + +#ifdef CONFIG_LIBDAEMON + if (this_is_the_daemon_process) { // this is the daemon that is exiting + debug(2,"libdaemon daemon exit"); + } else { + if (config.daemonise) + debug(2,"libdaemon parent exit"); + else + debug(2,"exit"); + } +#else + debug(2,"exit"); +#endif } // for removing zombie script processes @@ -1408,6 +1423,12 @@ void handle_sigchld(__attribute__((unused)) int sig) { errno = saved_errno; } +void main_thread_cleanup_handler(__attribute__((unused)) void *arg) { + debug(2,"main thread cleanup handler called"); + exit(EXIT_SUCCESS); +} + + int main(int argc, char **argv) { /* Check if we are called with -V or --version parameter */ if (argc >= 2 && ((strcmp(argv[1], "-V") == 0) || (strcmp(argv[1], "--version") == 0))) { @@ -1562,10 +1583,17 @@ int main(int argc, char **argv) { /* Kill daemon with SIGTERM */ /* Check if the new function daemon_pid_file_kill_wait() is available, if it is, use it. */ - if ((ret = daemon_pid_file_kill_wait(SIGTERM, 5)) < 0) - daemon_log(LOG_WARNING, "Failed to kill daemon: %s", strerror(errno)); - else { - // debug(1,"Successfully killed the shairport sync daemon."); + if ((ret = daemon_pid_file_kill_wait(SIGTERM, 5)) < 0) { + if (errno == ENOENT) + daemon_log(LOG_WARNING, "Failed to kill %s daemon: PID file not found.", config.appName); + else + daemon_log(LOG_WARNING, "Failed to kill %s daemon: \"%s\", errno %u.", config.appName, strerror(errno), errno); + } else { + // debug(1,"Successfully killed the %s daemon.", config.appName); + if (daemon_pid_file_remove() == 0) + debug(2, "killed the %s daemon.", config.appName); + else + daemon_log(LOG_WARNING, "killed the %s deamon, but cannot remove old PID file: \"%s\", errno %u.", config.appName, strerror(errno), errno); } return ret < 0 ? 1 : 0; #else @@ -1578,7 +1606,7 @@ int main(int argc, char **argv) { #ifdef CONFIG_LIBDAEMON /* If we are going to daemonise, check that the daemon is not running already.*/ if ((config.daemonise) && ((pid = daemon_pid_file_is_running()) >= 0)) { - daemon_log(LOG_ERR, "Daemon already running on PID file %u", pid); + daemon_log(LOG_ERR, "The %s daemon is already running as PID %u", config.appName, pid); return 1; } @@ -1613,16 +1641,16 @@ int main(int argc, char **argv) { break; case 1: daemon_log(LOG_ERR, - "daemon failed to launch: could not close open file descriptors after forking."); + "the %s daemon failed to launch: could not close open file descriptors after forking.", config.appName); break; case 2: - daemon_log(LOG_ERR, "daemon failed to launch: could not create PID file."); + daemon_log(LOG_ERR, "the %s daemon failed to launch: could not create PID file.", config.appName); break; case 3: - daemon_log(LOG_ERR, "daemon failed to launch: could not create or access PID directory."); + daemon_log(LOG_ERR, "the %s daemon failed to launch: could not create or access PID directory.", config.appName); break; default: - daemon_log(LOG_ERR, "daemon failed to launch, error %i.", ret); + daemon_log(LOG_ERR, "the %s daemon failed to launch, error %i.", config.appName, ret); } return ret; } else { /* pid == 0 means we are the daemon */ @@ -1710,7 +1738,7 @@ int main(int argc, char **argv) { } config.output->init(argc - audio_arg, argv + audio_arg); - // pthread_cleanup_push(main_cleanup_handler, NULL); + pthread_cleanup_push(main_thread_cleanup_handler, NULL); // daemon_log(LOG_NOTICE, "startup"); @@ -1927,5 +1955,6 @@ int main(int argc, char **argv) { activity_monitor_start(); rtsp_listen_loop(); + pthread_cleanup_pop(1); return 0; } From 0782aae77dbefa100106bc601f805b0b76920953 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 21 Jun 2020 13:35:22 +0100 Subject: [PATCH 164/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index 3c3fee99..a4c7227c 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,8 @@ +Version 3.3.7d11 +==== +**Bug Fixes** +* Fixed a number of bugs that prevented Shairport Sync from terminating cleanly in response to the `MPRIS` interface's `Quit()` command. Thanks to [João Gabriel](https://github.com/jgabriel98) for reporting this issue. + Version 3.3.7d10 ==== **Bug Fix** From 8d3e894172feb6763a31bb38ded2e1890cb2113e Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 21 Jun 2020 13:36:49 +0100 Subject: [PATCH 165/186] Bug fixes. Fix bugs preventing clean termination on MPRIS Quit(). --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index d8bd7dc6..41b2f72f 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.7d10], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.7d11], [mikebrady@eircom.net]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 5fac4bec0bbc01d94ccf9811ef438ee7eaaa7830 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 21 Jun 2020 13:37:33 +0100 Subject: [PATCH 166/186] Change blame address! --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 41b2f72f..4ec41ce2 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.7d11], [mikebrady@eircom.net]) +AC_INIT([shairport-sync], [3.3.7d11], [mikebradydublin@icloud.com]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 98005fd18fb8e63ec40d7f7f648d857ef52395b0 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 21 Jun 2020 13:50:21 +0100 Subject: [PATCH 167/186] quieten some debug messages --- metadata_hub.c | 6 +++--- rtsp.c | 8 ++++---- 2 files changed, 7 insertions(+), 7 deletions(-) diff --git a/metadata_hub.c b/metadata_hub.c index f1a02a22..190895c0 100644 --- a/metadata_hub.c +++ b/metadata_hub.c @@ -136,12 +136,12 @@ void _metadata_hub_modify_prolog(const char *filename, const int linenumber) { // debug(1, "locking metadata hub for writing"); if (pthread_rwlock_trywrlock(&metadata_hub_re_lock) != 0) { if (last_metadata_hub_modify_prolog_file) - debug(1, "Metadata_hub write lock at \"%s:%d\" is already taken at \"%s:%d\" -- must wait.", filename, linenumber, last_metadata_hub_modify_prolog_file, last_metadata_hub_modify_prolog_line); + debug(2, "Metadata_hub write lock at \"%s:%d\" is already taken at \"%s:%d\" -- must wait.", filename, linenumber, last_metadata_hub_modify_prolog_file, last_metadata_hub_modify_prolog_line); else - debug(1, "Metadata_hub write lock is already taken by unknown -- must wait."); + debug(2, "Metadata_hub write lock is already taken by unknown -- must wait."); metadata_hub_re_lock_access_is_delayed = 0; pthread_rwlock_wrlock(&metadata_hub_re_lock); - debug(1, "Okay -- acquired the metadata_hub write lock at \"%s:%d\".", filename, linenumber); + debug(2, "Okay -- acquired the metadata_hub write lock at \"%s:%d\".", filename, linenumber); } else { if (last_metadata_hub_modify_prolog_file) { free(last_metadata_hub_modify_prolog_file); diff --git a/rtsp.c b/rtsp.c index 301a6b87..ba977bfb 100644 --- a/rtsp.c +++ b/rtsp.c @@ -368,11 +368,11 @@ static void track_thread(rtsp_conn_info *conn) { void cancel_all_RTSP_threads(void) { int i; for (i = 0; i < nconns; i++) { - debug(1, "Connection %d: cancelling.", conns[i]->connection_number); + debug(2, "Connection %d: cancelling.", conns[i]->connection_number); pthread_cancel(conns[i]->thread); } for (i = 0; i < nconns; i++) { - debug(1, "Connection %d: joining.", conns[i]->connection_number); + debug(2, "Connection %d: joining.", conns[i]->connection_number); pthread_join(conns[i]->thread, NULL); free(conns[i]); } @@ -1482,13 +1482,13 @@ void metadata_process(uint32_t type, uint32_t code, char *data, uint32_t length) } void metadata_pack_cleanup_function(void *arg) { - debug(1, "metadata_pack_cleanup_function called"); + // debug(1, "metadata_pack_cleanup_function called"); metadata_package *pack = (metadata_package *)arg; if (pack->carrier) msg_free(&pack->carrier); // release the message else if (pack->data) free(pack->data); - debug(1, "metadata_pack_cleanup_function exit"); + // debug(1, "metadata_pack_cleanup_function exit"); } void metadata_thread_cleanup_function(__attribute__((unused)) void *arg) { From 9079dea96f8985068f819548dbc288aab3f406ac Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 21 Jun 2020 15:36:09 +0100 Subject: [PATCH 168/186] slightly change debug message --- audio_dummy.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/audio_dummy.c b/audio_dummy.c index 5d64ecf2..5d16bf95 100644 --- a/audio_dummy.c +++ b/audio_dummy.c @@ -39,7 +39,7 @@ static int init(__attribute__((unused)) int argc, __attribute__((unused)) char * static void deinit(void) {} static void start(int sample_rate, __attribute__((unused)) int sample_format) { - debug(1, "dummy audio output started at Fs=%d Hz\n", sample_rate); + debug(1, "dummy audio output started at %d frames per second.", sample_rate); } static int play(__attribute__((unused)) void *buf, __attribute__((unused)) int samples) { From 7e01da543ce10f7b42bcfe259dea3d1d7b5c34ae Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Sun, 21 Jun 2020 15:36:56 +0100 Subject: [PATCH 169/186] Fix the default audio_backend_buffer_desired_length and move the exit messages to debiug level 1 --- shairport.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/shairport.c b/shairport.c index 1553bf23..808f3c6c 100644 --- a/shairport.c +++ b/shairport.c @@ -1400,15 +1400,15 @@ Actually, there is no stop_mpris_service() function. #ifdef CONFIG_LIBDAEMON if (this_is_the_daemon_process) { // this is the daemon that is exiting - debug(2,"libdaemon daemon exit"); + debug(1,"libdaemon daemon exit"); } else { if (config.daemonise) - debug(2,"libdaemon parent exit"); + debug(1,"libdaemon parent exit"); else - debug(2,"exit"); + debug(1,"exit"); } #else - debug(2,"exit"); + debug(1,"exit"); #endif } @@ -1528,7 +1528,7 @@ int main(int argc, char **argv) { // snprintf(config.service_name, 20 + 100, "Shairport Sync on %s", hostname); set_requested_connection_state_to_output( 1); // we expect to be able to connect to the output device - config.audio_backend_buffer_desired_length = 6615; // 0.15 seconds. + config.audio_backend_buffer_desired_length = 0.15; // seconds config.udp_port_base = 6001; config.udp_port_range = 10; config.output_format = SPS_FORMAT_S16_LE; // default From deee20758a5e7d68a3629ac2c2b853ac4f9ec630 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 22 Jun 2020 10:15:35 +0100 Subject: [PATCH 170/186] quieten a debug message --- rtsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/rtsp.c b/rtsp.c index ba977bfb..9a8acb11 100644 --- a/rtsp.c +++ b/rtsp.c @@ -154,7 +154,7 @@ typedef struct { void pc_queue_init(pc_queue *the_queue, char *items, size_t item_size, uint32_t number_of_items, const char* name) { if (name) - debug(1, "Creating metadata queue \"%s\".", name); + debug(2, "Creating metadata queue \"%s\".", name); else debug(1, "Creating an unnamed metadata queue."); pthread_mutex_init(&the_queue->pc_queue_lock, NULL); From 05beeaab22d00cf0a8ec4cbcd9ae4f215c5deda0 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 22 Jun 2020 10:16:37 +0100 Subject: [PATCH 171/186] Use exit() instead of abort() when a fatal error occurs, but skip the cleanup before exit, as this can cause trouble. --- common.c | 4 +- common.h | 1 + shairport.c | 150 +++++++++++++++++++++++++++------------------------- 3 files changed, 81 insertions(+), 74 deletions(-) diff --git a/common.c b/common.c index a129792a..f8fe03b7 100644 --- a/common.c +++ b/common.c @@ -90,6 +90,7 @@ void set_alsa_out_dev(char *); #endif config_t config_file_stuff; +int emergency_exit; pthread_t main_thread_id; uint64_t ns_time_at_startup, ns_time_at_last_debug_message; @@ -317,7 +318,8 @@ void _die(const char *filename, const int linenumber, const char *format, ...) { va_end(args); sps_log(LOG_ERR, "%s", b); pthread_setcancelstate(oldState, NULL); - abort(); // exit() doesn't always work, by heaven. + emergency_exit = 1; + exit(EXIT_FAILURE); } void _warn(const char *filename, const int linenumber, const char *format, ...) { diff --git a/common.h b/common.h index dce7a00a..1f5d1c87 100644 --- a/common.h +++ b/common.h @@ -380,6 +380,7 @@ extern pthread_t main_thread_id; extern shairport_cfg config; extern config_t config_file_stuff; +extern int emergency_exit; int config_set_lookup_bool(config_t *cfg, char *where, int *dst); diff --git a/shairport.c b/shairport.c index 808f3c6c..14e50725 100644 --- a/shairport.c +++ b/shairport.c @@ -1306,110 +1306,113 @@ const char *pid_file_proc(void) { void exit_function() { -// the following is to ensure that if libdaemon has been included -// that most of this code will be skipped when the parent process is exiting -// exec + if (emergency_exit == 0) { + // the following is to ensure that if libdaemon has been included + // that most of this code will be skipped when the parent process is exiting + // exec #ifdef CONFIG_LIBDAEMON - if ((this_is_the_daemon_process) || (config.daemonise == 0)) { // if this is the daemon process that is exiting or it's not actually deamonised at all + if ((this_is_the_daemon_process) || (config.daemonise == 0)) { // if this is the daemon process that is exiting or it's not actually deamonised at all #endif - debug(2, "exit function called..."); - - /* - Actually, there is no terminate_mqtt() function. - #ifdef CONFIG_MQTT - if (config.mqtt_enabled) { - terminate_mqtt(); - } - #endif - */ + debug(2, "exit function called..."); + /* + Actually, there is no terminate_mqtt() function. + #ifdef CONFIG_MQTT + if (config.mqtt_enabled) { + terminate_mqtt(); + } + #endif + */ #if defined(CONFIG_DBUS_INTERFACE) || defined(CONFIG_MPRIS_INTERFACE) - -/* -Actually, there is no stop_mpris_service() function. -#ifdef CONFIG_MPRIS_INTERFACE - stop_mpris_service(); -#endif -*/ + /* + Actually, there is no stop_mpris_service() function. + #ifdef CONFIG_MPRIS_INTERFACE + stop_mpris_service(); + #endif + */ #ifdef CONFIG_DBUS_INTERFACE - stop_dbus_service(); + stop_dbus_service(); #endif - if (g_main_loop) { - debug(2, "Stopping DBUS Loop Thread"); - g_main_loop_quit(g_main_loop); - pthread_join(dbus_thread, NULL); - } + if (g_main_loop) { + debug(2, "Stopping DBUS Loop Thread"); + g_main_loop_quit(g_main_loop); + pthread_join(dbus_thread, NULL); + } #endif #ifdef CONFIG_DACP_CLIENT - debug(2, "Stopping DACP Monitor"); - dacp_monitor_stop(); + debug(2, "Stopping DACP Monitor"); + dacp_monitor_stop(); #endif #ifdef CONFIG_METADATA_HUB - debug(2, "Stopping metadata hub"); - metadata_hub_stop(); + debug(2, "Stopping metadata hub"); + metadata_hub_stop(); #endif #ifdef CONFIG_METADATA - metadata_stop(); // close down the metadata pipe + metadata_stop(); // close down the metadata pipe #endif - activity_monitor_stop(0); + activity_monitor_stop(0); - if ((config.output) && (config.output->deinit)) { - debug(2, "Deinitialise the audio backend."); - config.output->deinit(); - } + if ((config.output) && (config.output->deinit)) { + debug(2, "Deinitialise the audio backend."); + config.output->deinit(); + } #ifdef CONFIG_SOXR - // be careful -- not sure if the thread can be cancelled cleanly, so wait for it to shut down - pthread_join(soxr_time_check_thread, NULL); + // be careful -- not sure if the thread can be cancelled cleanly, so wait for it to shut down + pthread_join(soxr_time_check_thread, NULL); #endif - if (conns) - free(conns); // make sure the connections have been deleted first + if (conns) + free(conns); // make sure the connections have been deleted first - if (config.service_name) - free(config.service_name); + if (config.service_name) + free(config.service_name); #ifdef CONFIG_CONVOLUTION - if (config.convolution_ir_file) - free(config.convolution_ir_file); + if (config.convolution_ir_file) + free(config.convolution_ir_file); #endif - if (config.regtype) - free(config.regtype); + if (config.regtype) + free(config.regtype); #ifdef CONFIG_LIBDAEMON - if (this_is_the_daemon_process) { - daemon_retval_send(0); - daemon_pid_file_remove(); - daemon_signal_done(); - if (config.computed_piddir) - free(config.computed_piddir); - } - } + if (this_is_the_daemon_process) { + daemon_retval_send(0); + daemon_pid_file_remove(); + daemon_signal_done(); + if (config.computed_piddir) + free(config.computed_piddir); + } + } #endif - if (config.cfg) - config_destroy(config.cfg); - if (config.appName) - free(config.appName); - // probably should be freeing malloc'ed memory here, including strdup-created strings... - + if (config.cfg) + config_destroy(config.cfg); + if (config.appName) + free(config.appName); + // probably should be freeing malloc'ed memory here, including strdup-created strings... + + #ifdef CONFIG_LIBDAEMON - if (this_is_the_daemon_process) { // this is the daemon that is exiting - debug(1,"libdaemon daemon exit"); - } else { - if (config.daemonise) - debug(1,"libdaemon parent exit"); - else - debug(1,"exit"); - } + if (this_is_the_daemon_process) { // this is the daemon that is exiting + debug(1,"libdaemon daemon exit"); + } else { + if (config.daemonise) + debug(1,"libdaemon parent exit"); + else + debug(1,"exit"); + } #else - debug(1,"exit"); + debug(1,"exit"); #endif + } else { + debug(1,"emergency exit"); + } } // for removing zombie script processes @@ -1466,6 +1469,7 @@ int main(int argc, char **argv) { setlogmask(LOG_UPTO(LOG_DEBUG)); openlog(NULL, 0, LOG_DAEMON); #endif + emergency_exit = 0; // what to do or skip in the exit_function atexit(exit_function); // set defaults @@ -1584,8 +1588,8 @@ int main(int argc, char **argv) { /* Kill daemon with SIGTERM */ /* Check if the new function daemon_pid_file_kill_wait() is available, if it is, use it. */ if ((ret = daemon_pid_file_kill_wait(SIGTERM, 5)) < 0) { - if (errno == ENOENT) - daemon_log(LOG_WARNING, "Failed to kill %s daemon: PID file not found.", config.appName); + if (errno == ENOENT) + daemon_log(LOG_WARNING, "Failed to kill %s daemon: PID file not found.", config.appName); else daemon_log(LOG_WARNING, "Failed to kill %s daemon: \"%s\", errno %u.", config.appName, strerror(errno), errno); } else { @@ -1593,7 +1597,7 @@ int main(int argc, char **argv) { if (daemon_pid_file_remove() == 0) debug(2, "killed the %s daemon.", config.appName); else - daemon_log(LOG_WARNING, "killed the %s deamon, but cannot remove old PID file: \"%s\", errno %u.", config.appName, strerror(errno), errno); + daemon_log(LOG_WARNING, "killed the %s deamon, but cannot remove old PID file: \"%s\", errno %u.", config.appName, strerror(errno), errno); } return ret < 0 ? 1 : 0; #else From 19ec4dfd8941cb6ea8c5f2b04cc940da1a17f9e0 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 22 Jun 2020 10:32:07 +0100 Subject: [PATCH 172/186] clean up a fatal error message --- rtsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/rtsp.c b/rtsp.c index 9a8acb11..2febacd0 100644 --- a/rtsp.c +++ b/rtsp.c @@ -2981,7 +2981,7 @@ void rtsp_listen_loop(void) { } while (1); pthread_cleanup_pop(1); // should never happen } else { - warn("could not establish a service on port %d -- program terminating. Is another instance of " + die("could not establish a service on port %d -- program terminating. Is another instance of " "Shairport Sync running on this device?", config.port); } From 8e215cb34456ee7560640b38113cc4ccc3dea7bf Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 22 Jun 2020 10:41:20 +0100 Subject: [PATCH 173/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 1 + 1 file changed, 1 insertion(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index a4c7227c..c68afca5 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -2,6 +2,7 @@ Version 3.3.7d11 ==== **Bug Fixes** * Fixed a number of bugs that prevented Shairport Sync from terminating cleanly in response to the `MPRIS` interface's `Quit()` command. Thanks to [João Gabriel](https://github.com/jgabriel98) for reporting this issue. +* Related to the above, the code used to terminate the application after a fatal error has been cleaned up. It now uses the correct `exit()` call rather than the rather hacky `abort()` call, returning the value of the constant `EXIT_FAILURE` (typically `1`) to the caller . Version 3.3.7d10 ==== From 4fad3d24b12f3f1c88bce13a6288695d7ce3e020 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Mon, 22 Jun 2020 12:01:26 +0100 Subject: [PATCH 174/186] remove the cause of a FreeBSD warning --- dacp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/dacp.c b/dacp.c index 773ba065..c6b9968f 100644 --- a/dacp.c +++ b/dacp.c @@ -556,7 +556,7 @@ void *dacp_monitor_thread_code(__attribute__((unused)) void *na) { pthread_cleanup_pop(1); if (result == 490) { // 490 means no port was specified - if ((dacp_server.dacp_id != NULL) && (strlen(dacp_server.dacp_id) != 0)) { + if (((char *)dacp_server.dacp_id != NULL) && (strlen(dacp_server.dacp_id) != 0)) { // debug(1,"mdns_dacp_monitor_set_id"); mdns_dacp_monitor_set_id(dacp_server.dacp_id); } From bcddc6722e818eab3b6b437e85d69feca867f8bc Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 24 Jun 2020 09:48:46 +0100 Subject: [PATCH 175/186] Add the string "-alac" to the version string in '-V' if the Apple ALAC decoder is included. --- common.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/common.c b/common.c index f8fe03b7..39c4d62e 100644 --- a/common.c +++ b/common.c @@ -1427,6 +1427,9 @@ char *get_version_string() { if (version_string) { strcpy(version_string, PACKAGE_VERSION); +#ifdef CONFIG_APPLE_ALAC + strcat(version_string, "-alac"); +#endif #ifdef CONFIG_LIBDAEMON strcat(version_string, "-libdaemon"); #endif From 6b613dfc78d2b2c4e00811ef00fb7894ec09b7fe Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 24 Jun 2020 16:04:18 +0100 Subject: [PATCH 176/186] Simple Docker configuration --- docker/Dockerfile | 68 ++++++++++++++++++++++++++++++++++++++ docker/shairport-sync.conf | 13 ++++++++ docker/start.sh | 13 ++++++++ 3 files changed, 94 insertions(+) create mode 100644 docker/Dockerfile create mode 100644 docker/shairport-sync.conf create mode 100755 docker/start.sh diff --git a/docker/Dockerfile b/docker/Dockerfile new file mode 100644 index 00000000..9de0a929 --- /dev/null +++ b/docker/Dockerfile @@ -0,0 +1,68 @@ +FROM alpine AS builder-base + +RUN apk -U add \ + git \ + build-base \ + autoconf \ + automake \ + libtool \ + dbus \ + su-exec \ + alsa-lib-dev \ + libdaemon-dev \ + popt-dev \ + mbedtls-dev \ + soxr-dev \ + avahi-dev \ + libconfig-dev \ + libsndfile-dev \ + mosquitto-dev \ + xmltoman + +FROM builder-base AS builder-alac + +RUN git clone --recursive https://github.com/mikebrady/alac +WORKDIR alac +RUN autoreconf -fi +RUN ./configure +RUN make +RUN make install + +FROM builder-base AS builder-sps + +COPY --from=builder-alac /usr/local/lib/libalac.* /usr/local/lib/ +COPY --from=builder-alac /usr/local/lib/pkgconfig/alac.pc /usr/local/lib/pkgconfig/alac.pc +COPY --from=builder-alac /usr/local/include /usr/local/include + +RUN git clone --recursive https://github.com/mikebrady/shairport-sync +WORKDIR shairport-sync +RUN git checkout development +RUN autoreconf -fi +RUN ./configure \ + --with-alsa \ + --with-dummy \ + --with-pipe \ + --with-stdout \ + --with-avahi \ + --with-ssl=mbedtls \ + --with-soxr \ + --sysconfdir=/etc \ + --with-dbus-interface \ + --with-mpris-interface \ + --with-mqtt-client \ + --with-apple-alac \ + --with-convolution +RUN make -j $(nproc) +RUN make install + +RUN addgroup shairport-sync +RUN adduser -D shairport-sync -G shairport-sync + +# Add the shairport-sync user to the pre-existing audio group, which has ID 29, for access to the ALSA stuff +RUN addgroup -g 29 docker_audio && addgroup shairport-sync docker_audio + +COPY shairport-sync.conf /etc/shairport-sync.conf +COPY start.sh /start.sh + +ENTRYPOINT [ "/start.sh" ] + diff --git a/docker/shairport-sync.conf b/docker/shairport-sync.conf new file mode 100644 index 00000000..55531593 --- /dev/null +++ b/docker/shairport-sync.conf @@ -0,0 +1,13 @@ +// Shairport Sync has a rich collection of configuration settings. + +// This very simple configuration file simply gives the AirPlay service +// the default name "Docker". + +// Please look at /etc/shairport-sync.conf.sample for lots more settings and information +// Or go to the sample configuration file on the home respository: +// https://github.com/mikebrady/shairport-sync/blob/master/scripts/shairport-sync.conf + +general = +{ + name = "Docker"; // This is the name the service will advertise to iTunes. +} diff --git a/docker/start.sh b/docker/start.sh new file mode 100755 index 00000000..40d408d3 --- /dev/null +++ b/docker/start.sh @@ -0,0 +1,13 @@ +#!/bin/sh + +set -e + +rm -rf /var/run/dbus.pid +#mkdir -p /var/run/dbus + +dbus-uuidgen --ensure +dbus-daemon --system + +avahi-daemon --daemonize --no-chroot + +shairport-sync $@ From 905382d065788f5470e7ca8dbf5eb075eee7dd5e Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 24 Jun 2020 16:26:23 +0100 Subject: [PATCH 177/186] Create BEWARE.md --- docker/BEWARE.md | 1 + 1 file changed, 1 insertion(+) create mode 100644 docker/BEWARE.md diff --git a/docker/BEWARE.md b/docker/BEWARE.md new file mode 100644 index 00000000..6595988d --- /dev/null +++ b/docker/BEWARE.md @@ -0,0 +1 @@ +This is very flaky. Please do not rely on this. From db3cadaf3f2f7fa0803ccae93555f4cf70c88e9c Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 24 Jun 2020 16:27:32 +0100 Subject: [PATCH 178/186] Update BEWARE.md --- docker/BEWARE.md | 3 +++ 1 file changed, 3 insertions(+) diff --git a/docker/BEWARE.md b/docker/BEWARE.md index 6595988d..6143a185 100644 --- a/docker/BEWARE.md +++ b/docker/BEWARE.md @@ -1 +1,4 @@ +Beware +====== + This is very flaky. Please do not rely on this. From 12dd4aa4e7598cb30f2e75ef9b189b4d15f88e70 Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 24 Jun 2020 16:27:52 +0100 Subject: [PATCH 179/186] Update BEWARE.md --- docker/BEWARE.md | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/docker/BEWARE.md b/docker/BEWARE.md index 6143a185..4b6ab428 100644 --- a/docker/BEWARE.md +++ b/docker/BEWARE.md @@ -1,4 +1,4 @@ Beware ====== -This is very flaky. Please do not rely on this. +This is very flaky. Please do not rely on this Docker stuff. From 4d179691b215d697af206bb5167e35ece7dc331c Mon Sep 17 00:00:00 2001 From: Mike Brady Date: Wed, 24 Jun 2020 16:36:19 +0100 Subject: [PATCH 180/186] Move Docker stuff to another repository --- docker/BEWARE.md | 4 --- docker/Dockerfile | 68 -------------------------------------- docker/shairport-sync.conf | 13 -------- docker/start.sh | 13 -------- 4 files changed, 98 deletions(-) delete mode 100644 docker/BEWARE.md delete mode 100644 docker/Dockerfile delete mode 100644 docker/shairport-sync.conf delete mode 100755 docker/start.sh diff --git a/docker/BEWARE.md b/docker/BEWARE.md deleted file mode 100644 index 4b6ab428..00000000 --- a/docker/BEWARE.md +++ /dev/null @@ -1,4 +0,0 @@ -Beware -====== - -This is very flaky. Please do not rely on this Docker stuff. diff --git a/docker/Dockerfile b/docker/Dockerfile deleted file mode 100644 index 9de0a929..00000000 --- a/docker/Dockerfile +++ /dev/null @@ -1,68 +0,0 @@ -FROM alpine AS builder-base - -RUN apk -U add \ - git \ - build-base \ - autoconf \ - automake \ - libtool \ - dbus \ - su-exec \ - alsa-lib-dev \ - libdaemon-dev \ - popt-dev \ - mbedtls-dev \ - soxr-dev \ - avahi-dev \ - libconfig-dev \ - libsndfile-dev \ - mosquitto-dev \ - xmltoman - -FROM builder-base AS builder-alac - -RUN git clone --recursive https://github.com/mikebrady/alac -WORKDIR alac -RUN autoreconf -fi -RUN ./configure -RUN make -RUN make install - -FROM builder-base AS builder-sps - -COPY --from=builder-alac /usr/local/lib/libalac.* /usr/local/lib/ -COPY --from=builder-alac /usr/local/lib/pkgconfig/alac.pc /usr/local/lib/pkgconfig/alac.pc -COPY --from=builder-alac /usr/local/include /usr/local/include - -RUN git clone --recursive https://github.com/mikebrady/shairport-sync -WORKDIR shairport-sync -RUN git checkout development -RUN autoreconf -fi -RUN ./configure \ - --with-alsa \ - --with-dummy \ - --with-pipe \ - --with-stdout \ - --with-avahi \ - --with-ssl=mbedtls \ - --with-soxr \ - --sysconfdir=/etc \ - --with-dbus-interface \ - --with-mpris-interface \ - --with-mqtt-client \ - --with-apple-alac \ - --with-convolution -RUN make -j $(nproc) -RUN make install - -RUN addgroup shairport-sync -RUN adduser -D shairport-sync -G shairport-sync - -# Add the shairport-sync user to the pre-existing audio group, which has ID 29, for access to the ALSA stuff -RUN addgroup -g 29 docker_audio && addgroup shairport-sync docker_audio - -COPY shairport-sync.conf /etc/shairport-sync.conf -COPY start.sh /start.sh - -ENTRYPOINT [ "/start.sh" ] - diff --git a/docker/shairport-sync.conf b/docker/shairport-sync.conf deleted file mode 100644 index 55531593..00000000 --- a/docker/shairport-sync.conf +++ /dev/null @@ -1,13 +0,0 @@ -// Shairport Sync has a rich collection of configuration settings. - -// This very simple configuration file simply gives the AirPlay service -// the default name "Docker". - -// Please look at /etc/shairport-sync.conf.sample for lots more settings and information -// Or go to the sample configuration file on the home respository: -// https://github.com/mikebrady/shairport-sync/blob/master/scripts/shairport-sync.conf - -general = -{ - name = "Docker"; // This is the name the service will advertise to iTunes. -} diff --git a/docker/start.sh b/docker/start.sh deleted file mode 100755 index 40d408d3..00000000 --- a/docker/start.sh +++ /dev/null @@ -1,13 +0,0 @@ -#!/bin/sh - -set -e - -rm -rf /var/run/dbus.pid -#mkdir -p /var/run/dbus - -dbus-uuidgen --ensure -dbus-daemon --system - -avahi-daemon --daemonize --no-chroot - -shairport-sync $@ From 88a77916ca5f84394a335669896f1426aaf24455 Mon Sep 17 00:00:00 2001 From: Mike Brady <4265913+mikebrady@users.noreply.github.com> Date: Fri, 3 Jul 2020 17:06:54 +0100 Subject: [PATCH 181/186] Update RELEASENOTES-DEVELOPMENT.md --- RELEASENOTES-DEVELOPMENT.md | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/RELEASENOTES-DEVELOPMENT.md b/RELEASENOTES-DEVELOPMENT.md index c68afca5..4814b7db 100644 --- a/RELEASENOTES-DEVELOPMENT.md +++ b/RELEASENOTES-DEVELOPMENT.md @@ -1,3 +1,8 @@ +Version 3.3.7d12 +==== +**Enhancements** +* Added the the string `-alac` to the the version string in '-V' if Apple ALAC decoder support is included. + Version 3.3.7d11 ==== **Bug Fixes** From ec5ff0fcfbc374bda0361dd793d4d5caba14f5d6 Mon Sep 17 00:00:00 2001 From: Mike Brady <4265913+mikebrady@users.noreply.github.com> Date: Fri, 3 Jul 2020 17:32:44 +0100 Subject: [PATCH 182/186] 3.3.7d12 -- getting near to release --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 4ec41ce2..553085a0 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.7d11], [mikebradydublin@icloud.com]) +AC_INIT([shairport-sync], [3.3.7d12], [4265913+mikebrady@users.noreply.github.com]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From 282871eaba2b61dc56d6451509d78e9fd9351abd Mon Sep 17 00:00:00 2001 From: Mike Brady <4265913+mikebrady@users.noreply.github.com> Date: Wed, 8 Jul 2020 10:20:15 +0100 Subject: [PATCH 183/186] Remove leading zeroes from an incoming DACP ID in another place --- mdns_avahi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/mdns_avahi.c b/mdns_avahi.c index f3c1f1c9..fcbbe484 100644 --- a/mdns_avahi.c +++ b/mdns_avahi.c @@ -152,6 +152,8 @@ static void browse_callback(AvahiServiceBrowser *b, AvahiIfIndex interface, Avah char *dacpid = strstr(name, "iTunes_Ctrl_"); if (dacpid) { dacpid += strlen("iTunes_Ctrl_"); + while (*dacpid == '0') + dacpid++; // remove any leading zeroes if ((dbs->dacp_id) && (strcmp(dacpid, dbs->dacp_id) == 0)) dacp_monitor_port_update_callback(dbs->dacp_id, 0); // say the port is withdrawn } else { From 1028f0c734820c71f9508da61efd0617485d8171 Mon Sep 17 00:00:00 2001 From: Mike Brady <4265913+mikebrady@users.noreply.github.com> Date: Wed, 8 Jul 2020 10:20:33 +0100 Subject: [PATCH 184/186] Remove leading zeroes from an incoming DACP ID in another place --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 553085a0..182d3303 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ # Process this file with autoconf to produce a configure script. AC_PREREQ([2.50]) -AC_INIT([shairport-sync], [3.3.7d12], [4265913+mikebrady@users.noreply.github.com]) +AC_INIT([shairport-sync], [3.3.7d13], [4265913+mikebrady@users.noreply.github.com]) AM_INIT_AUTOMAKE AC_CONFIG_SRCDIR([shairport.c]) AC_CONFIG_HEADERS([config.h]) From df7d6df0adb1ece2da5f4018942bafa62bc00854 Mon Sep 17 00:00:00 2001 From: Mike Brady <4265913+mikebrady@users.noreply.github.com> Date: Wed, 8 Jul 2020 21:21:42 +0100 Subject: [PATCH 185/186] Change a slightly misleading comment --- mdns_avahi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/mdns_avahi.c b/mdns_avahi.c index fcbbe484..47ff8a1d 100644 --- a/mdns_avahi.c +++ b/mdns_avahi.c @@ -97,7 +97,7 @@ static void resolve_callback(AvahiServiceResolver *r, AVAHI_GCC_UNUSED AvahiIfIn if (dacpid) { dacpid += strlen("iTunes_Ctrl_"); while (*dacpid == '0') - dacpid++; // remove any leading zeroes + dacpid++; // skip any leading zeroes if (strcmp(dacpid, dbs->dacp_id) == 0) { debug(3, "resolve_callback: client dacp_id \"%s\" dacp port: %u.", dbs->dacp_id, port); #ifdef CONFIG_DACP_CLIENT @@ -153,7 +153,7 @@ static void browse_callback(AvahiServiceBrowser *b, AvahiIfIndex interface, Avah if (dacpid) { dacpid += strlen("iTunes_Ctrl_"); while (*dacpid == '0') - dacpid++; // remove any leading zeroes + dacpid++; // skip any leading zeroes if ((dbs->dacp_id) && (strcmp(dacpid, dbs->dacp_id) == 0)) dacp_monitor_port_update_callback(dbs->dacp_id, 0); // say the port is withdrawn } else { From 0dbc37ea9421972c03e8e7c6410244b491b6c922 Mon Sep 17 00:00:00 2001 From: Mike Brady <4265913+mikebrady@users.noreply.github.com> Date: Wed, 8 Jul 2020 21:24:44 +0100 Subject: [PATCH 186/186] Quieten a noisy flush debug message --- player.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/player.c b/player.c index 0ce0df85..dcef6c4e 100644 --- a/player.c +++ b/player.c @@ -995,7 +995,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) { } } } else { - debug(2, "flush request: flush frame %u -- buffer not synced or empty: synced: %d, ab_read: %u, ab_write: %u", conn->flush_rtp_timestamp, conn->ab_synced, conn->ab_read, conn->ab_write); + debug(3, "flush request: flush frame %u -- buffer not synced or empty: synced: %d, ab_read: %u, ab_write: %u", conn->flush_rtp_timestamp, conn->ab_synced, conn->ab_read, conn->ab_write); // leave flush request pending and don't do a buffer flush, because there isn't one } }