nils/shairport-sync
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Clean up some of the wording and layout of the configuration file

Browse Source 
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.
This commit is contained in:
Mike Brady
2020-05-18 16:55:09 +01:00
parent 012dd26c77
commit 54d761ff7d
23 changed files with 907 additions and 903 deletions
Download Patch File Download Diff File Expand all files Collapse all files
alac.h
-1
View File
@@ -27,7 +27,6 @@
*
*/
#ifndef __ALAC__DECOMP_H
#define __ALAC__DECOMP_H
audio.c
+12 -10
View File
@@ -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;
audio_alsa.c
+134 -126
View File
@@ -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 = &parameters; // 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 = &parameters; // 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 = &parameters; // 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 = &parameters; // 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 = &parameters; // 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 = &parameters; // 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);
}
audio_jack.c
+17 -36
View File
@@ -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
audio_sndio.c
+3 -3
View File
@@ -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);
}
audio_soundio.c
+3 -2
View File
@@ -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) {
common.c
+6 -4
View File
@@ -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;
common.h
+11 -10
View File
@@ -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
dacp.c
+17 -16
View File
@@ -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) {
dbus-service.c
+34 -31
View File
@@ -24,10 +24,10 @@
* OTHER DEALINGS IN THE SOFTWARE.
*/
#include <inttypes.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <inttypes.h>
#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);
mdns_avahi.c
+1 -1
View File
@@ -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
mdns_dns_sd.c
+1 -1
View File
@@ -24,8 +24,8 @@
* OTHER DEALINGS IN THE SOFTWARE.
*/
#include "mdns.h"
#include "common.h"
#include "mdns.h"
#include <arpa/inet.h>
#include <dns_sd.h>
#include <stdlib.h>
mdns_external.c
+1 -1
View File
@@ -25,8 +25,8 @@
* OTHER DEALINGS IN THE SOFTWARE.
*/
#include "mdns.h"
#include "common.h"
#include "mdns.h"
#include <errno.h>
#include <fcntl.h>
#include <stdio.h>
metadata_hub.c
+45 -44
View File
@@ -35,10 +35,10 @@
#include <dirent.h>
#include <errno.h>
#include <fcntl.h>
#include <inttypes.h>
#include <sys/stat.h>
#include <sys/types.h>
#include <unistd.h>
#include <inttypes.h>
#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;
//}
mpris-service.c
+9 -8
View File
@@ -23,9 +23,9 @@
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
* OTHER DEALINGS IN THE SOFTWARE.
*/
#include <inttypes.h>
#include <stdio.h>
#include <string.h>
#include <inttypes.h>
#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,
player.c
+388 -378
View File
File diff suppressed because it is too large Load Diff
player.h
+5 -4
View File
@@ -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
rtp.c
+26 -21
View File
@@ -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) {
rtsp.c
+42 -33
View File
@@ -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");
}
scripts/shairport-sync.conf
+28 -17
View File
@@ -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 = <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"
// 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.
shairport-sync-dbus-test-client.c
+16 -10
View File
@@ -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);
shairport.c
+104 -143
View File
@@ -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);
tinysvcmdns.c
+4 -3
View File
@@ -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;