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

Format tidy-up

Browse Source
This commit is contained in:
Mike Brady
2018-10-21 17:10:23 +01:00
parent e5259e8764
commit e0a4e5cde8
7 changed files with 160 additions and 130 deletions
Download Patch File Download Diff File Expand all files Collapse all files
audio_alsa.c
+3 -3
View File
@@ -894,15 +894,15 @@ int delay(long *the_delay) {
}
}
} else {
reply = -EIO; // shomething is wrong
reply = -EIO; // shomething is wrong
frame_index = 0; // we'll be starting over...
measurement_data_is_valid = 0;
if (dac_state == SND_PCM_STATE_PREPARED) {
debug(1,"delay not available -- state is SND_PCM_STATE_PREPARED");
debug(1, "delay not available -- state is SND_PCM_STATE_PREPARED");
} else {
if (dac_state == SND_PCM_STATE_XRUN) {
debug(1,"delay not available -- state is SND_PCM_STATE_XRUN");
debug(1, "delay not available -- state is SND_PCM_STATE_XRUN");
} else {
debug(1, "Error -- ALSA delay(): bad state: %d.", dac_state);
audio_pa.c
+5 -3
View File
@@ -187,12 +187,14 @@ static void start(__attribute__((unused)) int sample_rate,
if (config.pa_sink) {
// Connect stream to the sink specified in the config
connect_result = pa_stream_connect_playback(stream, config.pa_sink, &buffer_attr, stream_flags, NULL, NULL);
connect_result =
pa_stream_connect_playback(stream, config.pa_sink, &buffer_attr, stream_flags, NULL, NULL);
} else {
// Connect stream to the default audio output sink
connect_result = pa_stream_connect_playback(stream, NULL, &buffer_attr, stream_flags, NULL, NULL);
connect_result =
pa_stream_connect_playback(stream, NULL, &buffer_attr, stream_flags, NULL, NULL);
}
if (connect_result != 0)
die("could not connect to the pulseaudio playback stream -- the error message is \"%s\".",
pa_strerror(pa_context_errno(context)));
common.h
+3 -3
View File
@@ -84,9 +84,9 @@ typedef struct {
#ifdef CONFIG_PA
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.
// Defaults to "Shairport Sync". Shairport Sync must be playing to see it.
char *pa_sink; // the name (or id) of the sink that Shairport Sync will play on.
char *pa_sink; // the name (or id) of the sink that Shairport Sync will play on.
#endif
#ifdef CONFIG_METADATA
int metadata_enabled;
@@ -136,7 +136,7 @@ typedef struct {
int buffer_start_fill;
uint32_t userSuppliedLatency; // overrides all other latencies -- use with caution
uint32_t fixedLatencyOffset; // add this to all automatic latencies supplied to get the actual
// total latency
// total latency
// the total latency will be limited to the min and max-latency values, if supplied
int daemonise;
int daemonise_store_pid; // don't try to save a PID file
player.c
+111 -89
View File
@@ -100,10 +100,10 @@
#define BUFIDX(seqno) ((seq_t)(seqno) % BUFFER_FRAMES)
uint32_t rtp_frame_offset(uint32_t from, uint32_t to) {
if (from <= to)
return to - from;
else
return UINT32_MAX - from + to + 1;
if (from <= to)
return to - from;
else
return UINT32_MAX - from + to + 1;
}
void do_flush(uint32_t timestamp, rtsp_conn_info *conn);
@@ -120,33 +120,34 @@ static void ab_resync(rtsp_conn_info *conn) {
conn->ab_buffering = 1;
}
// given starting and ending points as unsigned 32-bit integers running modulo 2^32, returns the position of x in the interval in *pos
// given starting and ending points as unsigned 32-bit integers running modulo 2^32, returns the
// position of x in the interval in *pos
// returns true if x is actually within the buffer
int position_in_modulo_uint32_t_buffer(uint32_t x, uint32_t start, uint32_t end, uint32_t *pos ) {
int response = 0; // not in the buffer
if (start <= end ) {
if (x < start) {
if (pos)
*pos = UINT32_MAX - start + 1 + x;
} else {
if (pos)
*pos = x - start;
if (x < end)
response = 1;
}
} else if ((x >= start) && (x <= UINT32_MAX)) {
response = 1;
if (pos)
*pos = x - start;
} else {
if (pos)
*pos = UINT32_MAX - start + 1 + x;
if (x < end) {
response = 1;
}
}
return response;
int position_in_modulo_uint32_t_buffer(uint32_t x, uint32_t start, uint32_t end, uint32_t *pos) {
int response = 0; // not in the buffer
if (start <= end) {
if (x < start) {
if (pos)
*pos = UINT32_MAX - start + 1 + x;
} else {
if (pos)
*pos = x - start;
if (x < end)
response = 1;
}
} else if ((x >= start) && (x <= UINT32_MAX)) {
response = 1;
if (pos)
*pos = x - start;
} else {
if (pos)
*pos = UINT32_MAX - start + 1 + x;
if (x < end) {
response = 1;
}
}
return response;
}
// this is used.
@@ -441,8 +442,8 @@ static void free_audio_buffers(rtsp_conn_info *conn) {
free(conn->audio_buffer[i].data);
}
void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data,
int len, rtsp_conn_info *conn) {
void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, int len,
rtsp_conn_info *conn) {
// ignore a request to flush that has been made before the first packet...
if (conn->packet_count == 0) {
@@ -460,26 +461,31 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data,
// if (flush_rtp_timestamp != 0)
// debug(1,"Flush_rtp_timestamp is %u",flush_rtp_timestamp);
// now, if a flush_rtp_timestamp has been defined and the incoming timestamp is "before" it, drop it…
if ((conn->flush_rtp_timestamp != 0) && (rtp_frame_offset(actual_timestamp,conn->flush_rtp_timestamp) < 8820)) {// if it's less than 0.2 seconds
debug(2, "Dropping flushed packet in player_put_packet, seqno %u, timestamp %" PRIu32
", flushing to "
"timestamp: %" PRIu32 ".",
seqno, actual_timestamp, conn->flush_rtp_timestamp);
conn->initial_reference_time = 0;
conn->initial_reference_timestamp = 0;
// now, if a flush_rtp_timestamp has been defined and the incoming timestamp is "before" it,
// drop it…
if ((conn->flush_rtp_timestamp != 0) &&
(rtp_frame_offset(actual_timestamp, conn->flush_rtp_timestamp) <
8820)) { // if it's less than 0.2 seconds
debug(2, "Dropping flushed packet in player_put_packet, seqno %u, timestamp %" PRIu32
", flushing to "
"timestamp: %" PRIu32 ".",
seqno, actual_timestamp, conn->flush_rtp_timestamp);
conn->initial_reference_time = 0;
conn->initial_reference_timestamp = 0;
} else {
if ((conn->flush_rtp_timestamp != 0) && (rtp_frame_offset(conn->flush_rtp_timestamp,actual_timestamp) > 8820) && (rtp_frame_offset(conn->flush_rtp_timestamp,actual_timestamp) < 44100)) {
debug(2,"Dropping flush request");
if ((conn->flush_rtp_timestamp != 0) &&
(rtp_frame_offset(conn->flush_rtp_timestamp, actual_timestamp) > 8820) &&
(rtp_frame_offset(conn->flush_rtp_timestamp, actual_timestamp) < 44100)) {
debug(2, "Dropping flush request");
conn->flush_rtp_timestamp = 0;
}
abuf_t *abuf = 0;
if (!conn->ab_synced) {
// if this is the first packet…
// if this is the first packet…
debug(3, "syncing to seqno %u.", seqno);
conn->ab_write = seqno;
conn->ab_read = seqno;
@@ -500,7 +506,8 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data,
conn->resend_interval = resend_interval;
}
if (conn->ab_write == seqno) { // if this is the expected packet (which could be the first packet…)
if (conn->ab_write ==
seqno) { // if this is the expected packet (which could be the first packet…)
uint64_t reception_time = get_absolute_time_in_fp();
if (conn->input_frame_rate_starting_point_is_valid == 0) {
if ((conn->packet_count_since_flush >= 500) && (conn->packet_count_since_flush <= 510)) {
@@ -602,7 +609,7 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data,
j)) { // prevent multiple requests from the same level of lookback
check_buf->resend_level = j;
if (config.disable_resend_requests == 0) {
debug_mutex_unlock(&conn->ab_mutex, 3);
debug_mutex_unlock(&conn->ab_mutex, 3);
rtp_request_resend(next, 1, conn);
conn->resend_requests++;
debug_mutex_lock(&conn->ab_mutex, 20000, 1);
@@ -853,12 +860,14 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
debug(1, "Inconsistent sequence numbers detected");
}
}
if ((conn->flush_rtp_timestamp != 0) && (rtp_frame_offset(curframe->given_timestamp,conn->flush_rtp_timestamp) < 8820)) {// if it's less than 0.2 seconds
debug(2, "Dropping flushed packet in buffer_get_frame, seqno %u, timestamp %" PRIu32
", flushing to "
"timestamp: %" PRIu32 ".",
curframe->sequence_number, curframe->given_timestamp, conn->flush_rtp_timestamp);
if ((conn->flush_rtp_timestamp != 0) &&
(rtp_frame_offset(curframe->given_timestamp, conn->flush_rtp_timestamp) <
8820)) { // if it's less than 0.2 seconds
debug(2, "Dropping flushed packet in buffer_get_frame, seqno %u, timestamp %" PRIu32
", flushing to "
"timestamp: %" PRIu32 ".",
curframe->sequence_number, curframe->given_timestamp, conn->flush_rtp_timestamp);
curframe->ready = 0;
curframe->resend_level = 0;
flush_limit += curframe->length;
@@ -866,10 +875,12 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
conn->initial_reference_time = 0;
conn->initial_reference_timestamp = 0;
}
if ((conn->flush_rtp_timestamp != 0) && (rtp_frame_offset(conn->flush_rtp_timestamp,curframe->given_timestamp) > 8820) && (rtp_frame_offset(conn->flush_rtp_timestamp,curframe->given_timestamp) < 44100)) {
debug(2,"Dropping flush request");
conn->flush_rtp_timestamp = 0;
}
if ((conn->flush_rtp_timestamp != 0) &&
(rtp_frame_offset(conn->flush_rtp_timestamp, curframe->given_timestamp) > 8820) &&
(rtp_frame_offset(conn->flush_rtp_timestamp, curframe->given_timestamp) < 44100)) {
debug(2, "Dropping flush request");
conn->flush_rtp_timestamp = 0;
}
}
} while ((conn->flush_rtp_timestamp != 0) && (flush_limit <= 8820) && (curframe->ready == 0));
@@ -901,7 +912,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
// debug(1,"First frame seen with timestamp...");
conn->first_packet_timestamp =
curframe->given_timestamp; // we will keep buffering until we are
// supposed to start playing this
// supposed to start playing this
have_sent_prefiller_silence = 0;
conn->packet_stream_established = 1;
@@ -943,10 +954,11 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
// frame should be played"
uint64_t should_be_time;
frame_to_local_time(
conn->first_packet_timestamp + conn->latency +
(uint32_t)(config.audio_backend_latency_offset * conn->input_rate), // this will go modulo 2^32
&should_be_time, conn);
frame_to_local_time(conn->first_packet_timestamp + conn->latency +
(uint32_t)(config.audio_backend_latency_offset *
conn->input_rate), // this will go modulo 2^32
&should_be_time,
conn);
conn->first_packet_time_to_play = should_be_time;
@@ -964,10 +976,11 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
// recalculate conn->first_packet_time_to_play -- the latency might change
uint64_t should_be_time;
frame_to_local_time(
conn->first_packet_timestamp + conn->latency +
(uint32_t)(config.audio_backend_latency_offset * conn->input_rate), // this should go modulo 2^32
&should_be_time, conn);
frame_to_local_time(conn->first_packet_timestamp + conn->latency +
(uint32_t)(config.audio_backend_latency_offset *
conn->input_rate), // this should go modulo 2^32
&should_be_time,
conn);
conn->first_packet_time_to_play = should_be_time;
@@ -979,9 +992,9 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
int64_t max_dac_delay = config.output_rate / 10; // so the lead-in time must be greater
// than this, say 0.2 sec, to allow for
// dynamic adjustment
// dynamic adjustment
int64_t filler_size = max_dac_delay;
if (local_time_now >= conn->first_packet_time_to_play) {
// debug(1,"Gone past starting time");
have_sent_prefiller_silence = 1;
@@ -1006,34 +1019,38 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
int64_t lead_time = conn->first_packet_time_to_play - local_time_now;
// an audio_backend_silent_lead_in_time of less than zero means start filling ASAP
int64_t lead_in_time = -1;
if (config.audio_backend_silent_lead_in_time>=0)
if (config.audio_backend_silent_lead_in_time >= 0)
lead_in_time =
(int64_t)(config.audio_backend_silent_lead_in_time * (int64_t)0x100000000);
(int64_t)(config.audio_backend_silent_lead_in_time * (int64_t)0x100000000);
// debug(1,"Lead time is %llx at fpttp
// %llx.",lead_time,conn->first_packet_time_to_play);
if ((lead_in_time < 0) || (lead_time <= lead_in_time)) {
// debug(1,"Lead time is %" PRIx64 ", lead-in time is %" PRIx64 " at fpttp %llx.",lead_time,conn->first_packet_time_to_play);
// debug(1,"Lead time is %" PRIx64 ", lead-in time is %" PRIx64 " at fpttp
// %llx.",lead_time,conn->first_packet_time_to_play);
// debug(1,"Checking");
if (config.output->delay) {
// conn->first_packet_time_to_play is definitely later than local_time_now
int resp = 0;
dac_delay = 0;
if (have_sent_prefiller_silence != 0)
if (have_sent_prefiller_silence != 0)
resp = config.output->delay(&dac_delay);
if (resp == 0) {
int64_t gross_frame_gap =
((conn->first_packet_time_to_play - local_time_now) * config.output_rate) >>
32;
int64_t exact_frame_gap = gross_frame_gap - dac_delay;
// debug(1,"Exact and gross frame gaps are %" PRId64 " and %" PRId64 " frames, and the dac delay is %ld.", exact_frame_gap, gross_frame_gap, dac_delay);
// debug(1,"Exact and gross frame gaps are %" PRId64 " and %" PRId64 " frames,
// and the dac delay is %ld.", exact_frame_gap, gross_frame_gap, dac_delay);
if (exact_frame_gap < 0) {
// we've gone past the time...
// debug(1,"Run past time.");
// this might happen if a big clock adjustment was made at just the wrong time.
debug(1,"Run a bit past the exact start time by %" PRId64 " frames.",-exact_frame_gap);
// this might happen if a big clock adjustment was made at just the wrong
// time.
debug(1, "Run a bit past the exact start time by %" PRId64 " frames.",
-exact_frame_gap);
if (config.output->flush)
config.output->flush();
ab_resync(conn);
@@ -1044,7 +1061,9 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
if (fs > (max_dac_delay - dac_delay))
fs = max_dac_delay - dac_delay;
if (fs < 0) {
// this could happen if the dac delay mysteriously grows between samples, which could happen in a transition between having no interpolation and having interpolated buffer numbers.
// this could happen if the dac delay mysteriously grows between samples,
// which could happen in a transition between having no interpolation and
// having interpolated buffer numbers.
debug(2,
"frame size (fs) < 0 with max_dac_delay of %lld and dac_delay of %ld",
max_dac_delay, dac_delay);
@@ -1062,7 +1081,8 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
}
void *silence;
// if (fs==0)
// debug(2,"Zero length silence buffer needed with gross_frame_gap of %lld and
// debug(2,"Zero length silence buffer needed with gross_frame_gap of %lld
// and
// dac_delay of %lld.",gross_frame_gap,dac_delay);
// the fs (number of frames of silence to play) can be zero in the DAC doesn't
// start
@@ -1162,11 +1182,12 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
if (have_timestamp_timing_information(conn)) { // if we have a reference time
uint64_t time_to_play;
frame_to_local_time(
curframe->given_timestamp + conn->latency +
(uint32_t)(config.audio_backend_latency_offset * conn->input_rate) -
(uint32_t)(config.audio_backend_buffer_desired_length * conn->input_rate), // this will go modulo 2^32
&time_to_play, conn);
frame_to_local_time(curframe->given_timestamp + conn->latency +
(uint32_t)(config.audio_backend_latency_offset * conn->input_rate) -
(uint32_t)(config.audio_backend_buffer_desired_length *
conn->input_rate), // this will go modulo 2^32
&time_to_play,
conn);
if (local_time_now >= time_to_play) {
do_wait = 0;
@@ -1909,8 +1930,8 @@ void *player_thread_func(void *arg) {
get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time,
&remote_reference_timestamp_time, conn); // types okay
int64_t rt, nt;
rt = reference_timestamp; // uint32_t to int64_t
nt = inframe->given_timestamp; // uint32_t to int64_t
rt = reference_timestamp; // uint32_t to int64_t
nt = inframe->given_timestamp; // uint32_t to int64_t
rt *= conn->output_sample_ratio;
nt *= conn->output_sample_ratio;
@@ -2041,11 +2062,12 @@ void *player_thread_func(void *arg) {
(config.resyncthreshold > 0.0) &&
(abs_sync_error > config.resyncthreshold * config.output_rate)) {
if (abs_sync_error > 3 * config.output_rate) {
warn("Very large sync error: %" PRId64 " frames, with should_be_frame: %" PRId64
", nt: %" PRId64
", current_delay: %" PRId64 ", given timestamp %" PRIX32 ", reference timestamp %" PRIX32 ", should_be_frame %" PRIX32 ".",
sync_error, should_be_frame, nt, current_delay, inframe->given_timestamp, reference_timestamp, should_be_frame_32);
", nt: %" PRId64 ", current_delay: %" PRId64 ", given timestamp %" PRIX32
", reference timestamp %" PRIX32 ", should_be_frame %" PRIX32 ".",
sync_error, should_be_frame, nt, current_delay, inframe->given_timestamp,
reference_timestamp, should_be_frame_32);
}
sync_error_out_of_bounds++;
} else {
player.h
+7 -7
View File
@@ -68,14 +68,14 @@ typedef struct {
} stream_cfg;
typedef struct {
int connection_number; // for debug ID purposes, nothing else...
int resend_interval; // this is really just for debugging
int AirPlayVersion; // zero if not an AirPlay session. Used to help calculate latency
int connection_number; // for debug ID purposes, nothing else...
int resend_interval; // this is really just for debugging
int AirPlayVersion; // zero if not an AirPlay session. Used to help calculate latency
uint32_t latency; // the actual latency used for this play session
uint32_t minimum_latency; // set if an a=min-latency: line appears in the ANNOUNCE message; zero
// otherwise
// otherwise
uint32_t maximum_latency; // set if an a=max-latency: line appears in the ANNOUNCE message; zero
// otherwise
// otherwise
int fd;
int authorized; // set if a password is required and has been supplied
@@ -250,8 +250,8 @@ int player_stop(rtsp_conn_info *conn);
void player_volume(double f, rtsp_conn_info *conn);
void player_volume_without_notification(double f, rtsp_conn_info *conn);
void player_flush(uint32_t timestamp, rtsp_conn_info *conn);
void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data,
int len, rtsp_conn_info *conn);
void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, int len,
rtsp_conn_info *conn);
int64_t monotonic_timestamp(uint32_t timestamp,
rtsp_conn_info *conn); // add an epoch to the timestamp. The monotonic
// timestamp guaranteed to start between 2^32 2^33
rtp.c
+27 -20
View File
@@ -329,10 +329,14 @@ void *rtp_control_receiver(void *arg) {
// Sigh, it would be nice to have a published protocol...
uint16_t flags = nctohs(&packet[2]);
uint32_t la = sync_rtp_timestamp - rtp_timestamp_less_latency; // note, this might loop around in modulo. Not sure if you'll get an error!
uint32_t la = sync_rtp_timestamp - rtp_timestamp_less_latency; // note, this might
// loop around in
// modulo. Not sure if
// you'll get an error!
// debug(3, "Latency derived just from the sync packet is %" PRIu32 " frames.", la);
if ((flags == 7) || ((conn->AirPlayVersion > 0) && (conn->AirPlayVersion <= 353)) || ((conn->AirPlayVersion > 0) && (conn->AirPlayVersion >= 371))) {
if ((flags == 7) || ((conn->AirPlayVersion > 0) && (conn->AirPlayVersion <= 353)) ||
((conn->AirPlayVersion > 0) && (conn->AirPlayVersion >= 371))) {
la += config.fixedLatencyOffset;
// debug(3, "A fixed latency offset of %d frames has been added, giving a latency of
// "
@@ -350,7 +354,8 @@ void *rtp_control_receiver(void *arg) {
if (la > max_frames) {
warn("An out-of-range latency request of %" PRIu32
" frames was ignored. Must be %" PRIu32 " frames or less (44,100 frames per second). "
" frames was ignored. Must be %" PRIu32
" frames or less (44,100 frames per second). "
"Latency remains at %" PRIu32 " frames.",
la, max_frames, conn->latency);
} else {
@@ -379,7 +384,8 @@ void *rtp_control_receiver(void *arg) {
if (remote_frame_time_interval) {
conn->remote_frame_rate =
(1.0 * (conn->reference_timestamp - conn->initial_reference_timestamp)) /
remote_frame_time_interval; // an IEEE double calculation with a 32-bit numerator and 64-bit denominator
remote_frame_time_interval; // an IEEE double calculation with a 32-bit
// numerator and 64-bit denominator
// integers
conn->remote_frame_rate = conn->remote_frame_rate *
(uint64_t)0x100000000; // this should just change the
@@ -1040,8 +1046,9 @@ int sanitised_source_rate_information(uint32_t *frames, uint64_t *time, rtsp_con
*time = (uint64_t)(0x100000000); // one second in fp form
if ((conn->packet_stream_established) && (conn->initial_reference_time) &&
(conn->initial_reference_timestamp)) {
// uint32_t local_frames = conn->reference_timestamp - conn->initial_reference_timestamp;
uint32_t local_frames = rtp_frame_offset(conn->initial_reference_timestamp, conn->reference_timestamp);
// uint32_t local_frames = conn->reference_timestamp - conn->initial_reference_timestamp;
uint32_t local_frames =
rtp_frame_offset(conn->initial_reference_timestamp, conn->reference_timestamp);
uint64_t local_time = conn->remote_reference_timestamp_time - conn->initial_reference_time;
if ((local_frames == 0) || (local_time == 0) || (use_nominal_rate)) {
result = 1;
@@ -1073,24 +1080,24 @@ int frame_to_local_time(uint32_t timestamp, uint64_t *time, rtsp_conn_info *conn
uint64_t timestamp_interval_time;
uint64_t remote_time_of_timestamp;
uint32_t timestamp_interval = rtp_frame_offset(conn->reference_timestamp,timestamp);
if (timestamp_interval <= conn->input_rate*10) { // i.e. timestamp was really after the reference timestamp
timestamp_interval_time =
(timestamp_interval * time_difference) /
(frame_difference); // this is the nominal time, based on the
// fps specified between current and
// previous sync frame.
uint32_t timestamp_interval = rtp_frame_offset(conn->reference_timestamp, timestamp);
if (timestamp_interval <=
conn->input_rate * 10) { // i.e. timestamp was really after the reference timestamp
timestamp_interval_time = (timestamp_interval * time_difference) /
(frame_difference); // this is the nominal time, based on the
// fps specified between current and
// previous sync frame.
remote_time_of_timestamp = conn->remote_reference_timestamp_time +
timestamp_interval_time; // based on the reference timestamp time
// plus the time interval calculated based
// on the specified fps.
} else { // i.e. timestamp was actually after the reference timestamp
timestamp_interval = rtp_frame_offset(timestamp,conn->reference_timestamp); // fix the calculation
timestamp_interval_time =
(timestamp_interval * time_difference) /
(frame_difference); // this is the nominal time, based on the
// fps specified between current and
// previous sync frame.
timestamp_interval =
rtp_frame_offset(timestamp, conn->reference_timestamp); // fix the calculation
timestamp_interval_time = (timestamp_interval * time_difference) /
(frame_difference); // this is the nominal time, based on the
// fps specified between current and
// previous sync frame.
remote_time_of_timestamp = conn->remote_reference_timestamp_time -
timestamp_interval_time; // based on the reference timestamp time
// plus the time interval calculated based
rtsp.c
+4 -5
View File
@@ -507,8 +507,7 @@ fail:
return 0;
}
enum rtsp_read_request_response rtsp_read_request(rtsp_conn_info *conn,
rtsp_message **the_packet) {
enum rtsp_read_request_response rtsp_read_request(rtsp_conn_info *conn, rtsp_message **the_packet) {
enum rtsp_read_request_response reply = rtsp_read_request_response_ok;
ssize_t buflen = 4096;
char *buf = malloc(buflen + 1);
@@ -738,7 +737,7 @@ void handle_record(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp)
}
void handle_options(rtsp_conn_info *conn, __attribute__((unused)) rtsp_message *req,
rtsp_message *resp) {
rtsp_message *resp) {
debug(3, "Connection %d: OPTIONS", conn->connection_number);
resp->respcode = 200;
msg_add_header(resp, "Public", "ANNOUNCE, SETUP, RECORD, "
@@ -747,7 +746,7 @@ void handle_options(rtsp_conn_info *conn, __attribute__((unused)) rtsp_message *
}
void handle_teardown(rtsp_conn_info *conn, __attribute__((unused)) rtsp_message *req,
rtsp_message *resp) {
rtsp_message *resp) {
debug(2, "Connection %d: TEARDOWN", conn->connection_number);
// if (!rtsp_playing())
// debug(1, "This RTSP connection thread (%d) doesn't think it's playing, but "
@@ -904,7 +903,7 @@ static void handle_ignore(rtsp_conn_info *conn, rtsp_message *req, rtsp_message
*/
void handle_set_parameter_parameter(rtsp_conn_info *conn, rtsp_message *req,
__attribute__((unused)) rtsp_message *resp) {
__attribute__((unused)) rtsp_message *resp) {
char *cp = req->content;
int cp_left = req->contentlength;
char *next;