static void
gst_pulsesrc_stream_latency_update_cb (pa_stream * s, void *userdata)
{
const pa_timing_info *info;
pa_usec_t source_usec;
info = pa_stream_get_timing_info (s);
if (!info) {
GST_LOG_OBJECT (GST_PULSESRC_CAST (userdata),
"latency update (information unknown)");
return;
}
#ifdef HAVE_PULSE_0_9_11
source_usec = info->configured_source_usec;
#else
source_usec = 0;
#endif
GST_LOG_OBJECT (GST_PULSESRC_CAST (userdata),
"latency_update, %" G_GUINT64_FORMAT ", %d:%" G_GINT64_FORMAT ", %d:%"
G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT,
GST_TIMEVAL_TO_TIME (info->timestamp), info->write_index_corrupt,
info->write_index, info->read_index_corrupt, info->read_index,
info->source_usec, source_usec);
}
void CAESinkPULSE::GetDelay(AEDelayStatus& status)
{
if (!m_IsAllocated)
{
status.SetDelay(0);
return;
}
pa_threaded_mainloop_lock(m_MainLoop);
const pa_timing_info* pti = pa_stream_get_timing_info(m_Stream);
// only incorporate local sink delay + internal PA transport delay
double sink_delay = (pti->configured_sink_usec / 1000000.0);
double transport_delay = pti->transport_usec / 1000000.0;
uint64_t diff = CurrentHostCounter() - m_lastPackageStamp;
unsigned int bytes_played = (unsigned int) ((double) diff * (double) m_BytesPerSecond / (double) CurrentHostFrequency() + 0.5);
int buffer_delay = m_filled_bytes - bytes_played;
if (buffer_delay < 0)
buffer_delay = 0;
pa_threaded_mainloop_unlock(m_MainLoop);
double delay = buffer_delay / (double) m_BytesPerSecond + sink_delay + transport_delay;
status.SetDelay(delay);
}
// Return the current latency in seconds
static float get_delay(struct ao *ao)
{
/* This code basically does what pa_stream_get_latency() _should_
* do, but doesn't due to multiple known bugs in PulseAudio (at
* PulseAudio version 2.1). In particular, the timing interpolation
* mode (PA_STREAM_INTERPOLATE_TIMING) can return completely bogus
* values, and the non-interpolating code has a bug causing too
* large results at end of stream (so a stream never seems to finish).
* This code can still return wrong values in some cases due to known
* PulseAudio bugs that can not be worked around on the client side.
*
* We always query the server for latest timing info. This may take
* too long to work well with remote audio servers, but at least
* this should be enough to fix the normal local playback case.
*/
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
if (!waitop(priv, pa_stream_update_timing_info(priv->stream, NULL, NULL))) {
GENERIC_ERR_MSG(priv->context, "pa_stream_update_timing_info() failed");
return 0;
}
pa_threaded_mainloop_lock(priv->mainloop);
const pa_timing_info *ti = pa_stream_get_timing_info(priv->stream);
if (!ti) {
pa_threaded_mainloop_unlock(priv->mainloop);
GENERIC_ERR_MSG(priv->context, "pa_stream_get_timing_info() failed");
return 0;
}
const struct pa_sample_spec *ss = pa_stream_get_sample_spec(priv->stream);
if (!ss) {
pa_threaded_mainloop_unlock(priv->mainloop);
GENERIC_ERR_MSG(priv->context, "pa_stream_get_sample_spec() failed");
return 0;
}
// data left in PulseAudio's main buffers (not written to sink yet)
int64_t latency = pa_bytes_to_usec(ti->write_index - ti->read_index, ss);
// since this info may be from a while ago, playback has progressed since
latency -= ti->transport_usec;
// data already moved from buffers to sink, but not played yet
int64_t sink_latency = ti->sink_usec;
if (!ti->playing)
/* At the end of a stream, part of the data "left" in the sink may
* be padding silence after the end; that should be subtracted to
* get the amount of real audio from our stream. This adjustment
* is missing from Pulseaudio's own get_latency calculations
* (as of PulseAudio 2.1). */
sink_latency -= pa_bytes_to_usec(ti->since_underrun, ss);
if (sink_latency > 0)
latency += sink_latency;
if (latency < 0)
latency = 0;
pa_threaded_mainloop_unlock(priv->mainloop);
return latency / 1e6;
}
/*
* update pulseaudio latency
* args:
* s - pointer to pa_stream
*
* asserts:
* none
*
* returns:none
*/
static void get_latency(pa_stream *s)
{
pa_usec_t l;
int negative;
pa_stream_get_timing_info(s);
if (pa_stream_get_latency(s, &l, &negative) != 0)
{
fprintf(stderr, "AUDIO: Pulseaudio pa_stream_get_latency() failed\n");
return;
}
//latency = l * (negative?-1:1);
latency = l; /*can only be negative in monitoring streams*/
//printf("AUDIO: pulseaudio latency is %0.0f usec \r", (float)latency);
}
static int pulse_playing(const pa_timing_info *the_timing_info) {
ENTER(__FUNCTION__);
int r = 0;
const pa_timing_info *i;
assert(the_timing_info);
CHECK_CONNECTED(0);
pa_threaded_mainloop_lock(mainloop);
for (;;) {
CHECK_DEAD_GOTO(fail, 1);
if ((i = pa_stream_get_timing_info(stream)))
{
break;
}
if (pa_context_errno(context) != PA_ERR_NODATA) {
SHOW("pa_stream_get_timing_info() failed: %s", pa_strerror(pa_context_errno(context)));
goto fail;
}
pa_threaded_mainloop_wait(mainloop);
}
r = i->playing;
memcpy((void*)the_timing_info, (void*)i, sizeof(pa_timing_info));
// display_timing_info(i);
fail:
pa_threaded_mainloop_unlock(mainloop);
return r;
}
}
static void outputStreamOverflowCallback(pa_stream *stream, void *userdata)
{
Q_UNUSED(stream)
Q_UNUSED(userdata)
qWarning() << "Got a buffer overflow!";
}
static void outputStreamLatencyCallback(pa_stream *stream, void *userdata)
{
Q_UNUSED(stream)
Q_UNUSED(userdata)
#ifdef DEBUG_PULSE
const pa_timing_info *info = pa_stream_get_timing_info(stream);
qDebug() << "Write index corrupt: " << info->write_index_corrupt;
qDebug() << "Write index: " << info->write_index;
qDebug() << "Read index corrupt: " << info->read_index_corrupt;
qDebug() << "Read index: " << info->read_index;
qDebug() << "Sink usec: " << info->sink_usec;
qDebug() << "Configured sink usec: " << info->configured_sink_usec;
#endif
}
static void outputStreamSuccessCallback(pa_stream *stream, int success, void *userdata)
{
Q_UNUSED(stream);
Q_UNUSED(success);
Q_UNUSED(userdata);
static int
pulseaudio_audio_deliver(audio_decoder_t *ad, int samples,
int64_t pts, int epoch)
{
decoder_t *d = (decoder_t *)ad;
size_t bytes;
if(ad->ad_spdif_muxer != NULL) {
bytes = ad->ad_spdif_frame_size;
} else {
bytes = samples * d->framesize;
}
void *buf;
assert(d->s != NULL);
pa_threaded_mainloop_lock(mainloop);
int writable = pa_stream_writable_size(d->s);
if(writable == 0) {
d->blocked = 1;
pa_threaded_mainloop_unlock(mainloop);
return 1;
}
pa_stream_begin_write(d->s, &buf, &bytes);
assert(bytes > 0);
if(ad->ad_spdif_muxer != NULL) {
memcpy(buf, ad->ad_spdif_frame, ad->ad_spdif_frame_size);
} else {
int rsamples = bytes / d->framesize;
uint8_t *data[8] = {0};
data[0] = (uint8_t *)buf;
assert(rsamples <= samples);
avresample_read(ad->ad_avr, data, rsamples);
}
if(pts != AV_NOPTS_VALUE) {
media_pipe_t *mp = ad->ad_mp;
hts_mutex_lock(&mp->mp_clock_mutex);
const pa_timing_info *ti = pa_stream_get_timing_info(d->s);
if(ti != NULL) {
mp->mp_audio_clock_avtime =
ti->timestamp.tv_sec * 1000000LL + ti->timestamp.tv_usec;
int64_t busec = pa_bytes_to_usec(ti->write_index - ti->read_index,
&d->ss);
int64_t delay = ti->sink_usec + busec + ti->transport_usec;
mp->mp_audio_clock = pts - delay;
mp->mp_audio_clock_epoch = epoch;
}
hts_mutex_unlock(&mp->mp_clock_mutex);
}
pa_stream_write(d->s, buf, bytes, NULL, 0LL, PA_SEEK_RELATIVE);
ad->ad_spdif_frame_size = 0;
pa_threaded_mainloop_unlock(mainloop);
return 0;
}
/* This is called whenever the context status changes */
static void context_state_callback(pa_context *c, void *userdata) {
fail_unless(c != NULL);
switch (pa_context_get_state(c)) {
case PA_CONTEXT_CONNECTING:
case PA_CONTEXT_AUTHORIZING:
case PA_CONTEXT_SETTING_NAME:
break;
case PA_CONTEXT_READY: {
pa_stream_flags_t flags = PA_STREAM_AUTO_TIMING_UPDATE;
pa_buffer_attr attr;
static const pa_sample_spec ss = {
.format = PA_SAMPLE_S16LE,
.rate = 44100,
.channels = 2
};
pa_zero(attr);
attr.maxlength = (uint32_t) -1;
attr.tlength = latency > 0 ? (uint32_t) pa_usec_to_bytes(latency, &ss) : (uint32_t) -1;
attr.prebuf = (uint32_t) -1;
attr.minreq = (uint32_t) -1;
attr.fragsize = (uint32_t) -1;
#ifdef INTERPOLATE
flags |= PA_STREAM_INTERPOLATE_TIMING;
#endif
if (latency > 0)
flags |= PA_STREAM_ADJUST_LATENCY;
pa_log("Connection established");
stream = pa_stream_new(c, "interpol-test", &ss, NULL);
fail_unless(stream != NULL);
if (playback) {
pa_assert_se(pa_stream_connect_playback(stream, NULL, &attr, flags, NULL, NULL) == 0);
pa_stream_set_write_callback(stream, stream_write_cb, NULL);
} else {
pa_assert_se(pa_stream_connect_record(stream, NULL, &attr, flags) == 0);
pa_stream_set_read_callback(stream, stream_read_cb, NULL);
}
pa_stream_set_latency_update_callback(stream, stream_latency_cb, NULL);
break;
}
case PA_CONTEXT_TERMINATED:
break;
case PA_CONTEXT_FAILED:
default:
pa_log_error("Context error: %s", pa_strerror(pa_context_errno(c)));
ck_abort();
}
}
START_TEST (interpol_test) {
pa_threaded_mainloop* m = NULL;
int k;
struct timeval start, last_info = { 0, 0 };
pa_usec_t old_t = 0, old_rtc = 0;
#ifdef CORK
bool corked = false;
#endif
/* Set up a new main loop */
m = pa_threaded_mainloop_new();
fail_unless(m != NULL);
mainloop_api = pa_threaded_mainloop_get_api(m);
fail_unless(mainloop_api != NULL);
context = pa_context_new(mainloop_api, bname);
fail_unless(context != NULL);
pa_context_set_state_callback(context, context_state_callback, NULL);
fail_unless(pa_context_connect(context, NULL, 0, NULL) >= 0);
pa_gettimeofday(&start);
fail_unless(pa_threaded_mainloop_start(m) >= 0);
/* #ifdef CORK */
for (k = 0; k < 20000; k++)
/* #else */
/* for (k = 0; k < 2000; k++) */
/* #endif */
{
bool success = false, changed = false;
pa_usec_t t, rtc, d;
struct timeval now, tv;
bool playing = false;
pa_threaded_mainloop_lock(m);
if (stream) {
const pa_timing_info *info;
if (pa_stream_get_time(stream, &t) >= 0 &&
pa_stream_get_latency(stream, &d, NULL) >= 0)
success = true;
if ((info = pa_stream_get_timing_info(stream))) {
if (memcmp(&last_info, &info->timestamp, sizeof(struct timeval))) {
changed = true;
last_info = info->timestamp;
}
if (info->playing)
playing = true;
}
}
pa_threaded_mainloop_unlock(m);
pa_gettimeofday(&now);
if (success) {
#ifdef CORK
bool cork_now;
#endif
rtc = pa_timeval_diff(&now, &start);
pa_log_info("%i\t%llu\t%llu\t%llu\t%llu\t%lli\t%u\t%u\t%llu\t%llu\n", k,
(unsigned long long) rtc,
(unsigned long long) t,
(unsigned long long) (rtc-old_rtc),
(unsigned long long) (t-old_t),
(signed long long) rtc - (signed long long) t,
changed,
playing,
(unsigned long long) latency,
(unsigned long long) d);
fflush(stdout);
old_t = t;
old_rtc = rtc;
#ifdef CORK
cork_now = (rtc / (2*PA_USEC_PER_SEC)) % 2 == 1;
if (corked != cork_now) {
pa_threaded_mainloop_lock(m);
pa_operation_unref(pa_stream_cork(stream, cork_now, NULL, NULL));
pa_threaded_mainloop_unlock(m);
pa_log(cork_now ? "Corking" : "Uncorking");
corked = cork_now;
}
#endif
}
/* Spin loop, ugly but normal usleep() is just too badly grained */
tv = now;
while (pa_timeval_diff(pa_gettimeofday(&now), &tv) < 1000)
pa_thread_yield();
}
if (m)
pa_threaded_mainloop_stop(m);
if (stream) {
pa_stream_disconnect(stream);
pa_stream_unref(stream);
}
if (context) {
pa_context_disconnect(context);
pa_context_unref(context);
}
if (m)
pa_threaded_mainloop_free(m);
}
END_TEST
int main(int argc, char *argv[]) {
int failed = 0;
Suite *s;
TCase *tc;
SRunner *sr;
if (!getenv("MAKE_CHECK"))
pa_log_set_level(PA_LOG_DEBUG);
bname = argv[0];
playback = argc <= 1 || !pa_streq(argv[1], "-r");
latency = (argc >= 2 && !pa_streq(argv[1], "-r")) ? atoi(argv[1]) : (argc >= 3 ? atoi(argv[2]) : 0);
s = suite_create("Interpol");
tc = tcase_create("interpol");
tcase_add_test(tc, interpol_test);
tcase_set_timeout(tc, 5 * 60);
suite_add_tcase(s, tc);
sr = srunner_create(s);
srunner_run_all(sr, CK_NORMAL);
failed = srunner_ntests_failed(sr);
srunner_free(sr);
return (failed == 0) ? EXIT_SUCCESS : EXIT_FAILURE;
}
