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Fix issue #42 and changes for faster a/v sync
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@jojo61
jojo61 committed Mar 31, 2020
1 parent fecb814 commit a6e65d9
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Showing 4 changed files with 83 additions and 526 deletions.
80 changes: 24 additions & 56 deletions audio.c
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Original file line number Diff line number Diff line change
Expand Up @@ -86,6 +86,8 @@
#define __USE_GNU
#endif
#include <pthread.h>
#include <sys/syscall.h>
#include <sys/resource.h>
#ifndef HAVE_PTHREAD_NAME
/// only available with newer glibc
#define pthread_setname_np(thread, name)
Expand Down Expand Up @@ -688,7 +690,7 @@ static int AudioRingAdd(unsigned sample_rate, int channels, int passthrough)
AudioRing[AudioRingWrite].InChannels = channels;
AudioRing[AudioRingWrite].HwSampleRate = sample_rate;
AudioRing[AudioRingWrite].HwChannels = AudioChannelMatrix[u][channels];
AudioRing[AudioRingWrite].PTS = INT64_C(0x8000000000000000);
AudioRing[AudioRingWrite].PTS = AV_NOPTS_VALUE;
RingBufferReset(AudioRing[AudioRingWrite].RingBuffer);

Debug(3, "audio: %d ring buffer prepared\n", atomic_read(&AudioRingFilled) + 1);
Expand Down Expand Up @@ -1156,7 +1158,7 @@ static int64_t AlsaGetDelay(void)
//Debug(3, "audio/alsa: %ld frames delay ok, but not running\n", delay);
#endif
}
//Debug(3, "audio/alsa: %ld frames hw delay\n", delay);
Debug(4, "audio/alsa: %ld frames hw delay\n", delay);

// delay can be negative, when underrun occur
if (delay < 0) {
Expand Down Expand Up @@ -1291,7 +1293,6 @@ static int AlsaSetup(int *freq, int *channels, int passthrough)
Info(_("audio/alsa: start delay %ums\n"), (AudioStartThreshold * 1000)
/ (*freq * *channels * AudioBytesProSample));
}

return 0;
}

Expand Down Expand Up @@ -1989,7 +1990,7 @@ static int AudioNextRing(void)

// stop, if not enough in next buffer
used = RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer);
if (AudioStartThreshold * 10 < used || (AudioVideoIsReady && AudioStartThreshold < used)) {
if (AudioStartThreshold * 4 < used || (AudioVideoIsReady && AudioStartThreshold < used)) {
return 0;
}
return 1;
Expand All @@ -2004,6 +2005,7 @@ static void *AudioPlayHandlerThread(void *dummy)
{
Debug(3, "audio: play thread started\n");
prctl(PR_SET_NAME, "cuvid audio", 0, 0, 0);

for (;;) {
// check if we should stop the thread
if (AudioThreadStop) {
Expand All @@ -2020,9 +2022,9 @@ static void *AudioPlayHandlerThread(void *dummy)
} while (!AudioRunning);
pthread_mutex_unlock(&AudioMutex);

Debug(3, "audio: ----> %dms start\n", (AudioUsedBytes() * 1000)
Debug(3, "audio: ----> %dms %d start\n", (AudioUsedBytes() * 1000)
/ (!AudioRing[AudioRingWrite].HwSampleRate + !AudioRing[AudioRingWrite].HwChannels +
AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample));
AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample),AudioUsedBytes());

do {
int filled;
Expand Down Expand Up @@ -2056,10 +2058,8 @@ static void *AudioPlayHandlerThread(void *dummy)
AudioUsedModule->FlushBuffers();
atomic_sub(flush, &AudioRingFilled);
if (AudioNextRing()) {
Debug(3, "audio: HandlerThread break after flush\n");
break;
}
Debug(3, "audio: continue after flush\n");
}
// try to play some samples
err = 0;
Expand Down Expand Up @@ -2252,7 +2252,7 @@ void AudioEnqueue(const void *samples, int count)
AudioNormalizer(buffer, count);
}
}

n = RingBufferWrite(AudioRing[AudioRingWrite].RingBuffer, buffer, count);
if (n != (size_t)count) {
Error(_("audio: can't place %d samples in ring buffer\n"), count);
Expand Down Expand Up @@ -2284,18 +2284,18 @@ void AudioEnqueue(const void *samples, int count)
}
// forced start or enough video + audio buffered
// for some exotic channels * 4 too small
if (AudioStartThreshold * 10 < n || (AudioVideoIsReady
if (AudioStartThreshold * 4 < n || (AudioVideoIsReady
// if ((AudioVideoIsReady
&& AudioStartThreshold < n)) {
// restart play-back
// no lock needed, can wakeup next time
AudioRunning = 1;
pthread_cond_signal(&AudioStartCond);
Debug(3, "Start on AudioEnque\n");
Debug(3, "Start on AudioEnque Threshold %d n %d\n",AudioStartThreshold,n);
}
}
// Update audio clock (stupid gcc developers thinks INT64_C is unsigned)
if (AudioRing[AudioRingWrite].PTS != (int64_t) INT64_C(0x8000000000000000)) {
if (AudioRing[AudioRingWrite].PTS != (int64_t) AV_NOPTS_VALUE) {
AudioRing[AudioRingWrite].PTS += ((int64_t) count * 90 * 1000)
/ (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample);
}
Expand All @@ -2311,21 +2311,21 @@ void AudioVideoReady(int64_t pts)
int64_t audio_pts;
size_t used;

if (pts == (int64_t) INT64_C(0x8000000000000000)) {
if (pts == (int64_t) AV_NOPTS_VALUE) {
Debug(3, "audio: a/v start, no valid video\n");
return;
}
// no valid audio known
if (!AudioRing[AudioRingWrite].HwSampleRate || !AudioRing[AudioRingWrite].HwChannels
|| AudioRing[AudioRingWrite].PTS == (int64_t) INT64_C(0x8000000000000000)) {
|| AudioRing[AudioRingWrite].PTS == (int64_t) AV_NOPTS_VALUE) {
Debug(3, "audio: a/v start, no valid audio\n");
AudioVideoIsReady = 1;
return;
}
// Audio.PTS = next written sample time stamp

used = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer);
audio_pts =
audio_pts =
AudioRing[AudioRingWrite].PTS -
(used * 90 * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels *
AudioBytesProSample);
Expand All @@ -2337,12 +2337,11 @@ void AudioVideoReady(int64_t pts)

if (!AudioRunning) {
int skip;

// buffer ~15 video frames
// FIXME: HDTV can use smaller video buffer
skip = pts - 15 * 20 * 90 - AudioBufferTime * 90 - audio_pts + VideoAudioDelay;
skip = pts - 0 * 20 * 90 - AudioBufferTime * 90 - audio_pts + VideoAudioDelay;
#ifdef DEBUG
fprintf(stderr, "%dms %dms %dms\n", (int)(pts - audio_pts) / 90, VideoAudioDelay / 90, skip / 90);
// fprintf(stderr, "a/v-diff %dms a/v-delay %dms skip %dms Audiobuffer %d\n", (int)(pts - audio_pts) / 90, VideoAudioDelay / 90, skip / 90,AudioBufferTime);
#endif
// guard against old PTS
if (skip > 0 && skip < 4000 * 90) {
Expand All @@ -2353,17 +2352,16 @@ void AudioVideoReady(int64_t pts)
AudioSkip = skip - used;
skip = used;
}
Debug(3, "audio: sync advance %dms %d/%zd\n",
Debug(3, "audio: sync advance %dms %d/%zd Rest %d\n",
(skip * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels *
AudioBytesProSample), skip, used);
AudioBytesProSample), skip, used, AudioSkip);
RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer, skip);

used = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer);
} else {
Debug(3, "No audio skip -> should skip %d\n", skip / 90);
}
// FIXME: skip<0 we need bigger audio buffer

// enough video + audio buffered
if (AudioStartThreshold < used) {
AudioRunning = 1;
Expand All @@ -2373,38 +2371,7 @@ void AudioVideoReady(int64_t pts)
}

AudioVideoIsReady = 1;
#if 0
if (AudioRing[AudioRingWrite].HwSampleRate && AudioRing[AudioRingWrite].HwChannels) {
if (pts != (int64_t) INT64_C(0x8000000000000000)
&& AudioRing[AudioRingWrite].PTS != (int64_t) INT64_C(0x8000000000000000)) {
Debug(3, "audio: a/v %d %s\n", (int)(pts - AudioRing[AudioRingWrite].PTS) / 90,
AudioRunning ? "running" : "stopped");
}
Debug(3, "audio: start %4zdms %s|%s video ready\n",
(RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer) * 1000)
/ (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample),
Timestamp2String(pts), Timestamp2String(AudioRing[AudioRingWrite].PTS));

if (!AudioRunning) {
size_t used;

used = RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer);
// enough video + audio buffered
if (AudioStartThreshold < used) {
// too much audio buffered, skip it
if (AudioStartThreshold < used) {
Debug(3, "audio: start %4zdms skip video ready\n", ((used - AudioStartThreshold) * 1000)
/ (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels *
AudioBytesProSample));
RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer, used - AudioStartThreshold);
}
AudioRunning = 1;
pthread_cond_signal(&AudioStartCond);
}
}
}
AudioVideoIsReady = 1;
#endif
}

/**
Expand Down Expand Up @@ -2439,7 +2406,7 @@ void AudioFlushBuffers(void)
AudioRing[AudioRingWrite].HwChannels = AudioRing[old].HwChannels;
AudioRing[AudioRingWrite].InSampleRate = AudioRing[old].InSampleRate;
AudioRing[AudioRingWrite].InChannels = AudioRing[old].InChannels;
AudioRing[AudioRingWrite].PTS = INT64_C(0x8000000000000000);
AudioRing[AudioRingWrite].PTS = AV_NOPTS_VALUE;
RingBufferReadAdvance(AudioRing[AudioRingWrite].RingBuffer,
RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer));
Debug(3, "audio: reset video ready\n");
Expand Down Expand Up @@ -2512,7 +2479,7 @@ int64_t AudioGetDelay(void)
pts += ((int64_t) RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer)
* 90 * 1000) / (AudioRing[AudioRingRead].HwSampleRate * AudioRing[AudioRingRead].HwChannels *
AudioBytesProSample);
Debug(4, "audio: hw+sw delay %zd %" PRId64 "ms\n", RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer),
Debug(4,"audio: hw+sw delay %zd %" PRId64 "ms\n", RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer),
pts / 90);

return pts;
Expand All @@ -2529,6 +2496,7 @@ void AudioSetClock(int64_t pts)
Debug(4, "audio: set clock %s -> %s pts\n", Timestamp2String(AudioRing[AudioRingWrite].PTS),
Timestamp2String(pts));
}
// printf("Audiosetclock pts %#012" PRIx64 " %d\n",pts,RingBufferUsedBytes(AudioRing[AudioRingWrite].RingBuffer));
AudioRing[AudioRingWrite].PTS = pts;
}

Expand All @@ -2540,7 +2508,7 @@ void AudioSetClock(int64_t pts)
int64_t AudioGetClock(void)
{
// (cast) needed for the evil gcc
if (AudioRing[AudioRingRead].PTS != (int64_t) INT64_C(0x8000000000000000)) {
if (AudioRing[AudioRingRead].PTS != (int64_t) AV_NOPTS_VALUE) {
int64_t delay;

// delay zero, if no valid time stamp
Expand All @@ -2551,7 +2519,7 @@ int64_t AudioGetClock(void)
return AudioRing[AudioRingRead].PTS + 0 * 90 - delay;
}
}
return INT64_C(0x8000000000000000);
return AV_NOPTS_VALUE;
}

/**
Expand Down
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