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Code Issues Releases Wiki Activity

Try to restart alsa after underrun.

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This commit is contained in:
Johns 2012-02-25 18:10:19 +01:00
parent aa426cd8b2
commit 769f00b4f6
1 changed files with 43 additions and 15 deletions
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58
audio.c
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@ -140,6 +140,8 @@ static volatile char AudioPaused; ///< audio paused
static unsigned AudioSampleRate; ///< audio sample rate in hz
static unsigned AudioChannels; ///< number of audio channels
static const int AudioBytesProSample = 2; ///< number of bytes per sample
static uint32_t AudioTicks; ///< audio ticks
static uint32_t AudioTickPTS; ///< audio pts of tick
static int64_t AudioPTS; ///< audio pts clock
static int AudioBufferTime = 336; ///< audio buffer time in ms
@ -290,12 +292,6 @@ static int AlsaAddToRingbuffer(const void *samples, int count)
// too many bytes are lost
// FIXME: should skip more, longer skip, but less often?
}
// Update audio clock (stupid gcc developers thinks INT64_C is unsigned)
if (AudioPTS != (int64_t) INT64_C(0x8000000000000000)) {
AudioPTS +=
((int64_t) count * 90000) / (AudioSampleRate * AudioChannels *
AudioBytesProSample);
}
if (!AudioRunning) {
Debug(3, "audio/alsa: start %zd ms %d\n",
@ -353,6 +349,15 @@ static int AlsaPlayRingbuffer(void)
if (AudioThread) {
if (!AudioAlsaDriverBroken) {
Error(_("audio/alsa: broken driver %d\n"), avail);
Error("audio/alsa: state %s\n",
snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));
}
if (snd_pcm_state(AlsaPCMHandle)
== SND_PCM_STATE_PREPARED) {
if ((err = snd_pcm_start(AlsaPCMHandle)) < 0) {
Error(_("audio/alsa: snd_pcm_start(): %s\n"),
snd_strerror(err));
}
}
usleep(5 * 1000);
}
@ -364,6 +369,9 @@ static int AlsaPlayRingbuffer(void)
n = RingBufferGetReadPointer(AlsaRingBuffer, &p);
if (!n) { // ring buffer empty
if (first) { // only error on first loop
Debug(4, "audio/alsa: empty buffers %d\n", avail);
// ring buffer empty
// AlsaLowWaterMark = 1;
return 1;
}
return 0;
@ -438,6 +446,7 @@ static void AlsaFlushBuffers(void)
}
}
AudioRunning = 0;
AudioTicks = 0;
AudioPTS = INT64_C(0x8000000000000000);
}
@ -584,8 +593,6 @@ static void AlsaEnqueue(const void *samples, int count)
Debug(3, "audio/alsa: unpaused\n");
}
}
// Update audio clock
// AudioPTS += (size * 90000) / (AudioSampleRate * AudioChannels * AudioBytesProSample);
}
#endif
@ -670,7 +677,7 @@ static void AlsaThread(void)
break;
}
pthread_yield();
usleep(20 * 1000); // let fill/empty the buffers
usleep(24 * 1000); // let fill/empty the buffers
}
}
}
@ -1294,12 +1301,6 @@ static int OssAddToRingbuffer(const void *samples, int count)
// too many bytes are lost
// FIXME: should skip more, longer skip, but less often?
}
// Update audio clock (stupid gcc developers thinks INT64_C is unsigned)
if (AudioPTS != (int64_t) INT64_C(0x8000000000000000)) {
AudioPTS +=
((int64_t) count * 90000) / (AudioSampleRate * AudioChannels *
AudioBytesProSample);
}
if (!AudioRunning) {
Debug(3, "audio/oss: start %zd ms %d\n",
@ -1386,6 +1387,7 @@ static void OssFlushBuffers(void)
}
}
AudioRunning = 0;
AudioTicks = 0;
AudioPTS = INT64_C(0x8000000000000000);
}
@ -2111,6 +2113,13 @@ void AudioEnqueue(const void *samples, int count)
last = tick;
}
AudioUsedModule->Enqueue(samples, count);
// Update audio clock (stupid gcc developers thinks INT64_C is unsigned)
if (AudioPTS != (int64_t) INT64_C(0x8000000000000000)) {
AudioPTS +=
((int64_t) count * 90000) / (AudioSampleRate * AudioChannels *
AudioBytesProSample);
}
}
/**
@ -2176,6 +2185,25 @@ int64_t AudioGetClock(void)
int64_t delay;
if ((delay = AudioGetDelay())) {
#if 0
int64_t pts;
uint32_t ticks;
pts = AudioPTS - delay;
ticks = GetMsTicks();
if (AudioTicks) {
static int64_t drift;
drift += ((pts - AudioTickPTS) - (ticks - AudioTicks) * 90);
drift /= 2;
if (abs(drift) > 90) {
printf("audio-drift: %d\n", (int)drift);
}
}
AudioTicks = ticks;
AudioTickPTS = pts;
return pts;
#endif
return AudioPTS - delay;
}
}