diff options
Diffstat (limited to 'audio.c')
| -rw-r--r-- | audio.c | 73 |
1 files changed, 55 insertions, 18 deletions
@@ -152,7 +152,9 @@ static int AlsaPlayRingbuffer(void) // happens with broken alsa drivers Error(_("audio/alsa: broken driver %d\n"), avail); } - //break; + Debug(4, "audio/alsa: break state %s\n", + snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle))); + break; } n = RingBufferGetReadPointer(AlsaRingBuffer, &p); @@ -320,8 +322,7 @@ void AudioEnqueue(const void *samples, int count) } #endif state = snd_pcm_state(AlsaPCMHandle); - Debug(3, "audio/alsa: state %d - %s\n", state, - snd_pcm_state_name(state)); + Debug(3, "audio/alsa: state %s\n", snd_pcm_state_name(state)); Debug(3, "audio/alsa: unpaused\n"); AudioPaused = 0; } @@ -381,10 +382,15 @@ static void *AudioPlayHandlerThread(void *dummy) Debug(3, "audio/alsa: flushing buffers\n"); RingBufferReadAdvance(AlsaRingBuffer, RingBufferUsedBytes(AlsaRingBuffer)); - if ((err = snd_pcm_drain(AlsaPCMHandle))) { - Error(_("audio: snd_pcm_drain(): %s\n"), +#if 1 + if ((err = snd_pcm_drop(AlsaPCMHandle))) { + Error(_("audio: snd_pcm_drop(): %s\n"), snd_strerror(err)); + } + if ((err = snd_pcm_prepare(AlsaPCMHandle))) { + Error(_("audio: snd_pcm_prepare(): %s\n"), snd_strerror(err)); } +#endif AlsaFlushBuffer = 0; break; } @@ -504,10 +510,11 @@ static void AlsaInitPCM(void) SND_PCM_NONBLOCK)) < 0) { Fatal(_("audio/alsa: playback open '%s' error: %s\n"), device, snd_strerror(err)); + // FIXME: no fatal error for plugins! } AlsaPCMHandle = handle; - if ((err = snd_pcm_nonblock(handle, SND_PCM_NONBLOCK)) < 0) { + if ((err = snd_pcm_nonblock(handle, 0)) < 0) { Error(_("audio/alsa: can't set block mode: %s\n"), snd_strerror(err)); } @@ -610,7 +617,7 @@ static void AlsaInitMixer(void) void AudioSetClock(int64_t pts) { if (AudioPTS != pts) { - Debug(3, "audio: set clock to %#012" PRIx64 " %#012" PRIx64 " pts\n", + Debug(4, "audio: set clock to %#012" PRIx64 " %#012" PRIx64 " pts\n", AudioPTS, pts); AudioPTS = pts; @@ -618,6 +625,20 @@ void AudioSetClock(int64_t pts) } /** +** Get current audio clock. +*/ +int64_t AudioGetClock(void) +{ + int64_t delay; + + delay = AudioGetDelay(); + if (delay) { + return AudioPTS - delay; + } + return INT64_C(0x8000000000000000); +} + +/** ** Get audio delay in time stamps. */ uint64_t AudioGetDelay(void) @@ -626,17 +647,19 @@ uint64_t AudioGetDelay(void) snd_pcm_sframes_t delay; uint64_t pts; + // delay in frames in alsa + kernel buffers if ((err = snd_pcm_delay(AlsaPCMHandle, &delay)) < 0) { //Debug(3, "audio/alsa: no hw delay\n"); delay = 0UL; } else if (snd_pcm_state(AlsaPCMHandle) != SND_PCM_STATE_RUNNING) { - //Debug(3, "audio/alsa: %ld delay ok, but not running\n", delay); + //Debug(3, "audio/alsa: %ld frames delay ok, but not running\n", delay); } - - pts = ((uint64_t) delay * 90000) / AudioSampleRate; - pts += ((uint64_t) RingBufferUsedBytes(AlsaRingBuffer) * 90000) - / (AudioSampleRate * AudioChannels); - //Debug(3, "audio/alsa: hw+sw delay %"PRId64" ms\n", pts / 90); + //Debug(3, "audio/alsa: %ld frames hw delay\n", delay); + pts = ((uint64_t) delay * 90 * 1000) / AudioSampleRate; + pts += ((uint64_t) RingBufferUsedBytes(AlsaRingBuffer) * 90 * 1000) + / (AudioSampleRate * AudioChannels * 2); + Debug(4, "audio/alsa: hw+sw delay %zd %" PRId64 " ms\n", + RingBufferUsedBytes(AlsaRingBuffer), pts / 90); return pts; } @@ -676,13 +699,18 @@ int AudioSetup(int *freq, int *channels) // flush any buffered data #ifdef USE_AUDIO_THREAD if (AudioRunning) { - AlsaFlushBuffer = 1; + while (AudioRunning) { + AlsaFlushBuffer = 1; + usleep(1 * 1000); + } + AlsaFlushBuffer = 0; } else #endif { RingBufferReadAdvance(AlsaRingBuffer, RingBufferUsedBytes(AlsaRingBuffer)); } + AudioPTS = INT64_C(0x8000000000000000); ret = 0; try_again: @@ -692,6 +720,15 @@ int AudioSetup(int *freq, int *channels) SND_PCM_ACCESS_RW_INTERLEAVED, *channels, *freq, 1, 125 * 1000))) { Error(_("audio/alsa: set params error: %s\n"), snd_strerror(err)); + + /* + if ( err == -EBADFD ) { + snd_pcm_close(AlsaPCMHandle); + AlsaPCMHandle = NULL; + goto try_again; + } + */ + switch (*channels) { case 1: // FIXME: enable channel upmix @@ -795,10 +832,10 @@ int AudioSetup(int *freq, int *channels) Debug(3, "audio/alsa: state %s\n", snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle))); - AlsaStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, buffer_size); - // min 500ms - if (AlsaStartThreshold < (*freq * *channels * 2U) / 2) { - AlsaStartThreshold = (*freq * *channels * 2U) / 2; + AlsaStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size); + // min 333ms + if (AlsaStartThreshold < (*freq * *channels * 2U) / 3) { + AlsaStartThreshold = (*freq * *channels * 2U) / 3; } Debug(3, "audio/alsa: delay %u ms\n", (AlsaStartThreshold * 1000) / (AudioSampleRate * AudioChannels * 2)); |