diff options
| -rw-r--r-- | audio.c | 55 | ||||
| -rw-r--r-- | misc.h | 6 |
2 files changed, 46 insertions, 15 deletions
@@ -223,7 +223,7 @@ static void AudioRingInit(void) for (i = 0; i < AUDIO_RING_MAX; ++i) { // FIXME: - //AlsaRingBuffer = RingBufferNew(48000 * 8 * 2); // ~1s 8ch 16bit + //AlsaRingBuffer = RingBufferNew(2 * 48000 * 8 * 2); // ~2s 8ch 16bit } // one slot always reservered AudioRingWrite = 1; @@ -432,8 +432,24 @@ static void AlsaFlushBuffers(void) snd_pcm_state_t state; if (AlsaRingBuffer && AlsaPCMHandle) { +#ifdef DEBUG + const void *r; + void *w; +#endif + RingBufferReadAdvance(AlsaRingBuffer, RingBufferUsedBytes(AlsaRingBuffer)); +#ifdef DEBUG + RingBufferGetWritePointer(AlsaRingBuffer, &w); + RingBufferGetReadPointer(AlsaRingBuffer, &r); + if (r != w) { + Fatal(_("audio/alsa: ringbuffer out of sync %zd-%zd\n"), + RingBufferGetWritePointer(AlsaRingBuffer, &w), + RingBufferGetReadPointer(AlsaRingBuffer, &r)); + abort(); + } +#endif + state = snd_pcm_state(AlsaPCMHandle); Debug(3, "audio/alsa: flush state %s\n", snd_pcm_state_name(state)); if (state != SND_PCM_STATE_OPEN) { @@ -722,19 +738,31 @@ static void AlsaThreadEnqueue(const void *samples, int count) */ static void AlsaVideoReady(void) { - if (AudioSampleRate && AudioChannels) { - Debug(3, "audio/alsa: start %4zdms video start\n", - (RingBufferUsedBytes(AlsaRingBuffer) * 1000) - / (AudioSampleRate * AudioChannels * AudioBytesProSample)); - } - if (!AudioRunning) { + size_t used; + + used = RingBufferUsedBytes(AlsaRingBuffer); // enough video + audio buffered - if (AlsaStartThreshold < RingBufferUsedBytes(AlsaRingBuffer)) { + if (AlsaStartThreshold < used) { + // too much audio buffered, skip it + if (AlsaStartThreshold * 2 < used) { + Debug(3, "audio/alsa: start %4zdms skip ready\n", + ((used - AlsaStartThreshold * 2) * 1000) + / (AudioSampleRate * AudioChannels * AudioBytesProSample)); + RingBufferReadAdvance(AlsaRingBuffer, + used - AlsaStartThreshold * 2); + } AudioRunning = 1; pthread_cond_signal(&AudioStartCond); } } + + if (AudioSampleRate && AudioChannels) { + Debug(3, "audio/alsa: start %4zdms video ready\n", + (RingBufferUsedBytes(AlsaRingBuffer) * 1000) + / (AudioSampleRate * AudioChannels * AudioBytesProSample)); + } + } /** @@ -976,6 +1004,9 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3) #if 1 // easy alsa hw setup way // flush any buffered data AudioFlushBuffers(); + Debug(3, "audio: %dms flush\n", (AudioUsedBytes() * 1000) + / (!AudioSampleRate + !AudioChannels + + AudioSampleRate * AudioChannels * AudioBytesProSample)); if (1) { // close+open to fix hdmi no sound bugs handle = AlsaPCMHandle; @@ -1241,7 +1272,7 @@ static void AlsaInit(void) #else (void)AlsaNoopCallback; #endif - AlsaRingBuffer = RingBufferNew(48000 * 8 * 2); // ~1s 8ch 16bit + AlsaRingBuffer = RingBufferNew(2 * 48000 * 8 * 2); // ~2s 8ch 16bit AlsaInitPCM(); AlsaInitMixer(); @@ -1914,7 +1945,7 @@ void OssPause(void) */ static void OssInit(void) { - OssRingBuffer = RingBufferNew(48000 * 8 * 2); // ~1s 8ch 16bit + OssRingBuffer = RingBufferNew(2 * 48000 * 8 * 2); // ~2s 8ch 16bit OssInitPCM(); OssInitMixer(); @@ -2270,8 +2301,8 @@ void AudioSetClock(int64_t pts) { #ifdef DEBUG if (AudioPTS != pts) { - Debug(4, "audio: set clock to %#012" PRIx64 " %#012" PRIx64 " pts\n", - AudioPTS, pts); + Debug(4, "audio: set clock %s -> %s pts\n", Timestamp2String(AudioPTS), + Timestamp2String(pts)); } #endif AudioPTS = pts; @@ -107,7 +107,9 @@ static inline void Syslog(const int level, const char *format, ...) #define Debug(level, fmt...) /* disabled */ #endif -#ifdef AV_NOPTS_VALUE +#ifndef AV_NOPTS_VALUE +#define AV_NOPTS_VALUE INT64_C(0x8000000000000000) +#endif /** ** Nice time-stamp string. @@ -130,8 +132,6 @@ static inline const char *Timestamp2String(int64_t ts) return buf[idx]; } -#endif - /** ** Get ticks in ms. ** |