mirror of
https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git
synced 2023-10-10 19:16:51 +02:00
More debug for flush buffers. Bigger audio buffer.
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parent
24ba8175a3
commit
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55
audio.c
55
audio.c
@ -223,7 +223,7 @@ static void AudioRingInit(void)
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for (i = 0; i < AUDIO_RING_MAX; ++i) {
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// FIXME:
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//AlsaRingBuffer = RingBufferNew(48000 * 8 * 2); // ~1s 8ch 16bit
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//AlsaRingBuffer = RingBufferNew(2 * 48000 * 8 * 2); // ~2s 8ch 16bit
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}
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// one slot always reservered
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AudioRingWrite = 1;
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@ -432,8 +432,24 @@ static void AlsaFlushBuffers(void)
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snd_pcm_state_t state;
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if (AlsaRingBuffer && AlsaPCMHandle) {
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#ifdef DEBUG
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const void *r;
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void *w;
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#endif
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RingBufferReadAdvance(AlsaRingBuffer,
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RingBufferUsedBytes(AlsaRingBuffer));
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#ifdef DEBUG
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RingBufferGetWritePointer(AlsaRingBuffer, &w);
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RingBufferGetReadPointer(AlsaRingBuffer, &r);
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if (r != w) {
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Fatal(_("audio/alsa: ringbuffer out of sync %zd-%zd\n"),
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RingBufferGetWritePointer(AlsaRingBuffer, &w),
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RingBufferGetReadPointer(AlsaRingBuffer, &r));
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abort();
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}
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#endif
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state = snd_pcm_state(AlsaPCMHandle);
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Debug(3, "audio/alsa: flush state %s\n", snd_pcm_state_name(state));
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if (state != SND_PCM_STATE_OPEN) {
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@ -722,19 +738,31 @@ static void AlsaThreadEnqueue(const void *samples, int count)
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*/
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static void AlsaVideoReady(void)
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{
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if (AudioSampleRate && AudioChannels) {
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Debug(3, "audio/alsa: start %4zdms video start\n",
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(RingBufferUsedBytes(AlsaRingBuffer) * 1000)
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/ (AudioSampleRate * AudioChannels * AudioBytesProSample));
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}
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if (!AudioRunning) {
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size_t used;
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used = RingBufferUsedBytes(AlsaRingBuffer);
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// enough video + audio buffered
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if (AlsaStartThreshold < RingBufferUsedBytes(AlsaRingBuffer)) {
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if (AlsaStartThreshold < used) {
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// too much audio buffered, skip it
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if (AlsaStartThreshold * 2 < used) {
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Debug(3, "audio/alsa: start %4zdms skip ready\n",
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((used - AlsaStartThreshold * 2) * 1000)
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/ (AudioSampleRate * AudioChannels * AudioBytesProSample));
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RingBufferReadAdvance(AlsaRingBuffer,
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used - AlsaStartThreshold * 2);
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}
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AudioRunning = 1;
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pthread_cond_signal(&AudioStartCond);
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}
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}
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if (AudioSampleRate && AudioChannels) {
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Debug(3, "audio/alsa: start %4zdms video ready\n",
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(RingBufferUsedBytes(AlsaRingBuffer) * 1000)
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/ (AudioSampleRate * AudioChannels * AudioBytesProSample));
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}
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}
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/**
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@ -976,6 +1004,9 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
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#if 1 // easy alsa hw setup way
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// flush any buffered data
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AudioFlushBuffers();
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Debug(3, "audio: %dms flush\n", (AudioUsedBytes() * 1000)
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/ (!AudioSampleRate + !AudioChannels +
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AudioSampleRate * AudioChannels * AudioBytesProSample));
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if (1) { // close+open to fix hdmi no sound bugs
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handle = AlsaPCMHandle;
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@ -1241,7 +1272,7 @@ static void AlsaInit(void)
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#else
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(void)AlsaNoopCallback;
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#endif
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AlsaRingBuffer = RingBufferNew(48000 * 8 * 2); // ~1s 8ch 16bit
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AlsaRingBuffer = RingBufferNew(2 * 48000 * 8 * 2); // ~2s 8ch 16bit
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AlsaInitPCM();
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AlsaInitMixer();
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@ -1914,7 +1945,7 @@ void OssPause(void)
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*/
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static void OssInit(void)
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{
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OssRingBuffer = RingBufferNew(48000 * 8 * 2); // ~1s 8ch 16bit
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OssRingBuffer = RingBufferNew(2 * 48000 * 8 * 2); // ~2s 8ch 16bit
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OssInitPCM();
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OssInitMixer();
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@ -2270,8 +2301,8 @@ void AudioSetClock(int64_t pts)
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{
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#ifdef DEBUG
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if (AudioPTS != pts) {
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Debug(4, "audio: set clock to %#012" PRIx64 " %#012" PRIx64 " pts\n",
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AudioPTS, pts);
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Debug(4, "audio: set clock %s -> %s pts\n", Timestamp2String(AudioPTS),
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Timestamp2String(pts));
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}
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#endif
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AudioPTS = pts;
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6
misc.h
6
misc.h
@ -107,7 +107,9 @@ static inline void Syslog(const int level, const char *format, ...)
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#define Debug(level, fmt...) /* disabled */
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#endif
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#ifdef AV_NOPTS_VALUE
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#ifndef AV_NOPTS_VALUE
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#define AV_NOPTS_VALUE INT64_C(0x8000000000000000)
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#endif
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/**
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** Nice time-stamp string.
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@ -130,8 +132,6 @@ static inline const char *Timestamp2String(int64_t ts)
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return buf[idx];
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}
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#endif
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/**
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** Get ticks in ms.
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**
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