diff options
Diffstat (limited to 'audio.c')
| -rw-r--r-- | audio.c | 39 |
1 files changed, 30 insertions, 9 deletions
@@ -546,6 +546,13 @@ static void AudioUpmix(const int16_t * in, int in_chan, int frames, /** ** Resample ffmpeg sample format to hardware format. ** +** FIXME: use libswresample for this and move it to codec. +** FIXME: ffmpeg to alsa conversion is already done in codec.c. +** +** ffmpeg L R C Ls Rs -> alsa L R Ls Rs C +** ffmpeg L R C LFE Ls Rs -> alsa L R Ls Rs C LFE +** ffmpeg L R C LFE Ls Rs Rl Rr -> alsa L R Ls Rs C LFE Rl Rr +** ** @param in input sample buffer ** @param in_chan nr. of input channels ** @param frames number of frames in sample buffer @@ -581,6 +588,9 @@ static void AudioResample(const int16_t * in, int in_chan, int frames, AudioSurround2Stereo(in, in_chan, frames, out); break; case 5 * 8 + 6: + case 3 * 8 + 8: + case 5 * 8 + 8: + case 6 * 8 + 8: AudioUpmix(in, in_chan, frames, out, out_chan); break; @@ -1956,6 +1966,7 @@ static int AudioNextRing(void) int use_ac3; int sample_rate; int channels; + size_t used; // update audio format // not always needed, but check if needed is too complex @@ -1975,12 +1986,18 @@ static int AudioNextRing(void) AudioResetCompressor(); AudioResetNormalizer(); + Debug(3, "audio: a/v next buf(%d,%4zdms)\n", atomic_read(&AudioRingFilled), + (RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer) * 1000) + / (AudioRing[AudioRingWrite].HwSampleRate * + AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample)); + // stop, if not enough in next buffer - if (AudioStartThreshold >= - RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer)) { - return 1; + used = RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer); + if (AudioStartThreshold * 4 < used || (AudioVideoIsReady + && AudioStartThreshold < used)) { + return 0; } - return 0; + return 1; } /** @@ -2029,10 +2046,13 @@ static void *AudioPlayHandlerThread(void *dummy) } if (flush) { + Debug(3, "audio: flush\n"); AudioUsedModule->FlushBuffers(); if (AudioNextRing()) { + Debug(3, "audio: break after flush\n"); break; } + Debug(3, "audio: continue after flush\n"); } // try to play some samples err = AudioUsedModule->Thread(); @@ -2275,11 +2295,12 @@ void AudioVideoReady(int64_t pts) (used * 90 * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample); - Debug(3, "audio: a/v buf:%4zdms %s|%s = %dms video ready\n", + Debug(3, "audio: a/v sync buf(%d,%4zdms) %s|%s = %dms %s\n", + atomic_read(&AudioRingFilled), (used * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample), - Timestamp2String(audio_pts), Timestamp2String(pts), - (int)(pts - audio_pts) / 90); + Timestamp2String(pts), Timestamp2String(audio_pts), + (int)(pts - audio_pts) / 90, AudioRunning ? "running" : "ready"); if (!AudioRunning) { int skip; @@ -2290,7 +2311,7 @@ void AudioVideoReady(int64_t pts) pts - 15 * 20 * 90 - AudioBufferTime * 90 - audio_pts + VideoAudioDelay; #ifdef DEBUG - printf("%dms %dms %dms\n", (int)(pts - audio_pts) / 90, + fprintf(stderr, "%dms %dms %dms\n", (int)(pts - audio_pts) / 90, VideoAudioDelay / 90, skip / 90); #endif // guard against old PTS @@ -2303,7 +2324,7 @@ void AudioVideoReady(int64_t pts) AudioSkip = skip - used; skip = used; } - Debug(3, "audio: advance %dms %d/%zd\n", + Debug(3, "audio: sync advance %dms %d/%zd\n", (skip * 1000) / (AudioRing[AudioRingWrite].HwSampleRate * AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample), skip, used); |