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Calculate audio sync once per read

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This commit is contained in:
Chris Robinson 2012-12-15 00:18:42 -08:00
parent 32b0a5d96e
commit 62a995d492
1 changed files with 15 additions and 21 deletions
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36
apps/openmw/mwrender/videoplayer.cpp
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@ -255,15 +255,17 @@ class MovieAudioDecoder : public MWSound::Sound_Decoder
ssize_t mFramePos;
ssize_t mFrameSize;
/* Add or subtract samples to get a better sync, return new
* audio buffer size */
int synchronize_audio(uint8_t *samples, int samples_size, double pts)
/* Add or subtract samples to get a better sync, return number of bytes to
* skip (negative means to duplicate). */
int synchronize_audio()
{
if(is->av_sync_type == AV_SYNC_AUDIO_MASTER)
return samples_size;
return 0;
int sample_skip = 0;
// accumulate the clock difference
double diff = is->get_audio_clock() - is->get_master_clock();
double diff = is->get_master_clock() - is->get_audio_clock();
is->audio_diff_cum = diff + is->audio_diff_avg_coef *
is->audio_diff_cum;
if(is->audio_diff_avg_count < AUDIO_DIFF_AVG_NB)
@ -275,19 +277,14 @@ class MovieAudioDecoder : public MWSound::Sound_Decoder
{
int n = av_get_bytes_per_sample(is->audio_st->codec->sample_fmt) *
is->audio_st->codec->channels;
int wanted_size = samples_size + ((int)(diff * is->audio_st->codec->sample_rate) * n);
wanted_size = std::max(0, wanted_size);
wanted_size = std::min(wanted_size, samples_size*2);
samples_size = wanted_size;
sample_skip = ((int)(diff * is->audio_st->codec->sample_rate) * n);
}
}
return samples_size;
return sample_skip;
}
int audio_decode_frame(AVFrame *frame, double *pts_ptr)
int audio_decode_frame(AVFrame *frame)
{
AVPacket *pkt = &is->audio_pkt;
@ -313,7 +310,6 @@ class MovieAudioDecoder : public MWSound::Sound_Decoder
if(!got_frame || frame->nb_samples <= 0)
continue;
*pts_ptr = is->audio_clock;
is->audio_clock += (double)frame->nb_samples /
(double)is->audio_st->codec->sample_rate;
@ -406,25 +402,23 @@ public:
size_t read(char *stream, size_t len)
{
int sample_skip = synchronize_audio();
size_t total = 0;
while(total < len)
{
if(mFramePos >= mFrameSize)
while(mFramePos >= mFrameSize)
{
int audio_size;
double pts;
/* We have already sent all our data; get more */
mFrameSize = audio_decode_frame(mFrame, &pts);
mFrameSize = audio_decode_frame(mFrame);
if(mFrameSize < 0)
{
/* If error, we're done */
break;
}
audio_size = synchronize_audio(mFrame->data[0], mFrameSize, pts);
mFramePos = mFrameSize - audio_size;
mFramePos = std::min(mFrameSize, sample_skip);
sample_skip -= mFramePos;
}
size_t len1 = len - total;