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Use a global to specify alffplay's audio sync threshold

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This commit is contained in:
Chris Robinson
2017-12-31 23:44:33 -08:00
parent 6dd1643a70
commit 258eadd003
1 changed files with 11 additions and 15 deletions
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examples/alffplay.cpp
+11 -15
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@@ -43,7 +43,8 @@ const std::string AppName("alffplay");
bool do_direct_out = false;
LPALGETSOURCEI64VSOFT alGetSourcei64vSOFT;
const std::chrono::duration<double> AVSyncThreshold(0.01);
const std::chrono::duration<double> VideoSyncThreshold(0.01);
const std::chrono::duration<double> AudioSyncThreshold(0.03);
const std::chrono::seconds AVNoSyncThreshold(10);
const std::chrono::milliseconds AudioBufferTime(20); /* Per-buffer */
#define AUDIO_BUFFER_QUEUE_SIZE 25 /* Number of buffers to queue (~500ms) */
@@ -121,10 +122,8 @@ struct AudioState {
/* Used for clock difference average computation */
struct {
double Accum;
std::chrono::duration<double> Accum;
double AvgCoeff;
double Threshold;
int AvgCount;
} mDiff;
/* Time (in nanoseconds) of the next sample to be buffered */
@@ -155,7 +154,7 @@ struct AudioState {
AudioState(MovieState *movie)
: mMovie(movie), mStream(nullptr), mCodecCtx(nullptr)
, mDiff{0.0, 0.0, 0.0, 0}, mCurrentPts(0), mDecodedFrame(nullptr)
, mDiff{{}, 0.0}, mCurrentPts(0), mDecodedFrame(nullptr)
, mSwresCtx(nullptr), mDstChanLayout(0), mDstSampleFmt(AV_SAMPLE_FMT_NONE)
, mSamples(nullptr), mSamplesLen(0), mSamplesPos(0), mSamplesMax(0)
, mFormat(AL_NONE), mFrameSize(0), mSource(0), mBufferIdx(0)
@@ -373,7 +372,6 @@ std::chrono::nanoseconds AudioState::getClock()
int AudioState::getSync()
{
using seconds = std::chrono::duration<double>;
double avg_diff;
if(mMovie->mAVSyncType == SyncMaster::Audio)
return 0;
@@ -383,15 +381,15 @@ int AudioState::getSync()
if(!(diff < AVNoSyncThreshold && diff > -AVNoSyncThreshold))
{
/* Difference is TOO big; reset diff stuff */
mDiff.Accum = 0.0;
/* Difference is TOO big; reset accumulated average */
mDiff.Accum = std::chrono::duration<double>::zero();
return 0;
}
/* Accumulate the diffs */
mDiff.Accum = mDiff.Accum*mDiff.AvgCoeff + diff.count();
avg_diff = mDiff.Accum*(1.0 - mDiff.AvgCoeff);
if(fabs(avg_diff) < mDiff.Threshold)
mDiff.Accum = mDiff.Accum*mDiff.AvgCoeff + diff;
auto avg_diff = mDiff.Accum*(1.0 - mDiff.AvgCoeff);
if(avg_diff < AudioSyncThreshold/2.0 && avg_diff > -AudioSyncThreshold)
return 0;
/* Constrain the per-update difference to avoid exceedingly large skips */
@@ -894,7 +892,7 @@ retry:
auto diff = seconds(vp->mPts - ref_clock);
/* Skip or repeat the frame. Take delay into account. */
auto sync_threshold = std::min(seconds(delay), AVSyncThreshold);
auto sync_threshold = std::min(seconds(delay), VideoSyncThreshold);
if(!(diff < AVNoSyncThreshold && diff > -AVNoSyncThreshold))
{
if(diff <= -sync_threshold)
@@ -907,7 +905,7 @@ retry:
mFrameTimer += delay;
/* Compute the REAL delay. */
auto actual_delay = mFrameTimer - std::chrono::microseconds(av_gettime());
if(!(actual_delay >= AVSyncThreshold))
if(!(actual_delay >= VideoSyncThreshold))
{
/* We don't have time to handle this picture, just skip to the next one. */
mPictQRead = (mPictQRead+1)%mPictQ.size();
@@ -1227,8 +1225,6 @@ int MovieState::streamComponentOpen(int stream_index)
/* Averaging filter for audio sync */
mAudio.mDiff.AvgCoeff = exp(log(0.01) / AUDIO_DIFF_AVG_NB);
/* Correct audio only if larger error than this */
mAudio.mDiff.Threshold = 0.050/* 50 ms */;
mAudioThread = std::thread(std::mem_fn(&AudioState::handler), &mAudio);
break;