public void write(VideoLib.AudioFrame frame)
{
if (audioBuffer == null || frame.Length == 0)
{
return;
}
// store pts for this frame and the byte offset at which this frame is
// written
pts = frame.Pts;
ptsPos = offsetBytes;
int playPos, writePos;
audioBuffer.GetCurrentPosition(out playPos, out writePos);
if (playPos <= offsetBytes && offsetBytes < writePos)
{
log.Warn("playpos:" + playPos.ToString() + " offset:" + offsetBytes.ToString() + " writePos:" + writePos.ToString() + " dataSize:" + frame.Length.ToString());
offsetBytes = writePos;
}
audioBuffer.Write(frame.Data, 0, frame.Length, offsetBytes, LockFlags.None);
offsetBytes = (offsetBytes + frame.Length) % bufferSizeBytes;
if (audioState == AudioState.START_PLAY_AFTER_NEXT_WRITE)
{
audioBuffer.Play(0, PlayFlags.Looping);
audioState = AudioState.PLAYING;
}
}
public void play(VideoLib.AudioFrame frame)
{
if (audioBuffer == null || frame.Length == 0)
{
return;
}
// store pts for this frame and the byte offset at which this frame is
// written
pts = frame.Pts;
ptsPos = offsetBytes;
int playPos, writePos;
audioBuffer.GetCurrentPosition(out playPos, out writePos);
if (playPos <= offsetBytes && offsetBytes < writePos)
{
//log.Warn("playpos:" + playPos.ToString() + " offset:" + offsetBytes.ToString() + " writePos:" + writePos.ToString() + " dataSize:" + frame.Length.ToString());
offsetBytes = writePos;
}
audioBuffer.Write(frame.Data, 0, frame.Length, offsetBytes, LockFlags.None);
offsetBytes = (offsetBytes + frame.Length) % bufferSizeBytes;
if (Status == BufferStatus.None)
{
// start playing
audioBuffer.Play(0, PlayFlags.Looping);
}
//System.Diagnostics.Debug.Print("AudioClock:" + getAudioClock().ToString());
}
void audioRefreshTimer_Tick(Object sender, EventArgs e)
{
restartaudio:
VideoLib.AudioFrame audioFrame = videoDecoder.FrameQueue.getDecodedAudioFrame();
if (audioFrame == null)
{
return;
}
//videoDebug.AudioFrames = videoDebug.AudioFrames + 1;
//videoDebug.AudioFrameLength = audioFrame.Length;
// if the audio is lagging behind too much, skip the buffer completely
double diff = getVideoClock() - audioFrame.Pts;
if (diff > 0.2 && diff < 3 && syncMode == SyncMode.AUDIO_SYNCS_TO_VIDEO)
{
log.Warn("dropping audio buffer, lagging behind: " + (getVideoClock() - audioFrame.Pts).ToString() + " seconds");
goto restartaudio;
}
//adjustAudioSamplesPerSecond(audioFrame);
adjustAudioLength(audioFrame);
audioPlayer.write(audioFrame);
int frameLength = audioFrame.Length;
double actualDelay = synchronizeAudio(frameLength);
if (actualDelay < 0)
{
// delay too small, play next frame as quickly as possible
//videoDebug.NrAudioFramesLaggingBehind = videoDebug.NrAudioFramesLaggingBehind + 1;
goto restartaudio;
}
// start timer with delay for next frame
audioRefreshTimer.Interval = (int)(actualDelay * 1000 + 0.5);
audioRefreshTimer.start();
}
void adjustAudioLength(VideoLib.AudioFrame frame)
{
//videoDebug.AudioFrameLengthAdjust = 0;
if (syncMode == SyncMode.AUDIO_SYNCS_TO_VIDEO)
{
int n = videoDecoder.NrChannels * videoDecoder.BytesPerSample;
double diff = audioPlayer.getAudioClock() - getVideoClock();
if (Math.Abs(diff) < AV_NOSYNC_THRESHOLD)
{
// accumulate the diffs
audioDiffCum = diff + audioDiffAvgCoef * audioDiffCum;
if (audioDiffAvgCount < AUDIO_DIFF_AVG_NB)
{
audioDiffAvgCount++;
}
else
{
double avgDiff = audioDiffCum * (1.0 - audioDiffAvgCoef);
// Shrinking/expanding buffer code....
if (Math.Abs(avgDiff) >= audioDiffThreshold)
{
int wantedSize = (int)(frame.Length + diff * videoDecoder.SamplesPerSecond * n);
// get a correction percent from 10 to 60 based on the avgDiff
// in order to converge a little faster
double correctionPercent = Misc.clamp(10 + (Math.Abs(avgDiff) - audioDiffThreshold) * 15, 10, 60);
//Util.DebugOut(correctionPercent);
//AUDIO_SAMPLE_CORRECTION_PERCENT_MAX
int minSize = (int)(frame.Length * ((100 - correctionPercent)
/ 100));
int maxSize = (int)(frame.Length * ((100 + correctionPercent)
/ 100));
if (wantedSize < minSize)
{
wantedSize = minSize;
}
else if (wantedSize > maxSize)
{
wantedSize = maxSize;
}
// make sure the samples stay aligned after resizing the buffer
wantedSize = (wantedSize / n) * n;
if (wantedSize < frame.Length)
{
// remove samples
//videoDebug.AudioFrameLengthAdjust = wantedSize - frame.Length;
frame.Length = wantedSize;
}
else if (wantedSize > frame.Length)
{
// add samples by copying final samples
int nrExtraSamples = wantedSize - frame.Length;
//videoDebug.AudioFrameLengthAdjust = nrExtraSamples;
byte[] lastSample = new byte[n];
for (int i = 0; i < n; i++)
{
lastSample[i] = frame.Data[frame.Length - n + i];
}
frame.Stream.Position = frame.Length;
while (nrExtraSamples > 0)
{
frame.Stream.Write(lastSample, 0, n);
nrExtraSamples -= n;
}
frame.Stream.Position = 0;
frame.Length = wantedSize;
}
}
}
}
else
{
// difference is TOO big; reset diff stuff
audioDiffAvgCount = 0;
audioDiffCum = 0;
}
}
}
void adjustAudioSamplesPerSecond(VideoLib.AudioFrame frame)
{
//videoDebug.AudioFrameLengthAdjust = 0;
if (syncMode == SyncMode.AUDIO_SYNCS_TO_VIDEO)
{
int n = videoDecoder.NrChannels * videoDecoder.BytesPerSample;
double diff = audioPlayer.getAudioClock() - getVideoClock();
if (Math.Abs(diff) < AV_NOSYNC_THRESHOLD)
{
// accumulate the diffs
audioDiffCum = diff + audioDiffAvgCoef * audioDiffCum;
if (audioDiffAvgCount < AUDIO_DIFF_AVG_NB)
{
audioDiffAvgCount++;
}
else
{
double avgDiff = audioDiffCum * (1.0 - audioDiffAvgCoef);
// Shrinking/expanding buffer code....
if (Math.Abs(avgDiff) >= audioDiffThreshold)
{
int wantedSize = (int)(frame.Length + diff * videoDecoder.SamplesPerSecond * n);
// get a correction percent from 10 to 60 based on the avgDiff
// in order to converge a little faster
double correctionPercent = Misc.clamp(10 + (Math.Abs(avgDiff) - audioDiffThreshold) * 15, 10, 60);
//Util.DebugOut(correctionPercent);
//AUDIO_SAMPLE_CORRECTION_PERCENT_MAX
int minSize = (int)(frame.Length * ((100 - correctionPercent)
/ 100));
int maxSize = (int)(frame.Length * ((100 + correctionPercent)
/ 100));
if (wantedSize < minSize)
{
wantedSize = minSize;
}
else if (wantedSize > maxSize)
{
wantedSize = maxSize;
}
// adjust samples per second to speed up or slow down the audio
Int64 length = frame.Length;
Int64 sps = videoDecoder.SamplesPerSecond;
int samplesPerSecond = (int)((length * sps) / wantedSize);
//videoDebug.AudioFrameLengthAdjust = samplesPerSecond;
audioPlayer.SamplesPerSecond = samplesPerSecond;
}
else
{
audioPlayer.SamplesPerSecond = videoDecoder.SamplesPerSecond;
}
}
}
else
{
// difference is TOO big; reset diff stuff
audioDiffAvgCount = 0;
audioDiffCum = 0;
}
}
}
