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Compensate for trimmed audio in buffer time check
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After a period transition the first buffer queued has the sum of previous period
durations added to its source presentation timestamp. These durations take into
account gapless edits, but the check on the timestamp was based on the submitted
frame count, not the frame count after trimming.

This change fixes an issue where audio/video would gradually drift apart due to
accumulated error in the audio track position, which could lead to freezing due
to the audio renderer stopping being ready and switching to the standalone media
clock.

Issue: #4559

-------------
Created by MOE: https://github.com/google/moe
MOE_MIGRATED_REVID=213819908
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andrewlewis authored and ojw28 committed Sep 21, 2018
1 parent 7d90396 commit 1c07b95
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3 changes: 3 additions & 0 deletions RELEASENOTES.md
Original file line number Diff line number Diff line change
Expand Up @@ -4,6 +4,9 @@

* Allow setting log level for ExoPlayer logcat output
([#4665](https://github.com/google/ExoPlayer/issues/4665)).
* Fix an issue where audio and video would desynchronize when playing
concatenations of gapless content
([#4559](https://github.com/google/ExoPlayer/issues/4559)).

### 2.9.0 ###

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Original file line number Diff line number Diff line change
Expand Up @@ -632,7 +632,9 @@ public boolean handleBuffer(ByteBuffer buffer, long presentationTimeUs)
} else {
// Sanity check that presentationTimeUs is consistent with the expected value.
long expectedPresentationTimeUs =
startMediaTimeUs + inputFramesToDurationUs(getSubmittedFrames());
startMediaTimeUs
+ inputFramesToDurationUs(
getSubmittedFrames() - trimmingAudioProcessor.getTrimmedFrameCount());
if (startMediaTimeState == START_IN_SYNC
&& Math.abs(expectedPresentationTimeUs - presentationTimeUs) > 200000) {
Log.e(TAG, "Discontinuity detected [expected " + expectedPresentationTimeUs + ", got "
Expand Down Expand Up @@ -955,6 +957,7 @@ public void reset() {
playbackParametersCheckpoints.clear();
playbackParametersOffsetUs = 0;
playbackParametersPositionUs = 0;
trimmingAudioProcessor.resetTrimmedFrameCount();
inputBuffer = null;
outputBuffer = null;
flushAudioProcessors();
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Expand Up @@ -25,18 +25,22 @@
/** Audio processor for trimming samples from the start/end of data. */
/* package */ final class TrimmingAudioProcessor implements AudioProcessor {

private static final int OUTPUT_ENCODING = C.ENCODING_PCM_16BIT;

private boolean isActive;
private int trimStartFrames;
private int trimEndFrames;
private int channelCount;
private int sampleRateHz;
private int bytesPerFrame;

private int pendingTrimStartBytes;
private ByteBuffer buffer;
private ByteBuffer outputBuffer;
private byte[] endBuffer;
private int endBufferSize;
private boolean inputEnded;
private long trimmedFrameCount;

/** Creates a new audio processor for trimming samples from the start/end of data. */
public TrimmingAudioProcessor() {
Expand All @@ -61,17 +65,34 @@ public void setTrimFrameCount(int trimStartFrames, int trimEndFrames) {
this.trimEndFrames = trimEndFrames;
}

/** Sets the trimmed frame count returned by {@link #getTrimmedFrameCount()} to zero. */
public void resetTrimmedFrameCount() {
trimmedFrameCount = 0;
}

/**
* Returns the number of audio frames trimmed since the last call to {@link
* #resetTrimmedFrameCount()}.
*/
public long getTrimmedFrameCount() {
return trimmedFrameCount;
}

@Override
public boolean configure(int sampleRateHz, int channelCount, @Encoding int encoding)
throws UnhandledFormatException {
if (encoding != C.ENCODING_PCM_16BIT) {
if (encoding != OUTPUT_ENCODING) {
throw new UnhandledFormatException(sampleRateHz, channelCount, encoding);
}
if (endBufferSize > 0) {
trimmedFrameCount += endBufferSize / bytesPerFrame;
}
this.channelCount = channelCount;
this.sampleRateHz = sampleRateHz;
endBuffer = new byte[trimEndFrames * channelCount * 2];
bytesPerFrame = Util.getPcmFrameSize(OUTPUT_ENCODING, channelCount);
endBuffer = new byte[trimEndFrames * bytesPerFrame];
endBufferSize = 0;
pendingTrimStartBytes = trimStartFrames * channelCount * 2;
pendingTrimStartBytes = trimStartFrames * bytesPerFrame;
boolean wasActive = isActive;
isActive = trimStartFrames != 0 || trimEndFrames != 0;
return wasActive != isActive;
Expand All @@ -89,7 +110,7 @@ public int getOutputChannelCount() {

@Override
public int getOutputEncoding() {
return C.ENCODING_PCM_16BIT;
return OUTPUT_ENCODING;
}

@Override
Expand All @@ -103,8 +124,13 @@ public void queueInput(ByteBuffer inputBuffer) {
int limit = inputBuffer.limit();
int remaining = limit - position;

if (remaining == 0) {
return;
}

// Trim any pending start bytes from the input buffer.
int trimBytes = Math.min(remaining, pendingTrimStartBytes);
trimmedFrameCount += trimBytes / bytesPerFrame;
pendingTrimStartBytes -= trimBytes;
inputBuffer.position(position + trimBytes);
if (pendingTrimStartBytes > 0) {
Expand Down Expand Up @@ -151,17 +177,34 @@ public void queueEndOfStream() {
inputEnded = true;
}

@SuppressWarnings("ReferenceEquality")
@Override
public ByteBuffer getOutput() {
ByteBuffer outputBuffer = this.outputBuffer;
if (inputEnded && endBufferSize > 0 && outputBuffer == EMPTY_BUFFER) {
// Because audio processors may be drained in the middle of the stream we assume that the
// contents of the end buffer need to be output. Gapless transitions don't involve a call to
// queueEndOfStream so won't be affected. When audio is actually ending we play the padding
// data which is incorrect. This behavior can be fixed once we have the timestamps associated
// with input buffers.
if (buffer.capacity() < endBufferSize) {
buffer = ByteBuffer.allocateDirect(endBufferSize).order(ByteOrder.nativeOrder());
} else {
buffer.clear();
}
buffer.put(endBuffer, 0, endBufferSize);
endBufferSize = 0;
buffer.flip();
outputBuffer = buffer;
}
this.outputBuffer = EMPTY_BUFFER;
return outputBuffer;
}

@SuppressWarnings("ReferenceEquality")
@Override
public boolean isEnded() {
return inputEnded && outputBuffer == EMPTY_BUFFER;
return inputEnded && endBufferSize == 0 && outputBuffer == EMPTY_BUFFER;
}

@Override
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