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Fix PTS produced by the default opus/flac encoders

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The default OPUS and FLAC encoders on Android rewrite the input PTS so
that they exactly match the number of samples.

As a consequence:
 - audio clock drift is not compensated
 - implicit silences (without packets) are ignored

To work around this behavior, generate new PTS based on the current time
(after encoding) and the packet duration.

PR #5870 <https://github.com/Genymobile/scrcpy/pull/5870>
This commit is contained in:
Romain Vimont 2025-03-02 17:09:43 +01:00
commit 245981281e
1 changed files with 31 additions and 0 deletions
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31
server/src/main/java/com/genymobile/scrcpy/audio/AudioEncoder.java
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@ -55,6 +55,9 @@ public final class AudioEncoder implements AsyncProcessor {
private final List<CodecOption> codecOptions; private final List<CodecOption> codecOptions;
private final String encoderName; private final String encoderName;
private boolean recreatePts;
private long previousPts;
// Capacity of 64 is in practice "infinite" (it is limited by the number of available MediaCodec buffers, typically 4). // Capacity of 64 is in practice "infinite" (it is limited by the number of available MediaCodec buffers, typically 4).
// So many pending tasks would lead to an unacceptable delay anyway. // So many pending tasks would lead to an unacceptable delay anyway.
private final BlockingQueue<InputTask> inputTasks = new ArrayBlockingQueue<>(64); private final BlockingQueue<InputTask> inputTasks = new ArrayBlockingQueue<>(64);
@ -118,6 +121,9 @@ public final class AudioEncoder implements AsyncProcessor {
OutputTask task = outputTasks.take(); OutputTask task = outputTasks.take();
ByteBuffer buffer = mediaCodec.getOutputBuffer(task.index); ByteBuffer buffer = mediaCodec.getOutputBuffer(task.index);
try { try {
if (recreatePts) {
fixTimestamp(task.bufferInfo);
}
streamer.writePacket(buffer, task.bufferInfo); streamer.writePacket(buffer, task.bufferInfo);
} finally { } finally {
mediaCodec.releaseOutputBuffer(task.index, false); mediaCodec.releaseOutputBuffer(task.index, false);
@ -125,6 +131,25 @@ public final class AudioEncoder implements AsyncProcessor {
} }
} }
private void fixTimestamp(MediaCodec.BufferInfo bufferInfo) {
assert recreatePts;
if ((bufferInfo.flags & MediaCodec.BUFFER_FLAG_CODEC_CONFIG) != 0) {
// Config packet, nothing to fix
return;
}
long pts = bufferInfo.presentationTimeUs;
if (previousPts != 0) {
long now = System.nanoTime() / 1000;
// This specific encoder produces PTS matching the exact number of samples
long duration = pts - previousPts;
bufferInfo.presentationTimeUs = now - duration;
}
previousPts = pts;
}
@Override @Override
public void start(TerminationListener listener) { public void start(TerminationListener listener) {
thread = new Thread(() -> { thread = new Thread(() -> {
@ -194,6 +219,12 @@ public final class AudioEncoder implements AsyncProcessor {
Codec codec = streamer.getCodec(); Codec codec = streamer.getCodec();
mediaCodec = createMediaCodec(codec, encoderName); mediaCodec = createMediaCodec(codec, encoderName);
// The default OPUS and FLAC encoders overwrite the input PTS with a value that matches the number of samples. This is not the behavior
// we want: it ignores any audio clock drift and hard silences (packets not produced on silence). To work around this behavior,
// regenerate PTS based on the current time and the packet duration.
String codecName = mediaCodec.getCanonicalName();
recreatePts = "c2.android.opus.encoder".equals(codecName) || "c2.android.flac.encoder".equals(codecName);
mediaCodecThread = new HandlerThread("media-codec"); mediaCodecThread = new HandlerThread("media-codec");
mediaCodecThread.start(); mediaCodecThread.start();