aaudio: fix timestamps and underruns
Start the client after we get valid timing data from the server.
That can take a while because of the long cold start times.
The client is synced with the current position of the service.
Now the client can start clean with no underruns.
Bug: 63918065
Test: test_timestamps.cpp
Change-Id: I5d01eb844e4b14cd5477d56ea1dd9e309abc1c52
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 31e0a40..f2e40a2 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -48,6 +48,7 @@
mClockModel.stop(AudioClock::getNanoseconds());
setState(AAUDIO_STREAM_STATE_PAUSING);
+ mAtomicTimestamp.clear();
return AAudioConvert_androidToAAudioResult(pauseWithStatus());
}
@@ -72,21 +73,25 @@
return mServiceInterface.flushStream(mServiceStreamHandle);
}
-void AudioStreamInternalPlay::onFlushFromServer() {
+void AudioStreamInternalPlay::advanceClientToMatchServerPosition() {
int64_t readCounter = mAudioEndpoint.getDataReadCounter();
int64_t writeCounter = mAudioEndpoint.getDataWriteCounter();
// Bump offset so caller does not see the retrograde motion in getFramesRead().
- int64_t framesFlushed = writeCounter - readCounter;
- mFramesOffsetFromService += framesFlushed;
- ALOGD("AudioStreamInternal::onFlushFromServer() readN = %lld, writeN = %lld, offset = %lld",
+ int64_t offset = writeCounter - readCounter;
+ mFramesOffsetFromService += offset;
+ ALOGD("advanceClientToMatchServerPosition() readN = %lld, writeN = %lld, offset = %lld",
(long long)readCounter, (long long)writeCounter, (long long)mFramesOffsetFromService);
- // Flush written frames by forcing writeCounter to readCounter.
- // This is because we cannot move the read counter in the hardware.
+ // Force writeCounter to match readCounter.
+ // This is because we cannot change the read counter in the hardware.
mAudioEndpoint.setDataWriteCounter(readCounter);
}
+void AudioStreamInternalPlay::onFlushFromServer() {
+ advanceClientToMatchServerPosition();
+}
+
// Write the data, block if needed and timeoutMillis > 0
aaudio_result_t AudioStreamInternalPlay::write(const void *buffer, int32_t numFrames,
int64_t timeoutNanoseconds)
@@ -106,6 +111,18 @@
const char *traceName = "aaWrNow";
ATRACE_BEGIN(traceName);
+ if (mClockModel.isStarting()) {
+ // Still haven't got any timestamps from server.
+ // Keep waiting until we get some valid timestamps then start writing to the
+ // current buffer position.
+ ALOGD("processDataNow() wait for valid timestamps");
+ // Sleep very briefly and hope we get a timestamp soon.
+ *wakeTimePtr = currentNanoTime + (2000 * AAUDIO_NANOS_PER_MICROSECOND);
+ ATRACE_END();
+ return 0;
+ }
+ // If we have gotten this far then we have at least one timestamp from server.
+
// If a DMA channel or DSP is reading the other end then we have to update the readCounter.
if (mAudioEndpoint.isFreeRunning()) {
// Update data queue based on the timing model.
@@ -114,6 +131,13 @@
mAudioEndpoint.setDataReadCounter(estimatedReadCounter);
}
+ if (mNeedCatchUp.isRequested()) {
+ // Catch an MMAP pointer that is already advancing.
+ // This will avoid initial underruns caused by a slow cold start.
+ advanceClientToMatchServerPosition();
+ mNeedCatchUp.acknowledge();
+ }
+
// If the read index passed the write index then consider it an underrun.
if (mAudioEndpoint.getFullFramesAvailable() < 0) {
mXRunCount++;
@@ -153,9 +177,9 @@
// Calculate frame position based off of the writeCounter because
// the readCounter might have just advanced in the background,
// causing us to sleep until a later burst.
- int64_t nextReadPosition = mAudioEndpoint.getDataWriteCounter() + mFramesPerBurst
+ int64_t nextPosition = mAudioEndpoint.getDataWriteCounter() + mFramesPerBurst
- mAudioEndpoint.getBufferSizeInFrames();
- wakeTime = mClockModel.convertPositionToTime(nextReadPosition);
+ wakeTime = mClockModel.convertPositionToTime(nextPosition);
}
break;
default: