aaudio: fix small underflow when a stream is stopped
Also remove inconsequential volume parameter.
Bug: 67910437
Test: test_aaudio_monkey.cpp and write_sine_callback.cpp
Change-Id: I6d11f3bfced3d579440f99c02d01a7d68af5c1e0
diff --git a/services/oboeservice/AAudioMixer.cpp b/services/oboeservice/AAudioMixer.cpp
index 952aa82..442653c 100644
--- a/services/oboeservice/AAudioMixer.cpp
+++ b/services/oboeservice/AAudioMixer.cpp
@@ -49,10 +49,9 @@
memset(mOutputBuffer, 0, mBufferSizeInBytes);
}
-bool AAudioMixer::mix(int trackIndex, FifoBuffer *fifo, float volume) {
+bool AAudioMixer::mix(int streamIndex, FifoBuffer *fifo, bool allowUnderflow) {
WrappingBuffer wrappingBuffer;
float *destination = mOutputBuffer;
- fifo_frames_t framesLeft = mFramesPerBurst;
#if AAUDIO_MIXER_ATRACE_ENABLED
ATRACE_BEGIN("aaMix");
@@ -63,35 +62,44 @@
#if AAUDIO_MIXER_ATRACE_ENABLED
if (ATRACE_ENABLED()) {
char rdyText[] = "aaMixRdy#";
- char letter = 'A' + (trackIndex % 26);
+ char letter = 'A' + (streamIndex % 26);
rdyText[sizeof(rdyText) - 2] = letter;
ATRACE_INT(rdyText, fullFrames);
}
#else /* MIXER_ATRACE_ENABLED */
(void) trackIndex;
- (void) fullFrames;
#endif /* AAUDIO_MIXER_ATRACE_ENABLED */
+ // If allowUnderflow then always advance by one burst even if we do not have the data.
+ // Otherwise the stream timing will drift whenever there is an underflow.
+ // This actual underflow can then be detected by the client for XRun counting.
+ //
+ // Generally, allowUnderflow will be false when stopping a stream and we want to
+ // use up whatever data is in the queue.
+ fifo_frames_t framesDesired = mFramesPerBurst;
+ if (!allowUnderflow && fullFrames < framesDesired) {
+ framesDesired = fullFrames; // just use what is available then stop
+ }
+
// Mix data in one or two parts.
int partIndex = 0;
+ int32_t framesLeft = framesDesired;
while (framesLeft > 0 && partIndex < WrappingBuffer::SIZE) {
- fifo_frames_t framesToMix = framesLeft;
- fifo_frames_t framesAvailable = wrappingBuffer.numFrames[partIndex];
- if (framesAvailable > 0) {
- if (framesToMix > framesAvailable) {
- framesToMix = framesAvailable;
+ fifo_frames_t framesToMixFromPart = framesLeft;
+ fifo_frames_t framesAvailableFromPart = wrappingBuffer.numFrames[partIndex];
+ if (framesAvailableFromPart > 0) {
+ if (framesToMixFromPart > framesAvailableFromPart) {
+ framesToMixFromPart = framesAvailableFromPart;
}
- mixPart(destination, (float *)wrappingBuffer.data[partIndex], framesToMix, volume);
+ mixPart(destination, (float *)wrappingBuffer.data[partIndex],
+ framesToMixFromPart);
- destination += framesToMix * mSamplesPerFrame;
- framesLeft -= framesToMix;
+ destination += framesToMixFromPart * mSamplesPerFrame;
+ framesLeft -= framesToMixFromPart;
}
partIndex++;
}
- // Always advance by one burst even if we do not have the data.
- // Otherwise the stream timing will drift whenever there is an underflow.
- // This actual underflow can then be detected by the client for XRun counting.
- fifo->getFifoControllerBase()->advanceReadIndex(mFramesPerBurst);
+ fifo->getFifoControllerBase()->advanceReadIndex(framesDesired);
#if AAUDIO_MIXER_ATRACE_ENABLED
ATRACE_END();
@@ -100,11 +108,11 @@
return (framesLeft > 0); // did not get all the frames we needed, ie. "underflow"
}
-void AAudioMixer::mixPart(float *destination, float *source, int32_t numFrames, float volume) {
+void AAudioMixer::mixPart(float *destination, float *source, int32_t numFrames) {
int32_t numSamples = numFrames * mSamplesPerFrame;
// TODO maybe optimize using SIMD
for (int sampleIndex = 0; sampleIndex < numSamples; sampleIndex++) {
- *destination++ += *source++ * volume;
+ *destination++ += *source++;
}
}
diff --git a/services/oboeservice/AAudioMixer.h b/services/oboeservice/AAudioMixer.h
index a8090bc..5625d4d 100644
--- a/services/oboeservice/AAudioMixer.h
+++ b/services/oboeservice/AAudioMixer.h
@@ -33,13 +33,14 @@
/**
* Mix from this FIFO
- * @param fifo
- * @param volume
- * @return true if underflowed
+ * @param streamIndex for marking stream variables in systrace
+ * @param fifo to read from
+ * @param allowUnderflow if true then allow mixer to advance read index past the write index
+ * @return true if actually underflowed
*/
- bool mix(int trackIndex, android::FifoBuffer *fifo, float volume);
+ bool mix(int streamIndex, android::FifoBuffer *fifo, bool allowUnderflow);
- void mixPart(float *destination, float *source, int32_t numFrames, float volume);
+ void mixPart(float *destination, float *source, int32_t numFrames);
float *getOutputBuffer();
@@ -50,5 +51,4 @@
int32_t mBufferSizeInBytes = 0;
};
-
#endif //AAUDIO_AAUDIO_MIXER_H
diff --git a/services/oboeservice/AAudioServiceEndpointPlay.cpp b/services/oboeservice/AAudioServiceEndpointPlay.cpp
index 9b1833a..472a336 100644
--- a/services/oboeservice/AAudioServiceEndpointPlay.cpp
+++ b/services/oboeservice/AAudioServiceEndpointPlay.cpp
@@ -82,9 +82,13 @@
std::lock_guard <std::mutex> lock(mLockStreams);
for (const auto clientStream : mRegisteredStreams) {
int64_t clientFramesRead = 0;
+ bool allowUnderflow = true;
- if (!clientStream->isRunning()) {
- continue;
+ aaudio_stream_state_t state = clientStream->getState();
+ if (state == AAUDIO_STREAM_STATE_STOPPING) {
+ allowUnderflow = false; // just read what is already in the FIFO
+ } else if (state != AAUDIO_STREAM_STATE_STARTED) {
+ continue; // this stream is not running so skip it.
}
sp<AAudioServiceStreamShared> streamShared =
@@ -104,8 +108,7 @@
int64_t positionOffset = mmapFramesWritten - clientFramesRead;
streamShared->setTimestampPositionOffset(positionOffset);
- float volume = 1.0; // to match legacy volume
- bool underflowed = mMixer.mix(index, fifo, volume);
+ bool underflowed = mMixer.mix(index, fifo, allowUnderflow);
if (underflowed) {
streamShared->incrementXRunCount();
}
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index e670129..6246e7e 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -231,6 +231,8 @@
return AAUDIO_ERROR_INVALID_STATE;
}
+ setState(AAUDIO_STREAM_STATE_STOPPING);
+
sendCurrentTimestamp(); // warning - this calls a virtual function
result = stopTimestampThread();
if (result != AAUDIO_OK) {