aaudio: write if there is room
Go ahead and write the data to the buffer if there is room,
even if it exceeds the setBufferSizeInFrames().
There is no point of holding it in the application buffer.
By making the data available to the DSP we can reduce glitches
or reduce latency.
Because this offers better glitch protection, we can allow
setBufferSizeInFrames() to accept a buffer size of zero.
This will provide the lowest possible latency.
Test: OboeTester glitch test
Change-Id: Ie706867c5dd57c29b4393c806cd51ae09198873e
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 2ece474..6614e5e 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -653,7 +653,7 @@
// Should we block?
if (timeoutNanoseconds == 0) {
break; // don't block
- } else if (framesLeft > 0) {
+ } else if (wakeTimeNanos != 0) {
if (!mAudioEndpoint.isFreeRunning()) {
// If there is software on the other end of the FIFO then it may get delayed.
// So wake up just a little after we expect it to be ready.
@@ -712,37 +712,38 @@
aaudio_result_t AudioStreamInternal::setBufferSize(int32_t requestedFrames) {
int32_t adjustedFrames = requestedFrames;
- int32_t actualFrames = 0;
- int32_t maximumSize = getBufferCapacity();
+ const int32_t maximumSize = getBufferCapacity() - mFramesPerBurst;
+ // The buffer size can be set to zero.
+ // This means that the callback may be called when the internal buffer becomes empty.
+ // This will be fine on some devices in ideal circumstances and will result in the
+ // lowest possible latency.
+ // If there are glitches then they should be detected as XRuns and the size can be increased.
+ static const int32_t minimumSize = 0;
// Clip to minimum size so that rounding up will work better.
- if (adjustedFrames < 1) {
- adjustedFrames = 1;
- }
+ adjustedFrames = std::max(minimumSize, adjustedFrames);
- if (adjustedFrames > maximumSize) {
- // Clip to maximum size.
+ // Prevent arithmetic overflow by clipping before we round.
+ if (adjustedFrames >= maximumSize) {
adjustedFrames = maximumSize;
} else {
// Round to the next highest burst size.
int32_t numBursts = (adjustedFrames + mFramesPerBurst - 1) / mFramesPerBurst;
adjustedFrames = numBursts * mFramesPerBurst;
- // Rounding may have gone above maximum.
- if (adjustedFrames > maximumSize) {
- adjustedFrames = maximumSize;
- }
}
- aaudio_result_t result = mAudioEndpoint.setBufferSizeInFrames(adjustedFrames, &actualFrames);
- if (result < 0) {
- return result;
- } else {
- return (aaudio_result_t) actualFrames;
- }
+ // Clip against the actual size from the endpoint.
+ int32_t actualFrames = 0;
+ mAudioEndpoint.setBufferSizeInFrames(maximumSize, &actualFrames);
+ // actualFrames should be <= maximumSize
+ adjustedFrames = std::min(actualFrames, adjustedFrames);
+
+ mBufferSizeInFrames = adjustedFrames;
+ return (aaudio_result_t) adjustedFrames;
}
int32_t AudioStreamInternal::getBufferSize() const {
- return mAudioEndpoint.getBufferSizeInFrames();
+ return mBufferSizeInFrames;
}
int32_t AudioStreamInternal::getBufferCapacity() const {