AudioTrack: get/setStartThresholdInFrames
setStartThresholdInFrames is used to set the start threshold
in frames for streaming AudioTrack playback.
Normally this is the entire buffer capacity in frames
but may be reduced for low latency playback,
compressed formats, and direct tracks.
Test: atest AudioTrackTest#testStartThresholdInFrames
Test: atest AudioTrackTest#testStartThresholdInFramesExceptions
Bug: 183003720
Change-Id: Iae10764130160de2380477134df27c5b139468bc
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 8788a86..6c9e85c 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -1283,6 +1283,46 @@
return finalBufferSize;
}
+ssize_t AudioTrack::getStartThresholdInFrames() const
+{
+ AutoMutex lock(mLock);
+ if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
+ return NO_INIT;
+ }
+ return (ssize_t) mProxy->getStartThresholdInFrames();
+}
+
+ssize_t AudioTrack::setStartThresholdInFrames(size_t startThresholdInFrames)
+{
+ if (startThresholdInFrames > INT32_MAX || startThresholdInFrames == 0) {
+ // contractually we could simply return the current threshold in frames
+ // to indicate the request was ignored, but we return an error here.
+ return BAD_VALUE;
+ }
+ AutoMutex lock(mLock);
+ // We do not permit calling setStartThresholdInFrames() between the AudioTrack
+ // default ctor AudioTrack() and set(...) but rather fail such an attempt.
+ // (To do so would require a cached mOrigStartThresholdInFrames and we may
+ // not have proper validation for the actual set value).
+ if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
+ return NO_INIT;
+ }
+ const uint32_t original = mProxy->getStartThresholdInFrames();
+ const uint32_t final = mProxy->setStartThresholdInFrames(startThresholdInFrames);
+ if (original != final) {
+ android::mediametrics::LogItem(mMetricsId)
+ .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETSTARTTHRESHOLD)
+ .set(AMEDIAMETRICS_PROP_STARTTHRESHOLDFRAMES, (int32_t)final)
+ .record();
+ if (original > final) {
+ // restart track if it was disabled by audioflinger due to previous underrun
+ // and we reduced the number of frames for the threshold.
+ restartIfDisabled();
+ }
+ }
+ return final;
+}
+
status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
{
if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
@@ -2602,6 +2642,10 @@
staticPosition = mStaticProxy->getPosition().unsignedValue();
}
+ // save the old startThreshold and framecount
+ const uint32_t originalStartThresholdInFrames = mProxy->getStartThresholdInFrames();
+ const uint32_t originalFrameCount = mProxy->frameCount();
+
// See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
// causes a lot of churn on the service side, and it can reject starting
// playback of a previously created track. May also apply to other cases.
@@ -2662,6 +2706,18 @@
return status;
});
+ // restore the original start threshold if different than frameCount.
+ if (originalStartThresholdInFrames != originalFrameCount) {
+ // Note: mProxy->setStartThresholdInFrames() call is in the Proxy
+ // and does not trigger a restart.
+ // (Also CBLK_DISABLED is not set, buffers are empty after track recreation).
+ // Any start would be triggered on the mState == ACTIVE check below.
+ const uint32_t currentThreshold =
+ mProxy->setStartThresholdInFrames(originalStartThresholdInFrames);
+ ALOGD_IF(originalStartThresholdInFrames != currentThreshold,
+ "%s(%d) startThresholdInFrames changing from %u to %u",
+ __func__, mPortId, originalStartThresholdInFrames, currentThreshold);
+ }
if (mState == STATE_ACTIVE) {
mAudioTrack->start(&result);
}