Fix theoretical race using TrackBase::sampleRate()
In two places we assumed that TrackBase::sampleRate() would return the
same value when it is called twice in the same function. This is not
guaranteed; sampleRate() reads from the control block so the return
value could change. To fix this, only call sampleRate() once and cache
the return value to get a consistent value.
This was only a theoretical race. In MixerThread::prepareTracks_l()
it would have no bad effect. In TimedTrack::getNextBuffer() it could
cause a real problem, but we don't currently support dynamic sample rate
ratios for timed tracks.
Change-Id: I8e5c33f0121fc058d1e70c2ab5e9135397d3e0b7
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 0773534..3b5727b 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -2427,7 +2427,7 @@
}
for (size_t i=0 ; i<count ; i++) {
- sp<Track> t = mActiveTracks[i].promote();
+ const sp<Track> t = mActiveTracks[i].promote();
if (t == 0) {
continue;
}
@@ -2597,11 +2597,12 @@
// hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
// during last round
size_t desiredFrames;
- if (t->sampleRate() == mSampleRate) {
+ uint32_t sr = track->sampleRate();
+ if (sr == mSampleRate) {
desiredFrames = mNormalFrameCount;
} else {
// +1 for rounding and +1 for additional sample needed for interpolation
- desiredFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
+ desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1;
// add frames already consumed but not yet released by the resampler
// because cblk->framesReady() will include these frames
desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index bfc197c..6aca95f 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1130,10 +1130,12 @@
}
}
+ uint32_t sr = sampleRate();
+
// adjust the head buffer's PTS to reflect the portion of the head buffer
// that has already been consumed
int64_t effectivePTS = headLocalPTS +
- ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate());
+ ((head.position() / mFrameSize) * mLocalTimeFreq / sr);
// Calculate the delta in samples between the head of the input buffer
// queue and the start of the next output buffer that will be written.
@@ -1165,7 +1167,7 @@
// the current output position is within this threshold, then we will
// concatenate the next input samples to the previous output
const int64_t kSampleContinuityThreshold =
- (static_cast<int64_t>(sampleRate()) << 32) / 250;
+ (static_cast<int64_t>(sr) << 32) / 250;
// if this is the first buffer of audio that we're emitting from this track
// then it should be almost exactly on time.