New control block for AudioTrack and AudioRecord
Main differences between old and new control block:
- removes the mutex, which was a potential source of priority inversion
- circular indices into shared buffer, which is now always a power-of-2 size
Change-Id: I4e9b7fa99858b488ac98a441fa70e31dbba1b865
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index ee52fcb..0773534 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -139,7 +139,7 @@
// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or
// N-buffering, so AudioFlinger could allocate the right amount of memory.
// See the client's minBufCount and mNotificationFramesAct calculations for details.
-static const int kFastTrackMultiplier = 2;
+static const int kFastTrackMultiplier = 1;
// ----------------------------------------------------------------------------
@@ -1327,7 +1327,7 @@
// the track is newly added, make sure it fills up all its
// buffers before playing. This is to ensure the client will
// effectively get the latency it requested.
- track->mFillingUpStatus = Track::FS_FILLING;
+ track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING;
track->mResetDone = false;
track->mPresentationCompleteFrames = 0;
mActiveTracks.add(track);
@@ -2596,24 +2596,35 @@
// app does not call stop() and relies on underrun to stop:
// hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
// during last round
+ size_t desiredFrames;
+ if (t->sampleRate() == mSampleRate) {
+ desiredFrames = mNormalFrameCount;
+ } else {
+ // +1 for rounding and +1 for additional sample needed for interpolation
+ desiredFrames = (mNormalFrameCount * t->sampleRate()) / 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());
+ // the minimum track buffer size is normally twice the number of frames necessary
+ // to fill one buffer and the resampler should not leave more than one buffer worth
+ // of unreleased frames after each pass, but just in case...
+ ALOG_ASSERT(desiredFrames <= cblk->frameCount_);
+ }
uint32_t minFrames = 1;
if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
(mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
- if (t->sampleRate() == mSampleRate) {
- minFrames = mNormalFrameCount;
- } else {
- // +1 for rounding and +1 for additional sample needed for interpolation
- minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
- // add frames already consumed but not yet released by the resampler
- // because cblk->framesReady() will include these frames
- minFrames += mAudioMixer->getUnreleasedFrames(track->name());
- // the minimum track buffer size is normally twice the number of frames necessary
- // to fill one buffer and the resampler should not leave more than one buffer worth
- // of unreleased frames after each pass, but just in case...
- ALOG_ASSERT(minFrames <= cblk->frameCount_);
- }
+ minFrames = desiredFrames;
}
- if ((track->framesReady() >= minFrames) && track->isReady() &&
+ // It's not safe to call framesReady() for a static buffer track, so assume it's ready
+ size_t framesReady;
+ if (track->sharedBuffer() == 0) {
+ framesReady = track->framesReady();
+ } else if (track->isStopped()) {
+ framesReady = 0;
+ } else {
+ framesReady = 1;
+ }
+ if ((framesReady >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server,
@@ -2664,7 +2675,7 @@
// read original volumes with volume control
float typeVolume = mStreamTypes[track->streamType()].volume;
float v = masterVolume * typeVolume;
- ServerProxy *proxy = track->mServerProxy;
+ AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
uint32_t vlr = proxy->getVolumeLR();
vl = vlr & 0xFFFF;
vr = vlr >> 16;
@@ -2737,7 +2748,7 @@
AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
// limit track sample rate to 2 x output sample rate, which changes at re-configuration
uint32_t maxSampleRate = mSampleRate * 2;
- uint32_t reqSampleRate = track->mServerProxy->getSampleRate();
+ uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate();
if (reqSampleRate == 0) {
reqSampleRate = mSampleRate;
} else if (reqSampleRate > maxSampleRate) {
@@ -2768,6 +2779,13 @@
mixerStatus = MIXER_TRACKS_READY;
}
} else {
+ // only implemented for normal tracks, not fast tracks
+ if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) {
+ // we missed desiredFrames whatever the actual number of frames missing was
+ cblk->u.mStreaming.mUnderrunFrames += desiredFrames;
+ // FIXME also wake futex so that underrun is noticed more quickly
+ (void) android_atomic_or(CBLK_UNDERRUN, &cblk->flags);
+ }
// clear effect chain input buffer if an active track underruns to avoid sending
// previous audio buffer again to effects
chain = getEffectChain_l(track->sessionId());
@@ -3170,7 +3188,7 @@
} else {
float typeVolume = mStreamTypes[track->streamType()].volume;
float v = mMasterVolume * typeVolume;
- uint32_t vlr = track->mServerProxy->getVolumeLR();
+ uint32_t vlr = track->mAudioTrackServerProxy->getVolumeLR();
float v_clamped = v * (vlr & 0xFFFF);
if (v_clamped > MAX_GAIN) {
v_clamped = MAX_GAIN;
@@ -3696,7 +3714,8 @@
}
buffer.frameCount = mFrameCount;
- if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+ status_t status = mActiveTrack->getNextBuffer(&buffer);
+ if (CC_LIKELY(status == NO_ERROR)) {
readOnce = true;
size_t framesOut = buffer.frameCount;
if (mResampler == NULL) {