diff options
Diffstat (limited to 'services/audioflinger/Threads.cpp')
| -rw-r--r-- | services/audioflinger/Threads.cpp | 272 |
1 files changed, 177 insertions, 95 deletions
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index 47dcca6..498ddb6 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -135,12 +135,12 @@ static const int kPriorityFastMixer = 3; // IAudioFlinger::createTrack() reports back to client the total size of shared memory area // for the track. The client then sub-divides this into smaller buffers for its use. -// Currently the client uses double-buffering by default, but doesn't tell us about that. -// So for now we just assume that client is double-buffered. -// 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. +// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. +// So for now we just assume that client is double-buffered for fast tracks. +// FIXME It would be better for client to tell AudioFlinger the value of N, +// so AudioFlinger could allocate the right amount of memory. // See the client's minBufCount and mNotificationFramesAct calculations for details. -static const int kFastTrackMultiplier = 1; +static const int kFastTrackMultiplier = 2; // ---------------------------------------------------------------------------- @@ -272,6 +272,7 @@ AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() mParamStatus(NO_ERROR), + //FIXME: mStandby should be true here. Is this some kind of hack? mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), // mName will be set by concrete (non-virtual) subclass @@ -424,7 +425,7 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) result.append(buffer); snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); result.append(buffer); - snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); + snprintf(buffer, SIZE, "HAL frame count: %zu\n", mFrameCount); result.append(buffer); snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); result.append(buffer); @@ -432,14 +433,14 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) result.append(buffer); snprintf(buffer, SIZE, "Format: %d\n", mFormat); result.append(buffer); - snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); + snprintf(buffer, SIZE, "Frame size: %zu\n", mFrameSize); result.append(buffer); snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); result.append(buffer); result.append(" Index Command"); for (size_t i = 0; i < mNewParameters.size(); ++i) { - snprintf(buffer, SIZE, "\n %02d ", i); + snprintf(buffer, SIZE, "\n %02zu ", i); result.append(buffer); result.append(mNewParameters[i]); } @@ -465,7 +466,7 @@ void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& char buffer[SIZE]; String8 result; - snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); + snprintf(buffer, SIZE, "\n- %zu Effect Chains:\n", mEffectChains.size()); write(fd, buffer, strlen(buffer)); for (size_t i = 0; i < mEffectChains.size(); ++i) { @@ -503,17 +504,7 @@ String16 AudioFlinger::ThreadBase::getWakeLockTag() void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) { - if (mPowerManager == 0) { - // use checkService() to avoid blocking if power service is not up yet - sp<IBinder> binder = - defaultServiceManager()->checkService(String16("power")); - if (binder == 0) { - ALOGW("Thread %s cannot connect to the power manager service", mName); - } else { - mPowerManager = interface_cast<IPowerManager>(binder); - binder->linkToDeath(mDeathRecipient); - } - } + getPowerManager_l(); if (mPowerManager != 0) { sp<IBinder> binder = new BBinder(); status_t status; @@ -553,6 +544,41 @@ void AudioFlinger::ThreadBase::releaseWakeLock_l() } } +void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { + Mutex::Autolock _l(mLock); + updateWakeLockUids_l(uids); +} + +void AudioFlinger::ThreadBase::getPowerManager_l() { + + if (mPowerManager == 0) { + // use checkService() to avoid blocking if power service is not up yet + sp<IBinder> binder = + defaultServiceManager()->checkService(String16("power")); + if (binder == 0) { + ALOGW("Thread %s cannot connect to the power manager service", mName); + } else { + mPowerManager = interface_cast<IPowerManager>(binder); + binder->linkToDeath(mDeathRecipient); + } + } +} + +void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { + + getPowerManager_l(); + if (mWakeLockToken == NULL) { + ALOGE("no wake lock to update!"); + return; + } + if (mPowerManager != 0) { + sp<IBinder> binder = new BBinder(); + status_t status; + status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); + ALOGV("acquireWakeLock_l() %s status %d", mName, status); + } +} + void AudioFlinger::ThreadBase::clearPowerManager() { Mutex::Autolock _l(mLock); @@ -977,6 +1003,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), mNormalFrameCount(0), mMixBuffer(NULL), mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), + mActiveTracksGeneration(0), // mStreamTypes[] initialized in constructor body mOutput(output), mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), @@ -1101,7 +1128,7 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); result.append(buffer); - snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); + snprintf(buffer, SIZE, "Normal frame count: %zu\n", mNormalFrameCount); result.append(buffer); snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); @@ -1160,6 +1187,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac int sessionId, IAudioFlinger::track_flags_t *flags, pid_t tid, + int uid, status_t *status) { sp<Track> track; @@ -1182,7 +1210,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac ( (tid != -1) && ((frameCount == 0) || - (frameCount >= (mFrameCount * kFastTrackMultiplier))) + (frameCount >= mFrameCount)) ) ) && // PCM data @@ -1190,10 +1218,8 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac // mono or stereo ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && -#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE // hardware sample rate (sampleRate == mSampleRate) && -#endif // normal mixer has an associated fast mixer hasFastMixer() && // there are sufficient fast track slots available @@ -1293,10 +1319,10 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac if (!isTimed) { track = new Track(this, client, streamType, sampleRate, format, - channelMask, frameCount, sharedBuffer, sessionId, *flags); + channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); } else { track = TimedTrack::create(this, client, streamType, sampleRate, format, - channelMask, frameCount, sharedBuffer, sessionId); + channelMask, frameCount, sharedBuffer, sessionId, uid); } if (track == 0 || track->getCblk() == NULL || track->name() < 0) { lStatus = NO_MEMORY; @@ -1432,6 +1458,9 @@ status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) track->mResetDone = false; track->mPresentationCompleteFrames = 0; mActiveTracks.add(track); + mWakeLockUids.add(track->uid()); + mActiveTracksGeneration++; + mLatestActiveTrack = track; sp<EffectChain> chain = getEffectChain_l(track->sessionId()); if (chain != 0) { ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), @@ -1687,7 +1716,7 @@ void AudioFlinger::PlaybackThread::readOutputParameters() } -status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) +status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) { if (halFrames == NULL || dspFrames == NULL) { return BAD_VALUE; @@ -1705,7 +1734,11 @@ status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; return NO_ERROR; } else { - return mOutput->stream->get_render_position(mOutput->stream, dspFrames); + status_t status; + uint32_t frames; + status = mOutput->stream->get_render_position(mOutput->stream, &frames); + *dspFrames = (size_t)frames; + return status; } } @@ -1905,7 +1938,7 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write() mNumWrites++; mInWrite = false; - + mStandby = false; return bytesWritten; } @@ -2127,6 +2160,8 @@ bool AudioFlinger::PlaybackThread::threadLoop() // FIXME could this be made local to while loop? writeFrames = 0; + int lastGeneration = 0; + cacheParameters_l(); sleepTime = idleSleepTime; @@ -2183,6 +2218,8 @@ bool AudioFlinger::PlaybackThread::threadLoop() break; } releaseWakeLock_l(); + mWakeLockUids.clear(); + mActiveTracksGeneration++; ALOGV("wait async completion"); mWaitWorkCV.wait(mLock); ALOGV("async completion/wake"); @@ -2213,6 +2250,8 @@ bool AudioFlinger::PlaybackThread::threadLoop() } releaseWakeLock_l(); + mWakeLockUids.clear(); + mActiveTracksGeneration++; // wait until we have something to do... ALOGV("%s going to sleep", myName.string()); mWaitWorkCV.wait(mLock); @@ -2237,11 +2276,18 @@ bool AudioFlinger::PlaybackThread::threadLoop() // mMixerStatusIgnoringFastTracks is also updated internally mMixerStatus = prepareTracks_l(&tracksToRemove); + // compare with previously applied list + if (lastGeneration != mActiveTracksGeneration) { + // update wakelock + updateWakeLockUids_l(mWakeLockUids); + lastGeneration = mActiveTracksGeneration; + } + // prevent any changes in effect chain list and in each effect chain // during mixing and effect process as the audio buffers could be deleted // or modified if an effect is created or deleted lockEffectChains_l(effectChains); - } + } // mLock scope ends if (mBytesRemaining == 0) { mCurrentWriteLength = 0; @@ -2315,7 +2361,6 @@ if (mType == MIXER) { } } - mStandby = false; } else { usleep(sleepTime); } @@ -2351,6 +2396,8 @@ if (mType == MIXER) { } releaseWakeLock(); + mWakeLockUids.clear(); + mActiveTracksGeneration++; ALOGV("Thread %p type %d exiting", this, mType); return false; @@ -2364,6 +2411,8 @@ void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tra for (size_t i=0 ; i<count ; i++) { const sp<Track>& track = tracksToRemove.itemAt(i); mActiveTracks.remove(track); + mWakeLockUids.remove(track->uid()); + mActiveTracksGeneration++; ALOGV("removeTracks_l removing track on session %d", track->sessionId()); sp<EffectChain> chain = getEffectChain_l(track->sessionId()); if (chain != 0) { @@ -2926,7 +2975,6 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac VolumeProvider *vp = track; fastTrack->mBufferProvider = eabp; fastTrack->mVolumeProvider = vp; - fastTrack->mSampleRate = track->mSampleRate; fastTrack->mChannelMask = track->mChannelMask; fastTrack->mGeneration++; state->mTrackMask |= 1 << j; @@ -2989,15 +3037,8 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { minFrames = desiredFrames; } - // 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; - } + + size_t framesReady = track->framesReady(); if ((framesReady >= minFrames) && track->isReady() && !track->isPaused() && !track->isTerminated()) { @@ -3110,9 +3151,9 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac mAudioMixer->setBufferProvider(name, track); mAudioMixer->enable(name); - mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); - mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); - mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); + mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); + mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); + mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); mAudioMixer->setParameter( name, AudioMixer::TRACK, @@ -3120,7 +3161,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac mAudioMixer->setParameter( name, AudioMixer::TRACK, - AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); + AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)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->mAudioTrackServerProxy->getSampleRate(); @@ -3133,7 +3174,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac name, AudioMixer::RESAMPLE, AudioMixer::SAMPLE_RATE, - (void *)reqSampleRate); + (void *)(uintptr_t)reqSampleRate); mAudioMixer->setParameter( name, AudioMixer::TRACK, @@ -3559,6 +3600,12 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep Track* const track = t.get(); audio_track_cblk_t* cblk = track->cblk(); + // Only consider last track started for volume and mixer state control. + // In theory an older track could underrun and restart after the new one starts + // but as we only care about the transition phase between two tracks on a + // direct output, it is not a problem to ignore the underrun case. + sp<Track> l = mLatestActiveTrack.promote(); + bool last = l.get() == track; // The first time a track is added we wait // for all its buffers to be filled before processing it @@ -3568,11 +3615,6 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep } else { minFrames = 1; } - // Only consider last track started for volume and mixer state control. - // This is the last entry in mActiveTracks unless a track underruns. - // As we only care about the transition phase between two tracks on a - // direct output, it is not a problem to ignore the underrun case. - bool last = (i == (count - 1)); if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && !track->isTerminated()) @@ -3599,7 +3641,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep } else { // clear effect chain input buffer if the last active track started underruns // to avoid sending previous audio buffer again to effects - if (!mEffectChains.isEmpty() && (i == (count -1))) { + if (!mEffectChains.isEmpty() && last) { mEffectChains[0]->clearInputBuffer(); } @@ -3611,7 +3653,8 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep // TODO: implement behavior for compressed audio size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; size_t framesWritten = mBytesWritten / mFrameSize; - if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { + if (mStandby || !last || + track->presentationComplete(framesWritten, audioHALFrames)) { if (track->isStopped()) { track->reset(); } @@ -3624,6 +3667,9 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep if (--(track->mRetryCount) <= 0) { ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); tracksToRemove->add(track); + // indicate to client process that the track was disabled because of underrun; + // it will then automatically call start() when data is available + android_atomic_or(CBLK_DISABLED, &cblk->mFlags); } else if (last) { mixerStatus = MIXER_TRACKS_ENABLED; } @@ -3809,7 +3855,12 @@ bool AudioFlinger::AsyncCallbackThread::threadLoop() { Mutex::Autolock _l(mLock); - mWaitWorkCV.wait(mLock); + while (!((mWriteAckSequence & 1) || + (mDrainSequence & 1) || + exitPending())) { + mWaitWorkCV.wait(mLock); + } + if (exitPending()) { break; } @@ -3886,11 +3937,8 @@ AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, mFlushPending(false), mPausedBytesRemaining(0) { -} - -AudioFlinger::OffloadThread::~OffloadThread() -{ - mPreviousTrack.clear(); + //FIXME: mStandby should be set to true by ThreadBase constructor + mStandby = true; } void AudioFlinger::OffloadThread::threadLoop_exit() @@ -3927,24 +3975,13 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTr } Track* const track = t.get(); audio_track_cblk_t* cblk = track->cblk(); - if (mPreviousTrack != NULL) { - if (t != mPreviousTrack) { - // Flush any data still being written from last track - mBytesRemaining = 0; - if (mPausedBytesRemaining) { - // Last track was paused so we also need to flush saved - // mixbuffer state and invalidate track so that it will - // re-submit that unwritten data when it is next resumed - mPausedBytesRemaining = 0; - // Invalidate is a bit drastic - would be more efficient - // to have a flag to tell client that some of the - // previously written data was lost - mPreviousTrack->invalidate(); - } - } - } - mPreviousTrack = t; - bool last = (i == (count - 1)); + // Only consider last track started for volume and mixer state control. + // In theory an older track could underrun and restart after the new one starts + // but as we only care about the transition phase between two tracks on a + // direct output, it is not a problem to ignore the underrun case. + sp<Track> l = mLatestActiveTrack.promote(); + bool last = l.get() == track; + if (track->isPausing()) { track->setPaused(); if (last) { @@ -3992,6 +4029,31 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTr } if (last) { + sp<Track> previousTrack = mPreviousTrack.promote(); + if (previousTrack != 0) { + if (track != previousTrack.get()) { + // Flush any data still being written from last track + mBytesRemaining = 0; + if (mPausedBytesRemaining) { + // Last track was paused so we also need to flush saved + // mixbuffer state and invalidate track so that it will + // re-submit that unwritten data when it is next resumed + mPausedBytesRemaining = 0; + // Invalidate is a bit drastic - would be more efficient + // to have a flag to tell client that some of the + // previously written data was lost + previousTrack->invalidate(); + } + // flush data already sent to the DSP if changing audio session as audio + // comes from a different source. Also invalidate previous track to force a + // seek when resuming. + if (previousTrack->sessionId() != track->sessionId()) { + previousTrack->invalidate(); + mFlushPending = true; + } + } + } + mPreviousTrack = track; // reset retry count track->mRetryCount = kMaxTrackRetriesOffload; mActiveTrack = t; @@ -4008,22 +4070,27 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTr // has been written ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); track->mState = TrackBase::STOPPING_2; // so presentation completes after drain - if (last) { - sleepTime = 0; - standbyTime = systemTime() + standbyDelay; - mixerStatus = MIXER_DRAIN_TRACK; - mDrainSequence += 2; + // do not drain if no data was ever sent to HAL (mStandby == true) + if (last && !mStandby) { + // do not modify drain sequence if we are already draining. This happens + // when resuming from pause after drain. + if ((mDrainSequence & 1) == 0) { + sleepTime = 0; + standbyTime = systemTime() + standbyDelay; + mixerStatus = MIXER_DRAIN_TRACK; + mDrainSequence += 2; + } if (mHwPaused) { // It is possible to move from PAUSED to STOPPING_1 without // a resume so we must ensure hardware is running - mOutput->stream->resume(mOutput->stream); + doHwResume = true; mHwPaused = false; } } } } else if (track->isStopping_2()) { - // Drain has completed, signal presentation complete - if (!(mDrainSequence & 1) || !last) { + // Drain has completed or we are in standby, signal presentation complete + if (!(mDrainSequence & 1) || !last || mStandby) { track->mState = TrackBase::STOPPED; size_t audioHALFrames = (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; @@ -4040,6 +4107,9 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTr ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", track->name()); tracksToRemove->add(track); + // indicate to client process that the track was disabled because of underrun; + // it will then automatically call start() when data is available + android_atomic_or(CBLK_DISABLED, &cblk->mFlags); } else if (last){ mixerStatus = MIXER_TRACKS_ENABLED; } @@ -4053,7 +4123,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTr // If a flush is pending and a track is active but the HW is not paused, force a HW pause // before flush and then resume HW. This can happen in case of pause/flush/resume // if resume is received before pause is executed. - if (doHwPause || (mFlushPending && !mHwPaused && (count != 0))) { + if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { mOutput->stream->pause(mOutput->stream); if (!doHwPause) { doHwResume = true; @@ -4063,7 +4133,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTr flushHw_l(); mFlushPending = false; } - if (doHwResume) { + if (!mStandby && doHwResume) { mOutput->stream->resume(mOutput->stream); } @@ -4185,6 +4255,7 @@ ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() for (size_t i = 0; i < outputTracks.size(); i++) { outputTracks[i]->write(mMixBuffer, writeFrames); } + mStandby = false; return (ssize_t)mixBufferSize; } @@ -4216,7 +4287,8 @@ void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) mSampleRate, mFormat, mChannelMask, - frameCount); + frameCount, + IPCThreadState::self()->getCallingUid()); if (outputTrack->cblk() != NULL) { thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); mOutputTracks.add(outputTrack); @@ -4318,7 +4390,6 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, snprintf(mName, kNameLength, "AudioIn_%X", id); readInputParameters(); - mClientUid = IPCThreadState::self()->getCallingUid(); } @@ -4350,7 +4421,11 @@ bool AudioFlinger::RecordThread::threadLoop() nsecs_t lastWarning = 0; inputStandBy(); - acquireWakeLock(mClientUid); + { + Mutex::Autolock _l(mLock); + activeTrack = mActiveTrack; + acquireWakeLock_l(activeTrack != 0 ? activeTrack->uid() : -1); + } // used to verify we've read at least once before evaluating how many bytes were read bool readOnce = false; @@ -4363,6 +4438,12 @@ bool AudioFlinger::RecordThread::threadLoop() { // scope for mLock Mutex::Autolock _l(mLock); checkForNewParameters_l(); + if (mActiveTrack != 0 && activeTrack != mActiveTrack) { + SortedVector<int> tmp; + tmp.add(mActiveTrack->uid()); + updateWakeLockUids_l(tmp); + } + activeTrack = mActiveTrack; if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { standby(); @@ -4375,7 +4456,7 @@ bool AudioFlinger::RecordThread::threadLoop() // go to sleep mWaitWorkCV.wait(mLock); ALOGV("RecordThread: loop starting"); - acquireWakeLock_l(mClientUid); + acquireWakeLock_l(mActiveTrack != 0 ? mActiveTrack->uid() : -1); continue; } if (mActiveTrack != 0) { @@ -4585,6 +4666,7 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR audio_channel_mask_t channelMask, size_t frameCount, int sessionId, + int uid, IAudioFlinger::track_flags_t *flags, pid_t tid, status_t *status) @@ -4604,7 +4686,7 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR ( (tid != -1) && ((frameCount == 0) || - (frameCount >= (mFrameCount * kFastTrackMultiplier))) + (frameCount >= mFrameCount)) ) && // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) // mono or stereo @@ -4654,7 +4736,7 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR Mutex::Autolock _l(mLock); track = new RecordTrack(this, client, sampleRate, - format, channelMask, frameCount, sessionId); + format, channelMask, frameCount, sessionId, uid); if (track->getCblk() == 0) { ALOGE("createRecordTrack_l() no control block"); @@ -4876,9 +4958,9 @@ void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& a result.append(buffer); if (mActiveTrack != 0) { - snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); + snprintf(buffer, SIZE, "In index: %zu\n", mRsmpInIndex); result.append(buffer); - snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); + snprintf(buffer, SIZE, "Buffer size: %zu bytes\n", mBufferSize); result.append(buffer); snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); result.append(buffer); |