diff options
Diffstat (limited to 'services/audioflinger')
| -rw-r--r-- | services/audioflinger/AudioFlinger.cpp | 15 | ||||
| -rw-r--r-- | services/audioflinger/AudioFlinger.h | 5 | ||||
| -rw-r--r-- | services/audioflinger/AudioMixer.cpp | 18 | ||||
| -rw-r--r-- | services/audioflinger/AudioPolicyService.cpp | 8 | ||||
| -rw-r--r-- | services/audioflinger/AudioResampler.cpp | 4 | ||||
| -rw-r--r-- | services/audioflinger/Configuration.h | 3 | ||||
| -rw-r--r-- | services/audioflinger/Effects.cpp | 24 | ||||
| -rw-r--r-- | services/audioflinger/FastMixer.cpp | 39 | ||||
| -rw-r--r-- | services/audioflinger/FastMixerState.cpp | 2 | ||||
| -rw-r--r-- | services/audioflinger/FastMixerState.h | 1 | ||||
| -rw-r--r-- | services/audioflinger/PlaybackTracks.h | 10 | ||||
| -rw-r--r-- | services/audioflinger/RecordTracks.h | 3 | ||||
| -rw-r--r-- | services/audioflinger/StateQueue.cpp | 8 | ||||
| -rw-r--r-- | services/audioflinger/Threads.cpp | 272 | ||||
| -rw-r--r-- | services/audioflinger/Threads.h | 15 | ||||
| -rw-r--r-- | services/audioflinger/TrackBase.h | 3 | ||||
| -rw-r--r-- | services/audioflinger/Tracks.cpp | 56 |
17 files changed, 308 insertions, 178 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index a9c9b56..e9c38e3 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -448,6 +448,7 @@ sp<IAudioTrack> AudioFlinger::createTrack( pid_t tid, int *sessionId, String8& name, + int clientUid, status_t *status) { sp<PlaybackThread::Track> track; @@ -483,6 +484,7 @@ sp<IAudioTrack> AudioFlinger::createTrack( } pid_t pid = IPCThreadState::self()->getCallingPid(); + client = registerPid_l(pid); ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); @@ -510,7 +512,7 @@ sp<IAudioTrack> AudioFlinger::createTrack( ALOGV("createTrack() lSessionId: %d", lSessionId); track = thread->createTrack_l(client, streamType, sampleRate, format, - channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); + channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); // move effect chain to this output thread if an effect on same session was waiting // for a track to be created @@ -1008,7 +1010,7 @@ size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t form return size; } -unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const +uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const { Mutex::Autolock _l(mLock); @@ -1040,7 +1042,7 @@ status_t AudioFlinger::setVoiceVolume(float value) return ret; } -status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames, +status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, audio_io_handle_t output) const { status_t status; @@ -1284,8 +1286,11 @@ sp<IAudioRecord> AudioFlinger::openRecord( } // create new record track. // The record track uses one track in mHardwareMixerThread by convention. + // TODO: the uid should be passed in as a parameter to openRecord recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, - frameCount, lSessionId, flags, tid, &lStatus); + frameCount, lSessionId, + IPCThreadState::self()->getCallingUid(), + flags, tid, &lStatus); LOG_ALWAYS_FATAL_IF((recordTrack != 0) != (lStatus == NO_ERROR)); } if (lStatus != NO_ERROR) { @@ -2335,6 +2340,7 @@ status_t AudioFlinger::moveEffectChain_l(int sessionId, strategy, sessionId, effect->id()); + AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); } effect = chain->getEffectFromId_l(0); } @@ -2349,6 +2355,7 @@ status_t AudioFlinger::moveEffectChain_l(int sessionId, strategy, sessionId, removed[i]->id()); + AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); } } } diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h index 2aeb263..7320144 100644 --- a/services/audioflinger/AudioFlinger.h +++ b/services/audioflinger/AudioFlinger.h @@ -109,6 +109,7 @@ public: pid_t tid, int *sessionId, String8& name, + int clientUid, status_t *status); virtual sp<IAudioRecord> openRecord( @@ -185,10 +186,10 @@ public: virtual status_t setVoiceVolume(float volume); - virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames, + virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, audio_io_handle_t output) const; - virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; + virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; virtual int newAudioSessionId(); diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp index df4e029..f92421e 100644 --- a/services/audioflinger/AudioMixer.cpp +++ b/services/audioflinger/AudioMixer.cpp @@ -421,15 +421,16 @@ void AudioMixer::setParameter(int name, int target, int param, void *value) ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name); track_t& track = mState.tracks[name]; - int valueInt = (int)value; - int32_t *valueBuf = (int32_t *)value; + int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value)); + int32_t *valueBuf = reinterpret_cast<int32_t*>(value); switch (target) { case TRACK: switch (param) { case CHANNEL_MASK: { - audio_channel_mask_t mask = (audio_channel_mask_t) value; + audio_channel_mask_t mask = + static_cast<audio_channel_mask_t>(reinterpret_cast<uintptr_t>(value)); if (track.channelMask != mask) { uint32_t channelCount = popcount(mask); ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount); @@ -1122,10 +1123,6 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) t.bufferProvider->getNextBuffer(&t.buffer, pts); t.frameCount = t.buffer.frameCount; t.in = t.buffer.raw; - // t.in == NULL can happen if the track was flushed just after having - // been enabled for mixing. - if (t.in == NULL) - enabledTracks &= ~(1<<i); } e0 = enabledTracks; @@ -1161,6 +1158,13 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) aux = t.auxBuffer + numFrames; } while (outFrames) { + // t.in == NULL can happen if the track was flushed just after having + // been enabled for mixing. + if (t.in == NULL) { + enabledTracks &= ~(1<<i); + e1 &= ~(1<<i); + break; + } size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount; if (inFrames) { t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp index 35e816b..646a317 100644 --- a/services/audioflinger/AudioPolicyService.cpp +++ b/services/audioflinger/AudioPolicyService.cpp @@ -1433,6 +1433,14 @@ status_t AudioPolicyService::loadPreProcessorConfig(const char *path) loadEffects(root, effects); loadInputSources(root, effects); + // delete effects to fix memory leak. + // as effects is local var and valgrind would treat this as memory leak + // and although it only did in mediaserver init, but free it in case mediaserver reboot + size_t i; + for (i = 0; i < effects.size(); i++) { + delete effects[i]; + } + config_free(root); free(root); free(data); diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp index 2c3c719..e5cceb1 100644 --- a/services/audioflinger/AudioResampler.cpp +++ b/services/audioflinger/AudioResampler.cpp @@ -526,7 +526,7 @@ void AudioResamplerOrder1::AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t " ldr r8, [sp, #" MO_PARAM5 " + 4]\n" // out " ldr r0, [sp, #" MO_PARAM5 " + 0]\n" // &outputIndex " ldr r0, [r0]\n" // outputIndex - " add r8, r0, asl #2\n" // curOut + " add r8, r8, r0, asl #2\n" // curOut " ldr r9, [sp, #" MO_PARAM5 " + 24]\n" // phaseIncrement " ldr r10, [sp, #" MO_PARAM5 " + 12]\n" // vl " ldr r11, [sp, #" MO_PARAM5 " + 16]\n" // vr @@ -636,7 +636,7 @@ void AudioResamplerOrder1::AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32 " ldr r8, [sp, #" ST_PARAM5 " + 4]\n" // out " ldr r0, [sp, #" ST_PARAM5 " + 0]\n" // &outputIndex " ldr r0, [r0]\n" // outputIndex - " add r8, r0, asl #2\n" // curOut + " add r8, r8, r0, asl #2\n" // curOut " ldr r9, [sp, #" ST_PARAM5 " + 24]\n" // phaseIncrement " ldr r10, [sp, #" ST_PARAM5 " + 12]\n" // vl " ldr r11, [sp, #" ST_PARAM5 " + 16]\n" // vr diff --git a/services/audioflinger/Configuration.h b/services/audioflinger/Configuration.h index bc2038a..0754d9d 100644 --- a/services/audioflinger/Configuration.h +++ b/services/audioflinger/Configuration.h @@ -32,9 +32,6 @@ // uncomment to enable fast mixer to take performance samples for later statistical analysis #define FAST_MIXER_STATISTICS -// uncomment to allow fast tracks at non-native sample rate -//#define FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE - // uncomment for debugging timing problems related to StateQueue::push() //#define STATE_QUEUE_DUMP diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp index a8a5169..010e233 100644 --- a/services/audioflinger/Effects.cpp +++ b/services/audioflinger/Effects.cpp @@ -820,8 +820,8 @@ void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) } result.append("\t\tSession Status State Engine:\n"); - snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", - mSessionId, mStatus, mState, (uint32_t)mEffectInterface); + snprintf(buffer, SIZE, "\t\t%05d %03d %03d %p\n", + mSessionId, mStatus, mState, mEffectInterface); result.append(buffer); result.append("\t\tDescriptor:\n"); @@ -850,26 +850,26 @@ void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) result.append(buffer); result.append("\t\t- Input configuration:\n"); - result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); - snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", - (uint32_t)mConfig.inputCfg.buffer.raw, + result.append("\t\t\tFrames Smp rate Channels Format Buffer\n"); + snprintf(buffer, SIZE, "\t\t\t%05zu %05d %08x %6d %p\n", mConfig.inputCfg.buffer.frameCount, mConfig.inputCfg.samplingRate, mConfig.inputCfg.channels, - mConfig.inputCfg.format); + mConfig.inputCfg.format, + mConfig.inputCfg.buffer.raw); result.append(buffer); result.append("\t\t- Output configuration:\n"); result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); - snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", - (uint32_t)mConfig.outputCfg.buffer.raw, + snprintf(buffer, SIZE, "\t\t\t%p %05zu %05d %08x %d\n", + mConfig.outputCfg.buffer.raw, mConfig.outputCfg.buffer.frameCount, mConfig.outputCfg.samplingRate, mConfig.outputCfg.channels, mConfig.outputCfg.format); result.append(buffer); - snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); + snprintf(buffer, SIZE, "\t\t%zu Clients:\n", mHandles.size()); result.append(buffer); result.append("\t\t\tPid Priority Ctrl Locked client server\n"); for (size_t i = 0; i < mHandles.size(); ++i) { @@ -1578,10 +1578,10 @@ void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) } result.append("\tNum fx In buffer Out buffer Active tracks:\n"); - snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", + snprintf(buffer, SIZE, "\t%02zu %p %p %d\n", mEffects.size(), - (uint32_t)mInBuffer, - (uint32_t)mOutBuffer, + mInBuffer, + mOutBuffer, mActiveTrackCnt); result.append(buffer); write(fd, result.string(), result.size()); diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp index f27ea17..85d637e 100644 --- a/services/audioflinger/FastMixer.cpp +++ b/services/audioflinger/FastMixer.cpp @@ -236,7 +236,6 @@ bool FastMixer::threadLoop() sampleRate = Format_sampleRate(format); ALOG_ASSERT(Format_channelCount(format) == FCC_2); } - dumpState->mSampleRate = sampleRate; } if ((format != previousFormat) || (frameCount != previous->mFrameCount)) { @@ -321,12 +320,8 @@ bool FastMixer::threadLoop() mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER, (void *) mixBuffer); // newly allocated track names default to full scale volume - if (fastTrack->mSampleRate != 0 && fastTrack->mSampleRate != sampleRate) { - mixer->setParameter(name, AudioMixer::RESAMPLE, - AudioMixer::SAMPLE_RATE, (void*) fastTrack->mSampleRate); - } mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK, - (void *) fastTrack->mChannelMask); + (void *)(uintptr_t)fastTrack->mChannelMask); mixer->enable(name); } generations[i] = fastTrack->mGeneration; @@ -353,16 +348,10 @@ bool FastMixer::threadLoop() mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, (void *)0x1000); } - if (fastTrack->mSampleRate != 0 && - fastTrack->mSampleRate != sampleRate) { - mixer->setParameter(name, AudioMixer::RESAMPLE, - AudioMixer::SAMPLE_RATE, (void*) fastTrack->mSampleRate); - } else { - mixer->setParameter(name, AudioMixer::RESAMPLE, - AudioMixer::REMOVE, NULL); - } + mixer->setParameter(name, AudioMixer::RESAMPLE, + AudioMixer::REMOVE, NULL); mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK, - (void *) fastTrack->mChannelMask); + (void *)(uintptr_t) fastTrack->mChannelMask); // already enabled } generations[i] = fastTrack->mGeneration; @@ -392,16 +381,8 @@ bool FastMixer::threadLoop() // Refresh the per-track timestamp if (timestampStatus == NO_ERROR) { - uint32_t trackFramesWrittenButNotPresented; - uint32_t trackSampleRate = fastTrack->mSampleRate; - // There is currently no sample rate conversion for fast tracks currently - if (trackSampleRate != 0 && trackSampleRate != sampleRate) { - trackFramesWrittenButNotPresented = - ((int64_t) nativeFramesWrittenButNotPresented * trackSampleRate) / - sampleRate; - } else { - trackFramesWrittenButNotPresented = nativeFramesWrittenButNotPresented; - } + uint32_t trackFramesWrittenButNotPresented = + nativeFramesWrittenButNotPresented; uint32_t trackFramesWritten = fastTrack->mBufferProvider->framesReleased(); // Can't provide an AudioTimestamp before first frame presented, // or during the brief 32-bit wraparound window @@ -419,9 +400,9 @@ bool FastMixer::threadLoop() if (fastTrack->mVolumeProvider != NULL) { uint32_t vlr = fastTrack->mVolumeProvider->getVolumeLR(); mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, - (void *)(vlr & 0xFFFF)); + (void *)(uintptr_t)(vlr & 0xFFFF)); mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, - (void *)(vlr >> 16)); + (void *)(uintptr_t)(vlr >> 16)); } // FIXME The current implementation of framesReady() for fast tracks // takes a tryLock, which can block @@ -750,7 +731,7 @@ void FastMixerDumpState::dump(int fd) const double mixPeriodSec = (double) mFrameCount / (double) mSampleRate; fdprintf(fd, "FastMixer command=%s writeSequence=%u framesWritten=%u\n" " numTracks=%u writeErrors=%u underruns=%u overruns=%u\n" - " sampleRate=%u frameCount=%u measuredWarmup=%.3g ms, warmupCycles=%u\n" + " sampleRate=%u frameCount=%zu measuredWarmup=%.3g ms, warmupCycles=%u\n" " mixPeriod=%.2f ms\n", string, mWriteSequence, mFramesWritten, mNumTracks, mWriteErrors, mUnderruns, mOverruns, @@ -864,7 +845,7 @@ void FastMixerDumpState::dump(int fd) const mostRecent = "?"; break; } - fdprintf(fd, "%5u %6s %4u %7u %5u %7s %5u\n", i, isActive ? "yes" : "no", + fdprintf(fd, "%5u %6s %4u %7u %5u %7s %5zu\n", i, isActive ? "yes" : "no", (underruns.mBitFields.mFull) & UNDERRUN_MASK, (underruns.mBitFields.mPartial) & UNDERRUN_MASK, (underruns.mBitFields.mEmpty) & UNDERRUN_MASK, diff --git a/services/audioflinger/FastMixerState.cpp b/services/audioflinger/FastMixerState.cpp index 737de97..43ff233 100644 --- a/services/audioflinger/FastMixerState.cpp +++ b/services/audioflinger/FastMixerState.cpp @@ -20,7 +20,7 @@ namespace android { FastTrack::FastTrack() : - mBufferProvider(NULL), mVolumeProvider(NULL), mSampleRate(0), + mBufferProvider(NULL), mVolumeProvider(NULL), mChannelMask(AUDIO_CHANNEL_OUT_STEREO), mGeneration(0) { } diff --git a/services/audioflinger/FastMixerState.h b/services/audioflinger/FastMixerState.h index f6e7903..9739fe9 100644 --- a/services/audioflinger/FastMixerState.h +++ b/services/audioflinger/FastMixerState.h @@ -43,7 +43,6 @@ struct FastTrack { ExtendedAudioBufferProvider* mBufferProvider; // must be NULL if inactive, or non-NULL if active VolumeProvider* mVolumeProvider; // optional; if NULL then full-scale - unsigned mSampleRate; // optional; if zero then use mixer sample rate audio_channel_mask_t mChannelMask; // AUDIO_CHANNEL_OUT_MONO or AUDIO_CHANNEL_OUT_STEREO int mGeneration; // increment when any field is assigned }; diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h index a2e2511..43b77f3 100644 --- a/services/audioflinger/PlaybackTracks.h +++ b/services/audioflinger/PlaybackTracks.h @@ -31,6 +31,7 @@ public: size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId, + int uid, IAudioFlinger::track_flags_t flags); virtual ~Track(); @@ -165,7 +166,8 @@ class TimedTrack : public Track { audio_channel_mask_t channelMask, size_t frameCount, const sp<IMemory>& sharedBuffer, - int sessionId); + int sessionId, + int uid); virtual ~TimedTrack(); class TimedBuffer { @@ -208,7 +210,8 @@ class TimedTrack : public Track { audio_channel_mask_t channelMask, size_t frameCount, const sp<IMemory>& sharedBuffer, - int sessionId); + int sessionId, + int uid); void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); void timedYieldSilence_l(uint32_t numFrames, @@ -255,7 +258,8 @@ public: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - size_t frameCount); + size_t frameCount, + int uid); virtual ~OutputTrack(); virtual status_t start(AudioSystem::sync_event_t event = diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h index cd8f70c..57de568 100644 --- a/services/audioflinger/RecordTracks.h +++ b/services/audioflinger/RecordTracks.h @@ -28,7 +28,8 @@ public: audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, - int sessionId); + int sessionId, + int uid); virtual ~RecordTrack(); virtual status_t start(AudioSystem::sync_event_t event, int triggerSession); diff --git a/services/audioflinger/StateQueue.cpp b/services/audioflinger/StateQueue.cpp index c2d3bbd..48399c0 100644 --- a/services/audioflinger/StateQueue.cpp +++ b/services/audioflinger/StateQueue.cpp @@ -58,7 +58,11 @@ template<typename T> StateQueue<T>::~StateQueue() template<typename T> const T* StateQueue<T>::poll() { +#ifdef __LP64__ + const T *next = (const T *) android_atomic_acquire_load64((volatile int64_t *) &mNext); +#else const T *next = (const T *) android_atomic_acquire_load((volatile int32_t *) &mNext); +#endif if (next != mCurrent) { mAck = next; // no additional barrier needed mCurrent = next; @@ -140,7 +144,11 @@ template<typename T> bool StateQueue<T>::push(StateQueue<T>::block_t block) } // publish +#ifdef __LP64__ + android_atomic_release_store64((int64_t) mMutating, (volatile int64_t *) &mNext); +#else android_atomic_release_store((int32_t) mMutating, (volatile int32_t *) &mNext); +#endif mExpecting = mMutating; // copy with circular wraparound 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); diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h index 802b784..a2fb874 100644 --- a/services/audioflinger/Threads.h +++ b/services/audioflinger/Threads.h @@ -241,6 +241,9 @@ protected: void acquireWakeLock_l(int uid = -1); void releaseWakeLock(); void releaseWakeLock_l(); + void updateWakeLockUids(const SortedVector<int> &uids); + void updateWakeLockUids_l(const SortedVector<int> &uids); + void getPowerManager_l(); void setEffectSuspended_l(const effect_uuid_t *type, bool suspend, int sessionId); @@ -421,6 +424,7 @@ public: int sessionId, IAudioFlinger::track_flags_t *flags, pid_t tid, + int uid, status_t *status); AudioStreamOut* getOutput() const; @@ -442,7 +446,7 @@ public: virtual String8 getParameters(const String8& keys); virtual void audioConfigChanged_l(int event, int param = 0); - status_t getRenderPosition(size_t *halFrames, size_t *dspFrames); + status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); int16_t *mixBuffer() const { return mMixBuffer; }; virtual void detachAuxEffect_l(int effectId); @@ -495,6 +499,9 @@ private: void setMasterMute_l(bool muted) { mMasterMute = muted; } protected: SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> + SortedVector<int> mWakeLockUids; + int mActiveTracksGeneration; + wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks // Allocate a track name for a given channel mask. // Returns name >= 0 if successful, -1 on failure. @@ -735,7 +742,7 @@ public: OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, audio_io_handle_t id, uint32_t device); - virtual ~OffloadThread(); + virtual ~OffloadThread() {}; protected: // threadLoop snippets @@ -755,7 +762,7 @@ private: bool mFlushPending; size_t mPausedWriteLength; // length in bytes of write interrupted by pause size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume - sp<Track> mPreviousTrack; // used to detect track switch + wp<Track> mPreviousTrack; // used to detect track switch }; class AsyncCallbackThread : public Thread { @@ -873,6 +880,7 @@ public: audio_channel_mask_t channelMask, size_t frameCount, int sessionId, + int uid, IAudioFlinger::track_flags_t *flags, pid_t tid, status_t *status); @@ -953,5 +961,4 @@ private: // For dumpsys const sp<NBAIO_Sink> mTeeSink; - int mClientUid; }; diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h index 523e4b2..cd201d9 100644 --- a/services/audioflinger/TrackBase.h +++ b/services/audioflinger/TrackBase.h @@ -45,6 +45,7 @@ public: size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId, + int uid, bool isOut); virtual ~TrackBase(); @@ -54,6 +55,7 @@ public: sp<IMemory> getCblk() const { return mCblkMemory; } audio_track_cblk_t* cblk() const { return mCblk; } int sessionId() const { return mSessionId; } + int uid() const { return mUid; } virtual status_t setSyncEvent(const sp<SyncEvent>& event); protected: @@ -132,6 +134,7 @@ protected: // openRecord(), and then adjusted as needed const int mSessionId; + int mUid; Vector < sp<SyncEvent> >mSyncEvents; const bool mIsOut; ServerProxy* mServerProxy; diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp index 9c6e724..fccc7b8 100644 --- a/services/audioflinger/Tracks.cpp +++ b/services/audioflinger/Tracks.cpp @@ -68,6 +68,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase( size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId, + int clientUid, bool isOut) : RefBase(), mThread(thread), @@ -88,6 +89,18 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase( mId(android_atomic_inc(&nextTrackId)), mTerminated(false) { + // if the caller is us, trust the specified uid + if (IPCThreadState::self()->getCallingPid() != getpid_cached || clientUid == -1) { + int newclientUid = IPCThreadState::self()->getCallingUid(); + if (clientUid != -1 && clientUid != newclientUid) { + ALOGW("uid %d tried to pass itself off as %d", newclientUid, clientUid); + } + clientUid = newclientUid; + } + // clientUid contains the uid of the app that is responsible for this track, so we can blame + // battery usage on it. + mUid = clientUid; + // client == 0 implies sharedBuffer == 0 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); @@ -313,9 +326,10 @@ AudioFlinger::PlaybackThread::Track::Track( size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId, + int uid, IAudioFlinger::track_flags_t flags) : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, - sessionId, true /*isOut*/), + sessionId, uid, true /*isOut*/), mFillingUpStatus(FS_INVALID), // mRetryCount initialized later when needed mSharedBuffer(sharedBuffer), @@ -473,8 +487,8 @@ void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) nowInUnderrun = '?'; break; } - snprintf(&buffer[7], size-7, " %6u %4u %08X %08X %7u %6u %1c %1d %5u %5.2g %5.2g " - "%08X %08X %08X 0x%03X %9u%c\n", + snprintf(&buffer[7], size-7, " %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g " + "%08X %p %p 0x%03X %9u%c\n", (mClient == 0) ? getpid_cached : mClient->pid(), mStreamType, mFormat, @@ -487,8 +501,8 @@ void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 20.0 * log10((vlr & 0xFFFF) / 4096.0), 20.0 * log10((vlr >> 16) / 4096.0), mCblk->mServer, - (int)mMainBuffer, - (int)mAuxBuffer, + mMainBuffer, + mAuxBuffer, mCblk->mFlags, mAudioTrackServerProxy->getUnderrunFrames(), nowInUnderrun); @@ -600,6 +614,15 @@ status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t ev // track was already in the active list, not a problem if (status == ALREADY_EXISTS) { status = NO_ERROR; + } else { + // Acknowledge any pending flush(), so that subsequent new data isn't discarded. + // It is usually unsafe to access the server proxy from a binder thread. + // But in this case we know the mixer thread (whether normal mixer or fast mixer) + // isn't looking at this track yet: we still hold the normal mixer thread lock, + // and for fast tracks the track is not yet in the fast mixer thread's active set. + ServerProxy::Buffer buffer; + buffer.mFrameCount = 1; + (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/); } } else { status = BAD_VALUE; @@ -829,6 +852,7 @@ status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) dstChain->strategy(), AUDIO_SESSION_OUTPUT_MIX, effect->id()); + AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); } status = playbackThread->attachAuxEffect(this, EffectId); } @@ -954,13 +978,14 @@ AudioFlinger::PlaybackThread::TimedTrack::create( audio_channel_mask_t channelMask, size_t frameCount, const sp<IMemory>& sharedBuffer, - int sessionId) { + int sessionId, + int uid) { if (!client->reserveTimedTrack()) return 0; return new TimedTrack( thread, client, streamType, sampleRate, format, channelMask, frameCount, - sharedBuffer, sessionId); + sharedBuffer, sessionId, uid); } AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( @@ -972,9 +997,10 @@ AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( audio_channel_mask_t channelMask, size_t frameCount, const sp<IMemory>& sharedBuffer, - int sessionId) + int sessionId, + int uid) : Track(thread, client, streamType, sampleRate, format, channelMask, - frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), + frameCount, sharedBuffer, sessionId, uid, IAudioFlinger::TRACK_TIMED), mQueueHeadInFlight(false), mTrimQueueHeadOnRelease(false), mFramesPendingInQueue(0), @@ -1467,9 +1493,10 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - size_t frameCount) + size_t frameCount, + int uid) : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, - NULL, 0, IAudioFlinger::TRACK_DEFAULT), + NULL, 0, uid, IAudioFlinger::TRACK_DEFAULT), mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) { @@ -1729,9 +1756,10 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack( audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, - int sessionId) + int sessionId, + int uid) : TrackBase(thread, client, sampleRate, format, - channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/), + channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, uid, false /*isOut*/), mOverflow(false) { ALOGV("RecordTrack constructor"); @@ -1822,7 +1850,7 @@ void AudioFlinger::RecordThread::RecordTrack::invalidate() void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) { - snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6u\n", + snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6zu\n", (mClient == 0) ? getpid_cached : mClient->pid(), mFormat, mChannelMask, |