diff options
Diffstat (limited to 'services/audioflinger/Threads.cpp')
-rw-r--r-- | services/audioflinger/Threads.cpp | 495 |
1 files changed, 271 insertions, 224 deletions
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index 4234965..6873c7e 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -269,8 +269,8 @@ AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio : Thread(false /*canCallJava*/), mType(type), mAudioFlinger(audioFlinger), - // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are - // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() + // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize + // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() mParamStatus(NO_ERROR), mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), @@ -296,6 +296,17 @@ AudioFlinger::ThreadBase::~ThreadBase() } } +status_t AudioFlinger::ThreadBase::readyToRun() +{ + status_t status = initCheck(); + if (status == NO_ERROR) { + ALOGI("AudioFlinger's thread %p ready to run", this); + } else { + ALOGE("No working audio driver found."); + } + return status; +} + void AudioFlinger::ThreadBase::exit() { ALOGV("ThreadBase::exit"); @@ -368,7 +379,13 @@ void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32 void AudioFlinger::ThreadBase::processConfigEvents() { - mLock.lock(); + Mutex::Autolock _l(mLock); + processConfigEvents_l(); +} + +// post condition: mConfigEvents.isEmpty() +void AudioFlinger::ThreadBase::processConfigEvents_l() +{ while (!mConfigEvents.isEmpty()) { ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); ConfigEvent *event = mConfigEvents[0]; @@ -376,32 +393,31 @@ void AudioFlinger::ThreadBase::processConfigEvents() // release mLock before locking AudioFlinger mLock: lock order is always // AudioFlinger then ThreadBase to avoid cross deadlock mLock.unlock(); - switch(event->type()) { - case CFG_EVENT_PRIO: { - PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); - // FIXME Need to understand why this has be done asynchronously - int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), - true /*asynchronous*/); - if (err != 0) { - ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " - "error %d", - prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); - } - } break; - case CFG_EVENT_IO: { - IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); - mAudioFlinger->mLock.lock(); + switch (event->type()) { + case CFG_EVENT_PRIO: { + PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); + // FIXME Need to understand why this has be done asynchronously + int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), + true /*asynchronous*/); + if (err != 0) { + ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", + prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); + } + } break; + case CFG_EVENT_IO: { + IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); + { + Mutex::Autolock _l(mAudioFlinger->mLock); audioConfigChanged_l(ioEvent->event(), ioEvent->param()); - mAudioFlinger->mLock.unlock(); - } break; - default: - ALOGE("processConfigEvents() unknown event type %d", event->type()); - break; + } + } break; + default: + ALOGE("processConfigEvents() unknown event type %d", event->type()); + break; } delete event; mLock.lock(); } - mLock.unlock(); } void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) @@ -426,6 +442,8 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) result.append(buffer); snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); result.append(buffer); + snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); + result.append(buffer); snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); result.append(buffer); snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); @@ -685,8 +703,7 @@ sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( int sessionId, effect_descriptor_t *desc, int *enabled, - status_t *status - ) + status_t *status) { sp<EffectModule> effect; sp<EffectHandle> handle; @@ -786,9 +803,7 @@ Exit: handle.clear(); } - if (status != NULL) { - *status = lStatus; - } + *status = lStatus; return handle; } @@ -932,7 +947,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge type_t type) : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), mNormalFrameCount(0), mMixBuffer(NULL), - mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), + mSuspended(0), mBytesWritten(0), // mStreamTypes[] initialized in constructor body mOutput(output), mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), @@ -988,7 +1003,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge AudioFlinger::PlaybackThread::~PlaybackThread() { mAudioFlinger->unregisterWriter(mNBLogWriter); - delete [] mAllocMixBuffer; + delete[] mMixBuffer; } void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) @@ -1078,16 +1093,6 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& } // Thread virtuals -status_t AudioFlinger::PlaybackThread::readyToRun() -{ - status_t status = initCheck(); - if (status == NO_ERROR) { - ALOGI("AudioFlinger's thread %p ready to run", this); - } else { - ALOGE("No working audio driver found."); - } - return status; -} void AudioFlinger::PlaybackThread::onFirstRef() { @@ -1253,8 +1258,12 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac track = TimedTrack::create(this, client, streamType, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId); } - if (track == 0 || track->getCblk() == NULL || track->name() < 0) { - lStatus = NO_MEMORY; + + // new Track always returns non-NULL, + // but TimedTrack::create() is a factory that could fail by returning NULL + lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; + if (lStatus != NO_ERROR) { + track.clear(); goto Exit; } @@ -1279,9 +1288,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac lStatus = NO_ERROR; Exit: - if (status) { - *status = lStatus; - } + *status = lStatus; return track; } @@ -1566,7 +1573,8 @@ void AudioFlinger::PlaybackThread::readOutputParameters() mFormat); } mFrameSize = audio_stream_frame_size(&mOutput->stream->common); - mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; + mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); + mFrameCount = mBufferSize / mFrameSize; if (mFrameCount & 15) { ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", mFrameCount); @@ -1622,11 +1630,11 @@ void AudioFlinger::PlaybackThread::readOutputParameters() ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); - delete[] mAllocMixBuffer; - size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; - mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; - mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); - memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); + delete[] mMixBuffer; + size_t normalBufferSize = mNormalFrameCount * mFrameSize; + // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) + mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; + memset(mMixBuffer, 0, normalBufferSize); // force reconfiguration of effect chains and engines to take new buffer size and audio // parameters into account @@ -1760,7 +1768,7 @@ void AudioFlinger::PlaybackThread::threadLoop_removeTracks( const Vector< sp<Track> >& tracksToRemove) { size_t count = tracksToRemove.size(); - if (count) { + if (count > 0) { for (size_t i = 0 ; i < count ; i++) { const sp<Track>& track = tracksToRemove.itemAt(i); if (!track->isOutputTrack()) { @@ -2305,7 +2313,7 @@ if (mType == MIXER) { void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) { size_t count = tracksToRemove.size(); - if (count) { + if (count > 0) { for (size_t i=0 ; i<count ; i++) { const sp<Track>& track = tracksToRemove.itemAt(i); mActiveTracks.remove(track); @@ -2680,7 +2688,7 @@ void AudioFlinger::MixerThread::threadLoop_sleepTime() sleepTime = idleSleepTime; } } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { - memset (mMixBuffer, 0, mixBufferSize); + memset(mMixBuffer, 0, mixBufferSize); sleepTime = 0; ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), "anticipated start"); @@ -2906,7 +2914,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac // +1 for rounding and +1 for additional sample needed for interpolation desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; // add frames already consumed but not yet released by the resampler - // because cblk->framesReady() will include these frames + // because mAudioTrackServerProxy->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 @@ -3244,6 +3252,7 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l() if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { status = BAD_VALUE; } else { + // no need to save value, since it's constant reconfig = true; } } @@ -3251,6 +3260,7 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l() if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { status = BAD_VALUE; } else { + // no need to save value, since it's constant reconfig = true; } } @@ -4224,7 +4234,7 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, ) : ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), - // mRsmpInIndex and mBufferSize set by readInputParameters() + // mRsmpInIndex set by readInputParameters() mReqChannelCount(popcount(channelMask)), mReqSampleRate(sampleRate) // mBytesRead is only meaningful while active, and so is cleared in start() @@ -4252,18 +4262,9 @@ void AudioFlinger::RecordThread::onFirstRef() run(mName, PRIORITY_URGENT_AUDIO); } -status_t AudioFlinger::RecordThread::readyToRun() -{ - status_t status = initCheck(); - ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); - return status; -} - bool AudioFlinger::RecordThread::threadLoop() { AudioBufferProvider::Buffer buffer; - sp<RecordTrack> activeTrack; - Vector< sp<EffectChain> > effectChains; nsecs_t lastWarning = 0; @@ -4273,21 +4274,34 @@ bool AudioFlinger::RecordThread::threadLoop() // used to verify we've read at least once before evaluating how many bytes were read bool readOnce = false; + // used to request a deferred sleep, to be executed later while mutex is unlocked + bool doSleep = false; + // start recording - while (!exitPending()) { + for (;;) { + sp<RecordTrack> activeTrack; + TrackBase::track_state activeTrackState; + Vector< sp<EffectChain> > effectChains; - processConfigEvents(); + // sleep with mutex unlocked + if (doSleep) { + doSleep = false; + usleep(kRecordThreadSleepUs); + } { // scope for mLock Mutex::Autolock _l(mLock); - checkForNewParameters_l(); - if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { + if (exitPending()) { + break; + } + processConfigEvents_l(); + // return value 'reconfig' is currently unused + bool reconfig = checkForNewParameters_l(); + // make a stable copy of mActiveTrack + activeTrack = mActiveTrack; + if (activeTrack == 0) { standby(); - - if (exitPending()) { - break; - } - + // exitPending() can't become true here releaseWakeLock_l(); ALOGV("RecordThread: loop stopping"); // go to sleep @@ -4296,172 +4310,191 @@ bool AudioFlinger::RecordThread::threadLoop() acquireWakeLock_l(); continue; } - if (mActiveTrack != 0) { - if (mActiveTrack->isTerminated()) { - removeTrack_l(mActiveTrack); - mActiveTrack.clear(); - } else if (mActiveTrack->mState == TrackBase::PAUSING) { - standby(); + + if (activeTrack->isTerminated()) { + removeTrack_l(activeTrack); + mActiveTrack.clear(); + continue; + } + + activeTrackState = activeTrack->mState; + switch (activeTrackState) { + case TrackBase::PAUSING: + standby(); + mActiveTrack.clear(); + mStartStopCond.broadcast(); + doSleep = true; + continue; + + case TrackBase::RESUMING: + mStandby = false; + if (mReqChannelCount != activeTrack->channelCount()) { mActiveTrack.clear(); mStartStopCond.broadcast(); - } else if (mActiveTrack->mState == TrackBase::RESUMING) { - if (mReqChannelCount != mActiveTrack->channelCount()) { + continue; + } + if (readOnce) { + mStartStopCond.broadcast(); + // record start succeeds only if first read from audio input succeeds + if (mBytesRead < 0) { mActiveTrack.clear(); - mStartStopCond.broadcast(); - } else if (readOnce) { - // record start succeeds only if first read from audio input - // succeeds - if (mBytesRead >= 0) { - mActiveTrack->mState = TrackBase::ACTIVE; - } else { - mActiveTrack.clear(); - } - mStartStopCond.broadcast(); + continue; } - mStandby = false; + activeTrack->mState = TrackBase::ACTIVE; } + break; + + case TrackBase::ACTIVE: + break; + + case TrackBase::IDLE: + doSleep = true; + continue; + + default: + LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); } + lockEffectChains_l(effectChains); } - if (mActiveTrack != 0) { - if (mActiveTrack->mState != TrackBase::ACTIVE && - mActiveTrack->mState != TrackBase::RESUMING) { - unlockEffectChains(effectChains); - usleep(kRecordThreadSleepUs); - continue; - } - for (size_t i = 0; i < effectChains.size(); i ++) { - effectChains[i]->process_l(); - } + // thread mutex is now unlocked, mActiveTrack unknown, activeTrack != 0, kept, immutable + // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING - buffer.frameCount = mFrameCount; - status_t status = mActiveTrack->getNextBuffer(&buffer); - if (status == NO_ERROR) { - readOnce = true; - size_t framesOut = buffer.frameCount; - if (mResampler == NULL) { - // no resampling - while (framesOut) { - size_t framesIn = mFrameCount - mRsmpInIndex; - if (framesIn) { - int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; - int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * - mActiveTrack->mFrameSize; - if (framesIn > framesOut) - framesIn = framesOut; - mRsmpInIndex += framesIn; - framesOut -= framesIn; - if (mChannelCount == mReqChannelCount) { - memcpy(dst, src, framesIn * mFrameSize); - } else { - if (mChannelCount == 1) { - upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, - (int16_t *)src, framesIn); - } else { - downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, - (int16_t *)src, framesIn); - } - } + for (size_t i = 0; i < effectChains.size(); i ++) { + // thread mutex is not locked, but effect chain is locked + effectChains[i]->process_l(); + } + + buffer.frameCount = mFrameCount; + status_t status = activeTrack->getNextBuffer(&buffer); + if (status == NO_ERROR) { + readOnce = true; + size_t framesOut = buffer.frameCount; + if (mResampler == NULL) { + // no resampling + while (framesOut) { + size_t framesIn = mFrameCount - mRsmpInIndex; + if (framesIn > 0) { + int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; + int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * + activeTrack->mFrameSize; + if (framesIn > framesOut) { + framesIn = framesOut; } - if (framesOut && mFrameCount == mRsmpInIndex) { - void *readInto; - if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { - readInto = buffer.raw; - framesOut = 0; + mRsmpInIndex += framesIn; + framesOut -= framesIn; + if (mChannelCount == mReqChannelCount) { + memcpy(dst, src, framesIn * mFrameSize); + } else { + if (mChannelCount == 1) { + upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, + (int16_t *)src, framesIn); } else { - readInto = mRsmpInBuffer; - mRsmpInIndex = 0; + downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, + (int16_t *)src, framesIn); } - mBytesRead = mInput->stream->read(mInput->stream, readInto, - mBufferSize); - if (mBytesRead <= 0) { - if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) - { - ALOGE("Error reading audio input"); - // Force input into standby so that it tries to - // recover at next read attempt - inputStandBy(); - usleep(kRecordThreadSleepUs); - } - mRsmpInIndex = mFrameCount; - framesOut = 0; - buffer.frameCount = 0; + } + } + if (framesOut > 0 && mFrameCount == mRsmpInIndex) { + void *readInto; + if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { + readInto = buffer.raw; + framesOut = 0; + } else { + readInto = mRsmpInBuffer; + mRsmpInIndex = 0; + } + mBytesRead = mInput->stream->read(mInput->stream, readInto, + mBufferSize); + if (mBytesRead <= 0) { + // TODO: verify that it's benign to use a stale track state + if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE)) + { + ALOGE("Error reading audio input"); + // Force input into standby so that it tries to + // recover at next read attempt + inputStandBy(); + doSleep = true; } + mRsmpInIndex = mFrameCount; + framesOut = 0; + buffer.frameCount = 0; + } #ifdef TEE_SINK - else if (mTeeSink != 0) { - (void) mTeeSink->write(readInto, - mBytesRead >> Format_frameBitShift(mTeeSink->format())); - } -#endif + else if (mTeeSink != 0) { + (void) mTeeSink->write(readInto, + mBytesRead >> Format_frameBitShift(mTeeSink->format())); } +#endif } + } + } else { + // resampling + + // resampler accumulates, but we only have one source track + memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); + // alter output frame count as if we were expecting stereo samples + if (mChannelCount == 1 && mReqChannelCount == 1) { + framesOut >>= 1; + } + mResampler->resample(mRsmpOutBuffer, framesOut, + this /* AudioBufferProvider* */); + // ditherAndClamp() works as long as all buffers returned by + // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. + if (mChannelCount == 2 && mReqChannelCount == 1) { + // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t + ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); + // the resampler always outputs stereo samples: + // do post stereo to mono conversion + downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, + framesOut); } else { - // resampling + ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); + } + // now done with mRsmpOutBuffer - // resampler accumulates, but we only have one source track - memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); - // alter output frame count as if we were expecting stereo samples - if (mChannelCount == 1 && mReqChannelCount == 1) { - framesOut >>= 1; - } - mResampler->resample(mRsmpOutBuffer, framesOut, - this /* AudioBufferProvider* */); - // ditherAndClamp() works as long as all buffers returned by - // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. - if (mChannelCount == 2 && mReqChannelCount == 1) { - // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t - ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); - // the resampler always outputs stereo samples: - // do post stereo to mono conversion - downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, - framesOut); - } else { - ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); + } + if (mFramestoDrop == 0) { + activeTrack->releaseBuffer(&buffer); + } else { + if (mFramestoDrop > 0) { + mFramestoDrop -= buffer.frameCount; + if (mFramestoDrop <= 0) { + clearSyncStartEvent(); } - // now done with mRsmpOutBuffer - - } - if (mFramestoDrop == 0) { - mActiveTrack->releaseBuffer(&buffer); } else { - if (mFramestoDrop > 0) { - mFramestoDrop -= buffer.frameCount; - if (mFramestoDrop <= 0) { - clearSyncStartEvent(); - } - } else { - mFramestoDrop += buffer.frameCount; - if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || - mSyncStartEvent->isCancelled()) { - ALOGW("Synced record %s, session %d, trigger session %d", - (mFramestoDrop >= 0) ? "timed out" : "cancelled", - mActiveTrack->sessionId(), - (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); - clearSyncStartEvent(); - } + mFramestoDrop += buffer.frameCount; + if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || + mSyncStartEvent->isCancelled()) { + ALOGW("Synced record %s, session %d, trigger session %d", + (mFramestoDrop >= 0) ? "timed out" : "cancelled", + activeTrack->sessionId(), + (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); + clearSyncStartEvent(); } } - mActiveTrack->clearOverflow(); } - // client isn't retrieving buffers fast enough - else { - if (!mActiveTrack->setOverflow()) { - nsecs_t now = systemTime(); - if ((now - lastWarning) > kWarningThrottleNs) { - ALOGW("RecordThread: buffer overflow"); - lastWarning = now; - } + activeTrack->clearOverflow(); + } + // client isn't retrieving buffers fast enough + else { + if (!activeTrack->setOverflow()) { + nsecs_t now = systemTime(); + if ((now - lastWarning) > kWarningThrottleNs) { + ALOGW("RecordThread: buffer overflow"); + lastWarning = now; } - // Release the processor for a while before asking for a new buffer. - // This will give the application more chance to read from the buffer and - // clear the overflow. - usleep(kRecordThreadSleepUs); } + // Release the processor for a while before asking for a new buffer. + // This will give the application more chance to read from the buffer and + // clear the overflow. + doSleep = true; } + // enable changes in effect chain unlockEffectChains(effectChains); - effectChains.clear(); + // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end } standby(); @@ -4495,7 +4528,7 @@ void AudioFlinger::RecordThread::inputStandBy() mInput->stream->common.standby(&mInput->stream->common); } -sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( +sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( const sp<AudioFlinger::Client>& client, uint32_t sampleRate, audio_format_t format, @@ -4574,8 +4607,9 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR track = new RecordTrack(this, client, sampleRate, format, channelMask, frameCount, sessionId); - if (track->getCblk() == 0) { - lStatus = NO_MEMORY; + lStatus = track->initCheck(); + if (lStatus != NO_ERROR) { + track.clear(); goto Exit; } mTracks.add(track); @@ -4596,9 +4630,7 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR lStatus = NO_ERROR; Exit: - if (status) { - *status = lStatus; - } + *status = lStatus; return track; } @@ -4629,6 +4661,7 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac } { + // This section is a rendezvous between binder thread executing start() and RecordThread AutoMutex lock(mLock); if (mActiveTrack != 0) { if (recordTrack != mActiveTrack.get()) { @@ -4639,11 +4672,13 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac return status; } + // FIXME why? already set in constructor, 'STARTING_1' would be more accurate recordTrack->mState = TrackBase::IDLE; mActiveTrack = recordTrack; mLock.unlock(); status_t status = AudioSystem::startInput(mId); mLock.lock(); + // FIXME should verify that mActiveTrack is still == recordTrack if (status != NO_ERROR) { mActiveTrack.clear(); clearSyncStartEvent(); @@ -4654,6 +4689,8 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac if (mResampler != NULL) { mResampler->reset(); } + // FIXME hijacking a playback track state name which was intended for start after pause; + // here 'STARTING_2' would be more accurate mActiveTrack->mState = TrackBase::RESUMING; // signal thread to start ALOGV("Signal record thread"); @@ -4664,6 +4701,7 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac status = INVALID_OPERATION; goto startError; } + // FIXME incorrect usage of wait: no explicit predicate or loop mStartStopCond.wait(mLock); if (mActiveTrack == 0) { ALOGV("Record failed to start"); @@ -4714,11 +4752,13 @@ bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { return false; } + // note that threadLoop may still be processing the track at this point [without lock] recordTrack->mState = TrackBase::PAUSING; // do not wait for mStartStopCond if exiting if (exitPending()) { return true; } + // FIXME incorrect usage of wait: no explicit predicate or loop mStartStopCond.wait(mLock); // if we have been restarted, recordTrack == mActiveTrack.get() here if (exitPending() || recordTrack != mActiveTrack.get()) { @@ -4854,6 +4894,7 @@ status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* // Force input into standby so that it tries to // recover at next read attempt inputStandBy(); + // FIXME an awkward place to sleep, consider using doSleep when this is pulled up usleep(kRecordThreadSleepUs); } buffer->raw = NULL; @@ -4896,7 +4937,7 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() int value; audio_format_t reqFormat = mFormat; uint32_t reqSamplingRate = mReqSampleRate; - uint32_t reqChannelCount = mReqChannelCount; + audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { reqSamplingRate = value; @@ -4911,8 +4952,13 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() } } if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { - reqChannelCount = popcount(value); - reconfig = true; + audio_channel_mask_t mask = (audio_channel_mask_t) value; + if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { + status = BAD_VALUE; + } else { + reqChannelMask = mask; + reconfig = true; + } } if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { // do not accept frame count changes if tracks are open as the track buffer @@ -4961,6 +5007,7 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() } mAudioSource = (audio_source_t)value; } + if (status == NO_ERROR) { status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); @@ -4977,7 +5024,8 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() <= (2 * reqSamplingRate)) && popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && - (reqChannelCount <= FCC_2)) { + (reqChannelMask == AUDIO_CHANNEL_IN_MONO || + reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { status = NO_ERROR; } if (status == NO_ERROR) { @@ -5054,8 +5102,7 @@ void AudioFlinger::RecordThread::readInputParameters() mFrameCount = mBufferSize / mFrameSize; mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; - if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) - { + if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { int channelCount; // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid // stereo to mono post process as the resampler always outputs stereo. |