diff options
Diffstat (limited to 'services/audioflinger/AudioFlinger.cpp')
| -rw-r--r-- | services/audioflinger/AudioFlinger.cpp | 361 |
1 files changed, 246 insertions, 115 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 76d6447..35bd431 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -417,6 +417,12 @@ status_t AudioFlinger::dump(int fd, const Vector<String16>& args) audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); dev->dump(dev, fd); } + + // dump the serially shared record tee sink + if (mRecordTeeSource != 0) { + dumpTee(fd, mRecordTeeSource); + } + if (locked) mLock.unlock(); } return NO_ERROR; @@ -941,8 +947,8 @@ status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const { -// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", -// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); + ALOGVV("getParameters() io %d, keys %s, tid %d, calling pid %d", + ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); Mutex::Autolock _l(mLock); @@ -1112,7 +1118,8 @@ void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, c // removeClient_l() must be called with AudioFlinger::mLock held void AudioFlinger::removeClient_l(pid_t pid) { - ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); + ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), + IPCThreadState::self()->getCallingPid()); mClients.removeItem(pid); } @@ -1215,7 +1222,8 @@ void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) { IoConfigEvent *ioEvent = new IoConfigEvent(event, param); mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); - ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); + ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, + param); mWaitWorkCV.signal(); } @@ -1244,7 +1252,8 @@ void AudioFlinger::ThreadBase::processConfigEvents() PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); if (err != 0) { - ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", + ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " + "error %d", prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); } } break; @@ -1661,7 +1670,8 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); result.append(buffer); - snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); + snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", + ns2ms(systemTime() - mLastWriteTime)); result.append(buffer); snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); result.append(buffer); @@ -1791,7 +1801,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac if (mType == DIRECT) { if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { - ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" + ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x " "for output %p with format %d", sampleRate, format, channelMask, mOutput, mFormat); lStatus = BAD_VALUE; @@ -1959,7 +1969,8 @@ status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) if (track->mainBuffer() != mMixBuffer) { sp<EffectChain> chain = getEffectChain_l(track->sessionId()); if (chain != 0) { - ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); + ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), + track->sessionId()); chain->incActiveTrackCnt(); } } @@ -2025,7 +2036,8 @@ void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { AudioSystem::OutputDescriptor desc; void *param2 = NULL; - ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); + ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, + param); switch (event) { case AudioSystem::OUTPUT_OPENED: @@ -2033,7 +2045,8 @@ void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { desc.channels = mChannelMask; desc.samplingRate = mSampleRate; desc.format = mFormat; - desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) + desc.frameCount = mNormalFrameCount; // FIXME see + // AudioFlinger::frameCount(audio_io_handle_t) desc.latency = latency(); param2 = &desc; break; @@ -2062,7 +2075,8 @@ void AudioFlinger::PlaybackThread::readOutputParameters() // Calculate size of normal mix buffer relative to the HAL output buffer size double multiplier = 1.0; - if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { + if (mType == MIXER && (kUseFastMixer == FastMixer_Static || + kUseFastMixer == FastMixer_Dynamic)) { size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer @@ -2081,9 +2095,10 @@ void AudioFlinger::PlaybackThread::readOutputParameters() multiplier = (double) maxNormalFrameCount / (double) mFrameCount; } } else { - // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC - // (it would be unusual for the normal mix buffer size to not be a multiple of fast - // track, but we sometimes have to do this to satisfy the maximum frame count constraint) + // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL + // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast + // track, but we sometimes have to do this to satisfy the maximum frame count + // constraint) // FIXME this rounding up should not be done if no HAL SRC uint32_t truncMult = (uint32_t) multiplier; if ((truncMult & 1)) { @@ -2097,7 +2112,8 @@ void AudioFlinger::PlaybackThread::readOutputParameters() mNormalFrameCount = multiplier * mFrameCount; // round up to nearest 16 frames to satisfy AudioMixer mNormalFrameCount = (mNormalFrameCount + 15) & ~15; - ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); + ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, + mNormalFrameCount); delete[] mMixBuffer; mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; @@ -2235,7 +2251,8 @@ bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; } -void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) +void AudioFlinger::PlaybackThread::threadLoop_removeTracks( + const Vector< sp<Track> >& tracksToRemove) { size_t count = tracksToRemove.size(); if (CC_UNLIKELY(count)) { @@ -2891,7 +2908,8 @@ void AudioFlinger::MixerThread::threadLoop_sleepTime() } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { memset (mMixBuffer, 0, mixBufferSize); sleepTime = 0; - ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); + ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), + "anticipated start"); } // TODO add standby time extension fct of effect tail } @@ -3125,7 +3143,8 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && !track->isTerminated()) { - //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); + ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, + this); mixedTracks++; @@ -3138,7 +3157,8 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac if (chain != 0) { tracksWithEffect++; } else { - ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", + ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " + "session %d", name, track->sessionId()); } } @@ -3268,7 +3288,8 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac chain->clearInputBuffer(); } - //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); + ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, + cblk->server, this); if ((track->sharedBuffer() != 0) || track->isTerminated() || track->isStopped() || track->isPaused()) { // We have consumed all the buffers of this track. @@ -3362,7 +3383,8 @@ track_is_ready: ; if (track->mainBuffer() != mMixBuffer) { chain = getEffectChain_l(track->sessionId()); if (chain != 0) { - ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); + ALOGV("stopping track on chain %p for session Id: %d", chain.get(), + track->sessionId()); chain->decActiveTrackCnt(); } } @@ -3375,7 +3397,8 @@ track_is_ready: ; // mix buffer must be cleared if all tracks are connected to an // effect chain as in this case the mixer will not write to // mix buffer and track effects will accumulate into it - if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { + if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || + (mixedTracks == 0 && fastTracks > 0)) { // FIXME as a performance optimization, should remember previous zero status memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); } @@ -3580,39 +3603,18 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l() return reconfig; } -void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) +void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) { - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - PlaybackThread::dumpInternals(fd, args); - - snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); - result.append(buffer); - write(fd, result.string(), result.size()); - - // Make a non-atomic copy of fast mixer dump state so it won't change underneath us - FastMixerDumpState copy = mFastMixerDumpState; - copy.dump(fd); - -#ifdef STATE_QUEUE_DUMP - // Similar for state queue - StateQueueObserverDump observerCopy = mStateQueueObserverDump; - observerCopy.dump(fd); - StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; - mutatorCopy.dump(fd); -#endif - - // Write the tee output to a .wav file - NBAIO_Source *teeSource = mTeeSource.get(); + NBAIO_Source *teeSource = source.get(); if (teeSource != NULL) { - char teePath[64]; + char teeTime[16]; struct timeval tv; gettimeofday(&tv, NULL); struct tm tm; localtime_r(&tv.tv_sec, &tm); - strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); + strftime(teeTime, sizeof(teeTime), "%T", &tm); + char teePath[64]; + sprintf(teePath, "/data/misc/media/%s_%d.wav", teeTime, id); int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); if (teeFd >= 0) { char wavHeader[44]; @@ -3660,6 +3662,34 @@ void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& ar fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); } } +} + +void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + PlaybackThread::dumpInternals(fd, args); + + snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); + result.append(buffer); + write(fd, result.string(), result.size()); + + // Make a non-atomic copy of fast mixer dump state so it won't change underneath us + FastMixerDumpState copy = mFastMixerDumpState; + copy.dump(fd); + +#ifdef STATE_QUEUE_DUMP + // Similar for state queue + StateQueueObserverDump observerCopy = mStateQueueObserverDump; + observerCopy.dump(fd); + StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; + mutatorCopy.dump(fd); +#endif + + // Write the tee output to a .wav file + dumpTee(fd, mTeeSource, mId); #ifdef AUDIO_WATCHDOG if (mAudioWatchdog != 0) { @@ -3731,7 +3761,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && !track->isTerminated()) { - //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); + ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); if (track->mFillingUpStatus == Track::FS_FILLED) { track->mFillingUpStatus = Track::FS_ACTIVE; @@ -3792,7 +3822,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep mEffectChains[0]->clearInputBuffer(); } - //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); + ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); if ((track->sharedBuffer() != 0) || track->isTerminated() || track->isStopped() || track->isPaused()) { // We have consumed all the buffers of this track. @@ -3982,7 +4012,8 @@ void AudioFlinger::DirectOutputThread::cacheParameters_l() AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) - : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), DUPLICATING), + : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), + DUPLICATING), mWaitTimeMs(UINT_MAX) { addOutputTrack(mainThread); @@ -4103,18 +4134,21 @@ void AudioFlinger::DuplicatingThread::updateWaitTime_l() } -bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) +bool AudioFlinger::DuplicatingThread::outputsReady( + const SortedVector< sp<OutputTrack> > &outputTracks) { for (size_t i = 0; i < outputTracks.size(); i++) { sp<ThreadBase> thread = outputTracks[i]->thread().promote(); if (thread == 0) { - ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); + ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", + outputTracks[i].get()); return false; } PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); // see note at standby() declaration if (playbackThread->standby() && !playbackThread->isSuspended()) { - ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); + ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), + thread.get()); return false; } } @@ -4161,7 +4195,8 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase( // mChannelCount // mChannelMask { - ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); + ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), + sharedBuffer->size()); // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); size_t size = sizeof(audio_track_cblk_t); @@ -4322,7 +4357,8 @@ AudioFlinger::PlaybackThread::Track::Track( const sp<IMemory>& sharedBuffer, int sessionId, IAudioFlinger::track_flags_t flags) - : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), + : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, + sessionId), mMute(false), mFillingUpStatus(FS_INVALID), // mRetryCount initialized later when needed @@ -4341,7 +4377,8 @@ AudioFlinger::PlaybackThread::Track::Track( if (mCblk != NULL) { // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack - mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); + mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : + sizeof(uint8_t); // to avoid leaking a track name, do not allocate one unless there is an mCblk mName = thread->getTrackName_l(channelMask, sessionId); mCblk->mName = mName; @@ -4366,7 +4403,8 @@ AudioFlinger::PlaybackThread::Track::Track( thread->mFastTrackAvailMask &= ~(1 << i); } } - ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); + ALOGV("Track constructor name %d, calling pid %d", mName, + IPCThreadState::self()->getCallingPid()); } AudioFlinger::PlaybackThread::Track::~Track() @@ -4408,8 +4446,8 @@ void AudioFlinger::PlaybackThread::Track::destroy() /*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) { - result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " - " Server User Main buf Aux Buf Flags Underruns\n"); + result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate " + "L dB R dB Server User Main buf Aux Buf Flags Underruns\n"); } void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) @@ -4636,7 +4674,8 @@ void AudioFlinger::PlaybackThread::Track::stop() // and then to STOPPED and reset() when presentation is complete mState = STOPPING_1; } - ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); + ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, + playbackThread); } if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { thread->mLock.unlock(); @@ -5395,7 +5434,8 @@ AudioFlinger::RecordThread::RecordTrack::~RecordTrack() } // AudioBufferProvider interface -status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) +status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, + int64_t pts) { audio_track_cblk_t* cblk = this->cblk(); uint32_t framesAvail; @@ -5587,7 +5627,8 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr mOutBuffer.frameCount = pInBuffer->frameCount; nsecs_t startTime = systemTime(); if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { - ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); + ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, + mThread.unsafe_get()); outputBufferFull = true; break; } @@ -5599,7 +5640,8 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr } } - uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; + uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : + pInBuffer->frameCount; memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); mCblk->stepUser(outFrames); pInBuffer->frameCount -= outFrames; @@ -5612,7 +5654,8 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr mBufferQueue.removeAt(0); delete [] pInBuffer->mBuffer; delete pInBuffer; - ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); + ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, + mThread.unsafe_get(), mBufferQueue.size()); } else { break; } @@ -5628,11 +5671,14 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; pInBuffer->frameCount = inBuffer.frameCount; pInBuffer->i16 = pInBuffer->mBuffer; - memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); + memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * + sizeof(int16_t)); mBufferQueue.add(pInBuffer); - ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); + ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, + mThread.unsafe_get(), mBufferQueue.size()); } else { - ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); + ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", + mThread.unsafe_get(), this); } } } @@ -5657,14 +5703,15 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr return outputBufferFull; } -status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) +status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( + AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) { int active; status_t result; audio_track_cblk_t* cblk = mCblk; uint32_t framesReq = buffer->frameCount; -// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); + ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); buffer->frameCount = 0; uint32_t framesAvail = cblk->framesAvailable(); @@ -5921,13 +5968,14 @@ sp<IAudioRecord> AudioFlinger::openRecord( *sessionId = lSessionId; } } - // create new record track. The record track uses one track in mHardwareMixerThread by convention. + // create new record track. + // The record track uses one track in mHardwareMixerThread by convention. recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, frameCount, lSessionId, flags, tid, &lStatus); } if (lStatus != NO_ERROR) { - // remove local strong reference to Client before deleting the RecordTrack so that the Client - // destructor is called by the TrackBase destructor with mLock held + // remove local strong reference to Client before deleting the RecordTrack so that the + // Client destructor is called by the TrackBase destructor with mLock held client.clear(); recordTrack.clear(); goto Exit; @@ -5946,7 +5994,8 @@ Exit: // ---------------------------------------------------------------------------- -AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) +AudioFlinger::RecordHandle::RecordHandle( + const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) : BnAudioRecord(), mRecordTrack(recordTrack) { @@ -5961,7 +6010,8 @@ sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { return mRecordTrack->getCblk(); } -status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, int triggerSession) { +status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, + int triggerSession) { ALOGV("RecordHandle::start()"); return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); } @@ -5988,18 +6038,21 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_io_handle_t id, - audio_devices_t device) : + audio_devices_t device, + const sp<NBAIO_Sink>& teeSink) : ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), // mRsmpInIndex and mInputBytes set by readInputParameters() mReqChannelCount(popcount(channelMask)), - mReqSampleRate(sampleRate) + mReqSampleRate(sampleRate), // mBytesRead is only meaningful while active, and so is cleared in start() // (but might be better to also clear here for dump?) + mTeeSink(teeSink) { snprintf(mName, kNameLength, "AudioIn_%X", id); readInputParameters(); + } @@ -6106,7 +6159,8 @@ bool AudioFlinger::RecordThread::threadLoop() size_t framesIn = mFrameCount - mRsmpInIndex; if (framesIn) { int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; - int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; + int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * + mActiveTrack->mCblk->frameSize; if (framesIn > framesOut) framesIn = framesOut; mRsmpInIndex += framesIn; @@ -6125,14 +6179,17 @@ bool AudioFlinger::RecordThread::threadLoop() } } if (framesOut && mFrameCount == mRsmpInIndex) { + void *readInto; if (framesOut == mFrameCount && - ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { - mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); + ((int)mChannelCount == mReqChannelCount || + mFormat != AUDIO_FORMAT_PCM_16_BIT)) { + readInto = buffer.raw; framesOut = 0; } else { - mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); + readInto = mRsmpInBuffer; mRsmpInIndex = 0; } + mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); if (mBytesRead <= 0) { if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { @@ -6145,6 +6202,9 @@ bool AudioFlinger::RecordThread::threadLoop() mRsmpInIndex = mFrameCount; framesOut = 0; buffer.frameCount = 0; + } else if (mTeeSink != 0) { + (void) mTeeSink->write(readInto, + mBytesRead >> Format_frameBitShift(mTeeSink->format())); } } } @@ -6156,12 +6216,14 @@ bool AudioFlinger::RecordThread::threadLoop() 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. + 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) { ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); - // the resampler always outputs stereo samples: do post stereo to mono conversion + // 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 { @@ -6635,7 +6697,8 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() status = BAD_VALUE; } else { mInDevice = value; - // disable AEC and NS if the device is a BT SCO headset supporting those pre processings + // disable AEC and NS if the device is a BT SCO headset supporting those + // pre processings if (mTracks.size() > 0) { bool suspend = audio_is_bluetooth_sco_device(mInDevice) && mAudioFlinger->btNrecIsOff(); @@ -6657,7 +6720,8 @@ 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()); + status = mInput->stream->common.set_parameters(&mInput->stream->common, + keyValuePair.string()); if (status == INVALID_OPERATION) { inputStandBy(); status = mInput->stream->common.set_parameters(&mInput->stream->common, @@ -6667,8 +6731,10 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() if (status == BAD_VALUE && reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && reqFormat == AUDIO_FORMAT_PCM_16_BIT && - ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && - popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && + ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) + <= (2 * reqSamplingRate)) && + popcount(mInput->stream->common.get_channels(&mInput->stream->common)) + <= FCC_2 && (reqChannelCount <= FCC_2)) { status = NO_ERROR; } @@ -6762,7 +6828,8 @@ void AudioFlinger::RecordThread::readInputParameters() mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); mRsmpOutBuffer = new int32_t[mFrameCount * 2]; - // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples + // optmization: if mono to mono, alter input frame count as if we were inputing + // stereo samples if (mChannelCount == 1 && mReqChannelCount == 1) { mFrameCount >>= 1; } @@ -6989,7 +7056,8 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, &outStream); mHardwareStatus = AUDIO_HW_IDLE; - ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", + ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, " + "Channels %x, status %d", outStream, config.sample_rate, config.format, @@ -7042,7 +7110,8 @@ audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, MixerThread *thread2 = checkMixerThread_l(output2); if (thread1 == NULL || thread2 == NULL) { - ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); + ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, + output2); return 0; } @@ -7077,7 +7146,8 @@ status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) if (thread->type() == ThreadBase::MIXER) { for (size_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { - DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); + DuplicatingThread *dupThread = + (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); dupThread->removeOutputTrack((MixerThread *)thread.get()); } } @@ -7164,16 +7234,17 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); - ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", + ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " + "status %d", inStream, config.sample_rate, config.format, config.channel_mask, status); - // If the input could not be opened with the requested parameters and we can handle the conversion internally, - // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo - // or stereo to mono conversions on 16 bit PCM inputs. + // If the input could not be opened with the requested parameters and we can handle the + // conversion internally, try to open again with the proposed parameters. The AudioFlinger can + // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. if (status == BAD_VALUE && reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && (config.sample_rate <= 2 * reqSamplingRate) && @@ -7184,18 +7255,66 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, } if (status == NO_ERROR && inStream != NULL) { + + // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, + // or (re-)create if current Pipe is idle and does not match the new format + sp<NBAIO_Sink> teeSink; +#ifdef TEE_SINK_INPUT_FRAMES + enum { + TEE_SINK_NO, // don't copy input + TEE_SINK_NEW, // copy input using a new pipe + TEE_SINK_OLD, // copy input using an existing pipe + } kind; + NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), + popcount(inStream->common.get_channels(&inStream->common))); + if (format == Format_Invalid) { + kind = TEE_SINK_NO; + } else if (mRecordTeeSink == 0) { + kind = TEE_SINK_NEW; + } else if (mRecordTeeSink->getStrongCount() != 1) { + kind = TEE_SINK_NO; + } else if (format == mRecordTeeSink->format()) { + kind = TEE_SINK_OLD; + } else { + kind = TEE_SINK_NEW; + } + switch (kind) { + case TEE_SINK_NEW: { + Pipe *pipe = new Pipe(TEE_SINK_INPUT_FRAMES, format); + size_t numCounterOffers = 0; + const NBAIO_Format offers[1] = {format}; + ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); + ALOG_ASSERT(index == 0); + PipeReader *pipeReader = new PipeReader(*pipe); + numCounterOffers = 0; + index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); + ALOG_ASSERT(index == 0); + mRecordTeeSink = pipe; + mRecordTeeSource = pipeReader; + teeSink = pipe; + } + break; + case TEE_SINK_OLD: + teeSink = mRecordTeeSink; + break; + case TEE_SINK_NO: + default: + break; + } +#endif AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); // Start record thread // RecorThread require both input and output device indication to forward to audio // pre processing modules audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); + thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id, - device); + device, teeSink); mRecordThreads.add(id, thread); ALOGV("openInput() created record thread: ID %d thread %p", id, thread); if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; @@ -8003,7 +8122,8 @@ status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& c for (size_t i = 0; i < mTracks.size(); ++i) { sp<Track> track = mTracks[i]; if (session == track->sessionId()) { - ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); + ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), + buffer); track->setMainBuffer(buffer); chain->incTrackCnt(); } @@ -8594,7 +8714,7 @@ status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, void *pReplyData) { Mutex::Autolock _l(mLock); -// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); + ALOGVV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); if (mState == DESTROYED || mEffectInterface == NULL) { return NO_INIT; @@ -8845,12 +8965,15 @@ void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) result.append("\t\tDescriptor:\n"); snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, - mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], + mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1], + mDescriptor.uuid.node[2], mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); result.append(buffer); snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", - mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, - mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], + mDescriptor.type.timeLow, mDescriptor.type.timeMid, + mDescriptor.type.timeHiAndVersion, + mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1], + mDescriptor.type.node[2], mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); result.append(buffer); snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", @@ -8934,7 +9057,8 @@ AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, mBuffer = (uint8_t *)mCblk + bufOffset; } } else { - ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); + ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + + sizeof(effect_param_cblk_t)); return; } } @@ -9049,8 +9173,8 @@ status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, uint32_t *replySize, void *pReplyData) { -// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", -// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); + ALOGVV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", + cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); // only get parameter command is permitted for applications not controlling the effect if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { @@ -9061,8 +9185,9 @@ status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, // handle commands that are not forwarded transparently to effect engine if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { - // No need to trylock() here as this function is executed in the binder thread serving a particular client process: - // no risk to block the whole media server process or mixer threads is we are stuck here + // No need to trylock() here as this function is executed in the binder thread serving a + // particular client process: no risk to block the whole media server process or mixer + // threads if we are stuck here Mutex::Autolock _l(mCblk->lock); if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { @@ -9202,7 +9327,8 @@ AudioFlinger::EffectChain::~EffectChain() } // getEffectFromDesc_l() must be called with ThreadBase::mLock held -sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) +sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l( + effect_descriptor_t *descriptor) { size_t size = mEffects.size(); @@ -9361,7 +9487,8 @@ status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) // check invalid effect chaining combinations if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { - ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); + ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", + desc.name, d.name); return INVALID_OPERATION; } // remember position of first insert effect and by default @@ -9412,7 +9539,8 @@ status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) } mEffects.insertAt(effect, idx_insert); - ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); + ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, + idx_insert); } effect->configure(); return NO_ERROR; @@ -9443,7 +9571,8 @@ size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) } } mEffects.removeAt(i); - ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); + ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), + this, i); break; } } @@ -9667,7 +9796,8 @@ void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) for (size_t i = 0; i < types.size(); i++) { setEffectSuspended_l(types[i], false); } - ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); + ALOGV("setEffectSuspendedAll_l() remove entry for %08x", + mSuspendedEffects.keyAt(index)); mSuspendedEffects.removeItem((int)kKeyForSuspendAll); } } @@ -9693,7 +9823,8 @@ bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descript return true; } -void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) +void AudioFlinger::EffectChain::getSuspendEligibleEffects( + Vector< sp<AudioFlinger::EffectModule> > &effects) { effects.clear(); for (size_t i = 0; i < mEffects.size(); i++) { |