| /* |
| ** |
| ** Copyright 2012, The Android Open Source Project |
| ** |
| ** Licensed under the Apache License, Version 2.0 (the "License"); |
| ** you may not use this file except in compliance with the License. |
| ** You may obtain a copy of the License at |
| ** |
| ** http://www.apache.org/licenses/LICENSE-2.0 |
| ** |
| ** Unless required by applicable law or agreed to in writing, software |
| ** distributed under the License is distributed on an "AS IS" BASIS, |
| ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| ** See the License for the specific language governing permissions and |
| ** limitations under the License. |
| */ |
| |
| |
| #define LOG_TAG "AudioFlinger" |
| //#define LOG_NDEBUG 0 |
| |
| #include "Configuration.h" |
| #include <linux/futex.h> |
| #include <math.h> |
| #include <sys/syscall.h> |
| #include <utils/Log.h> |
| |
| #include <private/media/AudioTrackShared.h> |
| |
| #include <common_time/cc_helper.h> |
| #include <common_time/local_clock.h> |
| |
| #include "AudioMixer.h" |
| #include "AudioFlinger.h" |
| #include "ServiceUtilities.h" |
| |
| #include <media/nbaio/Pipe.h> |
| #include <media/nbaio/PipeReader.h> |
| #include <audio_utils/minifloat.h> |
| |
| // ---------------------------------------------------------------------------- |
| |
| // Note: the following macro is used for extremely verbose logging message. In |
| // order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to |
| // 0; but one side effect of this is to turn all LOGV's as well. Some messages |
| // are so verbose that we want to suppress them even when we have ALOG_ASSERT |
| // turned on. Do not uncomment the #def below unless you really know what you |
| // are doing and want to see all of the extremely verbose messages. |
| //#define VERY_VERY_VERBOSE_LOGGING |
| #ifdef VERY_VERY_VERBOSE_LOGGING |
| #define ALOGVV ALOGV |
| #else |
| #define ALOGVV(a...) do { } while(0) |
| #endif |
| |
| namespace android { |
| |
| // ---------------------------------------------------------------------------- |
| // TrackBase |
| // ---------------------------------------------------------------------------- |
| |
| static volatile int32_t nextTrackId = 55; |
| |
| // TrackBase constructor must be called with AudioFlinger::mLock held |
| AudioFlinger::ThreadBase::TrackBase::TrackBase( |
| ThreadBase *thread, |
| const sp<Client>& client, |
| uint32_t sampleRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| size_t frameCount, |
| void *buffer, |
| int sessionId, |
| int clientUid, |
| IAudioFlinger::track_flags_t flags, |
| bool isOut, |
| alloc_type alloc, |
| track_type type) |
| : RefBase(), |
| mThread(thread), |
| mClient(client), |
| mCblk(NULL), |
| // mBuffer |
| mState(IDLE), |
| mSampleRate(sampleRate), |
| mFormat(format), |
| mChannelMask(channelMask), |
| mChannelCount(isOut ? |
| audio_channel_count_from_out_mask(channelMask) : |
| audio_channel_count_from_in_mask(channelMask)), |
| mFrameSize(audio_is_linear_pcm(format) ? |
| mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), |
| mFrameCount(frameCount), |
| mSessionId(sessionId), |
| mFlags(flags), |
| mIsOut(isOut), |
| mServerProxy(NULL), |
| mId(android_atomic_inc(&nextTrackId)), |
| mTerminated(false), |
| mType(type), |
| mThreadIoHandle(thread->id()) |
| { |
| // 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; |
| |
| // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); |
| size_t size = sizeof(audio_track_cblk_t); |
| size_t bufferSize = (buffer == NULL ? roundup(frameCount) : frameCount) * mFrameSize; |
| if (buffer == NULL && alloc == ALLOC_CBLK) { |
| size += bufferSize; |
| } |
| |
| if (client != 0) { |
| mCblkMemory = client->heap()->allocate(size); |
| if (mCblkMemory == 0 || |
| (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) { |
| ALOGE("not enough memory for AudioTrack size=%u", size); |
| client->heap()->dump("AudioTrack"); |
| mCblkMemory.clear(); |
| return; |
| } |
| } else { |
| // this syntax avoids calling the audio_track_cblk_t constructor twice |
| mCblk = (audio_track_cblk_t *) new uint8_t[size]; |
| // assume mCblk != NULL |
| } |
| |
| // construct the shared structure in-place. |
| if (mCblk != NULL) { |
| new(mCblk) audio_track_cblk_t(); |
| switch (alloc) { |
| case ALLOC_READONLY: { |
| const sp<MemoryDealer> roHeap(thread->readOnlyHeap()); |
| if (roHeap == 0 || |
| (mBufferMemory = roHeap->allocate(bufferSize)) == 0 || |
| (mBuffer = mBufferMemory->pointer()) == NULL) { |
| ALOGE("not enough memory for read-only buffer size=%zu", bufferSize); |
| if (roHeap != 0) { |
| roHeap->dump("buffer"); |
| } |
| mCblkMemory.clear(); |
| mBufferMemory.clear(); |
| return; |
| } |
| memset(mBuffer, 0, bufferSize); |
| } break; |
| case ALLOC_PIPE: |
| mBufferMemory = thread->pipeMemory(); |
| // mBuffer is the virtual address as seen from current process (mediaserver), |
| // and should normally be coming from mBufferMemory->pointer(). |
| // However in this case the TrackBase does not reference the buffer directly. |
| // It should references the buffer via the pipe. |
| // Therefore, to detect incorrect usage of the buffer, we set mBuffer to NULL. |
| mBuffer = NULL; |
| break; |
| case ALLOC_CBLK: |
| // clear all buffers |
| if (buffer == NULL) { |
| mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); |
| memset(mBuffer, 0, bufferSize); |
| } else { |
| mBuffer = buffer; |
| #if 0 |
| mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic |
| #endif |
| } |
| break; |
| case ALLOC_LOCAL: |
| mBuffer = calloc(1, bufferSize); |
| break; |
| case ALLOC_NONE: |
| mBuffer = buffer; |
| break; |
| } |
| |
| #ifdef TEE_SINK |
| if (mTeeSinkTrackEnabled) { |
| NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount, mFormat); |
| if (Format_isValid(pipeFormat)) { |
| Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat); |
| size_t numCounterOffers = 0; |
| const NBAIO_Format offers[1] = {pipeFormat}; |
| 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); |
| mTeeSink = pipe; |
| mTeeSource = pipeReader; |
| } |
| } |
| #endif |
| |
| } |
| } |
| |
| status_t AudioFlinger::ThreadBase::TrackBase::initCheck() const |
| { |
| status_t status; |
| if (mType == TYPE_OUTPUT || mType == TYPE_PATCH) { |
| status = cblk() != NULL ? NO_ERROR : NO_MEMORY; |
| } else { |
| status = getCblk() != 0 ? NO_ERROR : NO_MEMORY; |
| } |
| return status; |
| } |
| |
| AudioFlinger::ThreadBase::TrackBase::~TrackBase() |
| { |
| #ifdef TEE_SINK |
| dumpTee(-1, mTeeSource, mId); |
| #endif |
| // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference |
| delete mServerProxy; |
| if (mCblk != NULL) { |
| if (mClient == 0) { |
| delete mCblk; |
| } else { |
| mCblk->~audio_track_cblk_t(); // destroy our shared-structure. |
| } |
| } |
| mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to |
| if (mClient != 0) { |
| // Client destructor must run with AudioFlinger client mutex locked |
| Mutex::Autolock _l(mClient->audioFlinger()->mClientLock); |
| // If the client's reference count drops to zero, the associated destructor |
| // must run with AudioFlinger lock held. Thus the explicit clear() rather than |
| // relying on the automatic clear() at end of scope. |
| mClient.clear(); |
| } |
| // flush the binder command buffer |
| IPCThreadState::self()->flushCommands(); |
| } |
| |
| // AudioBufferProvider interface |
| // getNextBuffer() = 0; |
| // This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack |
| void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) |
| { |
| #ifdef TEE_SINK |
| if (mTeeSink != 0) { |
| (void) mTeeSink->write(buffer->raw, buffer->frameCount); |
| } |
| #endif |
| |
| ServerProxy::Buffer buf; |
| buf.mFrameCount = buffer->frameCount; |
| buf.mRaw = buffer->raw; |
| buffer->frameCount = 0; |
| buffer->raw = NULL; |
| mServerProxy->releaseBuffer(&buf); |
| } |
| |
| status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) |
| { |
| mSyncEvents.add(event); |
| return NO_ERROR; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // Playback |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) |
| : BnAudioTrack(), |
| mTrack(track) |
| { |
| } |
| |
| AudioFlinger::TrackHandle::~TrackHandle() { |
| // just stop the track on deletion, associated resources |
| // will be freed from the main thread once all pending buffers have |
| // been played. Unless it's not in the active track list, in which |
| // case we free everything now... |
| mTrack->destroy(); |
| } |
| |
| sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { |
| return mTrack->getCblk(); |
| } |
| |
| status_t AudioFlinger::TrackHandle::start() { |
| return mTrack->start(); |
| } |
| |
| void AudioFlinger::TrackHandle::stop() { |
| mTrack->stop(); |
| } |
| |
| void AudioFlinger::TrackHandle::flush() { |
| mTrack->flush(); |
| } |
| |
| void AudioFlinger::TrackHandle::pause() { |
| mTrack->pause(); |
| } |
| |
| status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) |
| { |
| return mTrack->attachAuxEffect(EffectId); |
| } |
| |
| status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, |
| sp<IMemory>* buffer) { |
| if (!mTrack->isTimedTrack()) |
| return INVALID_OPERATION; |
| |
| PlaybackThread::TimedTrack* tt = |
| reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); |
| return tt->allocateTimedBuffer(size, buffer); |
| } |
| |
| status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, |
| int64_t pts) { |
| if (!mTrack->isTimedTrack()) |
| return INVALID_OPERATION; |
| |
| if (buffer == 0 || buffer->pointer() == NULL) { |
| ALOGE("queueTimedBuffer() buffer is 0 or has NULL pointer()"); |
| return BAD_VALUE; |
| } |
| |
| PlaybackThread::TimedTrack* tt = |
| reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); |
| return tt->queueTimedBuffer(buffer, pts); |
| } |
| |
| status_t AudioFlinger::TrackHandle::setMediaTimeTransform( |
| const LinearTransform& xform, int target) { |
| |
| if (!mTrack->isTimedTrack()) |
| return INVALID_OPERATION; |
| |
| PlaybackThread::TimedTrack* tt = |
| reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); |
| return tt->setMediaTimeTransform( |
| xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); |
| } |
| |
| status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) { |
| return mTrack->setParameters(keyValuePairs); |
| } |
| |
| status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp) |
| { |
| return mTrack->getTimestamp(timestamp); |
| } |
| |
| |
| void AudioFlinger::TrackHandle::signal() |
| { |
| return mTrack->signal(); |
| } |
| |
| status_t AudioFlinger::TrackHandle::onTransact( |
| uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) |
| { |
| return BnAudioTrack::onTransact(code, data, reply, flags); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| // Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held |
| AudioFlinger::PlaybackThread::Track::Track( |
| PlaybackThread *thread, |
| const sp<Client>& client, |
| audio_stream_type_t streamType, |
| uint32_t sampleRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| size_t frameCount, |
| void *buffer, |
| const sp<IMemory>& sharedBuffer, |
| int sessionId, |
| int uid, |
| IAudioFlinger::track_flags_t flags, |
| track_type type) |
| : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, |
| (sharedBuffer != 0) ? sharedBuffer->pointer() : buffer, |
| sessionId, uid, flags, true /*isOut*/, |
| (type == TYPE_PATCH) ? ( buffer == NULL ? ALLOC_LOCAL : ALLOC_NONE) : ALLOC_CBLK, |
| type), |
| mFillingUpStatus(FS_INVALID), |
| // mRetryCount initialized later when needed |
| mSharedBuffer(sharedBuffer), |
| mStreamType(streamType), |
| mName(-1), // see note below |
| mMainBuffer(thread->mixBuffer()), |
| mAuxBuffer(NULL), |
| mAuxEffectId(0), mHasVolumeController(false), |
| mPresentationCompleteFrames(0), |
| mFastIndex(-1), |
| mCachedVolume(1.0), |
| mIsInvalid(false), |
| mAudioTrackServerProxy(NULL), |
| mResumeToStopping(false), |
| mFlushHwPending(false) |
| { |
| // client == 0 implies sharedBuffer == 0 |
| ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); |
| |
| ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), |
| sharedBuffer->size()); |
| |
| if (mCblk == NULL) { |
| return; |
| } |
| |
| if (sharedBuffer == 0) { |
| mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, |
| mFrameSize, !isExternalTrack(), sampleRate); |
| } else { |
| mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, |
| mFrameSize); |
| } |
| mServerProxy = mAudioTrackServerProxy; |
| |
| mName = thread->getTrackName_l(channelMask, format, sessionId); |
| if (mName < 0) { |
| ALOGE("no more track names available"); |
| return; |
| } |
| // only allocate a fast track index if we were able to allocate a normal track name |
| if (flags & IAudioFlinger::TRACK_FAST) { |
| mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); |
| ALOG_ASSERT(thread->mFastTrackAvailMask != 0); |
| int i = __builtin_ctz(thread->mFastTrackAvailMask); |
| ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); |
| // FIXME This is too eager. We allocate a fast track index before the |
| // fast track becomes active. Since fast tracks are a scarce resource, |
| // this means we are potentially denying other more important fast tracks from |
| // being created. It would be better to allocate the index dynamically. |
| mFastIndex = i; |
| thread->mFastTrackAvailMask &= ~(1 << i); |
| } |
| } |
| |
| AudioFlinger::PlaybackThread::Track::~Track() |
| { |
| ALOGV("PlaybackThread::Track destructor"); |
| |
| // The destructor would clear mSharedBuffer, |
| // but it will not push the decremented reference count, |
| // leaving the client's IMemory dangling indefinitely. |
| // This prevents that leak. |
| if (mSharedBuffer != 0) { |
| mSharedBuffer.clear(); |
| } |
| } |
| |
| status_t AudioFlinger::PlaybackThread::Track::initCheck() const |
| { |
| status_t status = TrackBase::initCheck(); |
| if (status == NO_ERROR && mName < 0) { |
| status = NO_MEMORY; |
| } |
| return status; |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::destroy() |
| { |
| // NOTE: destroyTrack_l() can remove a strong reference to this Track |
| // by removing it from mTracks vector, so there is a risk that this Tracks's |
| // destructor is called. As the destructor needs to lock mLock, |
| // we must acquire a strong reference on this Track before locking mLock |
| // here so that the destructor is called only when exiting this function. |
| // On the other hand, as long as Track::destroy() is only called by |
| // TrackHandle destructor, the TrackHandle still holds a strong ref on |
| // this Track with its member mTrack. |
| sp<Track> keep(this); |
| { // scope for mLock |
| bool wasActive = false; |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| Mutex::Autolock _l(thread->mLock); |
| PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); |
| wasActive = playbackThread->destroyTrack_l(this); |
| } |
| if (isExternalTrack() && !wasActive) { |
| AudioSystem::releaseOutput(mThreadIoHandle, mStreamType, (audio_session_t)mSessionId); |
| } |
| } |
| } |
| |
| /*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) |
| { |
| result.append(" Name Active Client Type Fmt Chn mask Session fCount S F SRate " |
| "L dB R dB Server Main buf Aux Buf Flags UndFrmCnt\n"); |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active) |
| { |
| gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR(); |
| if (isFastTrack()) { |
| sprintf(buffer, " F %2d", mFastIndex); |
| } else if (mName >= AudioMixer::TRACK0) { |
| sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); |
| } else { |
| sprintf(buffer, " none"); |
| } |
| track_state state = mState; |
| char stateChar; |
| if (isTerminated()) { |
| stateChar = 'T'; |
| } else { |
| switch (state) { |
| case IDLE: |
| stateChar = 'I'; |
| break; |
| case STOPPING_1: |
| stateChar = 's'; |
| break; |
| case STOPPING_2: |
| stateChar = '5'; |
| break; |
| case STOPPED: |
| stateChar = 'S'; |
| break; |
| case RESUMING: |
| stateChar = 'R'; |
| break; |
| case ACTIVE: |
| stateChar = 'A'; |
| break; |
| case PAUSING: |
| stateChar = 'p'; |
| break; |
| case PAUSED: |
| stateChar = 'P'; |
| break; |
| case FLUSHED: |
| stateChar = 'F'; |
| break; |
| default: |
| stateChar = '?'; |
| break; |
| } |
| } |
| char nowInUnderrun; |
| switch (mObservedUnderruns.mBitFields.mMostRecent) { |
| case UNDERRUN_FULL: |
| nowInUnderrun = ' '; |
| break; |
| case UNDERRUN_PARTIAL: |
| nowInUnderrun = '<'; |
| break; |
| case UNDERRUN_EMPTY: |
| nowInUnderrun = '*'; |
| break; |
| default: |
| nowInUnderrun = '?'; |
| break; |
| } |
| snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g " |
| "%08X %p %p 0x%03X %9u%c\n", |
| active ? "yes" : "no", |
| (mClient == 0) ? getpid_cached : mClient->pid(), |
| mStreamType, |
| mFormat, |
| mChannelMask, |
| mSessionId, |
| mFrameCount, |
| stateChar, |
| mFillingUpStatus, |
| mAudioTrackServerProxy->getSampleRate(), |
| 20.0 * log10(float_from_gain(gain_minifloat_unpack_left(vlr))), |
| 20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))), |
| mCblk->mServer, |
| mMainBuffer, |
| mAuxBuffer, |
| mCblk->mFlags, |
| mAudioTrackServerProxy->getUnderrunFrames(), |
| nowInUnderrun); |
| } |
| |
| uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { |
| return mAudioTrackServerProxy->getSampleRate(); |
| } |
| |
| // AudioBufferProvider interface |
| status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( |
| AudioBufferProvider::Buffer* buffer, int64_t pts __unused) |
| { |
| ServerProxy::Buffer buf; |
| size_t desiredFrames = buffer->frameCount; |
| buf.mFrameCount = desiredFrames; |
| status_t status = mServerProxy->obtainBuffer(&buf); |
| buffer->frameCount = buf.mFrameCount; |
| buffer->raw = buf.mRaw; |
| if (buf.mFrameCount == 0) { |
| mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); |
| } |
| return status; |
| } |
| |
| // releaseBuffer() is not overridden |
| |
| // ExtendedAudioBufferProvider interface |
| |
| // framesReady() may return an approximation of the number of frames if called |
| // from a different thread than the one calling Proxy->obtainBuffer() and |
| // Proxy->releaseBuffer(). Also note there is no mutual exclusion in the |
| // AudioTrackServerProxy so be especially careful calling with FastTracks. |
| size_t AudioFlinger::PlaybackThread::Track::framesReady() const { |
| if (mSharedBuffer != 0 && (isStopped() || isStopping())) { |
| // Static tracks return zero frames immediately upon stopping (for FastTracks). |
| // The remainder of the buffer is not drained. |
| return 0; |
| } |
| return mAudioTrackServerProxy->framesReady(); |
| } |
| |
| size_t AudioFlinger::PlaybackThread::Track::framesReleased() const |
| { |
| return mAudioTrackServerProxy->framesReleased(); |
| } |
| |
| // Don't call for fast tracks; the framesReady() could result in priority inversion |
| bool AudioFlinger::PlaybackThread::Track::isReady() const { |
| if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { |
| return true; |
| } |
| |
| if (isStopping()) { |
| if (framesReady() > 0) { |
| mFillingUpStatus = FS_FILLED; |
| } |
| return true; |
| } |
| |
| if (framesReady() >= mFrameCount || |
| (mCblk->mFlags & CBLK_FORCEREADY)) { |
| mFillingUpStatus = FS_FILLED; |
| android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); |
| return true; |
| } |
| return false; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused, |
| int triggerSession __unused) |
| { |
| status_t status = NO_ERROR; |
| ALOGV("start(%d), calling pid %d session %d", |
| mName, IPCThreadState::self()->getCallingPid(), mSessionId); |
| |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| if (isOffloaded()) { |
| Mutex::Autolock _laf(thread->mAudioFlinger->mLock); |
| Mutex::Autolock _lth(thread->mLock); |
| sp<EffectChain> ec = thread->getEffectChain_l(mSessionId); |
| if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() || |
| (ec != 0 && ec->isNonOffloadableEnabled())) { |
| invalidate(); |
| return PERMISSION_DENIED; |
| } |
| } |
| Mutex::Autolock _lth(thread->mLock); |
| track_state state = mState; |
| // here the track could be either new, or restarted |
| // in both cases "unstop" the track |
| |
| // initial state-stopping. next state-pausing. |
| // What if resume is called ? |
| |
| if (state == PAUSED || state == PAUSING) { |
| if (mResumeToStopping) { |
| // happened we need to resume to STOPPING_1 |
| mState = TrackBase::STOPPING_1; |
| ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); |
| } else { |
| mState = TrackBase::RESUMING; |
| ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); |
| } |
| } else { |
| mState = TrackBase::ACTIVE; |
| ALOGV("? => ACTIVE (%d) on thread %p", mName, this); |
| } |
| |
| PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); |
| if (isFastTrack()) { |
| // refresh fast track underruns on start because that field is never cleared |
| // by the fast mixer; furthermore, the same track can be recycled, i.e. start |
| // after stop. |
| mObservedUnderruns = playbackThread->getFastTrackUnderruns(mFastIndex); |
| } |
| status = playbackThread->addTrack_l(this); |
| if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { |
| triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); |
| // restore previous state if start was rejected by policy manager |
| if (status == PERMISSION_DENIED) { |
| mState = state; |
| } |
| } |
| // 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; |
| } |
| return status; |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::stop() |
| { |
| ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| Mutex::Autolock _l(thread->mLock); |
| track_state state = mState; |
| if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { |
| // If the track is not active (PAUSED and buffers full), flush buffers |
| PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); |
| if (playbackThread->mActiveTracks.indexOf(this) < 0) { |
| reset(); |
| mState = STOPPED; |
| } else if (!isFastTrack() && !isOffloaded() && !isDirect()) { |
| mState = STOPPED; |
| } else { |
| // For fast tracks prepareTracks_l() will set state to STOPPING_2 |
| // presentation is complete |
| // For an offloaded track this starts a drain and state will |
| // move to STOPPING_2 when drain completes and then STOPPED |
| mState = STOPPING_1; |
| } |
| playbackThread->broadcast_l(); |
| ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, |
| playbackThread); |
| } |
| } |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::pause() |
| { |
| ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| Mutex::Autolock _l(thread->mLock); |
| PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); |
| switch (mState) { |
| case STOPPING_1: |
| case STOPPING_2: |
| if (!isOffloaded()) { |
| /* nothing to do if track is not offloaded */ |
| break; |
| } |
| |
| // Offloaded track was draining, we need to carry on draining when resumed |
| mResumeToStopping = true; |
| // fall through... |
| case ACTIVE: |
| case RESUMING: |
| mState = PAUSING; |
| ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); |
| playbackThread->broadcast_l(); |
| break; |
| |
| default: |
| break; |
| } |
| } |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::flush() |
| { |
| ALOGV("flush(%d)", mName); |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| Mutex::Autolock _l(thread->mLock); |
| PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); |
| |
| if (isOffloaded()) { |
| // If offloaded we allow flush during any state except terminated |
| // and keep the track active to avoid problems if user is seeking |
| // rapidly and underlying hardware has a significant delay handling |
| // a pause |
| if (isTerminated()) { |
| return; |
| } |
| |
| ALOGV("flush: offload flush"); |
| reset(); |
| |
| if (mState == STOPPING_1 || mState == STOPPING_2) { |
| ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); |
| mState = ACTIVE; |
| } |
| |
| if (mState == ACTIVE) { |
| ALOGV("flush called in active state, resetting buffer time out retry count"); |
| mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; |
| } |
| |
| mFlushHwPending = true; |
| mResumeToStopping = false; |
| } else { |
| if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && |
| mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { |
| return; |
| } |
| // No point remaining in PAUSED state after a flush => go to |
| // FLUSHED state |
| mState = FLUSHED; |
| // do not reset the track if it is still in the process of being stopped or paused. |
| // this will be done by prepareTracks_l() when the track is stopped. |
| // prepareTracks_l() will see mState == FLUSHED, then |
| // remove from active track list, reset(), and trigger presentation complete |
| if (isDirect()) { |
| mFlushHwPending = true; |
| } |
| if (playbackThread->mActiveTracks.indexOf(this) < 0) { |
| reset(); |
| } |
| } |
| // Prevent flush being lost if the track is flushed and then resumed |
| // before mixer thread can run. This is important when offloading |
| // because the hardware buffer could hold a large amount of audio |
| playbackThread->broadcast_l(); |
| } |
| } |
| |
| // must be called with thread lock held |
| void AudioFlinger::PlaybackThread::Track::flushAck() |
| { |
| if (!isOffloaded() && !isDirect()) |
| return; |
| |
| mFlushHwPending = false; |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::reset() |
| { |
| // Do not reset twice to avoid discarding data written just after a flush and before |
| // the audioflinger thread detects the track is stopped. |
| if (!mResetDone) { |
| // Force underrun condition to avoid false underrun callback until first data is |
| // written to buffer |
| android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); |
| mFillingUpStatus = FS_FILLING; |
| mResetDone = true; |
| if (mState == FLUSHED) { |
| mState = IDLE; |
| } |
| } |
| } |
| |
| status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) |
| { |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread == 0) { |
| ALOGE("thread is dead"); |
| return FAILED_TRANSACTION; |
| } else if ((thread->type() == ThreadBase::DIRECT) || |
| (thread->type() == ThreadBase::OFFLOAD)) { |
| return thread->setParameters(keyValuePairs); |
| } else { |
| return PERMISSION_DENIED; |
| } |
| } |
| |
| status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp) |
| { |
| // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant |
| if (isFastTrack()) { |
| return INVALID_OPERATION; |
| } |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread == 0) { |
| return INVALID_OPERATION; |
| } |
| |
| Mutex::Autolock _l(thread->mLock); |
| PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); |
| |
| status_t result = INVALID_OPERATION; |
| if (!isOffloaded() && !isDirect()) { |
| if (!playbackThread->mLatchQValid) { |
| return INVALID_OPERATION; |
| } |
| // FIXME Not accurate under dynamic changes of sample rate and speed. |
| // Do not use track's mSampleRate as it is not current for mixer tracks. |
| uint32_t sampleRate = mAudioTrackServerProxy->getSampleRate(); |
| AudioPlaybackRate playbackRate = mAudioTrackServerProxy->getPlaybackRate(); |
| uint32_t unpresentedFrames = ((double) playbackThread->mLatchQ.mUnpresentedFrames * |
| sampleRate * playbackRate.mSpeed)/ playbackThread->mSampleRate; |
| // FIXME Since we're using a raw pointer as the key, it is theoretically possible |
| // for a brand new track to share the same address as a recently destroyed |
| // track, and thus for us to get the frames released of the wrong track. |
| // It is unlikely that we would be able to call getTimestamp() so quickly |
| // right after creating a new track. Nevertheless, the index here should |
| // be changed to something that is unique. Or use a completely different strategy. |
| ssize_t i = playbackThread->mLatchQ.mFramesReleased.indexOfKey(this); |
| uint32_t framesWritten = i >= 0 ? |
| playbackThread->mLatchQ.mFramesReleased[i] : |
| mAudioTrackServerProxy->framesReleased(); |
| if (framesWritten >= unpresentedFrames) { |
| timestamp.mPosition = framesWritten - unpresentedFrames; |
| timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime; |
| result = NO_ERROR; |
| } |
| } else { // offloaded or direct |
| result = playbackThread->getTimestamp_l(timestamp); |
| } |
| |
| return result; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) |
| { |
| status_t status = DEAD_OBJECT; |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); |
| sp<AudioFlinger> af = mClient->audioFlinger(); |
| |
| Mutex::Autolock _l(af->mLock); |
| |
| sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); |
| |
| if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { |
| Mutex::Autolock _dl(playbackThread->mLock); |
| Mutex::Autolock _sl(srcThread->mLock); |
| sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); |
| if (chain == 0) { |
| return INVALID_OPERATION; |
| } |
| |
| sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); |
| if (effect == 0) { |
| return INVALID_OPERATION; |
| } |
| srcThread->removeEffect_l(effect); |
| status = playbackThread->addEffect_l(effect); |
| if (status != NO_ERROR) { |
| srcThread->addEffect_l(effect); |
| return INVALID_OPERATION; |
| } |
| // removeEffect_l() has stopped the effect if it was active so it must be restarted |
| if (effect->state() == EffectModule::ACTIVE || |
| effect->state() == EffectModule::STOPPING) { |
| effect->start(); |
| } |
| |
| sp<EffectChain> dstChain = effect->chain().promote(); |
| if (dstChain == 0) { |
| srcThread->addEffect_l(effect); |
| return INVALID_OPERATION; |
| } |
| AudioSystem::unregisterEffect(effect->id()); |
| AudioSystem::registerEffect(&effect->desc(), |
| srcThread->id(), |
| dstChain->strategy(), |
| AUDIO_SESSION_OUTPUT_MIX, |
| effect->id()); |
| AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); |
| } |
| status = playbackThread->attachAuxEffect(this, EffectId); |
| } |
| return status; |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) |
| { |
| mAuxEffectId = EffectId; |
| mAuxBuffer = buffer; |
| } |
| |
| bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, |
| size_t audioHalFrames) |
| { |
| // a track is considered presented when the total number of frames written to audio HAL |
| // corresponds to the number of frames written when presentationComplete() is called for the |
| // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. |
| // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used |
| // to detect when all frames have been played. In this case framesWritten isn't |
| // useful because it doesn't always reflect whether there is data in the h/w |
| // buffers, particularly if a track has been paused and resumed during draining |
| ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", |
| mPresentationCompleteFrames, framesWritten); |
| if (mPresentationCompleteFrames == 0) { |
| mPresentationCompleteFrames = framesWritten + audioHalFrames; |
| ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", |
| mPresentationCompleteFrames, audioHalFrames); |
| } |
| |
| if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { |
| triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); |
| mAudioTrackServerProxy->setStreamEndDone(); |
| return true; |
| } |
| return false; |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) |
| { |
| for (size_t i = 0; i < mSyncEvents.size(); i++) { |
| if (mSyncEvents[i]->type() == type) { |
| mSyncEvents[i]->trigger(); |
| mSyncEvents.removeAt(i); |
| i--; |
| } |
| } |
| } |
| |
| // implement VolumeBufferProvider interface |
| |
| gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR() |
| { |
| // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs |
| ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); |
| gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR(); |
| float vl = float_from_gain(gain_minifloat_unpack_left(vlr)); |
| float vr = float_from_gain(gain_minifloat_unpack_right(vlr)); |
| // track volumes come from shared memory, so can't be trusted and must be clamped |
| if (vl > GAIN_FLOAT_UNITY) { |
| vl = GAIN_FLOAT_UNITY; |
| } |
| if (vr > GAIN_FLOAT_UNITY) { |
| vr = GAIN_FLOAT_UNITY; |
| } |
| // now apply the cached master volume and stream type volume; |
| // this is trusted but lacks any synchronization or barrier so may be stale |
| float v = mCachedVolume; |
| vl *= v; |
| vr *= v; |
| // re-combine into packed minifloat |
| vlr = gain_minifloat_pack(gain_from_float(vl), gain_from_float(vr)); |
| // FIXME look at mute, pause, and stop flags |
| return vlr; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) |
| { |
| if (isTerminated() || mState == PAUSED || |
| ((framesReady() == 0) && ((mSharedBuffer != 0) || |
| (mState == STOPPED)))) { |
| ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", |
| mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); |
| event->cancel(); |
| return INVALID_OPERATION; |
| } |
| (void) TrackBase::setSyncEvent(event); |
| return NO_ERROR; |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::invalidate() |
| { |
| // FIXME should use proxy, and needs work |
| audio_track_cblk_t* cblk = mCblk; |
| android_atomic_or(CBLK_INVALID, &cblk->mFlags); |
| android_atomic_release_store(0x40000000, &cblk->mFutex); |
| // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE |
| (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX); |
| mIsInvalid = true; |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::signal() |
| { |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| PlaybackThread *t = (PlaybackThread *)thread.get(); |
| Mutex::Autolock _l(t->mLock); |
| t->broadcast_l(); |
| } |
| } |
| |
| //To be called with thread lock held |
| bool AudioFlinger::PlaybackThread::Track::isResumePending() { |
| |
| if (mState == RESUMING) |
| return true; |
| /* Resume is pending if track was stopping before pause was called */ |
| if (mState == STOPPING_1 && |
| mResumeToStopping) |
| return true; |
| |
| return false; |
| } |
| |
| //To be called with thread lock held |
| void AudioFlinger::PlaybackThread::Track::resumeAck() { |
| |
| |
| if (mState == RESUMING) |
| mState = ACTIVE; |
| |
| // Other possibility of pending resume is stopping_1 state |
| // Do not update the state from stopping as this prevents |
| // drain being called. |
| if (mState == STOPPING_1) { |
| mResumeToStopping = false; |
| } |
| } |
| // ---------------------------------------------------------------------------- |
| |
| sp<AudioFlinger::PlaybackThread::TimedTrack> |
| AudioFlinger::PlaybackThread::TimedTrack::create( |
| PlaybackThread *thread, |
| const sp<Client>& client, |
| audio_stream_type_t streamType, |
| uint32_t sampleRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| size_t frameCount, |
| const sp<IMemory>& sharedBuffer, |
| int sessionId, |
| int uid) |
| { |
| if (!client->reserveTimedTrack()) |
| return 0; |
| |
| return new TimedTrack( |
| thread, client, streamType, sampleRate, format, channelMask, frameCount, |
| sharedBuffer, sessionId, uid); |
| } |
| |
| AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( |
| PlaybackThread *thread, |
| const sp<Client>& client, |
| audio_stream_type_t streamType, |
| uint32_t sampleRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| size_t frameCount, |
| const sp<IMemory>& sharedBuffer, |
| int sessionId, |
| int uid) |
| : Track(thread, client, streamType, sampleRate, format, channelMask, |
| frameCount, (sharedBuffer != 0) ? sharedBuffer->pointer() : NULL, sharedBuffer, |
| sessionId, uid, IAudioFlinger::TRACK_TIMED, TYPE_TIMED), |
| mQueueHeadInFlight(false), |
| mTrimQueueHeadOnRelease(false), |
| mFramesPendingInQueue(0), |
| mTimedSilenceBuffer(NULL), |
| mTimedSilenceBufferSize(0), |
| mTimedAudioOutputOnTime(false), |
| mMediaTimeTransformValid(false) |
| { |
| LocalClock lc; |
| mLocalTimeFreq = lc.getLocalFreq(); |
| |
| mLocalTimeToSampleTransform.a_zero = 0; |
| mLocalTimeToSampleTransform.b_zero = 0; |
| mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; |
| mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; |
| LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, |
| &mLocalTimeToSampleTransform.a_to_b_denom); |
| |
| mMediaTimeToSampleTransform.a_zero = 0; |
| mMediaTimeToSampleTransform.b_zero = 0; |
| mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; |
| mMediaTimeToSampleTransform.a_to_b_denom = 1000000; |
| LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, |
| &mMediaTimeToSampleTransform.a_to_b_denom); |
| } |
| |
| AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { |
| mClient->releaseTimedTrack(); |
| delete [] mTimedSilenceBuffer; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( |
| size_t size, sp<IMemory>* buffer) { |
| |
| Mutex::Autolock _l(mTimedBufferQueueLock); |
| |
| trimTimedBufferQueue_l(); |
| |
| // lazily initialize the shared memory heap for timed buffers |
| if (mTimedMemoryDealer == NULL) { |
| const int kTimedBufferHeapSize = 512 << 10; |
| |
| mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, |
| "AudioFlingerTimed"); |
| if (mTimedMemoryDealer == NULL) { |
| return NO_MEMORY; |
| } |
| } |
| |
| sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); |
| if (newBuffer == 0 || newBuffer->pointer() == NULL) { |
| return NO_MEMORY; |
| } |
| |
| *buffer = newBuffer; |
| return NO_ERROR; |
| } |
| |
| // caller must hold mTimedBufferQueueLock |
| void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { |
| int64_t mediaTimeNow; |
| { |
| Mutex::Autolock mttLock(mMediaTimeTransformLock); |
| if (!mMediaTimeTransformValid) |
| return; |
| |
| int64_t targetTimeNow; |
| status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) |
| ? mCCHelper.getCommonTime(&targetTimeNow) |
| : mCCHelper.getLocalTime(&targetTimeNow); |
| |
| if (OK != res) |
| return; |
| |
| if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, |
| &mediaTimeNow)) { |
| return; |
| } |
| } |
| |
| size_t trimEnd; |
| for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { |
| int64_t bufEnd; |
| |
| if ((trimEnd + 1) < mTimedBufferQueue.size()) { |
| // We have a next buffer. Just use its PTS as the PTS of the frame |
| // following the last frame in this buffer. If the stream is sparse |
| // (ie, there are deliberate gaps left in the stream which should be |
| // filled with silence by the TimedAudioTrack), then this can result |
| // in one extra buffer being left un-trimmed when it could have |
| // been. In general, this is not typical, and we would rather |
| // optimized away the TS calculation below for the more common case |
| // where PTSes are contiguous. |
| bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); |
| } else { |
| // We have no next buffer. Compute the PTS of the frame following |
| // the last frame in this buffer by computing the duration of of |
| // this frame in media time units and adding it to the PTS of the |
| // buffer. |
| int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() |
| / mFrameSize; |
| |
| if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, |
| &bufEnd)) { |
| ALOGE("Failed to convert frame count of %lld to media time" |
| " duration" " (scale factor %d/%u) in %s", |
| frameCount, |
| mMediaTimeToSampleTransform.a_to_b_numer, |
| mMediaTimeToSampleTransform.a_to_b_denom, |
| __PRETTY_FUNCTION__); |
| break; |
| } |
| bufEnd += mTimedBufferQueue[trimEnd].pts(); |
| } |
| |
| if (bufEnd > mediaTimeNow) |
| break; |
| |
| // Is the buffer we want to use in the middle of a mix operation right |
| // now? If so, don't actually trim it. Just wait for the releaseBuffer |
| // from the mixer which should be coming back shortly. |
| if (!trimEnd && mQueueHeadInFlight) { |
| mTrimQueueHeadOnRelease = true; |
| } |
| } |
| |
| size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; |
| if (trimStart < trimEnd) { |
| // Update the bookkeeping for framesReady() |
| for (size_t i = trimStart; i < trimEnd; ++i) { |
| updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); |
| } |
| |
| // Now actually remove the buffers from the queue. |
| mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); |
| } |
| } |
| |
| void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( |
| const char* logTag) { |
| ALOG_ASSERT(mTimedBufferQueue.size() > 0, |
| "%s called (reason \"%s\"), but timed buffer queue has no" |
| " elements to trim.", __FUNCTION__, logTag); |
| |
| updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); |
| mTimedBufferQueue.removeAt(0); |
| } |
| |
| void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( |
| const TimedBuffer& buf, |
| const char* logTag __unused) { |
| uint32_t bufBytes = buf.buffer()->size(); |
| uint32_t consumedAlready = buf.position(); |
| |
| ALOG_ASSERT(consumedAlready <= bufBytes, |
| "Bad bookkeeping while updating frames pending. Timed buffer is" |
| " only %u bytes long, but claims to have consumed %u" |
| " bytes. (update reason: \"%s\")", |
| bufBytes, consumedAlready, logTag); |
| |
| uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; |
| ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, |
| "Bad bookkeeping while updating frames pending. Should have at" |
| " least %u queued frames, but we think we have only %u. (update" |
| " reason: \"%s\")", |
| bufFrames, mFramesPendingInQueue, logTag); |
| |
| mFramesPendingInQueue -= bufFrames; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( |
| const sp<IMemory>& buffer, int64_t pts) { |
| |
| { |
| Mutex::Autolock mttLock(mMediaTimeTransformLock); |
| if (!mMediaTimeTransformValid) |
| return INVALID_OPERATION; |
| } |
| |
| Mutex::Autolock _l(mTimedBufferQueueLock); |
| |
| uint32_t bufFrames = buffer->size() / mFrameSize; |
| mFramesPendingInQueue += bufFrames; |
| mTimedBufferQueue.add(TimedBuffer(buffer, pts)); |
| |
| return NO_ERROR; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( |
| const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { |
| |
| ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", |
| xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, |
| target); |
| |
| if (!(target == TimedAudioTrack::LOCAL_TIME || |
| target == TimedAudioTrack::COMMON_TIME)) { |
| return BAD_VALUE; |
| } |
| |
| Mutex::Autolock lock(mMediaTimeTransformLock); |
| mMediaTimeTransform = xform; |
| mMediaTimeTransformTarget = target; |
| mMediaTimeTransformValid = true; |
| |
| return NO_ERROR; |
| } |
| |
| #define min(a, b) ((a) < (b) ? (a) : (b)) |
| |
| // implementation of getNextBuffer for tracks whose buffers have timestamps |
| status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( |
| AudioBufferProvider::Buffer* buffer, int64_t pts) |
| { |
| if (pts == AudioBufferProvider::kInvalidPTS) { |
| buffer->raw = NULL; |
| buffer->frameCount = 0; |
| mTimedAudioOutputOnTime = false; |
| return INVALID_OPERATION; |
| } |
| |
| Mutex::Autolock _l(mTimedBufferQueueLock); |
| |
| ALOG_ASSERT(!mQueueHeadInFlight, |
| "getNextBuffer called without releaseBuffer!"); |
| |
| while (true) { |
| |
| // if we have no timed buffers, then fail |
| if (mTimedBufferQueue.isEmpty()) { |
| buffer->raw = NULL; |
| buffer->frameCount = 0; |
| return NOT_ENOUGH_DATA; |
| } |
| |
| TimedBuffer& head = mTimedBufferQueue.editItemAt(0); |
| |
| // calculate the PTS of the head of the timed buffer queue expressed in |
| // local time |
| int64_t headLocalPTS; |
| { |
| Mutex::Autolock mttLock(mMediaTimeTransformLock); |
| |
| ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); |
| |
| if (mMediaTimeTransform.a_to_b_denom == 0) { |
| // the transform represents a pause, so yield silence |
| timedYieldSilence_l(buffer->frameCount, buffer); |
| return NO_ERROR; |
| } |
| |
| int64_t transformedPTS; |
| if (!mMediaTimeTransform.doForwardTransform(head.pts(), |
| &transformedPTS)) { |
| // the transform failed. this shouldn't happen, but if it does |
| // then just drop this buffer |
| ALOGW("timedGetNextBuffer transform failed"); |
| buffer->raw = NULL; |
| buffer->frameCount = 0; |
| trimTimedBufferQueueHead_l("getNextBuffer; no transform"); |
| return NO_ERROR; |
| } |
| |
| if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { |
| if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, |
| &headLocalPTS)) { |
| buffer->raw = NULL; |
| buffer->frameCount = 0; |
| return INVALID_OPERATION; |
| } |
| } else { |
| headLocalPTS = transformedPTS; |
| } |
| } |
| |
| uint32_t sr = sampleRate(); |
| |
| // adjust the head buffer's PTS to reflect the portion of the head buffer |
| // that has already been consumed |
| int64_t effectivePTS = headLocalPTS + |
| ((head.position() / mFrameSize) * mLocalTimeFreq / sr); |
| |
| // Calculate the delta in samples between the head of the input buffer |
| // queue and the start of the next output buffer that will be written. |
| // If the transformation fails because of over or underflow, it means |
| // that the sample's position in the output stream is so far out of |
| // whack that it should just be dropped. |
| int64_t sampleDelta; |
| if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { |
| ALOGV("*** head buffer is too far from PTS: dropped buffer"); |
| trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" |
| " mix"); |
| continue; |
| } |
| if (!mLocalTimeToSampleTransform.doForwardTransform( |
| (effectivePTS - pts) << 32, &sampleDelta)) { |
| ALOGV("*** too late during sample rate transform: dropped buffer"); |
| trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); |
| continue; |
| } |
| |
| ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" |
| " sampleDelta=[%d.%08x]", |
| head.pts(), head.position(), pts, |
| static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) |
| + (sampleDelta >> 32)), |
| static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); |
| |
| // if the delta between the ideal placement for the next input sample and |
| // the current output position is within this threshold, then we will |
| // concatenate the next input samples to the previous output |
| const int64_t kSampleContinuityThreshold = |
| (static_cast<int64_t>(sr) << 32) / 250; |
| |
| // if this is the first buffer of audio that we're emitting from this track |
| // then it should be almost exactly on time. |
| const int64_t kSampleStartupThreshold = 1LL << 32; |
| |
| if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || |
| (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { |
| // the next input is close enough to being on time, so concatenate it |
| // with the last output |
| timedYieldSamples_l(buffer); |
| |
| ALOGVV("*** on time: head.pos=%d frameCount=%u", |
| head.position(), buffer->frameCount); |
| return NO_ERROR; |
| } |
| |
| // Looks like our output is not on time. Reset our on timed status. |
| // Next time we mix samples from our input queue, then should be within |
| // the StartupThreshold. |
| mTimedAudioOutputOnTime = false; |
| if (sampleDelta > 0) { |
| // the gap between the current output position and the proper start of |
| // the next input sample is too big, so fill it with silence |
| uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; |
| |
| timedYieldSilence_l(framesUntilNextInput, buffer); |
| ALOGV("*** silence: frameCount=%u", buffer->frameCount); |
| return NO_ERROR; |
| } else { |
| // the next input sample is late |
| uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); |
| size_t onTimeSamplePosition = |
| head.position() + lateFrames * mFrameSize; |
| |
| if (onTimeSamplePosition > head.buffer()->size()) { |
| // all the remaining samples in the head are too late, so |
| // drop it and move on |
| ALOGV("*** too late: dropped buffer"); |
| trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); |
| continue; |
| } else { |
| // skip over the late samples |
| head.setPosition(onTimeSamplePosition); |
| |
| // yield the available samples |
| timedYieldSamples_l(buffer); |
| |
| ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); |
| return NO_ERROR; |
| } |
| } |
| } |
| } |
| |
| // Yield samples from the timed buffer queue head up to the given output |
| // buffer's capacity. |
| // |
| // Caller must hold mTimedBufferQueueLock |
| void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( |
| AudioBufferProvider::Buffer* buffer) { |
| |
| const TimedBuffer& head = mTimedBufferQueue[0]; |
| |
| buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + |
| head.position()); |
| |
| uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / |
| mFrameSize); |
| size_t framesRequested = buffer->frameCount; |
| buffer->frameCount = min(framesLeftInHead, framesRequested); |
| |
| mQueueHeadInFlight = true; |
| mTimedAudioOutputOnTime = true; |
| } |
| |
| // Yield samples of silence up to the given output buffer's capacity |
| // |
| // Caller must hold mTimedBufferQueueLock |
| void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( |
| uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { |
| |
| // lazily allocate a buffer filled with silence |
| if (mTimedSilenceBufferSize < numFrames * mFrameSize) { |
| delete [] mTimedSilenceBuffer; |
| mTimedSilenceBufferSize = numFrames * mFrameSize; |
| mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; |
| memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); |
| } |
| |
| buffer->raw = mTimedSilenceBuffer; |
| size_t framesRequested = buffer->frameCount; |
| buffer->frameCount = min(numFrames, framesRequested); |
| |
| mTimedAudioOutputOnTime = false; |
| } |
| |
| // AudioBufferProvider interface |
| void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( |
| AudioBufferProvider::Buffer* buffer) { |
| |
| Mutex::Autolock _l(mTimedBufferQueueLock); |
| |
| // If the buffer which was just released is part of the buffer at the head |
| // of the queue, be sure to update the amt of the buffer which has been |
| // consumed. If the buffer being returned is not part of the head of the |
| // queue, its either because the buffer is part of the silence buffer, or |
| // because the head of the timed queue was trimmed after the mixer called |
| // getNextBuffer but before the mixer called releaseBuffer. |
| if (buffer->raw == mTimedSilenceBuffer) { |
| ALOG_ASSERT(!mQueueHeadInFlight, |
| "Queue head in flight during release of silence buffer!"); |
| goto done; |
| } |
| |
| ALOG_ASSERT(mQueueHeadInFlight, |
| "TimedTrack::releaseBuffer of non-silence buffer, but no queue" |
| " head in flight."); |
| |
| if (mTimedBufferQueue.size()) { |
| TimedBuffer& head = mTimedBufferQueue.editItemAt(0); |
| |
| void* start = head.buffer()->pointer(); |
| void* end = reinterpret_cast<void*>( |
| reinterpret_cast<uint8_t*>(head.buffer()->pointer()) |
| + head.buffer()->size()); |
| |
| ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), |
| "released buffer not within the head of the timed buffer" |
| " queue; qHead = [%p, %p], released buffer = %p", |
| start, end, buffer->raw); |
| |
| head.setPosition(head.position() + |
| (buffer->frameCount * mFrameSize)); |
| mQueueHeadInFlight = false; |
| |
| ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, |
| "Bad bookkeeping during releaseBuffer! Should have at" |
| " least %u queued frames, but we think we have only %u", |
| buffer->frameCount, mFramesPendingInQueue); |
| |
| mFramesPendingInQueue -= buffer->frameCount; |
| |
| if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) |
| || mTrimQueueHeadOnRelease) { |
| trimTimedBufferQueueHead_l("releaseBuffer"); |
| mTrimQueueHeadOnRelease = false; |
| } |
| } else { |
| LOG_ALWAYS_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" |
| " buffers in the timed buffer queue"); |
| } |
| |
| done: |
| buffer->raw = 0; |
| buffer->frameCount = 0; |
| } |
| |
| size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { |
| Mutex::Autolock _l(mTimedBufferQueueLock); |
| return mFramesPendingInQueue; |
| } |
| |
| AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() |
| : mPTS(0), mPosition(0) {} |
| |
| AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( |
| const sp<IMemory>& buffer, int64_t pts) |
| : mBuffer(buffer), mPTS(pts), mPosition(0) {} |
| |
| |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( |
| PlaybackThread *playbackThread, |
| DuplicatingThread *sourceThread, |
| uint32_t sampleRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| size_t frameCount, |
| int uid) |
| : Track(playbackThread, NULL, AUDIO_STREAM_PATCH, |
| sampleRate, format, channelMask, frameCount, |
| NULL, 0, 0, uid, IAudioFlinger::TRACK_DEFAULT, TYPE_OUTPUT), |
| mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) |
| { |
| |
| if (mCblk != NULL) { |
| mOutBuffer.frameCount = 0; |
| playbackThread->mTracks.add(this); |
| ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " |
| "frameCount %u, mChannelMask 0x%08x", |
| mCblk, mBuffer, |
| frameCount, mChannelMask); |
| // since client and server are in the same process, |
| // the buffer has the same virtual address on both sides |
| mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, |
| true /*clientInServer*/); |
| mClientProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); |
| mClientProxy->setSendLevel(0.0); |
| mClientProxy->setSampleRate(sampleRate); |
| } else { |
| ALOGW("Error creating output track on thread %p", playbackThread); |
| } |
| } |
| |
| AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() |
| { |
| clearBufferQueue(); |
| delete mClientProxy; |
| // superclass destructor will now delete the server proxy and shared memory both refer to |
| } |
| |
| status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, |
| int triggerSession) |
| { |
| status_t status = Track::start(event, triggerSession); |
| if (status != NO_ERROR) { |
| return status; |
| } |
| |
| mActive = true; |
| mRetryCount = 127; |
| return status; |
| } |
| |
| void AudioFlinger::PlaybackThread::OutputTrack::stop() |
| { |
| Track::stop(); |
| clearBufferQueue(); |
| mOutBuffer.frameCount = 0; |
| mActive = false; |
| } |
| |
| bool AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t frames) |
| { |
| Buffer *pInBuffer; |
| Buffer inBuffer; |
| bool outputBufferFull = false; |
| inBuffer.frameCount = frames; |
| inBuffer.raw = data; |
| |
| uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); |
| |
| if (!mActive && frames != 0) { |
| (void) start(); |
| } |
| |
| while (waitTimeLeftMs) { |
| // First write pending buffers, then new data |
| if (mBufferQueue.size()) { |
| pInBuffer = mBufferQueue.itemAt(0); |
| } else { |
| pInBuffer = &inBuffer; |
| } |
| |
| if (pInBuffer->frameCount == 0) { |
| break; |
| } |
| |
| if (mOutBuffer.frameCount == 0) { |
| mOutBuffer.frameCount = pInBuffer->frameCount; |
| nsecs_t startTime = systemTime(); |
| status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); |
| if (status != NO_ERROR) { |
| ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, |
| mThread.unsafe_get(), status); |
| outputBufferFull = true; |
| break; |
| } |
| uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); |
| if (waitTimeLeftMs >= waitTimeMs) { |
| waitTimeLeftMs -= waitTimeMs; |
| } else { |
| waitTimeLeftMs = 0; |
| } |
| } |
| |
| uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : |
| pInBuffer->frameCount; |
| memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * mFrameSize); |
| Proxy::Buffer buf; |
| buf.mFrameCount = outFrames; |
| buf.mRaw = NULL; |
| mClientProxy->releaseBuffer(&buf); |
| pInBuffer->frameCount -= outFrames; |
| pInBuffer->raw = (int8_t *)pInBuffer->raw + outFrames * mFrameSize; |
| mOutBuffer.frameCount -= outFrames; |
| mOutBuffer.raw = (int8_t *)mOutBuffer.raw + outFrames * mFrameSize; |
| |
| if (pInBuffer->frameCount == 0) { |
| if (mBufferQueue.size()) { |
| mBufferQueue.removeAt(0); |
| free(pInBuffer->mBuffer); |
| delete pInBuffer; |
| ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, |
| mThread.unsafe_get(), mBufferQueue.size()); |
| } else { |
| break; |
| } |
| } |
| } |
| |
| // If we could not write all frames, allocate a buffer and queue it for next time. |
| if (inBuffer.frameCount) { |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0 && !thread->standby()) { |
| if (mBufferQueue.size() < kMaxOverFlowBuffers) { |
| pInBuffer = new Buffer; |
| pInBuffer->mBuffer = malloc(inBuffer.frameCount * mFrameSize); |
| pInBuffer->frameCount = inBuffer.frameCount; |
| pInBuffer->raw = pInBuffer->mBuffer; |
| memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * mFrameSize); |
| mBufferQueue.add(pInBuffer); |
| 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); |
| } |
| } |
| } |
| |
| // Calling write() with a 0 length buffer means that no more data will be written: |
| // We rely on stop() to set the appropriate flags to allow the remaining frames to play out. |
| if (frames == 0 && mBufferQueue.size() == 0 && mActive) { |
| stop(); |
| } |
| |
| return outputBufferFull; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( |
| AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) |
| { |
| ClientProxy::Buffer buf; |
| buf.mFrameCount = buffer->frameCount; |
| struct timespec timeout; |
| timeout.tv_sec = waitTimeMs / 1000; |
| timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; |
| status_t status = mClientProxy->obtainBuffer(&buf, &timeout); |
| buffer->frameCount = buf.mFrameCount; |
| buffer->raw = buf.mRaw; |
| return status; |
| } |
| |
| void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() |
| { |
| size_t size = mBufferQueue.size(); |
| |
| for (size_t i = 0; i < size; i++) { |
| Buffer *pBuffer = mBufferQueue.itemAt(i); |
| free(pBuffer->mBuffer); |
| delete pBuffer; |
| } |
| mBufferQueue.clear(); |
| } |
| |
| |
| AudioFlinger::PlaybackThread::PatchTrack::PatchTrack(PlaybackThread *playbackThread, |
| audio_stream_type_t streamType, |
| uint32_t sampleRate, |
| audio_channel_mask_t channelMask, |
| audio_format_t format, |
| size_t frameCount, |
| void *buffer, |
| IAudioFlinger::track_flags_t flags) |
| : Track(playbackThread, NULL, streamType, |
| sampleRate, format, channelMask, frameCount, |
| buffer, 0, 0, getuid(), flags, TYPE_PATCH), |
| mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true)) |
| { |
| uint64_t mixBufferNs = ((uint64_t)2 * playbackThread->frameCount() * 1000000000) / |
| playbackThread->sampleRate(); |
| mPeerTimeout.tv_sec = mixBufferNs / 1000000000; |
| mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000); |
| |
| ALOGV("PatchTrack %p sampleRate %d mPeerTimeout %d.%03d sec", |
| this, sampleRate, |
| (int)mPeerTimeout.tv_sec, |
| (int)(mPeerTimeout.tv_nsec / 1000000)); |
| } |
| |
| AudioFlinger::PlaybackThread::PatchTrack::~PatchTrack() |
| { |
| } |
| |
| // AudioBufferProvider interface |
| status_t AudioFlinger::PlaybackThread::PatchTrack::getNextBuffer( |
| AudioBufferProvider::Buffer* buffer, int64_t pts) |
| { |
| ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::getNextBuffer() called without peer proxy"); |
| Proxy::Buffer buf; |
| buf.mFrameCount = buffer->frameCount; |
| status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout); |
| ALOGV_IF(status != NO_ERROR, "PatchTrack() %p getNextBuffer status %d", this, status); |
| buffer->frameCount = buf.mFrameCount; |
| if (buf.mFrameCount == 0) { |
| return WOULD_BLOCK; |
| } |
| status = Track::getNextBuffer(buffer, pts); |
| return status; |
| } |
| |
| void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(AudioBufferProvider::Buffer* buffer) |
| { |
| ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::releaseBuffer() called without peer proxy"); |
| Proxy::Buffer buf; |
| buf.mFrameCount = buffer->frameCount; |
| buf.mRaw = buffer->raw; |
| mPeerProxy->releaseBuffer(&buf); |
| TrackBase::releaseBuffer(buffer); |
| } |
| |
| status_t AudioFlinger::PlaybackThread::PatchTrack::obtainBuffer(Proxy::Buffer* buffer, |
| const struct timespec *timeOut) |
| { |
| return mProxy->obtainBuffer(buffer, timeOut); |
| } |
| |
| void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(Proxy::Buffer* buffer) |
| { |
| mProxy->releaseBuffer(buffer); |
| if (android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags) & CBLK_DISABLED) { |
| ALOGW("PatchTrack::releaseBuffer() disabled due to previous underrun, restarting"); |
| start(); |
| } |
| android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // Record |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::RecordHandle::RecordHandle( |
| const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) |
| : BnAudioRecord(), |
| mRecordTrack(recordTrack) |
| { |
| } |
| |
| AudioFlinger::RecordHandle::~RecordHandle() { |
| stop_nonvirtual(); |
| mRecordTrack->destroy(); |
| } |
| |
| 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); |
| } |
| |
| void AudioFlinger::RecordHandle::stop() { |
| stop_nonvirtual(); |
| } |
| |
| void AudioFlinger::RecordHandle::stop_nonvirtual() { |
| ALOGV("RecordHandle::stop()"); |
| mRecordTrack->stop(); |
| } |
| |
| status_t AudioFlinger::RecordHandle::onTransact( |
| uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) |
| { |
| return BnAudioRecord::onTransact(code, data, reply, flags); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| // RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held |
| AudioFlinger::RecordThread::RecordTrack::RecordTrack( |
| RecordThread *thread, |
| const sp<Client>& client, |
| uint32_t sampleRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| size_t frameCount, |
| void *buffer, |
| int sessionId, |
| int uid, |
| IAudioFlinger::track_flags_t flags, |
| track_type type) |
| : TrackBase(thread, client, sampleRate, format, |
| channelMask, frameCount, buffer, sessionId, uid, |
| flags, false /*isOut*/, |
| (type == TYPE_DEFAULT) ? |
| ((flags & IAudioFlinger::TRACK_FAST) ? ALLOC_PIPE : ALLOC_CBLK) : |
| ((buffer == NULL) ? ALLOC_LOCAL : ALLOC_NONE), |
| type), |
| mOverflow(false), |
| mFramesToDrop(0), |
| mResamplerBufferProvider(NULL), // initialize in case of early constructor exit |
| mRecordBufferConverter(NULL) |
| { |
| if (mCblk == NULL) { |
| return; |
| } |
| |
| mRecordBufferConverter = new RecordBufferConverter( |
| thread->mChannelMask, thread->mFormat, thread->mSampleRate, |
| channelMask, format, sampleRate); |
| // Check if the RecordBufferConverter construction was successful. |
| // If not, don't continue with construction. |
| // |
| // NOTE: It would be extremely rare that the record track cannot be created |
| // for the current device, but a pending or future device change would make |
| // the record track configuration valid. |
| if (mRecordBufferConverter->initCheck() != NO_ERROR) { |
| ALOGE("RecordTrack unable to create record buffer converter"); |
| return; |
| } |
| |
| mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, |
| mFrameSize, !isExternalTrack()); |
| mResamplerBufferProvider = new ResamplerBufferProvider(this); |
| |
| if (flags & IAudioFlinger::TRACK_FAST) { |
| ALOG_ASSERT(thread->mFastTrackAvail); |
| thread->mFastTrackAvail = false; |
| } |
| } |
| |
| AudioFlinger::RecordThread::RecordTrack::~RecordTrack() |
| { |
| ALOGV("%s", __func__); |
| delete mRecordBufferConverter; |
| delete mResamplerBufferProvider; |
| } |
| |
| status_t AudioFlinger::RecordThread::RecordTrack::initCheck() const |
| { |
| status_t status = TrackBase::initCheck(); |
| if (status == NO_ERROR && mServerProxy == 0) { |
| status = BAD_VALUE; |
| } |
| return status; |
| } |
| |
| // AudioBufferProvider interface |
| status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, |
| int64_t pts __unused) |
| { |
| ServerProxy::Buffer buf; |
| buf.mFrameCount = buffer->frameCount; |
| status_t status = mServerProxy->obtainBuffer(&buf); |
| buffer->frameCount = buf.mFrameCount; |
| buffer->raw = buf.mRaw; |
| if (buf.mFrameCount == 0) { |
| // FIXME also wake futex so that overrun is noticed more quickly |
| (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); |
| } |
| return status; |
| } |
| |
| status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, |
| int triggerSession) |
| { |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| RecordThread *recordThread = (RecordThread *)thread.get(); |
| return recordThread->start(this, event, triggerSession); |
| } else { |
| return BAD_VALUE; |
| } |
| } |
| |
| void AudioFlinger::RecordThread::RecordTrack::stop() |
| { |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| RecordThread *recordThread = (RecordThread *)thread.get(); |
| if (recordThread->stop(this) && isExternalTrack()) { |
| AudioSystem::stopInput(mThreadIoHandle, (audio_session_t)mSessionId); |
| } |
| } |
| } |
| |
| void AudioFlinger::RecordThread::RecordTrack::destroy() |
| { |
| // see comments at AudioFlinger::PlaybackThread::Track::destroy() |
| sp<RecordTrack> keep(this); |
| { |
| if (isExternalTrack()) { |
| if (mState == ACTIVE || mState == RESUMING) { |
| AudioSystem::stopInput(mThreadIoHandle, (audio_session_t)mSessionId); |
| } |
| AudioSystem::releaseInput(mThreadIoHandle, (audio_session_t)mSessionId); |
| } |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| Mutex::Autolock _l(thread->mLock); |
| RecordThread *recordThread = (RecordThread *) thread.get(); |
| recordThread->destroyTrack_l(this); |
| } |
| } |
| } |
| |
| void AudioFlinger::RecordThread::RecordTrack::invalidate() |
| { |
| // FIXME should use proxy, and needs work |
| audio_track_cblk_t* cblk = mCblk; |
| android_atomic_or(CBLK_INVALID, &cblk->mFlags); |
| android_atomic_release_store(0x40000000, &cblk->mFutex); |
| // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE |
| (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX); |
| } |
| |
| |
| /*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) |
| { |
| result.append(" Active Client Fmt Chn mask Session S Server fCount SRate\n"); |
| } |
| |
| void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active) |
| { |
| snprintf(buffer, size, " %6s %6u %3u %08X %7u %1d %08X %6zu %5u\n", |
| active ? "yes" : "no", |
| (mClient == 0) ? getpid_cached : mClient->pid(), |
| mFormat, |
| mChannelMask, |
| mSessionId, |
| mState, |
| mCblk->mServer, |
| mFrameCount, |
| mSampleRate); |
| |
| } |
| |
| void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event) |
| { |
| if (event == mSyncStartEvent) { |
| ssize_t framesToDrop = 0; |
| sp<ThreadBase> threadBase = mThread.promote(); |
| if (threadBase != 0) { |
| // TODO: use actual buffer filling status instead of 2 buffers when info is available |
| // from audio HAL |
| framesToDrop = threadBase->mFrameCount * 2; |
| } |
| mFramesToDrop = framesToDrop; |
| } |
| } |
| |
| void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent() |
| { |
| if (mSyncStartEvent != 0) { |
| mSyncStartEvent->cancel(); |
| mSyncStartEvent.clear(); |
| } |
| mFramesToDrop = 0; |
| } |
| |
| |
| AudioFlinger::RecordThread::PatchRecord::PatchRecord(RecordThread *recordThread, |
| uint32_t sampleRate, |
| audio_channel_mask_t channelMask, |
| audio_format_t format, |
| size_t frameCount, |
| void *buffer, |
| IAudioFlinger::track_flags_t flags) |
| : RecordTrack(recordThread, NULL, sampleRate, format, channelMask, frameCount, |
| buffer, 0, getuid(), flags, TYPE_PATCH), |
| mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true)) |
| { |
| uint64_t mixBufferNs = ((uint64_t)2 * recordThread->frameCount() * 1000000000) / |
| recordThread->sampleRate(); |
| mPeerTimeout.tv_sec = mixBufferNs / 1000000000; |
| mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000); |
| |
| ALOGV("PatchRecord %p sampleRate %d mPeerTimeout %d.%03d sec", |
| this, sampleRate, |
| (int)mPeerTimeout.tv_sec, |
| (int)(mPeerTimeout.tv_nsec / 1000000)); |
| } |
| |
| AudioFlinger::RecordThread::PatchRecord::~PatchRecord() |
| { |
| } |
| |
| // AudioBufferProvider interface |
| status_t AudioFlinger::RecordThread::PatchRecord::getNextBuffer( |
| AudioBufferProvider::Buffer* buffer, int64_t pts) |
| { |
| ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::getNextBuffer() called without peer proxy"); |
| Proxy::Buffer buf; |
| buf.mFrameCount = buffer->frameCount; |
| status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout); |
| ALOGV_IF(status != NO_ERROR, |
| "PatchRecord() %p mPeerProxy->obtainBuffer status %d", this, status); |
| buffer->frameCount = buf.mFrameCount; |
| if (buf.mFrameCount == 0) { |
| return WOULD_BLOCK; |
| } |
| status = RecordTrack::getNextBuffer(buffer, pts); |
| return status; |
| } |
| |
| void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(AudioBufferProvider::Buffer* buffer) |
| { |
| ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::releaseBuffer() called without peer proxy"); |
| Proxy::Buffer buf; |
| buf.mFrameCount = buffer->frameCount; |
| buf.mRaw = buffer->raw; |
| mPeerProxy->releaseBuffer(&buf); |
| TrackBase::releaseBuffer(buffer); |
| } |
| |
| status_t AudioFlinger::RecordThread::PatchRecord::obtainBuffer(Proxy::Buffer* buffer, |
| const struct timespec *timeOut) |
| { |
| return mProxy->obtainBuffer(buffer, timeOut); |
| } |
| |
| void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(Proxy::Buffer* buffer) |
| { |
| mProxy->releaseBuffer(buffer); |
| } |
| |
| } // namespace android |
