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Diffstat (limited to 'services/audioflinger/Tracks.cpp')
| -rw-r--r-- | services/audioflinger/Tracks.cpp | 1789 |
1 files changed, 1789 insertions, 0 deletions
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp new file mode 100644 index 0000000..2c6ba8b --- /dev/null +++ b/services/audioflinger/Tracks.cpp @@ -0,0 +1,1789 @@ +/* +** +** 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 <math.h> +#include <cutils/compiler.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" + +// ---------------------------------------------------------------------------- + +// 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 +// ---------------------------------------------------------------------------- + +// 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, + const sp<IMemory>& sharedBuffer, + int sessionId) + : RefBase(), + mThread(thread), + mClient(client), + mCblk(NULL), + // mBuffer + // mBufferEnd + mStepCount(0), + mState(IDLE), + mSampleRate(sampleRate), + mFormat(format), + mChannelMask(channelMask), + mChannelCount(popcount(channelMask)), + mFrameSize(audio_is_linear_pcm(format) ? + mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), + mFrameCount(frameCount), + mStepServerFailed(false), + mSessionId(sessionId) +{ + // client == 0 implies sharedBuffer == 0 + ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); + + 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); + size_t bufferSize = frameCount * mFrameSize; + if (sharedBuffer == 0) { + size += bufferSize; + } + + if (client != 0) { + mCblkMemory = client->heap()->allocate(size); + if (mCblkMemory != 0) { + mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); + // can't assume mCblk != NULL + } else { + ALOGE("not enough memory for AudioTrack size=%u", size); + client->heap()->dump("AudioTrack"); + return; + } + } else { + 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(); + // clear all buffers + mCblk->frameCount_ = frameCount; + mCblk->sampleRate = sampleRate; +// uncomment the following lines to quickly test 32-bit wraparound +// mCblk->user = 0xffff0000; +// mCblk->server = 0xffff0000; +// mCblk->userBase = 0xffff0000; +// mCblk->serverBase = 0xffff0000; + if (sharedBuffer == 0) { + mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); + memset(mBuffer, 0, bufferSize); + // Force underrun condition to avoid false underrun callback until first data is + // written to buffer (other flags are cleared) + mCblk->flags = CBLK_UNDERRUN; + } else { + mBuffer = sharedBuffer->pointer(); + } + mBufferEnd = (uint8_t *)mBuffer + bufferSize; + } +} + +AudioFlinger::ThreadBase::TrackBase::~TrackBase() +{ + 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 mutex locked + Mutex::Autolock _l(mClient->audioFlinger()->mLock); + // 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(); + } +} + +// 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) +{ + buffer->raw = NULL; + mStepCount = buffer->frameCount; + // FIXME See note at getNextBuffer() + (void) step(); // ignore return value of step() + buffer->frameCount = 0; +} + +bool AudioFlinger::ThreadBase::TrackBase::step() { + bool result; + audio_track_cblk_t* cblk = this->cblk(); + + result = cblk->stepServer(mStepCount, mFrameCount, isOut()); + if (!result) { + ALOGV("stepServer failed acquiring cblk mutex"); + mStepServerFailed = true; + } + return result; +} + +void AudioFlinger::ThreadBase::TrackBase::reset() { + audio_track_cblk_t* cblk = this->cblk(); + + cblk->user = 0; + cblk->server = 0; + cblk->userBase = 0; + cblk->serverBase = 0; + mStepServerFailed = false; + ALOGV("TrackBase::reset"); +} + +uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const { + return mCblk->sampleRate; +} + +void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { + audio_track_cblk_t* cblk = this->cblk(); + int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize; + int8_t *bufferEnd = bufferStart + frames * mFrameSize; + + // Check validity of returned pointer in case the track control block would have been corrupted. + ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), + "TrackBase::getBuffer buffer out of range:\n" + " start: %p, end %p , mBuffer %p mBufferEnd %p\n" + " server %u, serverBase %u, user %u, userBase %u, frameSize %u", + bufferStart, bufferEnd, mBuffer, mBufferEnd, + cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize); + + return bufferStart; +} + +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::mute(bool e) { + mTrack->mute(e); +} + +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; + + 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::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, + const sp<IMemory>& sharedBuffer, + int sessionId, + IAudioFlinger::track_flags_t flags) + : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, + sessionId), + mMute(false), + 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), + mFlags(flags), + mFastIndex(-1), + mUnderrunCount(0), + mCachedVolume(1.0) +{ + if (mCblk != NULL) { + // to avoid leaking a track name, do not allocate one unless there is an mCblk + mName = thread->getTrackName_l(channelMask, sessionId); + mCblk->mName = mName; + 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) { + 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; + mCblk->mName = i; + // Read the initial underruns because this field is never cleared by the fast mixer + mObservedUnderruns = thread->getFastTrackUnderruns(i); + thread->mFastTrackAvailMask &= ~(1 << i); + } + } + ALOGV("Track constructor name %d, calling pid %d", mName, + IPCThreadState::self()->getCallingPid()); +} + +AudioFlinger::PlaybackThread::Track::~Track() +{ + ALOGV("PlaybackThread::Track destructor"); +} + +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 + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + if (!isOutputTrack()) { + if (mState == ACTIVE || mState == RESUMING) { + AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); + +#ifdef ADD_BATTERY_DATA + // to track the speaker usage + addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); +#endif + } + AudioSystem::releaseOutput(thread->id()); + } + Mutex::Autolock _l(thread->mLock); + PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); + playbackThread->destroyTrack_l(this); + } + } +} + +/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) +{ + result.append(" Name Client Type Fmt Chn mask Session StpCnt 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) +{ + uint32_t vlr = mCblk->getVolumeLR(); + if (isFastTrack()) { + sprintf(buffer, " F %2d", mFastIndex); + } else { + sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); + } + track_state state = mState; + char stateChar; + switch (state) { + case IDLE: + stateChar = 'I'; + break; + case TERMINATED: + stateChar = 'T'; + 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[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " + "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", + (mClient == 0) ? getpid_cached : mClient->pid(), + mStreamType, + mFormat, + mChannelMask, + mSessionId, + mStepCount, + mFrameCount, + stateChar, + mMute, + mFillingUpStatus, + mCblk->sampleRate, + 20.0 * log10((vlr & 0xFFFF) / 4096.0), + 20.0 * log10((vlr >> 16) / 4096.0), + mCblk->server, + mCblk->user, + (int)mMainBuffer, + (int)mAuxBuffer, + mCblk->flags, + mUnderrunCount, + nowInUnderrun); +} + +// AudioBufferProvider interface +status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( + AudioBufferProvider::Buffer* buffer, int64_t pts) +{ + audio_track_cblk_t* cblk = this->cblk(); + uint32_t framesReady; + uint32_t framesReq = buffer->frameCount; + + // Check if last stepServer failed, try to step now + if (mStepServerFailed) { + // FIXME When called by fast mixer, this takes a mutex with tryLock(). + // Since the fast mixer is higher priority than client callback thread, + // it does not result in priority inversion for client. + // But a non-blocking solution would be preferable to avoid + // fast mixer being unable to tryLock(), and + // to avoid the extra context switches if the client wakes up, + // discovers the mutex is locked, then has to wait for fast mixer to unlock. + if (!step()) goto getNextBuffer_exit; + ALOGV("stepServer recovered"); + mStepServerFailed = false; + } + + // FIXME Same as above + framesReady = cblk->framesReadyOut(); + + if (CC_LIKELY(framesReady)) { + uint32_t s = cblk->server; + uint32_t bufferEnd = cblk->serverBase + mFrameCount; + + bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; + if (framesReq > framesReady) { + framesReq = framesReady; + } + if (framesReq > bufferEnd - s) { + framesReq = bufferEnd - s; + } + + buffer->raw = getBuffer(s, framesReq); + buffer->frameCount = framesReq; + return NO_ERROR; + } + +getNextBuffer_exit: + buffer->raw = NULL; + buffer->frameCount = 0; + ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); + return NOT_ENOUGH_DATA; +} + +// Note that framesReady() takes a mutex on the control block using tryLock(). +// This could result in priority inversion if framesReady() is called by the normal mixer, +// as the normal mixer thread runs at lower +// priority than the client's callback thread: there is a short window within framesReady() +// during which the normal mixer could be preempted, and the client callback would block. +// Another problem can occur if framesReady() is called by the fast mixer: +// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. +// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. +size_t AudioFlinger::PlaybackThread::Track::framesReady() const { + return mCblk->framesReadyOut(); +} + +// 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 (framesReady() >= mFrameCount || + (mCblk->flags & CBLK_FORCEREADY)) { + mFillingUpStatus = FS_FILLED; + android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); + return true; + } + return false; +} + +status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, + int triggerSession) +{ + 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) { + Mutex::Autolock _l(thread->mLock); + track_state state = mState; + // here the track could be either new, or restarted + // in both cases "unstop" the track + if (mState == PAUSED) { + mState = TrackBase::RESUMING; + ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); + } else { + mState = TrackBase::ACTIVE; + ALOGV("? => ACTIVE (%d) on thread %p", mName, this); + } + + if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { + thread->mLock.unlock(); + status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); + thread->mLock.lock(); + +#ifdef ADD_BATTERY_DATA + // to track the speaker usage + if (status == NO_ERROR) { + addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); + } +#endif + } + if (status == NO_ERROR) { + PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); + playbackThread->addTrack_l(this); + } else { + mState = state; + triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); + } + } 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()) { + mState = STOPPED; + } else { + // prepareTracks_l() will set state to STOPPING_2 after next underrun, + // 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); + } + if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { + thread->mLock.unlock(); + AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); + thread->mLock.lock(); + +#ifdef ADD_BATTERY_DATA + // to track the speaker usage + addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); +#endif + } + } +} + +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); + if (mState == ACTIVE || mState == RESUMING) { + mState = PAUSING; + ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); + if (!isOutputTrack()) { + thread->mLock.unlock(); + AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); + thread->mLock.lock(); + +#ifdef ADD_BATTERY_DATA + // to track the speaker usage + addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); +#endif + } + } + } +} + +void AudioFlinger::PlaybackThread::Track::flush() +{ + ALOGV("flush(%d)", mName); + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + Mutex::Autolock _l(thread->mLock); + 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 + PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); + if (playbackThread->mActiveTracks.indexOf(this) < 0) { + reset(); + } + } +} + +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) { + TrackBase::reset(); + // Force underrun condition to avoid false underrun callback until first data is + // written to buffer + android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); + android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); + mFillingUpStatus = FS_FILLING; + mResetDone = true; + if (mState == FLUSHED) { + mState = IDLE; + } + } +} + +void AudioFlinger::PlaybackThread::Track::mute(bool muted) +{ + mMute = muted; +} + +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); + playbackThread->addEffect_l(effect); + // 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()); + } + 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. + if (mPresentationCompleteFrames == 0) { + mPresentationCompleteFrames = framesWritten + audioHalFrames; + ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", + mPresentationCompleteFrames, audioHalFrames); + } + if (framesWritten >= mPresentationCompleteFrames) { + ALOGV("presentationComplete() session %d complete: framesWritten %d", + mSessionId, framesWritten); + triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); + return true; + } + return false; +} + +void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) +{ + for (int i = 0; i < (int)mSyncEvents.size(); i++) { + if (mSyncEvents[i]->type() == type) { + mSyncEvents[i]->trigger(); + mSyncEvents.removeAt(i); + i--; + } + } +} + +// implement VolumeBufferProvider interface + +uint32_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)); + uint32_t vlr = mCblk->getVolumeLR(); + uint32_t vl = vlr & 0xFFFF; + uint32_t vr = vlr >> 16; + // track volumes come from shared memory, so can't be trusted and must be clamped + if (vl > MAX_GAIN_INT) { + vl = MAX_GAIN_INT; + } + if (vr > MAX_GAIN_INT) { + vr = MAX_GAIN_INT; + } + // 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 U4.16 + vlr = (vr << 16) | (vl & 0xFFFF); + // FIXME look at mute, pause, and stop flags + return vlr; +} + +status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) +{ + if (mState == TERMINATED || 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; +} + +bool AudioFlinger::PlaybackThread::Track::isOut() const +{ + return true; +} + +// ---------------------------------------------------------------------------- + +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) { + if (!client->reserveTimedTrack()) + return 0; + + return new TimedTrack( + thread, client, streamType, sampleRate, format, channelMask, frameCount, + sharedBuffer, sessionId); +} + +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) + : Track(thread, client, streamType, sampleRate, format, channelMask, + frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_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 == NULL) { + newBuffer = mTimedMemoryDealer->allocate(size); + if (newBuffer == 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) { + 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; + } + } + + // 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 / sampleRate()); + + // 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>(sampleRate()) << 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_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) + : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, + NULL, 0, IAudioFlinger::TRACK_DEFAULT), + mActive(false), mSourceThread(sourceThread), mBuffers(NULL) +{ + + if (mCblk != NULL) { + mBuffers = (char*)mCblk + sizeof(audio_track_cblk_t); + mOutBuffer.frameCount = 0; + playbackThread->mTracks.add(this); + ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mBuffers %p, " \ + "mCblk->frameCount %d, mCblk->sampleRate %u, mChannelMask 0x%08x mBufferEnd %p", + mCblk, mBuffer, mBuffers, + mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); + } else { + ALOGW("Error creating output track on thread %p", playbackThread); + } +} + +AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() +{ + clearBufferQueue(); +} + +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(int16_t* data, uint32_t frames) +{ + Buffer *pInBuffer; + Buffer inBuffer; + uint32_t channelCount = mChannelCount; + bool outputBufferFull = false; + inBuffer.frameCount = frames; + inBuffer.i16 = data; + + uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); + + if (!mActive && frames != 0) { + start(); + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + MixerThread *mixerThread = (MixerThread *)thread.get(); + if (mFrameCount > frames) { + if (mBufferQueue.size() < kMaxOverFlowBuffers) { + uint32_t startFrames = (mFrameCount - frames); + pInBuffer = new Buffer; + pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; + pInBuffer->frameCount = startFrames; + pInBuffer->i16 = pInBuffer->mBuffer; + memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); + mBufferQueue.add(pInBuffer); + } else { + ALOGW ("OutputTrack::write() %p no more buffers in queue", this); + } + } + } + } + + 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(); + if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { + ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, + mThread.unsafe_get()); + 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 * channelCount * sizeof(int16_t)); + mCblk->stepUserOut(outFrames, mFrameCount); + pInBuffer->frameCount -= outFrames; + pInBuffer->i16 += outFrames * channelCount; + mOutBuffer.frameCount -= outFrames; + mOutBuffer.i16 += outFrames * channelCount; + + if (pInBuffer->frameCount == 0) { + if (mBufferQueue.size()) { + mBufferQueue.removeAt(0); + delete [] 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 = 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)); + 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: + // If no more buffers are pending, fill output track buffer to make sure it is started + // by output mixer. + if (frames == 0 && mBufferQueue.size() == 0) { + if (mCblk->user < mFrameCount) { + frames = mFrameCount - mCblk->user; + pInBuffer = new Buffer; + pInBuffer->mBuffer = new int16_t[frames * channelCount]; + pInBuffer->frameCount = frames; + pInBuffer->i16 = pInBuffer->mBuffer; + memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); + mBufferQueue.add(pInBuffer); + } else if (mActive) { + stop(); + } + } + + return outputBufferFull; +} + +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; + + ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); + buffer->frameCount = 0; + + uint32_t framesAvail = cblk->framesAvailableOut(mFrameCount); + + + if (framesAvail == 0) { + Mutex::Autolock _l(cblk->lock); + goto start_loop_here; + while (framesAvail == 0) { + active = mActive; + if (CC_UNLIKELY(!active)) { + ALOGV("Not active and NO_MORE_BUFFERS"); + return NO_MORE_BUFFERS; + } + result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); + if (result != NO_ERROR) { + return NO_MORE_BUFFERS; + } + // read the server count again + start_loop_here: + framesAvail = cblk->framesAvailableOut_l(mFrameCount); + } + } + +// if (framesAvail < framesReq) { +// return NO_MORE_BUFFERS; +// } + + if (framesReq > framesAvail) { + framesReq = framesAvail; + } + + uint32_t u = cblk->user; + uint32_t bufferEnd = cblk->userBase + mFrameCount; + + if (framesReq > bufferEnd - u) { + framesReq = bufferEnd - u; + } + + buffer->frameCount = framesReq; + buffer->raw = cblk->buffer(mBuffers, mFrameSize, u); + return NO_ERROR; +} + + +void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() +{ + size_t size = mBufferQueue.size(); + + for (size_t i = 0; i < size; i++) { + Buffer *pBuffer = mBufferQueue.itemAt(i); + delete [] pBuffer->mBuffer; + delete pBuffer; + } + mBufferQueue.clear(); +} + + +// ---------------------------------------------------------------------------- +// Record +// ---------------------------------------------------------------------------- + +AudioFlinger::RecordHandle::RecordHandle( + const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) + : BnAudioRecord(), + mRecordTrack(recordTrack) +{ +} + +AudioFlinger::RecordHandle::~RecordHandle() { + stop_nonvirtual(); + mRecordTrack->destroy(); +} + +sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { + return mRecordTrack->getCblk(); +} + +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 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, + int sessionId) + : TrackBase(thread, client, sampleRate, format, + channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), + mOverflow(false) +{ + ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); +} + +AudioFlinger::RecordThread::RecordTrack::~RecordTrack() +{ + ALOGV("%s", __func__); +} + +// AudioBufferProvider interface +status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, + int64_t pts) +{ + audio_track_cblk_t* cblk = this->cblk(); + uint32_t framesAvail; + uint32_t framesReq = buffer->frameCount; + + // Check if last stepServer failed, try to step now + if (mStepServerFailed) { + if (!step()) { + goto getNextBuffer_exit; + } + ALOGV("stepServer recovered"); + mStepServerFailed = false; + } + + // FIXME lock is not actually held, so overrun is possible + framesAvail = cblk->framesAvailableIn_l(mFrameCount); + + if (CC_LIKELY(framesAvail)) { + uint32_t s = cblk->server; + uint32_t bufferEnd = cblk->serverBase + mFrameCount; + + if (framesReq > framesAvail) { + framesReq = framesAvail; + } + if (framesReq > bufferEnd - s) { + framesReq = bufferEnd - s; + } + + buffer->raw = getBuffer(s, framesReq); + buffer->frameCount = framesReq; + return NO_ERROR; + } + +getNextBuffer_exit: + buffer->raw = NULL; + buffer->frameCount = 0; + return NOT_ENOUGH_DATA; +} + +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(); + recordThread->mLock.lock(); + bool doStop = recordThread->stop_l(this); + if (doStop) { + TrackBase::reset(); + // Force overrun condition to avoid false overrun callback until first data is + // read from buffer + android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); + } + recordThread->mLock.unlock(); + if (doStop) { + AudioSystem::stopInput(recordThread->id()); + } + } +} + +void AudioFlinger::RecordThread::RecordTrack::destroy() +{ + // see comments at AudioFlinger::PlaybackThread::Track::destroy() + sp<RecordTrack> keep(this); + { + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + if (mState == ACTIVE || mState == RESUMING) { + AudioSystem::stopInput(thread->id()); + } + AudioSystem::releaseInput(thread->id()); + Mutex::Autolock _l(thread->mLock); + RecordThread *recordThread = (RecordThread *) thread.get(); + recordThread->destroyTrack_l(this); + } + } +} + + +/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) +{ + result.append(" Clien Fmt Chn mask Session Step S SRate Serv User FrameCount\n"); +} + +void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) +{ + snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x %05d\n", + (mClient == 0) ? getpid_cached : mClient->pid(), + mFormat, + mChannelMask, + mSessionId, + mStepCount, + mState, + mCblk->sampleRate, + mCblk->server, + mCblk->user, + mFrameCount); +} + +bool AudioFlinger::RecordThread::RecordTrack::isOut() const +{ + return false; +} + +}; // namespace android |