/* //device/extlibs/pv/android/AudioTrack.cpp ** ** Copyright 2007, 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_NDEBUG 0 #define LOG_TAG "AudioTrack" #include <stdint.h> #include <sys/types.h> #include <limits.h> #include <sched.h> #include <sys/resource.h> #include <private/media/AudioTrackShared.h> #include <media/AudioSystem.h> #include <media/AudioTrack.h> #include <utils/Log.h> #include <binder/MemoryDealer.h> #include <binder/Parcel.h> #include <binder/IPCThreadState.h> #include <utils/Timers.h> #include <cutils/atomic.h> #define LIKELY( exp ) (__builtin_expect( (exp) != 0, true )) #define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false )) namespace android { // --------------------------------------------------------------------------- AudioTrack::AudioTrack() : mStatus(NO_INIT) { } AudioTrack::AudioTrack( int streamType, uint32_t sampleRate, int format, int channels, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames) : mStatus(NO_INIT) { mStatus = set(streamType, sampleRate, format, channels, frameCount, flags, cbf, user, notificationFrames, 0); } AudioTrack::AudioTrack( int streamType, uint32_t sampleRate, int format, int channels, const sp<IMemory>& sharedBuffer, uint32_t flags, callback_t cbf, void* user, int notificationFrames) : mStatus(NO_INIT) { mStatus = set(streamType, sampleRate, format, channels, 0, flags, cbf, user, notificationFrames, sharedBuffer); } AudioTrack::~AudioTrack() { LOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer()); if (mStatus == NO_ERROR) { // Make sure that callback function exits in the case where // it is looping on buffer full condition in obtainBuffer(). // Otherwise the callback thread will never exit. stop(); if (mAudioTrackThread != 0) { mAudioTrackThread->requestExitAndWait(); mAudioTrackThread.clear(); } mAudioTrack.clear(); IPCThreadState::self()->flushCommands(); AudioSystem::releaseOutput(getOutput()); } } status_t AudioTrack::set( int streamType, uint32_t sampleRate, int format, int channels, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames, const sp<IMemory>& sharedBuffer, bool threadCanCallJava) { LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); if (mAudioTrack != 0) { LOGE("Track already in use"); return INVALID_OPERATION; } const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); if (audioFlinger == 0) { LOGE("Could not get audioflinger"); return NO_INIT; } int afSampleRate; if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { return NO_INIT; } int afFrameCount; if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { return NO_INIT; } uint32_t afLatency; if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { return NO_INIT; } // handle default values first. if (streamType == AudioSystem::DEFAULT) { streamType = AudioSystem::MUSIC; } if (sampleRate == 0) { sampleRate = afSampleRate; } // these below should probably come from the audioFlinger too... if (format == 0) { format = AudioSystem::PCM_16_BIT; } if (channels == 0) { channels = AudioSystem::CHANNEL_OUT_STEREO; } // validate parameters if (!AudioSystem::isValidFormat(format)) { LOGE("Invalid format"); return BAD_VALUE; } // force direct flag if format is not linear PCM if (!AudioSystem::isLinearPCM(format)) { flags |= AudioSystem::OUTPUT_FLAG_DIRECT; } if (!AudioSystem::isOutputChannel(channels)) { LOGE("Invalid channel mask"); return BAD_VALUE; } uint32_t channelCount = AudioSystem::popCount(channels); audio_io_handle_t output = AudioSystem::getOutput((AudioSystem::stream_type)streamType, sampleRate, format, channels, (AudioSystem::output_flags)flags); if (output == 0) { LOGE("Could not get audio output for stream type %d", streamType); return BAD_VALUE; } if (!AudioSystem::isLinearPCM(format)) { if (sharedBuffer != 0) { frameCount = sharedBuffer->size(); } } else { // Ensure that buffer depth covers at least audio hardware latency uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); if (minBufCount < 2) minBufCount = 2; int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; if (sharedBuffer == 0) { if (frameCount == 0) { frameCount = minFrameCount; } if (notificationFrames == 0) { notificationFrames = frameCount/2; } // Make sure that application is notified with sufficient margin // before underrun if (notificationFrames > frameCount/2) { notificationFrames = frameCount/2; } if (frameCount < minFrameCount) { LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount); return BAD_VALUE; } } else { // Ensure that buffer alignment matches channelcount if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) { LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount); return BAD_VALUE; } frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t); } } // create the track status_t status; sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), streamType, sampleRate, format, channelCount, frameCount, ((uint16_t)flags) << 16, sharedBuffer, output, &status); if (track == 0) { LOGE("AudioFlinger could not create track, status: %d", status); return status; } sp<IMemory> cblk = track->getCblk(); if (cblk == 0) { LOGE("Could not get control block"); return NO_INIT; } if (cbf != 0) { mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); if (mAudioTrackThread == 0) { LOGE("Could not create callback thread"); return NO_INIT; } } mStatus = NO_ERROR; mAudioTrack = track; mCblkMemory = cblk; mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); mCblk->out = 1; // Update buffer size in case it has been limited by AudioFlinger during track creation mFrameCount = mCblk->frameCount; if (sharedBuffer == 0) { mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); } else { mCblk->buffers = sharedBuffer->pointer(); // Force buffer full condition as data is already present in shared memory mCblk->stepUser(mFrameCount); } mCblk->volume[0] = mCblk->volume[1] = 0x1000; mVolume[LEFT] = 1.0f; mVolume[RIGHT] = 1.0f; mStreamType = streamType; mFormat = format; mChannels = channels; mChannelCount = channelCount; mSharedBuffer = sharedBuffer; mMuted = false; mActive = 0; mCbf = cbf; mNotificationFrames = notificationFrames; mRemainingFrames = notificationFrames; mUserData = user; mLatency = afLatency + (1000*mFrameCount) / sampleRate; mLoopCount = 0; mMarkerPosition = 0; mMarkerReached = false; mNewPosition = 0; mUpdatePeriod = 0; mFlags = flags; return NO_ERROR; } status_t AudioTrack::initCheck() const { return mStatus; } // ------------------------------------------------------------------------- uint32_t AudioTrack::latency() const { return mLatency; } int AudioTrack::streamType() const { return mStreamType; } int AudioTrack::format() const { return mFormat; } int AudioTrack::channelCount() const { return mChannelCount; } uint32_t AudioTrack::frameCount() const { return mFrameCount; } int AudioTrack::frameSize() const { if (AudioSystem::isLinearPCM(mFormat)) { return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t)); } else { return sizeof(uint8_t); } } sp<IMemory>& AudioTrack::sharedBuffer() { return mSharedBuffer; } // ------------------------------------------------------------------------- void AudioTrack::start() { sp<AudioTrackThread> t = mAudioTrackThread; LOGV("start %p", this); if (t != 0) { if (t->exitPending()) { if (t->requestExitAndWait() == WOULD_BLOCK) { LOGE("AudioTrack::start called from thread"); return; } } t->mLock.lock(); } if (android_atomic_or(1, &mActive) == 0) { AudioSystem::startOutput(getOutput(), (AudioSystem::stream_type)mStreamType); mNewPosition = mCblk->server + mUpdatePeriod; mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; mCblk->waitTimeMs = 0; if (t != 0) { t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT); } else { setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT); } mAudioTrack->start(); } if (t != 0) { t->mLock.unlock(); } } void AudioTrack::stop() { sp<AudioTrackThread> t = mAudioTrackThread; LOGV("stop %p", this); if (t != 0) { t->mLock.lock(); } if (android_atomic_and(~1, &mActive) == 1) { mCblk->cv.signal(); mAudioTrack->stop(); // Cancel loops (If we are in the middle of a loop, playback // would not stop until loopCount reaches 0). setLoop(0, 0, 0); // the playback head position will reset to 0, so if a marker is set, we need // to activate it again mMarkerReached = false; // Force flush if a shared buffer is used otherwise audioflinger // will not stop before end of buffer is reached. if (mSharedBuffer != 0) { flush(); } if (t != 0) { t->requestExit(); } else { setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL); } AudioSystem::stopOutput(getOutput(), (AudioSystem::stream_type)mStreamType); } if (t != 0) { t->mLock.unlock(); } } bool AudioTrack::stopped() const { return !mActive; } void AudioTrack::flush() { LOGV("flush"); // clear playback marker and periodic update counter mMarkerPosition = 0; mMarkerReached = false; mUpdatePeriod = 0; if (!mActive) { mAudioTrack->flush(); // Release AudioTrack callback thread in case it was waiting for new buffers // in AudioTrack::obtainBuffer() mCblk->cv.signal(); } } void AudioTrack::pause() { LOGV("pause"); if (android_atomic_and(~1, &mActive) == 1) { mActive = 0; mAudioTrack->pause(); AudioSystem::stopOutput(getOutput(), (AudioSystem::stream_type)mStreamType); } } void AudioTrack::mute(bool e) { mAudioTrack->mute(e); mMuted = e; } bool AudioTrack::muted() const { return mMuted; } void AudioTrack::setVolume(float left, float right) { mVolume[LEFT] = left; mVolume[RIGHT] = right; // write must be atomic mCblk->volumeLR = (int32_t(int16_t(left * 0x1000)) << 16) | int16_t(right * 0x1000); } void AudioTrack::getVolume(float* left, float* right) { *left = mVolume[LEFT]; *right = mVolume[RIGHT]; } status_t AudioTrack::setSampleRate(int rate) { int afSamplingRate; if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { return NO_INIT; } // Resampler implementation limits input sampling rate to 2 x output sampling rate. if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE; mCblk->sampleRate = rate; return NO_ERROR; } uint32_t AudioTrack::getSampleRate() { return mCblk->sampleRate; } status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) { audio_track_cblk_t* cblk = mCblk; Mutex::Autolock _l(cblk->lock); if (loopCount == 0) { cblk->loopStart = UINT_MAX; cblk->loopEnd = UINT_MAX; cblk->loopCount = 0; mLoopCount = 0; return NO_ERROR; } if (loopStart >= loopEnd || loopEnd - loopStart > mFrameCount) { LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, mFrameCount, cblk->user); return BAD_VALUE; } if ((mSharedBuffer != 0) && (loopEnd > mFrameCount)) { LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d", loopStart, loopEnd, mFrameCount); return BAD_VALUE; } cblk->loopStart = loopStart; cblk->loopEnd = loopEnd; cblk->loopCount = loopCount; mLoopCount = loopCount; return NO_ERROR; } status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount) { if (loopStart != 0) { *loopStart = mCblk->loopStart; } if (loopEnd != 0) { *loopEnd = mCblk->loopEnd; } if (loopCount != 0) { if (mCblk->loopCount < 0) { *loopCount = -1; } else { *loopCount = mCblk->loopCount; } } return NO_ERROR; } status_t AudioTrack::setMarkerPosition(uint32_t marker) { if (mCbf == 0) return INVALID_OPERATION; mMarkerPosition = marker; mMarkerReached = false; return NO_ERROR; } status_t AudioTrack::getMarkerPosition(uint32_t *marker) { if (marker == 0) return BAD_VALUE; *marker = mMarkerPosition; return NO_ERROR; } status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) { if (mCbf == 0) return INVALID_OPERATION; uint32_t curPosition; getPosition(&curPosition); mNewPosition = curPosition + updatePeriod; mUpdatePeriod = updatePeriod; return NO_ERROR; } status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) { if (updatePeriod == 0) return BAD_VALUE; *updatePeriod = mUpdatePeriod; return NO_ERROR; } status_t AudioTrack::setPosition(uint32_t position) { Mutex::Autolock _l(mCblk->lock); if (!stopped()) return INVALID_OPERATION; if (position > mCblk->user) return BAD_VALUE; mCblk->server = position; mCblk->forceReady = 1; return NO_ERROR; } status_t AudioTrack::getPosition(uint32_t *position) { if (position == 0) return BAD_VALUE; *position = mCblk->server; return NO_ERROR; } status_t AudioTrack::reload() { if (!stopped()) return INVALID_OPERATION; flush(); mCblk->stepUser(mFrameCount); return NO_ERROR; } audio_io_handle_t AudioTrack::getOutput() { return AudioSystem::getOutput((AudioSystem::stream_type)mStreamType, mCblk->sampleRate, mFormat, mChannels, (AudioSystem::output_flags)mFlags); } // ------------------------------------------------------------------------- status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) { int active; int timeout = 0; status_t result; audio_track_cblk_t* cblk = mCblk; uint32_t framesReq = audioBuffer->frameCount; uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS; audioBuffer->frameCount = 0; audioBuffer->size = 0; uint32_t framesAvail = cblk->framesAvailable(); if (framesAvail == 0) { Mutex::Autolock _l(cblk->lock); goto start_loop_here; while (framesAvail == 0) { active = mActive; if (UNLIKELY(!active)) { LOGV("Not active and NO_MORE_BUFFERS"); return NO_MORE_BUFFERS; } if (UNLIKELY(!waitCount)) return WOULD_BLOCK; timeout = 0; result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); if (__builtin_expect(result!=NO_ERROR, false)) { cblk->waitTimeMs += waitTimeMs; if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) { // timing out when a loop has been set and we have already written upto loop end // is a normal condition: no need to wake AudioFlinger up. if (cblk->user < cblk->loopEnd) { LOGW( "obtainBuffer timed out (is the CPU pegged?) %p " "user=%08x, server=%08x", this, cblk->user, cblk->server); //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) cblk->lock.unlock(); mAudioTrack->start(); cblk->lock.lock(); timeout = 1; } cblk->waitTimeMs = 0; } if (--waitCount == 0) { return TIMED_OUT; } } // read the server count again start_loop_here: framesAvail = cblk->framesAvailable_l(); } } cblk->waitTimeMs = 0; if (framesReq > framesAvail) { framesReq = framesAvail; } uint32_t u = cblk->user; uint32_t bufferEnd = cblk->userBase + cblk->frameCount; if (u + framesReq > bufferEnd) { framesReq = bufferEnd - u; } LOGW_IF(timeout, "*** SERIOUS WARNING *** obtainBuffer() timed out " "but didn't need to be locked. We recovered, but " "this shouldn't happen (user=%08x, server=%08x)", cblk->user, cblk->server); audioBuffer->flags = mMuted ? Buffer::MUTE : 0; audioBuffer->channelCount = mChannelCount; audioBuffer->frameCount = framesReq; audioBuffer->size = framesReq * cblk->frameSize; if (AudioSystem::isLinearPCM(mFormat)) { audioBuffer->format = AudioSystem::PCM_16_BIT; } else { audioBuffer->format = mFormat; } audioBuffer->raw = (int8_t *)cblk->buffer(u); active = mActive; return active ? status_t(NO_ERROR) : status_t(STOPPED); } void AudioTrack::releaseBuffer(Buffer* audioBuffer) { audio_track_cblk_t* cblk = mCblk; cblk->stepUser(audioBuffer->frameCount); } // ------------------------------------------------------------------------- ssize_t AudioTrack::write(const void* buffer, size_t userSize) { if (mSharedBuffer != 0) return INVALID_OPERATION; if (ssize_t(userSize) < 0) { // sanity-check. user is most-likely passing an error code. LOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); return BAD_VALUE; } LOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive); ssize_t written = 0; const int8_t *src = (const int8_t *)buffer; Buffer audioBuffer; do { audioBuffer.frameCount = userSize/frameSize(); // Calling obtainBuffer() with a negative wait count causes // an (almost) infinite wait time. status_t err = obtainBuffer(&audioBuffer, -1); if (err < 0) { // out of buffers, return #bytes written if (err == status_t(NO_MORE_BUFFERS)) break; return ssize_t(err); } size_t toWrite; if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) { // Divide capacity by 2 to take expansion into account toWrite = audioBuffer.size>>1; // 8 to 16 bit conversion int count = toWrite; int16_t *dst = (int16_t *)(audioBuffer.i8); while(count--) { *dst++ = (int16_t)(*src++^0x80) << 8; } } else { toWrite = audioBuffer.size; memcpy(audioBuffer.i8, src, toWrite); src += toWrite; } userSize -= toWrite; written += toWrite; releaseBuffer(&audioBuffer); } while (userSize); return written; } // ------------------------------------------------------------------------- bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) { Buffer audioBuffer; uint32_t frames; size_t writtenSize; // Manage underrun callback if (mActive && (mCblk->framesReady() == 0)) { LOGV("Underrun user: %x, server: %x, flowControlFlag %d", mCblk->user, mCblk->server, mCblk->flowControlFlag); if (mCblk->flowControlFlag == 0) { mCbf(EVENT_UNDERRUN, mUserData, 0); if (mCblk->server == mCblk->frameCount) { mCbf(EVENT_BUFFER_END, mUserData, 0); } mCblk->flowControlFlag = 1; if (mSharedBuffer != 0) return false; } } // Manage loop end callback while (mLoopCount > mCblk->loopCount) { int loopCount = -1; mLoopCount--; if (mLoopCount >= 0) loopCount = mLoopCount; mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount); } // Manage marker callback if (!mMarkerReached && (mMarkerPosition > 0)) { if (mCblk->server >= mMarkerPosition) { mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition); mMarkerReached = true; } } // Manage new position callback if (mUpdatePeriod > 0) { while (mCblk->server >= mNewPosition) { mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition); mNewPosition += mUpdatePeriod; } } // If Shared buffer is used, no data is requested from client. if (mSharedBuffer != 0) { frames = 0; } else { frames = mRemainingFrames; } do { audioBuffer.frameCount = frames; // Calling obtainBuffer() with a wait count of 1 // limits wait time to WAIT_PERIOD_MS. This prevents from being // stuck here not being able to handle timed events (position, markers, loops). status_t err = obtainBuffer(&audioBuffer, 1); if (err < NO_ERROR) { if (err != TIMED_OUT) { LOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up."); return false; } break; } if (err == status_t(STOPPED)) return false; // Divide buffer size by 2 to take into account the expansion // due to 8 to 16 bit conversion: the callback must fill only half // of the destination buffer if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) { audioBuffer.size >>= 1; } size_t reqSize = audioBuffer.size; mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); writtenSize = audioBuffer.size; // Sanity check on returned size if (ssize_t(writtenSize) <= 0) { // The callback is done filling buffers // Keep this thread going to handle timed events and // still try to get more data in intervals of WAIT_PERIOD_MS // but don't just loop and block the CPU, so wait usleep(WAIT_PERIOD_MS*1000); break; } if (writtenSize > reqSize) writtenSize = reqSize; if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) { // 8 to 16 bit conversion const int8_t *src = audioBuffer.i8 + writtenSize-1; int count = writtenSize; int16_t *dst = audioBuffer.i16 + writtenSize-1; while(count--) { *dst-- = (int16_t)(*src--^0x80) << 8; } writtenSize <<= 1; } audioBuffer.size = writtenSize; // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for // 8 bit PCM data: in this case, mCblk->frameSize is based on a sampel size of // 16 bit. audioBuffer.frameCount = writtenSize/mCblk->frameSize; frames -= audioBuffer.frameCount; releaseBuffer(&audioBuffer); } while (frames); if (frames == 0) { mRemainingFrames = mNotificationFrames; } else { mRemainingFrames = frames; } return true; } status_t AudioTrack::dump(int fd, const Vector<String16>& args) const { const size_t SIZE = 256; char buffer[SIZE]; String8 result; result.append(" AudioTrack::dump\n"); snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]); result.append(buffer); snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mFrameCount); result.append(buffer); snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted); result.append(buffer); snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); result.append(buffer); ::write(fd, result.string(), result.size()); return NO_ERROR; } // ========================================================================= AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) : Thread(bCanCallJava), mReceiver(receiver) { } bool AudioTrack::AudioTrackThread::threadLoop() { return mReceiver.processAudioBuffer(this); } status_t AudioTrack::AudioTrackThread::readyToRun() { return NO_ERROR; } void AudioTrack::AudioTrackThread::onFirstRef() { } // ========================================================================= audio_track_cblk_t::audio_track_cblk_t() : lock(Mutex::SHARED), user(0), server(0), userBase(0), serverBase(0), buffers(0), frameCount(0), loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0), flowControlFlag(1), forceReady(0) { } uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount) { uint32_t u = this->user; u += frameCount; // Ensure that user is never ahead of server for AudioRecord if (out) { // If stepServer() has been called once, switch to normal obtainBuffer() timeout period if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; } } else if (u > this->server) { LOGW("stepServer occured after track reset"); u = this->server; } if (u >= userBase + this->frameCount) { userBase += this->frameCount; } this->user = u; // Clear flow control error condition as new data has been written/read to/from buffer. flowControlFlag = 0; return u; } bool audio_track_cblk_t::stepServer(uint32_t frameCount) { // the code below simulates lock-with-timeout // we MUST do this to protect the AudioFlinger server // as this lock is shared with the client. status_t err; err = lock.tryLock(); if (err == -EBUSY) { // just wait a bit usleep(1000); err = lock.tryLock(); } if (err != NO_ERROR) { // probably, the client just died. return false; } uint32_t s = this->server; s += frameCount; if (out) { // Mark that we have read the first buffer so that next time stepUser() is called // we switch to normal obtainBuffer() timeout period if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) { bufferTimeoutMs = MAX_RUN_TIMEOUT_MS - 1; } // It is possible that we receive a flush() // while the mixer is processing a block: in this case, // stepServer() is called After the flush() has reset u & s and // we have s > u if (s > this->user) { LOGW("stepServer occured after track reset"); s = this->user; } } if (s >= loopEnd) { LOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd); s = loopStart; if (--loopCount == 0) { loopEnd = UINT_MAX; loopStart = UINT_MAX; } } if (s >= serverBase + this->frameCount) { serverBase += this->frameCount; } this->server = s; cv.signal(); lock.unlock(); return true; } void* audio_track_cblk_t::buffer(uint32_t offset) const { return (int8_t *)this->buffers + (offset - userBase) * this->frameSize; } uint32_t audio_track_cblk_t::framesAvailable() { Mutex::Autolock _l(lock); return framesAvailable_l(); } uint32_t audio_track_cblk_t::framesAvailable_l() { uint32_t u = this->user; uint32_t s = this->server; if (out) { uint32_t limit = (s < loopStart) ? s : loopStart; return limit + frameCount - u; } else { return frameCount + u - s; } } uint32_t audio_track_cblk_t::framesReady() { uint32_t u = this->user; uint32_t s = this->server; if (out) { if (u < loopEnd) { return u - s; } else { Mutex::Autolock _l(lock); if (loopCount >= 0) { return (loopEnd - loopStart)*loopCount + u - s; } else { return UINT_MAX; } } } else { return s - u; } } // ------------------------------------------------------------------------- }; // namespace android