| /* //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/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 { |
| // --------------------------------------------------------------------------- |
| |
| // static |
| status_t AudioTrack::getMinFrameCount( |
| int* frameCount, |
| int streamType, |
| uint32_t sampleRate) |
| { |
| 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; |
| } |
| |
| // Ensure that buffer depth covers at least audio hardware latency |
| uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); |
| if (minBufCount < 2) minBufCount = 2; |
| |
| *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : |
| afFrameCount * minBufCount * sampleRate / afSampleRate; |
| return NO_ERROR; |
| } |
| |
| // --------------------------------------------------------------------------- |
| |
| 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, |
| int sessionId) |
| : mStatus(NO_INIT) |
| { |
| mStatus = set(streamType, sampleRate, format, channels, |
| frameCount, flags, cbf, user, notificationFrames, |
| 0, false, sessionId); |
| } |
| |
| 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, |
| int sessionId) |
| : mStatus(NO_INIT) |
| { |
| mStatus = set(streamType, sampleRate, format, channels, |
| 0, flags, cbf, user, notificationFrames, |
| sharedBuffer, false, sessionId); |
| } |
| |
| 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(); |
| } |
| } |
| |
| 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, |
| int sessionId) |
| { |
| |
| LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); |
| |
| if (mAudioTrack != 0) { |
| LOGE("Track already in use"); |
| return INVALID_OPERATION; |
| } |
| |
| int afSampleRate; |
| if (AudioSystem::getOutputSamplingRate(&afSampleRate, 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; |
| } |
| |
| mVolume[LEFT] = 1.0f; |
| mVolume[RIGHT] = 1.0f; |
| mSendLevel = 0; |
| mFrameCount = frameCount; |
| mNotificationFramesReq = notificationFrames; |
| mSessionId = sessionId; |
| mAuxEffectId = 0; |
| |
| // create the IAudioTrack |
| status_t status = createTrack(streamType, sampleRate, format, channelCount, |
| frameCount, flags, sharedBuffer, output, true); |
| |
| if (status != NO_ERROR) { |
| return status; |
| } |
| |
| if (cbf != 0) { |
| mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); |
| if (mAudioTrackThread == 0) { |
| LOGE("Could not create callback thread"); |
| return NO_INIT; |
| } |
| } |
| |
| mStatus = NO_ERROR; |
| |
| mStreamType = streamType; |
| mFormat = format; |
| mChannels = channels; |
| mChannelCount = channelCount; |
| mSharedBuffer = sharedBuffer; |
| mMuted = false; |
| mActive = 0; |
| mCbf = cbf; |
| mUserData = user; |
| 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 mCblk->frameCount; |
| } |
| |
| 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; |
| status_t status; |
| |
| 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) { |
| mNewPosition = mCblk->server + mUpdatePeriod; |
| mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; |
| mCblk->waitTimeMs = 0; |
| mCblk->flags &= ~CBLK_DISABLED_ON; |
| if (t != 0) { |
| t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT); |
| } else { |
| setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT); |
| } |
| |
| if (mCblk->flags & CBLK_INVALID_MSK) { |
| LOGW("start() track %p invalidated, creating a new one", this); |
| // no need to clear the invalid flag as this cblk will not be used anymore |
| // force new track creation |
| status = DEAD_OBJECT; |
| } else { |
| status = mAudioTrack->start(); |
| } |
| if (status == DEAD_OBJECT) { |
| LOGV("start() dead IAudioTrack: creating a new one"); |
| status = createTrack(mStreamType, mCblk->sampleRate, mFormat, mChannelCount, |
| mFrameCount, mFlags, mSharedBuffer, getOutput(), false); |
| if (status == NO_ERROR) { |
| status = mAudioTrack->start(); |
| if (status == NO_ERROR) { |
| mNewPosition = mCblk->server + mUpdatePeriod; |
| } |
| } |
| } |
| if (status != NO_ERROR) { |
| LOGV("start() failed"); |
| android_atomic_and(~1, &mActive); |
| if (t != 0) { |
| t->requestExit(); |
| } else { |
| setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL); |
| } |
| } |
| } |
| |
| 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); |
| } |
| } |
| |
| 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) { |
| mAudioTrack->pause(); |
| } |
| } |
| |
| void AudioTrack::mute(bool e) |
| { |
| mAudioTrack->mute(e); |
| mMuted = e; |
| } |
| |
| bool AudioTrack::muted() const |
| { |
| return mMuted; |
| } |
| |
| status_t AudioTrack::setVolume(float left, float right) |
| { |
| if (left > 1.0f || right > 1.0f) { |
| return BAD_VALUE; |
| } |
| |
| mVolume[LEFT] = left; |
| mVolume[RIGHT] = right; |
| |
| // write must be atomic |
| mCblk->volumeLR = (uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000); |
| |
| return NO_ERROR; |
| } |
| |
| void AudioTrack::getVolume(float* left, float* right) |
| { |
| if (left != NULL) { |
| *left = mVolume[LEFT]; |
| } |
| if (right != NULL) { |
| *right = mVolume[RIGHT]; |
| } |
| } |
| |
| status_t AudioTrack::setAuxEffectSendLevel(float level) |
| { |
| LOGV("setAuxEffectSendLevel(%f)", level); |
| if (level > 1.0f) { |
| return BAD_VALUE; |
| } |
| |
| mSendLevel = level; |
| |
| mCblk->sendLevel = uint16_t(level * 0x1000); |
| |
| return NO_ERROR; |
| } |
| |
| void AudioTrack::getAuxEffectSendLevel(float* level) |
| { |
| if (level != NULL) { |
| *level = mSendLevel; |
| } |
| } |
| |
| 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 > cblk->frameCount) { |
| LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user); |
| return BAD_VALUE; |
| } |
| |
| if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) { |
| LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d", |
| loopStart, loopEnd, cblk->frameCount); |
| 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->flags |= CBLK_FORCEREADY_ON; |
| |
| 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(mCblk->frameCount); |
| |
| return NO_ERROR; |
| } |
| |
| audio_io_handle_t AudioTrack::getOutput() |
| { |
| return AudioSystem::getOutput((AudioSystem::stream_type)mStreamType, |
| mCblk->sampleRate, mFormat, mChannels, (AudioSystem::output_flags)mFlags); |
| } |
| |
| int AudioTrack::getSessionId() |
| { |
| return mSessionId; |
| } |
| |
| status_t AudioTrack::attachAuxEffect(int effectId) |
| { |
| LOGV("attachAuxEffect(%d)", effectId); |
| status_t status = mAudioTrack->attachAuxEffect(effectId); |
| if (status == NO_ERROR) { |
| mAuxEffectId = effectId; |
| } |
| return status; |
| } |
| |
| // ------------------------------------------------------------------------- |
| |
| status_t AudioTrack::createTrack( |
| int streamType, |
| uint32_t sampleRate, |
| int format, |
| int channelCount, |
| int frameCount, |
| uint32_t flags, |
| const sp<IMemory>& sharedBuffer, |
| audio_io_handle_t output, |
| bool enforceFrameCount) |
| { |
| status_t status; |
| 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; |
| } |
| |
| mNotificationFramesAct = mNotificationFramesReq; |
| 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 (mNotificationFramesAct == 0) { |
| mNotificationFramesAct = frameCount/2; |
| } |
| // Make sure that application is notified with sufficient margin |
| // before underrun |
| if (mNotificationFramesAct > (uint32_t)frameCount/2) { |
| mNotificationFramesAct = frameCount/2; |
| } |
| if (frameCount < minFrameCount) { |
| if (enforceFrameCount) { |
| LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount); |
| return BAD_VALUE; |
| } else { |
| frameCount = minFrameCount; |
| } |
| } |
| } 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); |
| } |
| } |
| |
| sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), |
| streamType, |
| sampleRate, |
| format, |
| channelCount, |
| frameCount, |
| ((uint16_t)flags) << 16, |
| sharedBuffer, |
| output, |
| &mSessionId, |
| &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; |
| } |
| mAudioTrack.clear(); |
| mAudioTrack = track; |
| mCblkMemory.clear(); |
| mCblkMemory = cblk; |
| mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); |
| mCblk->flags |= CBLK_DIRECTION_OUT; |
| 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(mCblk->frameCount); |
| } |
| |
| mCblk->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000); |
| mCblk->sendLevel = uint16_t(mSendLevel * 0x1000); |
| mAudioTrack->attachAuxEffect(mAuxEffectId); |
| mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; |
| mCblk->waitTimeMs = 0; |
| mRemainingFrames = mNotificationFramesAct; |
| mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate; |
| return NO_ERROR; |
| } |
| |
| status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) |
| { |
| int active; |
| 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) { |
| cblk->lock.lock(); |
| goto start_loop_here; |
| while (framesAvail == 0) { |
| active = mActive; |
| if (UNLIKELY(!active)) { |
| LOGV("Not active and NO_MORE_BUFFERS"); |
| cblk->lock.unlock(); |
| return NO_MORE_BUFFERS; |
| } |
| if (UNLIKELY(!waitCount)) { |
| cblk->lock.unlock(); |
| return WOULD_BLOCK; |
| } |
| if (!(cblk->flags & CBLK_INVALID_MSK)) { |
| result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); |
| } |
| if (cblk->flags & CBLK_INVALID_MSK) { |
| LOGW("obtainBuffer() track %p invalidated, creating a new one", this); |
| // no need to clear the invalid flag as this cblk will not be used anymore |
| cblk->lock.unlock(); |
| goto create_new_track; |
| } |
| 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(); |
| result = mAudioTrack->start(); |
| if (result == DEAD_OBJECT) { |
| LOGW("obtainBuffer() dead IAudioTrack: creating a new one"); |
| create_new_track: |
| result = createTrack(mStreamType, cblk->sampleRate, mFormat, mChannelCount, |
| mFrameCount, mFlags, mSharedBuffer, getOutput(), false); |
| if (result == NO_ERROR) { |
| cblk = mCblk; |
| cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; |
| mAudioTrack->start(); |
| } |
| } |
| cblk->lock.lock(); |
| } |
| cblk->waitTimeMs = 0; |
| } |
| |
| if (--waitCount == 0) { |
| cblk->lock.unlock(); |
| return TIMED_OUT; |
| } |
| } |
| // read the server count again |
| start_loop_here: |
| framesAvail = cblk->framesAvailable_l(); |
| } |
| cblk->lock.unlock(); |
| } |
| |
| // restart track if it was disabled by audioflinger due to previous underrun |
| if (cblk->flags & CBLK_DISABLED_MSK) { |
| cblk->flags &= ~CBLK_DISABLED_ON; |
| LOGW("obtainBuffer() track %p disabled, restarting", this); |
| mAudioTrack->start(); |
| } |
| |
| 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; |
| } |
| |
| 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, flags %04x", mCblk->user, mCblk->server, mCblk->flags); |
| if ((mCblk->flags & CBLK_UNDERRUN_MSK) == CBLK_UNDERRUN_OFF) { |
| mCbf(EVENT_UNDERRUN, mUserData, 0); |
| if (mCblk->server == mCblk->frameCount) { |
| mCbf(EVENT_BUFFER_END, mUserData, 0); |
| } |
| mCblk->flags |= CBLK_UNDERRUN_ON; |
| 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 = mNotificationFramesAct; |
| } 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, mCblk->frameCount); |
| 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), cv(Condition::SHARED), user(0), server(0), |
| userBase(0), serverBase(0), buffers(0), frameCount(0), |
| loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0), |
| flags(0), sendLevel(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 (flags & CBLK_DIRECTION_MSK) { |
| // 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. |
| flags &= ~CBLK_UNDERRUN_MSK; |
| |
| 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 (flags & CBLK_DIRECTION_MSK) { |
| // 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_STARTUP_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 (flags & CBLK_DIRECTION_MSK) { |
| 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 (flags & CBLK_DIRECTION_MSK) { |
| 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 |
| |