/* ** ** 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 #include #include #include #include #include #include #define WAIT_PERIOD_MS 10 #define WAIT_STREAM_END_TIMEOUT_SEC 120 namespace android { // --------------------------------------------------------------------------- // static status_t AudioTrack::getMinFrameCount( size_t* frameCount, audio_stream_type_t streamType, uint32_t sampleRate) { if (frameCount == NULL) { return BAD_VALUE; } // FIXME merge with similar code in createTrack_l(), except we're missing // some information here that is available in createTrack_l(): // audio_io_handle_t output // audio_format_t format // audio_channel_mask_t channelMask // audio_output_flags_t flags uint32_t afSampleRate; status_t status; status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); if (status != NO_ERROR) { ALOGE("Unable to query output sample rate for stream type %d; status %d", streamType, status); return status; } size_t afFrameCount; status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); if (status != NO_ERROR) { ALOGE("Unable to query output frame count for stream type %d; status %d", streamType, status); return status; } uint32_t afLatency; status = AudioSystem::getOutputLatency(&afLatency, streamType); if (status != NO_ERROR) { ALOGE("Unable to query output latency for stream type %d; status %d", streamType, status); return status; } // 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; // The formula above should always produce a non-zero value, but return an error // in the unlikely event that it does not, as that's part of the API contract. if (*frameCount == 0) { ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d", streamType, sampleRate); return BAD_VALUE; } ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); return NO_ERROR; } // --------------------------------------------------------------------------- AudioTrack::AudioTrack() : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT) { } AudioTrack::AudioTrack( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, audio_output_flags_t flags, callback_t cbf, void* user, uint32_t notificationFrames, int sessionId, transfer_type transferType, const audio_offload_info_t *offloadInfo, int uid, pid_t pid) : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT) { mStatus = set(streamType, sampleRate, format, channelMask, frameCount, flags, cbf, user, notificationFrames, 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, uid, pid); } AudioTrack::AudioTrack( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, const sp& sharedBuffer, audio_output_flags_t flags, callback_t cbf, void* user, uint32_t notificationFrames, int sessionId, transfer_type transferType, const audio_offload_info_t *offloadInfo, int uid, pid_t pid) : mStatus(NO_INIT), mIsTimed(false), mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT) { mStatus = set(streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags, cbf, user, notificationFrames, sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, uid, pid); } AudioTrack::~AudioTrack() { 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) { mProxy->interrupt(); mAudioTrackThread->requestExit(); // see comment in AudioTrack.h mAudioTrackThread->requestExitAndWait(); mAudioTrackThread.clear(); } mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); mAudioTrack.clear(); IPCThreadState::self()->flushCommands(); ALOGV("~AudioTrack, releasing session id from %d on behalf of %d", IPCThreadState::self()->getCallingPid(), mClientPid); AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); } } status_t AudioTrack::set( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, audio_output_flags_t flags, callback_t cbf, void* user, uint32_t notificationFrames, const sp& sharedBuffer, bool threadCanCallJava, int sessionId, transfer_type transferType, const audio_offload_info_t *offloadInfo, int uid, pid_t pid) { ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " "flags #%x, notificationFrames %u, sessionId %d, transferType %d", streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, sessionId, transferType); switch (transferType) { case TRANSFER_DEFAULT: if (sharedBuffer != 0) { transferType = TRANSFER_SHARED; } else if (cbf == NULL || threadCanCallJava) { transferType = TRANSFER_SYNC; } else { transferType = TRANSFER_CALLBACK; } break; case TRANSFER_CALLBACK: if (cbf == NULL || sharedBuffer != 0) { ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); return BAD_VALUE; } break; case TRANSFER_OBTAIN: case TRANSFER_SYNC: if (sharedBuffer != 0) { ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); return BAD_VALUE; } break; case TRANSFER_SHARED: if (sharedBuffer == 0) { ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); return BAD_VALUE; } break; default: ALOGE("Invalid transfer type %d", transferType); return BAD_VALUE; } mSharedBuffer = sharedBuffer; mTransfer = transferType; ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); AutoMutex lock(mLock); // invariant that mAudioTrack != 0 is true only after set() returns successfully if (mAudioTrack != 0) { ALOGE("Track already in use"); return INVALID_OPERATION; } // handle default values first. if (streamType == AUDIO_STREAM_DEFAULT) { streamType = AUDIO_STREAM_MUSIC; } if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { ALOGE("Invalid stream type %d", streamType); return BAD_VALUE; } mStreamType = streamType; status_t status; if (sampleRate == 0) { status = AudioSystem::getOutputSamplingRate(&sampleRate, streamType); if (status != NO_ERROR) { ALOGE("Could not get output sample rate for stream type %d; status %d", streamType, status); return status; } } mSampleRate = sampleRate; // these below should probably come from the audioFlinger too... if (format == AUDIO_FORMAT_DEFAULT) { format = AUDIO_FORMAT_PCM_16_BIT; } // validate parameters if (!audio_is_valid_format(format)) { ALOGE("Invalid format %#x", format); return BAD_VALUE; } mFormat = format; if (!audio_is_output_channel(channelMask)) { ALOGE("Invalid channel mask %#x", channelMask); return BAD_VALUE; } mChannelMask = channelMask; uint32_t channelCount = popcount(channelMask); mChannelCount = channelCount; // AudioFlinger does not currently support 8-bit data in shared memory if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { ALOGE("8-bit data in shared memory is not supported"); return BAD_VALUE; } // force direct flag if format is not linear PCM // or offload was requested if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) || !audio_is_linear_pcm(format)) { ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) ? "Offload request, forcing to Direct Output" : "Not linear PCM, forcing to Direct Output"); flags = (audio_output_flags_t) // FIXME why can't we allow direct AND fast? ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); } // only allow deep buffering for music stream type if (streamType != AUDIO_STREAM_MUSIC) { flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); } if (audio_is_linear_pcm(format)) { mFrameSize = channelCount * audio_bytes_per_sample(format); mFrameSizeAF = channelCount * sizeof(int16_t); } else { mFrameSize = sizeof(uint8_t); mFrameSizeAF = sizeof(uint8_t); } // Make copy of input parameter offloadInfo so that in the future: // (a) createTrack_l doesn't need it as an input parameter // (b) we can support re-creation of offloaded tracks if (offloadInfo != NULL) { mOffloadInfoCopy = *offloadInfo; mOffloadInfo = &mOffloadInfoCopy; } else { mOffloadInfo = NULL; } mVolume[LEFT] = 1.0f; mVolume[RIGHT] = 1.0f; mSendLevel = 0.0f; // mFrameCount is initialized in createTrack_l mReqFrameCount = frameCount; mNotificationFramesReq = notificationFrames; mNotificationFramesAct = 0; mSessionId = sessionId; int callingpid = IPCThreadState::self()->getCallingPid(); int mypid = getpid(); if (uid == -1 || (callingpid != mypid)) { mClientUid = IPCThreadState::self()->getCallingUid(); } else { mClientUid = uid; } if (pid == -1 || (callingpid != mypid)) { mClientPid = callingpid; } else { mClientPid = pid; } mAuxEffectId = 0; mFlags = flags; mCbf = cbf; if (cbf != NULL) { mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); } // create the IAudioTrack status = createTrack_l(0 /*epoch*/); if (status != NO_ERROR) { if (mAudioTrackThread != 0) { mAudioTrackThread->requestExit(); // see comment in AudioTrack.h mAudioTrackThread->requestExitAndWait(); mAudioTrackThread.clear(); } // Use of direct and offloaded output streams is ref counted by audio policy manager. #if 0 // FIXME This should no longer be needed //Use of direct and offloaded output streams is ref counted by audio policy manager. // As getOutput was called above and resulted in an output stream to be opened, // we need to release it. if (mOutput != 0) { AudioSystem::releaseOutput(mOutput); mOutput = 0; } #endif return status; } mStatus = NO_ERROR; mState = STATE_STOPPED; mUserData = user; mLoopPeriod = 0; mMarkerPosition = 0; mMarkerReached = false; mNewPosition = 0; mUpdatePeriod = 0; AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); mSequence = 1; mObservedSequence = mSequence; mInUnderrun = false; return NO_ERROR; } // ------------------------------------------------------------------------- status_t AudioTrack::start() { AutoMutex lock(mLock); if (mState == STATE_ACTIVE) { return INVALID_OPERATION; } mInUnderrun = true; State previousState = mState; if (previousState == STATE_PAUSED_STOPPING) { mState = STATE_STOPPING; } else { mState = STATE_ACTIVE; } if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { // reset current position as seen by client to 0 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); // force refresh of remaining frames by processAudioBuffer() as last // write before stop could be partial. mRefreshRemaining = true; } mNewPosition = mProxy->getPosition() + mUpdatePeriod; int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); sp t = mAudioTrackThread; if (t != 0) { if (previousState == STATE_STOPPING) { mProxy->interrupt(); } else { t->resume(); } } else { mPreviousPriority = getpriority(PRIO_PROCESS, 0); get_sched_policy(0, &mPreviousSchedulingGroup); androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); } status_t status = NO_ERROR; if (!(flags & CBLK_INVALID)) { status = mAudioTrack->start(); if (status == DEAD_OBJECT) { flags |= CBLK_INVALID; } } if (flags & CBLK_INVALID) { status = restoreTrack_l("start"); } if (status != NO_ERROR) { ALOGE("start() status %d", status); mState = previousState; if (t != 0) { if (previousState != STATE_STOPPING) { t->pause(); } } else { setpriority(PRIO_PROCESS, 0, mPreviousPriority); set_sched_policy(0, mPreviousSchedulingGroup); } } return status; } void AudioTrack::stop() { AutoMutex lock(mLock); if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { return; } if (isOffloaded_l()) { mState = STATE_STOPPING; } else { mState = STATE_STOPPED; } mProxy->interrupt(); mAudioTrack->stop(); // the playback head position will reset to 0, so if a marker is set, we need // to activate it again mMarkerReached = false; #if 0 // Force flush if a shared buffer is used otherwise audioflinger // will not stop before end of buffer is reached. // It may be needed to make sure that we stop playback, likely in case looping is on. if (mSharedBuffer != 0) { flush_l(); } #endif sp t = mAudioTrackThread; if (t != 0) { if (!isOffloaded_l()) { t->pause(); } } else { setpriority(PRIO_PROCESS, 0, mPreviousPriority); set_sched_policy(0, mPreviousSchedulingGroup); } } bool AudioTrack::stopped() const { AutoMutex lock(mLock); return mState != STATE_ACTIVE; } void AudioTrack::flush() { if (mSharedBuffer != 0) { return; } AutoMutex lock(mLock); if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { return; } flush_l(); } void AudioTrack::flush_l() { ALOG_ASSERT(mState != STATE_ACTIVE); // clear playback marker and periodic update counter mMarkerPosition = 0; mMarkerReached = false; mUpdatePeriod = 0; mRefreshRemaining = true; mState = STATE_FLUSHED; if (isOffloaded_l()) { mProxy->interrupt(); } mProxy->flush(); mAudioTrack->flush(); } void AudioTrack::pause() { AutoMutex lock(mLock); if (mState == STATE_ACTIVE) { mState = STATE_PAUSED; } else if (mState == STATE_STOPPING) { mState = STATE_PAUSED_STOPPING; } else { return; } mProxy->interrupt(); mAudioTrack->pause(); } status_t AudioTrack::setVolume(float left, float right) { if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { return BAD_VALUE; } AutoMutex lock(mLock); mVolume[LEFT] = left; mVolume[RIGHT] = right; mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); if (isOffloaded_l()) { mAudioTrack->signal(); } return NO_ERROR; } status_t AudioTrack::setVolume(float volume) { return setVolume(volume, volume); } status_t AudioTrack::setAuxEffectSendLevel(float level) { if (level < 0.0f || level > 1.0f) { return BAD_VALUE; } AutoMutex lock(mLock); mSendLevel = level; mProxy->setSendLevel(level); return NO_ERROR; } void AudioTrack::getAuxEffectSendLevel(float* level) const { if (level != NULL) { *level = mSendLevel; } } status_t AudioTrack::setSampleRate(uint32_t rate) { if (mIsTimed || isOffloaded()) { return INVALID_OPERATION; } uint32_t 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; } AutoMutex lock(mLock); mSampleRate = rate; mProxy->setSampleRate(rate); return NO_ERROR; } uint32_t AudioTrack::getSampleRate() const { if (mIsTimed) { return 0; } AutoMutex lock(mLock); // sample rate can be updated during playback by the offloaded decoder so we need to // query the HAL and update if needed. // FIXME use Proxy return channel to update the rate from server and avoid polling here if (isOffloaded_l()) { if (mOutput != 0) { uint32_t sampleRate = 0; status_t status = AudioSystem::getSamplingRate(mOutput, mStreamType, &sampleRate); if (status == NO_ERROR) { mSampleRate = sampleRate; } } } return mSampleRate; } status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) { if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { return INVALID_OPERATION; } if (loopCount == 0) { ; } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && loopEnd - loopStart >= MIN_LOOP) { ; } else { return BAD_VALUE; } AutoMutex lock(mLock); // See setPosition() regarding setting parameters such as loop points or position while active if (mState == STATE_ACTIVE) { return INVALID_OPERATION; } setLoop_l(loopStart, loopEnd, loopCount); return NO_ERROR; } void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) { // FIXME If setting a loop also sets position to start of loop, then // this is correct. Otherwise it should be removed. mNewPosition = mProxy->getPosition() + mUpdatePeriod; mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; mStaticProxy->setLoop(loopStart, loopEnd, loopCount); } status_t AudioTrack::setMarkerPosition(uint32_t marker) { // The only purpose of setting marker position is to get a callback if (mCbf == NULL || isOffloaded()) { return INVALID_OPERATION; } AutoMutex lock(mLock); mMarkerPosition = marker; mMarkerReached = false; return NO_ERROR; } status_t AudioTrack::getMarkerPosition(uint32_t *marker) const { if (isOffloaded()) { return INVALID_OPERATION; } if (marker == NULL) { return BAD_VALUE; } AutoMutex lock(mLock); *marker = mMarkerPosition; return NO_ERROR; } status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) { // The only purpose of setting position update period is to get a callback if (mCbf == NULL || isOffloaded()) { return INVALID_OPERATION; } AutoMutex lock(mLock); mNewPosition = mProxy->getPosition() + updatePeriod; mUpdatePeriod = updatePeriod; return NO_ERROR; } status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const { if (isOffloaded()) { return INVALID_OPERATION; } if (updatePeriod == NULL) { return BAD_VALUE; } AutoMutex lock(mLock); *updatePeriod = mUpdatePeriod; return NO_ERROR; } status_t AudioTrack::setPosition(uint32_t position) { if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { return INVALID_OPERATION; } if (position > mFrameCount) { return BAD_VALUE; } AutoMutex lock(mLock); // Currently we require that the player is inactive before setting parameters such as position // or loop points. Otherwise, there could be a race condition: the application could read the // current position, compute a new position or loop parameters, and then set that position or // loop parameters but it would do the "wrong" thing since the position has continued to advance // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app // to specify how it wants to handle such scenarios. if (mState == STATE_ACTIVE) { return INVALID_OPERATION; } mNewPosition = mProxy->getPosition() + mUpdatePeriod; mLoopPeriod = 0; // FIXME Check whether loops and setting position are incompatible in old code. // If we use setLoop for both purposes we lose the capability to set the position while looping. mStaticProxy->setLoop(position, mFrameCount, 0); return NO_ERROR; } status_t AudioTrack::getPosition(uint32_t *position) const { if (position == NULL) { return BAD_VALUE; } AutoMutex lock(mLock); if (isOffloaded_l()) { uint32_t dspFrames = 0; if (mOutput != 0) { uint32_t halFrames; AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); } *position = dspFrames; } else { // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : mProxy->getPosition(); } return NO_ERROR; } status_t AudioTrack::getBufferPosition(uint32_t *position) { if (mSharedBuffer == 0 || mIsTimed) { return INVALID_OPERATION; } if (position == NULL) { return BAD_VALUE; } AutoMutex lock(mLock); *position = mStaticProxy->getBufferPosition(); return NO_ERROR; } status_t AudioTrack::reload() { if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) { return INVALID_OPERATION; } AutoMutex lock(mLock); // See setPosition() regarding setting parameters such as loop points or position while active if (mState == STATE_ACTIVE) { return INVALID_OPERATION; } mNewPosition = mUpdatePeriod; mLoopPeriod = 0; // FIXME The new code cannot reload while keeping a loop specified. // Need to check how the old code handled this, and whether it's a significant change. mStaticProxy->setLoop(0, mFrameCount, 0); return NO_ERROR; } audio_io_handle_t AudioTrack::getOutput() const { AutoMutex lock(mLock); return mOutput; } status_t AudioTrack::attachAuxEffect(int effectId) { AutoMutex lock(mLock); status_t status = mAudioTrack->attachAuxEffect(effectId); if (status == NO_ERROR) { mAuxEffectId = effectId; } return status; } // ------------------------------------------------------------------------- // must be called with mLock held status_t AudioTrack::createTrack_l(size_t epoch) { status_t status; const sp& audioFlinger = AudioSystem::get_audio_flinger(); if (audioFlinger == 0) { ALOGE("Could not get audioflinger"); return NO_INIT; } audio_io_handle_t output = AudioSystem::getOutput(mStreamType, mSampleRate, mFormat, mChannelMask, mFlags, mOffloadInfo); if (output == 0) { ALOGE("Could not get audio output for stream type %d, sample rate %u, format %#x, " "channel mask %#x, flags %#x", mStreamType, mSampleRate, mFormat, mChannelMask, mFlags); return BAD_VALUE; } { // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, // we must release it ourselves if anything goes wrong. // Not all of these values are needed under all conditions, but it is easier to get them all uint32_t afLatency; status = AudioSystem::getLatency(output, mStreamType, &afLatency); if (status != NO_ERROR) { ALOGE("getLatency(%d) failed status %d", output, status); goto release; } size_t afFrameCount; status = AudioSystem::getFrameCount(output, mStreamType, &afFrameCount); if (status != NO_ERROR) { ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, mStreamType, status); goto release; } uint32_t afSampleRate; status = AudioSystem::getSamplingRate(output, mStreamType, &afSampleRate); if (status != NO_ERROR) { ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, mStreamType, status); goto release; } // Client decides whether the track is TIMED (see below), but can only express a preference // for FAST. Server will perform additional tests. if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(( // either of these use cases: // use case 1: shared buffer (mSharedBuffer != 0) || // use case 2: callback transfer mode (mTransfer == TRANSFER_CALLBACK)) && // matching sample rate (mSampleRate == afSampleRate))) { ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); // once denied, do not request again if IAudioTrack is re-created mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); } ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where // n = 1 fast track with single buffering; nBuffering is ignored // n = 2 fast track with double buffering // n = 2 normal track, no sample rate conversion // n = 3 normal track, with sample rate conversion // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) // n > 3 very high latency or very small notification interval; nBuffering is ignored const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3; mNotificationFramesAct = mNotificationFramesReq; size_t frameCount = mReqFrameCount; if (!audio_is_linear_pcm(mFormat)) { if (mSharedBuffer != 0) { // Same comment as below about ignoring frameCount parameter for set() frameCount = mSharedBuffer->size(); } else if (frameCount == 0) { frameCount = afFrameCount; } if (mNotificationFramesAct != frameCount) { mNotificationFramesAct = frameCount; } } else if (mSharedBuffer != 0) { // Ensure that buffer alignment matches channel count // 8-bit data in shared memory is not currently supported by AudioFlinger size_t alignment = /* mFormat == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; if (mChannelCount > 1) { // More than 2 channels does not require stronger alignment than stereo alignment <<= 1; } if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { ALOGE("Invalid buffer alignment: address %p, channel count %u", mSharedBuffer->pointer(), mChannelCount); status = BAD_VALUE; goto release; } // When initializing a shared buffer AudioTrack via constructors, // there's no frameCount parameter. // But when initializing a shared buffer AudioTrack via set(), // there _is_ a frameCount parameter. We silently ignore it. frameCount = mSharedBuffer->size()/mChannelCount/sizeof(int16_t); } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { // FIXME move these calculations and associated checks to server // Ensure that buffer depth covers at least audio hardware latency uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", afFrameCount, minBufCount, afSampleRate, afLatency); if (minBufCount <= nBuffering) { minBufCount = nBuffering; } size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate; ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" ", afLatency=%d", minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency); if (frameCount == 0) { frameCount = minFrameCount; } else if (frameCount < minFrameCount) { // not ALOGW because it happens all the time when playing key clicks over A2DP ALOGV("Minimum buffer size corrected from %d to %d", frameCount, minFrameCount); frameCount = minFrameCount; } // Make sure that application is notified with sufficient margin before underrun if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { mNotificationFramesAct = frameCount/nBuffering; } } else { // For fast tracks, the frame count calculations and checks are done by server } IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; if (mIsTimed) { trackFlags |= IAudioFlinger::TRACK_TIMED; } pid_t tid = -1; if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { trackFlags |= IAudioFlinger::TRACK_FAST; if (mAudioTrackThread != 0) { tid = mAudioTrackThread->getTid(); } } if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { trackFlags |= IAudioFlinger::TRACK_OFFLOAD; } size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, // but we will still need the original value also sp track = audioFlinger->createTrack(mStreamType, mSampleRate, // AudioFlinger only sees 16-bit PCM mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat, mChannelMask, &temp, &trackFlags, mSharedBuffer, output, tid, &mSessionId, mName, mClientUid, &status); if (status != NO_ERROR) { ALOGE("AudioFlinger could not create track, status: %d", status); goto release; } ALOG_ASSERT(track != 0); // AudioFlinger now owns the reference to the I/O handle, // so we are no longer responsible for releasing it. sp iMem = track->getCblk(); if (iMem == 0) { ALOGE("Could not get control block"); return NO_INIT; } void *iMemPointer = iMem->pointer(); if (iMemPointer == NULL) { ALOGE("Could not get control block pointer"); return NO_INIT; } // invariant that mAudioTrack != 0 is true only after set() returns successfully if (mAudioTrack != 0) { mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); mDeathNotifier.clear(); } mAudioTrack = track; mCblkMemory = iMem; audio_track_cblk_t* cblk = static_cast(iMemPointer); mCblk = cblk; // note that temp is the (possibly revised) value of frameCount if (temp < frameCount || (frameCount == 0 && temp == 0)) { // In current design, AudioTrack client checks and ensures frame count validity before // passing it to AudioFlinger so AudioFlinger should not return a different value except // for fast track as it uses a special method of assigning frame count. ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); } frameCount = temp; mAwaitBoost = false; if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { if (trackFlags & IAudioFlinger::TRACK_FAST) { ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); mAwaitBoost = true; if (mSharedBuffer == 0) { // Theoretically double-buffering is not required for fast tracks, // due to tighter scheduling. But in practice, to accommodate kernels with // scheduling jitter, and apps with computation jitter, we use double-buffering. if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { mNotificationFramesAct = frameCount/nBuffering; } } } else { ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); // once denied, do not request again if IAudioTrack is re-created mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); if (mSharedBuffer == 0) { if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { mNotificationFramesAct = frameCount/nBuffering; } } } } if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); } else { ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); // FIXME This is a warning, not an error, so don't return error status //return NO_INIT; } } // We retain a copy of the I/O handle, but don't own the reference mOutput = output; mRefreshRemaining = true; // Starting address of buffers in shared memory. If there is a shared buffer, buffers // is the value of pointer() for the shared buffer, otherwise buffers points // immediately after the control block. This address is for the mapping within client // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. void* buffers; if (mSharedBuffer == 0) { buffers = (char*)cblk + sizeof(audio_track_cblk_t); } else { buffers = mSharedBuffer->pointer(); } mAudioTrack->attachAuxEffect(mAuxEffectId); // FIXME don't believe this lie mLatency = afLatency + (1000*frameCount) / mSampleRate; mFrameCount = frameCount; // If IAudioTrack is re-created, don't let the requested frameCount // decrease. This can confuse clients that cache frameCount(). if (frameCount > mReqFrameCount) { mReqFrameCount = frameCount; } // update proxy if (mSharedBuffer == 0) { mStaticProxy.clear(); mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); } else { mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); mProxy = mStaticProxy; } mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000)); mProxy->setSendLevel(mSendLevel); mProxy->setSampleRate(mSampleRate); mProxy->setEpoch(epoch); mProxy->setMinimum(mNotificationFramesAct); mDeathNotifier = new DeathNotifier(this); mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); return NO_ERROR; } release: AudioSystem::releaseOutput(output); if (status == NO_ERROR) { status = NO_INIT; } return status; } status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) { if (audioBuffer == NULL) { return BAD_VALUE; } if (mTransfer != TRANSFER_OBTAIN) { audioBuffer->frameCount = 0; audioBuffer->size = 0; audioBuffer->raw = NULL; return INVALID_OPERATION; } const struct timespec *requested; struct timespec timeout; if (waitCount == -1) { requested = &ClientProxy::kForever; } else if (waitCount == 0) { requested = &ClientProxy::kNonBlocking; } else if (waitCount > 0) { long long ms = WAIT_PERIOD_MS * (long long) waitCount; timeout.tv_sec = ms / 1000; timeout.tv_nsec = (int) (ms % 1000) * 1000000; requested = &timeout; } else { ALOGE("%s invalid waitCount %d", __func__, waitCount); requested = NULL; } return obtainBuffer(audioBuffer, requested); } status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, struct timespec *elapsed, size_t *nonContig) { // previous and new IAudioTrack sequence numbers are used to detect track re-creation uint32_t oldSequence = 0; uint32_t newSequence; Proxy::Buffer buffer; status_t status = NO_ERROR; static const int32_t kMaxTries = 5; int32_t tryCounter = kMaxTries; do { // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to // keep them from going away if another thread re-creates the track during obtainBuffer() sp proxy; sp iMem; { // start of lock scope AutoMutex lock(mLock); newSequence = mSequence; // did previous obtainBuffer() fail due to media server death or voluntary invalidation? if (status == DEAD_OBJECT) { // re-create track, unless someone else has already done so if (newSequence == oldSequence) { status = restoreTrack_l("obtainBuffer"); if (status != NO_ERROR) { buffer.mFrameCount = 0; buffer.mRaw = NULL; buffer.mNonContig = 0; break; } } } oldSequence = newSequence; // Keep the extra references proxy = mProxy; iMem = mCblkMemory; if (mState == STATE_STOPPING) { status = -EINTR; buffer.mFrameCount = 0; buffer.mRaw = NULL; buffer.mNonContig = 0; break; } // Non-blocking if track is stopped or paused if (mState != STATE_ACTIVE) { requested = &ClientProxy::kNonBlocking; } } // end of lock scope buffer.mFrameCount = audioBuffer->frameCount; // FIXME starts the requested timeout and elapsed over from scratch status = proxy->obtainBuffer(&buffer, requested, elapsed); } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); audioBuffer->frameCount = buffer.mFrameCount; audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; audioBuffer->raw = buffer.mRaw; if (nonContig != NULL) { *nonContig = buffer.mNonContig; } return status; } void AudioTrack::releaseBuffer(Buffer* audioBuffer) { if (mTransfer == TRANSFER_SHARED) { return; } size_t stepCount = audioBuffer->size / mFrameSizeAF; if (stepCount == 0) { return; } Proxy::Buffer buffer; buffer.mFrameCount = stepCount; buffer.mRaw = audioBuffer->raw; AutoMutex lock(mLock); mInUnderrun = false; mProxy->releaseBuffer(&buffer); // restart track if it was disabled by audioflinger due to previous underrun if (mState == STATE_ACTIVE) { audio_track_cblk_t* cblk = mCblk; if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting", this, mName.string()); // FIXME ignoring status mAudioTrack->start(); } } } // ------------------------------------------------------------------------- ssize_t AudioTrack::write(const void* buffer, size_t userSize) { if (mTransfer != TRANSFER_SYNC || mIsTimed) { return INVALID_OPERATION; } if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { // Sanity-check: user is most-likely passing an error code, and it would // make the return value ambiguous (actualSize vs error). ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); return BAD_VALUE; } size_t written = 0; Buffer audioBuffer; while (userSize >= mFrameSize) { audioBuffer.frameCount = userSize / mFrameSize; status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); if (err < 0) { if (written > 0) { break; } return ssize_t(err); } size_t toWrite; if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { // Divide capacity by 2 to take expansion into account toWrite = audioBuffer.size >> 1; memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); } else { toWrite = audioBuffer.size; memcpy(audioBuffer.i8, buffer, toWrite); } buffer = ((const char *) buffer) + toWrite; userSize -= toWrite; written += toWrite; releaseBuffer(&audioBuffer); } return written; } // ------------------------------------------------------------------------- TimedAudioTrack::TimedAudioTrack() { mIsTimed = true; } status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp* buffer) { AutoMutex lock(mLock); status_t result = UNKNOWN_ERROR; #if 1 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed // while we are accessing the cblk sp audioTrack = mAudioTrack; sp iMem = mCblkMemory; #endif // If the track is not invalid already, try to allocate a buffer. alloc // fails indicating that the server is dead, flag the track as invalid so // we can attempt to restore in just a bit. audio_track_cblk_t* cblk = mCblk; if (!(cblk->mFlags & CBLK_INVALID)) { result = mAudioTrack->allocateTimedBuffer(size, buffer); if (result == DEAD_OBJECT) { android_atomic_or(CBLK_INVALID, &cblk->mFlags); } } // If the track is invalid at this point, attempt to restore it. and try the // allocation one more time. if (cblk->mFlags & CBLK_INVALID) { result = restoreTrack_l("allocateTimedBuffer"); if (result == NO_ERROR) { result = mAudioTrack->allocateTimedBuffer(size, buffer); } } return result; } status_t TimedAudioTrack::queueTimedBuffer(const sp& buffer, int64_t pts) { status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); { AutoMutex lock(mLock); audio_track_cblk_t* cblk = mCblk; // restart track if it was disabled by audioflinger due to previous underrun if (buffer->size() != 0 && status == NO_ERROR && (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); ALOGW("queueTimedBuffer() track %p disabled, restarting", this); // FIXME ignoring status mAudioTrack->start(); } } return status; } status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, TargetTimeline target) { return mAudioTrack->setMediaTimeTransform(xform, target); } // ------------------------------------------------------------------------- nsecs_t AudioTrack::processAudioBuffer() { // Currently the AudioTrack thread is not created if there are no callbacks. // Would it ever make sense to run the thread, even without callbacks? // If so, then replace this by checks at each use for mCbf != NULL. LOG_ALWAYS_FATAL_IF(mCblk == NULL); mLock.lock(); if (mAwaitBoost) { mAwaitBoost = false; mLock.unlock(); static const int32_t kMaxTries = 5; int32_t tryCounter = kMaxTries; uint32_t pollUs = 10000; do { int policy = sched_getscheduler(0); if (policy == SCHED_FIFO || policy == SCHED_RR) { break; } usleep(pollUs); pollUs <<= 1; } while (tryCounter-- > 0); if (tryCounter < 0) { ALOGE("did not receive expected priority boost on time"); } // Run again immediately return 0; } // Can only reference mCblk while locked int32_t flags = android_atomic_and( ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); // Check for track invalidation if (flags & CBLK_INVALID) { // for offloaded tracks restoreTrack_l() will just update the sequence and clear // AudioSystem cache. We should not exit here but after calling the callback so // that the upper layers can recreate the track if (!isOffloaded_l() || (mSequence == mObservedSequence)) { status_t status = restoreTrack_l("processAudioBuffer"); mLock.unlock(); // Run again immediately, but with a new IAudioTrack return 0; } } bool waitStreamEnd = mState == STATE_STOPPING; bool active = mState == STATE_ACTIVE; // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() bool newUnderrun = false; if (flags & CBLK_UNDERRUN) { #if 0 // Currently in shared buffer mode, when the server reaches the end of buffer, // the track stays active in continuous underrun state. It's up to the application // to pause or stop the track, or set the position to a new offset within buffer. // This was some experimental code to auto-pause on underrun. Keeping it here // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. if (mTransfer == TRANSFER_SHARED) { mState = STATE_PAUSED; active = false; } #endif if (!mInUnderrun) { mInUnderrun = true; newUnderrun = true; } } // Get current position of server size_t position = mProxy->getPosition(); // Manage marker callback bool markerReached = false; size_t markerPosition = mMarkerPosition; // FIXME fails for wraparound, need 64 bits if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { mMarkerReached = markerReached = true; } // Determine number of new position callback(s) that will be needed, while locked size_t newPosCount = 0; size_t newPosition = mNewPosition; size_t updatePeriod = mUpdatePeriod; // FIXME fails for wraparound, need 64 bits if (updatePeriod > 0 && position >= newPosition) { newPosCount = ((position - newPosition) / updatePeriod) + 1; mNewPosition += updatePeriod * newPosCount; } // Cache other fields that will be needed soon uint32_t loopPeriod = mLoopPeriod; uint32_t sampleRate = mSampleRate; uint32_t notificationFrames = mNotificationFramesAct; if (mRefreshRemaining) { mRefreshRemaining = false; mRemainingFrames = notificationFrames; mRetryOnPartialBuffer = false; } size_t misalignment = mProxy->getMisalignment(); uint32_t sequence = mSequence; // These fields don't need to be cached, because they are assigned only by set(): // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags // mFlags is also assigned by createTrack_l(), but not the bit we care about. mLock.unlock(); if (waitStreamEnd) { AutoMutex lock(mLock); sp proxy = mProxy; sp iMem = mCblkMemory; struct timespec timeout; timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; timeout.tv_nsec = 0; mLock.unlock(); status_t status = mProxy->waitStreamEndDone(&timeout); mLock.lock(); switch (status) { case NO_ERROR: case DEAD_OBJECT: case TIMED_OUT: mLock.unlock(); mCbf(EVENT_STREAM_END, mUserData, NULL); mLock.lock(); if (mState == STATE_STOPPING) { mState = STATE_STOPPED; if (status != DEAD_OBJECT) { return NS_INACTIVE; } } return 0; default: return 0; } } // perform callbacks while unlocked if (newUnderrun) { mCbf(EVENT_UNDERRUN, mUserData, NULL); } // FIXME we will miss loops if loop cycle was signaled several times since last call // to processAudioBuffer() if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { mCbf(EVENT_LOOP_END, mUserData, NULL); } if (flags & CBLK_BUFFER_END) { mCbf(EVENT_BUFFER_END, mUserData, NULL); } if (markerReached) { mCbf(EVENT_MARKER, mUserData, &markerPosition); } while (newPosCount > 0) { size_t temp = newPosition; mCbf(EVENT_NEW_POS, mUserData, &temp); newPosition += updatePeriod; newPosCount--; } if (mObservedSequence != sequence) { mObservedSequence = sequence; mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); // for offloaded tracks, just wait for the upper layers to recreate the track if (isOffloaded()) { return NS_INACTIVE; } } // if inactive, then don't run me again until re-started if (!active) { return NS_INACTIVE; } // Compute the estimated time until the next timed event (position, markers, loops) // FIXME only for non-compressed audio uint32_t minFrames = ~0; if (!markerReached && position < markerPosition) { minFrames = markerPosition - position; } if (loopPeriod > 0 && loopPeriod < minFrames) { minFrames = loopPeriod; } if (updatePeriod > 0 && updatePeriod < minFrames) { minFrames = updatePeriod; } // If > 0, poll periodically to recover from a stuck server. A good value is 2. static const uint32_t kPoll = 0; if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { minFrames = kPoll * notificationFrames; } // Convert frame units to time units nsecs_t ns = NS_WHENEVER; if (minFrames != (uint32_t) ~0) { // This "fudge factor" avoids soaking CPU, and compensates for late progress by server static const nsecs_t kFudgeNs = 10000000LL; // 10 ms ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; } // If not supplying data by EVENT_MORE_DATA, then we're done if (mTransfer != TRANSFER_CALLBACK) { return ns; } struct timespec timeout; const struct timespec *requested = &ClientProxy::kForever; if (ns != NS_WHENEVER) { timeout.tv_sec = ns / 1000000000LL; timeout.tv_nsec = ns % 1000000000LL; ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); requested = &timeout; } while (mRemainingFrames > 0) { Buffer audioBuffer; audioBuffer.frameCount = mRemainingFrames; size_t nonContig; status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); requested = &ClientProxy::kNonBlocking; size_t avail = audioBuffer.frameCount + nonContig; ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d", mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); if (err != NO_ERROR) { if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || (isOffloaded() && (err == DEAD_OBJECT))) { return 0; } ALOGE("Error %d obtaining an audio buffer, giving up.", err); return NS_NEVER; } if (mRetryOnPartialBuffer && !isOffloaded()) { mRetryOnPartialBuffer = false; if (avail < mRemainingFrames) { int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; if (ns < 0 || myns < ns) { ns = myns; } return ns; } } // 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 == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { audioBuffer.size >>= 1; } size_t reqSize = audioBuffer.size; mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); size_t writtenSize = audioBuffer.size; // Sanity check on returned size if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", reqSize, (int) writtenSize); return NS_NEVER; } if (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 return WAIT_PERIOD_MS * 1000000LL; } if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { // 8 to 16 bit conversion, note that source and destination are the same address memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); audioBuffer.size <<= 1; } size_t releasedFrames = audioBuffer.size / mFrameSizeAF; audioBuffer.frameCount = releasedFrames; mRemainingFrames -= releasedFrames; if (misalignment >= releasedFrames) { misalignment -= releasedFrames; } else { misalignment = 0; } releaseBuffer(&audioBuffer); // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer // if callback doesn't like to accept the full chunk if (writtenSize < reqSize) { continue; } // There could be enough non-contiguous frames available to satisfy the remaining request if (mRemainingFrames <= nonContig) { continue; } #if 0 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA // that total to a sum == notificationFrames. if (0 < misalignment && misalignment <= mRemainingFrames) { mRemainingFrames = misalignment; return (mRemainingFrames * 1100000000LL) / sampleRate; } #endif } mRemainingFrames = notificationFrames; mRetryOnPartialBuffer = true; // A lot has transpired since ns was calculated, so run again immediately and re-calculate return 0; } status_t AudioTrack::restoreTrack_l(const char *from) { ALOGW("dead IAudioTrack, %s, creating a new one from %s()", isOffloaded_l() ? "Offloaded" : "PCM", from); ++mSequence; status_t result; // refresh the audio configuration cache in this process to make sure we get new // output parameters in createTrack_l() AudioSystem::clearAudioConfigCache(); if (isOffloaded_l()) { // FIXME re-creation of offloaded tracks is not yet implemented return DEAD_OBJECT; } // if the new IAudioTrack is created, createTrack_l() will modify the // following member variables: mAudioTrack, mCblkMemory and mCblk. // It will also delete the strong references on previous IAudioTrack and IMemory // take the frames that will be lost by track recreation into account in saved position size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; result = createTrack_l(position /*epoch*/); if (result == NO_ERROR) { // continue playback from last known position, but // don't attempt to restore loop after invalidation; it's difficult and not worthwhile if (mStaticProxy != NULL) { mLoopPeriod = 0; mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); } // FIXME How do we simulate the fact that all frames present in the buffer at the time of // track destruction have been played? This is critical for SoundPool implementation // This must be broken, and needs to be tested/debugged. #if 0 // restore write index and set other indexes to reflect empty buffer status if (!strcmp(from, "start")) { // Make sure that a client relying on callback events indicating underrun or // the actual amount of audio frames played (e.g SoundPool) receives them. if (mSharedBuffer == 0) { // restart playback even if buffer is not completely filled. android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); } } #endif if (mState == STATE_ACTIVE) { result = mAudioTrack->start(); } } if (result != NO_ERROR) { // Use of direct and offloaded output streams is ref counted by audio policy manager. #if 0 // FIXME This should no longer be needed //Use of direct and offloaded output streams is ref counted by audio policy manager. // As getOutput was called above and resulted in an output stream to be opened, // we need to release it. if (mOutput != 0) { AudioSystem::releaseOutput(mOutput); mOutput = 0; } #endif ALOGW("restoreTrack_l() failed status %d", result); mState = STATE_STOPPED; } return result; } status_t AudioTrack::setParameters(const String8& keyValuePairs) { AutoMutex lock(mLock); return mAudioTrack->setParameters(keyValuePairs); } status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) { AutoMutex lock(mLock); // FIXME not implemented for fast tracks; should use proxy and SSQ if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { return INVALID_OPERATION; } if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { return INVALID_OPERATION; } status_t status = mAudioTrack->getTimestamp(timestamp); if (status == NO_ERROR) { timestamp.mPosition += mProxy->getEpoch(); } return status; } String8 AudioTrack::getParameters(const String8& keys) { audio_io_handle_t output = getOutput(); if (output != 0) { return AudioSystem::getParameters(output, keys); } else { return String8::empty(); } } bool AudioTrack::isOffloaded() const { AutoMutex lock(mLock); return isOffloaded_l(); } status_t AudioTrack::dump(int fd, const Vector& args __unused) 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(%zu)\n", mFormat, mChannelCount, mFrameCount); result.append(buffer); snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); result.append(buffer); snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); result.append(buffer); ::write(fd, result.string(), result.size()); return NO_ERROR; } uint32_t AudioTrack::getUnderrunFrames() const { AutoMutex lock(mLock); return mProxy->getUnderrunFrames(); } // ========================================================================= void AudioTrack::DeathNotifier::binderDied(const wp& who __unused) { sp audioTrack = mAudioTrack.promote(); if (audioTrack != 0) { AutoMutex lock(audioTrack->mLock); audioTrack->mProxy->binderDied(); } } // ========================================================================= AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), mIgnoreNextPausedInt(false) { } AudioTrack::AudioTrackThread::~AudioTrackThread() { } bool AudioTrack::AudioTrackThread::threadLoop() { { AutoMutex _l(mMyLock); if (mPaused) { mMyCond.wait(mMyLock); // caller will check for exitPending() return true; } if (mIgnoreNextPausedInt) { mIgnoreNextPausedInt = false; mPausedInt = false; } if (mPausedInt) { if (mPausedNs > 0) { (void) mMyCond.waitRelative(mMyLock, mPausedNs); } else { mMyCond.wait(mMyLock); } mPausedInt = false; return true; } } nsecs_t ns = mReceiver.processAudioBuffer(); switch (ns) { case 0: return true; case NS_INACTIVE: pauseInternal(); return true; case NS_NEVER: return false; case NS_WHENEVER: // FIXME increase poll interval, or make event-driven ns = 1000000000LL; // fall through default: LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); pauseInternal(ns); return true; } } void AudioTrack::AudioTrackThread::requestExit() { // must be in this order to avoid a race condition Thread::requestExit(); resume(); } void AudioTrack::AudioTrackThread::pause() { AutoMutex _l(mMyLock); mPaused = true; } void AudioTrack::AudioTrackThread::resume() { AutoMutex _l(mMyLock); mIgnoreNextPausedInt = true; if (mPaused || mPausedInt) { mPaused = false; mPausedInt = false; mMyCond.signal(); } } void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) { AutoMutex _l(mMyLock); mPausedInt = true; mPausedNs = ns; } }; // namespace android