diff options
Diffstat (limited to 'media/libmedia/AudioTrack.cpp')
| -rw-r--r-- | media/libmedia/AudioTrack.cpp | 601 |
1 files changed, 314 insertions, 287 deletions
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index aec8c4a..e40895a 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -50,11 +50,13 @@ namespace android { // static status_t AudioTrack::getMinFrameCount( - int* frameCount, + size_t* frameCount, audio_stream_type_t streamType, uint32_t sampleRate) { - if (frameCount == NULL) return BAD_VALUE; + if (frameCount == NULL) { + return BAD_VALUE; + } // default to 0 in case of error *frameCount = 0; @@ -65,11 +67,11 @@ status_t AudioTrack::getMinFrameCount( // audio_format_t format // audio_channel_mask_t channelMask // audio_output_flags_t flags - int afSampleRate; + uint32_t afSampleRate; if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { return NO_INIT; } - int afFrameCount; + size_t afFrameCount; if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { return NO_INIT; } @@ -120,28 +122,6 @@ AudioTrack::AudioTrack( 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId); } -// DEPRECATED -AudioTrack::AudioTrack( - int streamType, - uint32_t sampleRate, - int format, - int channelMask, - int frameCount, - uint32_t flags, - callback_t cbf, - void* user, - int notificationFrames, - int sessionId) - : mStatus(NO_INIT), - mIsTimed(false), - mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT) -{ - mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format, - (audio_channel_mask_t) channelMask, - frameCount, (audio_output_flags_t)flags, cbf, user, notificationFrames, - 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId); -} - AudioTrack::AudioTrack( audio_stream_type_t streamType, uint32_t sampleRate, @@ -188,7 +168,7 @@ status_t AudioTrack::set( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + int frameCountInt, audio_output_flags_t flags, callback_t cbf, void* user, @@ -197,10 +177,17 @@ status_t AudioTrack::set( bool threadCanCallJava, int sessionId) { + // FIXME "int" here is legacy and will be replaced by size_t later + if (frameCountInt < 0) { + ALOGE("Invalid frame count %d", frameCountInt); + return BAD_VALUE; + } + size_t frameCount = frameCountInt; - ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); + ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), + sharedBuffer->size()); - ALOGV("set() streamType %d frameCount %d flags %04x", streamType, frameCount, flags); + ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); AutoMutex lock(mLock); if (mAudioTrack != 0) { @@ -214,7 +201,7 @@ status_t AudioTrack::set( } if (sampleRate == 0) { - int afSampleRate; + uint32_t afSampleRate; if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { return NO_INIT; } @@ -256,7 +243,9 @@ status_t AudioTrack::set( ALOGE("Invalid channel mask %#x", channelMask); return BAD_VALUE; } + mChannelMask = channelMask; uint32_t channelCount = popcount(channelMask); + mChannelCount = channelCount; audio_io_handle_t output = AudioSystem::getOutput( streamType, @@ -272,6 +261,7 @@ status_t AudioTrack::set( mVolume[RIGHT] = 1.0f; mSendLevel = 0.0f; mFrameCount = frameCount; + mReqFrameCount = frameCount; mNotificationFramesReq = notificationFrames; mSessionId = sessionId; mAuxEffectId = 0; @@ -287,7 +277,6 @@ status_t AudioTrack::set( status_t status = createTrack_l(streamType, sampleRate, format, - channelMask, frameCount, flags, sharedBuffer, @@ -305,8 +294,15 @@ status_t AudioTrack::set( mStreamType = streamType; mFormat = format; - mChannelMask = channelMask; - mChannelCount = channelCount; + + 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); + } + mSharedBuffer = sharedBuffer; mMuted = false; mActive = false; @@ -318,7 +314,6 @@ status_t AudioTrack::set( mUpdatePeriod = 0; mFlushed = false; AudioSystem::acquireAudioSessionId(mSessionId); - mRestoreStatus = NO_ERROR; return NO_ERROR; } @@ -344,23 +339,14 @@ audio_format_t AudioTrack::format() const return mFormat; } -int AudioTrack::channelCount() const +uint32_t AudioTrack::channelCount() const { return mChannelCount; } -uint32_t AudioTrack::frameCount() const +size_t AudioTrack::frameCount() const { - return mCblk->frameCount; -} - -size_t AudioTrack::frameSize() const -{ - if (audio_is_linear_pcm(mFormat)) { - return channelCount()*audio_bytes_per_sample(mFormat); - } else { - return sizeof(uint8_t); - } + return mFrameCount; } sp<IMemory>& AudioTrack::sharedBuffer() @@ -390,7 +376,7 @@ void AudioTrack::start() cblk->lock.lock(); cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; cblk->waitTimeMs = 0; - android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags); + android_atomic_and(~CBLK_DISABLED, &cblk->flags); if (t != 0) { t->resume(); } else { @@ -399,19 +385,21 @@ void AudioTrack::start() androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); } - ALOGV("start %p before lock cblk %p", this, mCblk); + ALOGV("start %p before lock cblk %p", this, cblk); status_t status = NO_ERROR; - if (!(cblk->flags & CBLK_INVALID_MSK)) { + if (!(cblk->flags & CBLK_INVALID)) { cblk->lock.unlock(); ALOGV("mAudioTrack->start()"); status = mAudioTrack->start(); cblk->lock.lock(); if (status == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID_ON, &cblk->flags); + android_atomic_or(CBLK_INVALID, &cblk->flags); } } - if (cblk->flags & CBLK_INVALID_MSK) { - status = restoreTrack_l(cblk, true); + if (cblk->flags & CBLK_INVALID) { + audio_track_cblk_t* temp = cblk; + status = restoreTrack_l(temp, true /*fromStart*/); + cblk = temp; } cblk->lock.unlock(); if (status != NO_ERROR) { @@ -528,14 +516,9 @@ status_t AudioTrack::setVolume(float left, float right) return NO_ERROR; } -void AudioTrack::getVolume(float* left, float* right) const +status_t AudioTrack::setVolume(float volume) { - if (left != NULL) { - *left = mVolume[LEFT]; - } - if (right != NULL) { - *right = mVolume[RIGHT]; - } + return setVolume(volume, volume); } status_t AudioTrack::setAuxEffectSendLevel(float level) @@ -560,9 +543,9 @@ void AudioTrack::getAuxEffectSendLevel(float* level) const } } -status_t AudioTrack::setSampleRate(int rate) +status_t AudioTrack::setSampleRate(uint32_t rate) { - int afSamplingRate; + uint32_t afSamplingRate; if (mIsTimed) { return INVALID_OPERATION; @@ -572,7 +555,9 @@ status_t AudioTrack::setSampleRate(int rate) return NO_INIT; } // Resampler implementation limits input sampling rate to 2 x output sampling rate. - if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE; + if (rate == 0 || rate > afSamplingRate*2 ) { + return BAD_VALUE; + } AutoMutex lock(mLock); mCblk->sampleRate = rate; @@ -582,7 +567,7 @@ status_t AudioTrack::setSampleRate(int rate) uint32_t AudioTrack::getSampleRate() const { if (mIsTimed) { - return INVALID_OPERATION; + return 0; } AutoMutex lock(mLock); @@ -615,15 +600,17 @@ status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCou } if (loopStart >= loopEnd || - loopEnd - loopStart > cblk->frameCount || + loopEnd - loopStart > mFrameCount || cblk->server > loopStart) { - ALOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user); + ALOGE("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 > cblk->frameCount)) { - ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d", - loopStart, loopEnd, cblk->frameCount); + if ((mSharedBuffer != 0) && (loopEnd > mFrameCount)) { + ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, " + "framecount %d", + loopStart, loopEnd, mFrameCount); return BAD_VALUE; } @@ -637,7 +624,9 @@ status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCou status_t AudioTrack::setMarkerPosition(uint32_t marker) { - if (mCbf == NULL) return INVALID_OPERATION; + if (mCbf == NULL) { + return INVALID_OPERATION; + } mMarkerPosition = marker; mMarkerReached = false; @@ -647,7 +636,9 @@ status_t AudioTrack::setMarkerPosition(uint32_t marker) status_t AudioTrack::getMarkerPosition(uint32_t *marker) const { - if (marker == NULL) return BAD_VALUE; + if (marker == NULL) { + return BAD_VALUE; + } *marker = mMarkerPosition; @@ -656,7 +647,9 @@ status_t AudioTrack::getMarkerPosition(uint32_t *marker) const status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) { - if (mCbf == NULL) return INVALID_OPERATION; + if (mCbf == NULL) { + return INVALID_OPERATION; + } uint32_t curPosition; getPosition(&curPosition); @@ -668,7 +661,9 @@ status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const { - if (updatePeriod == NULL) return BAD_VALUE; + if (updatePeriod == NULL) { + return BAD_VALUE; + } *updatePeriod = mUpdatePeriod; @@ -677,25 +672,34 @@ status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const status_t AudioTrack::setPosition(uint32_t position) { - if (mIsTimed) return INVALID_OPERATION; + if (mIsTimed) { + return INVALID_OPERATION; + } AutoMutex lock(mLock); - if (!stopped_l()) return INVALID_OPERATION; + if (!stopped_l()) { + return INVALID_OPERATION; + } - Mutex::Autolock _l(mCblk->lock); + audio_track_cblk_t* cblk = mCblk; + Mutex::Autolock _l(cblk->lock); - if (position > mCblk->user) return BAD_VALUE; + if (position > cblk->user) { + return BAD_VALUE; + } - mCblk->server = position; - android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); + cblk->server = position; + android_atomic_or(CBLK_FORCEREADY, &cblk->flags); return NO_ERROR; } status_t AudioTrack::getPosition(uint32_t *position) { - if (position == NULL) return BAD_VALUE; + if (position == NULL) { + return BAD_VALUE; + } AutoMutex lock(mLock); *position = mFlushed ? 0 : mCblk->server; @@ -706,11 +710,14 @@ status_t AudioTrack::reload() { AutoMutex lock(mLock); - if (!stopped_l()) return INVALID_OPERATION; + if (!stopped_l()) { + return INVALID_OPERATION; + } flush_l(); - mCblk->stepUser(mCblk->frameCount); + audio_track_cblk_t* cblk = mCblk; + cblk->stepUserOut(mFrameCount, mFrameCount); return NO_ERROR; } @@ -750,8 +757,7 @@ status_t AudioTrack::createTrack_l( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, - audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, audio_output_flags_t flags, const sp<IMemory>& sharedBuffer, audio_io_handle_t output) @@ -791,7 +797,7 @@ status_t AudioTrack::createTrack_l( // Same comment as below about ignoring frameCount parameter for set() frameCount = sharedBuffer->size(); } else if (frameCount == 0) { - int afFrameCount; + size_t afFrameCount; if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { return NO_INIT; } @@ -800,17 +806,16 @@ status_t AudioTrack::createTrack_l( } else if (sharedBuffer != 0) { - // Ensure that buffer alignment matches channelCount - int channelCount = popcount(channelMask); + // Ensure that buffer alignment matches channel count // 8-bit data in shared memory is not currently supported by AudioFlinger size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; - if (channelCount > 1) { + if (mChannelCount > 1) { // More than 2 channels does not require stronger alignment than stereo alignment <<= 1; } - if (((uint32_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { - ALOGE("Invalid buffer alignment: address %p, channelCount %d", - sharedBuffer->pointer(), channelCount); + if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { + ALOGE("Invalid buffer alignment: address %p, channel count %u", + sharedBuffer->pointer(), mChannelCount); return BAD_VALUE; } @@ -818,16 +823,16 @@ status_t AudioTrack::createTrack_l( // there's no frameCount parameter. // But when initializing a shared buffer AudioTrack via set(), // there _is_ a frameCount parameter. We silently ignore it. - frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t); + frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { // FIXME move these calculations and associated checks to server - int afSampleRate; + uint32_t afSampleRate; if (AudioSystem::getSamplingRate(output, streamType, &afSampleRate) != NO_ERROR) { return NO_INIT; } - int afFrameCount; + size_t afFrameCount; if (AudioSystem::getFrameCount(output, streamType, &afFrameCount) != NO_ERROR) { return NO_INIT; } @@ -836,8 +841,8 @@ status_t AudioTrack::createTrack_l( uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); if (minBufCount < 2) minBufCount = 2; - int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; - ALOGV("minFrameCount: %d, afFrameCount=%d, minBufCount=%d, sampleRate=%d, afSampleRate=%d" + size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; + ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" ", afLatency=%d", minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); @@ -849,7 +854,7 @@ status_t AudioTrack::createTrack_l( } // Make sure that application is notified with sufficient margin // before underrun - if (mNotificationFramesAct > (uint32_t)frameCount/2) { + if (mNotificationFramesAct > frameCount/2) { mNotificationFramesAct = frameCount/2; } if (frameCount < minFrameCount) { @@ -879,10 +884,12 @@ status_t AudioTrack::createTrack_l( sp<IAudioTrack> track = audioFlinger->createTrack(getpid(), streamType, sampleRate, - format, - channelMask, + // AudioFlinger only sees 16-bit PCM + format == AUDIO_FORMAT_PCM_8_BIT ? + AUDIO_FORMAT_PCM_16_BIT : format, + mChannelMask, frameCount, - trackFlags, + &trackFlags, sharedBuffer, output, tid, @@ -893,49 +900,58 @@ status_t AudioTrack::createTrack_l( ALOGE("AudioFlinger could not create track, status: %d", status); return status; } - sp<IMemory> cblk = track->getCblk(); - if (cblk == 0) { + sp<IMemory> iMem = track->getCblk(); + if (iMem == 0) { ALOGE("Could not get control block"); return NO_INIT; } mAudioTrack = track; - mCblkMemory = cblk; - mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); - // old has the previous value of mCblk->flags before the "or" operation - int32_t old = android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags); + mCblkMemory = iMem; + audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); + mCblk = cblk; + size_t temp = cblk->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; if (flags & AUDIO_OUTPUT_FLAG_FAST) { - if (old & CBLK_FAST) { - ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", mCblk->frameCount); + if (trackFlags & IAudioFlinger::TRACK_FAST) { + ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); } else { - ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", mCblk->frameCount); + ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); // once denied, do not request again if IAudioTrack is re-created flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); mFlags = flags; } if (sharedBuffer == 0) { - mNotificationFramesAct = mCblk->frameCount/2; + mNotificationFramesAct = frameCount/2; } } if (sharedBuffer == 0) { - mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); + mBuffers = (char*)cblk + sizeof(audio_track_cblk_t); } else { - mCblk->buffers = sharedBuffer->pointer(); + mBuffers = sharedBuffer->pointer(); // Force buffer full condition as data is already present in shared memory - mCblk->stepUser(mCblk->frameCount); + cblk->stepUserOut(frameCount, frameCount); } - mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000)); - mCblk->setSendLevel(mSendLevel); + cblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | + uint16_t(mVolume[LEFT] * 0x1000)); + cblk->setSendLevel(mSendLevel); mAudioTrack->attachAuxEffect(mAuxEffectId); - mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; - mCblk->waitTimeMs = 0; + cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; + cblk->waitTimeMs = 0; mRemainingFrames = mNotificationFramesAct; // FIXME don't believe this lie - mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate; + mLatency = afLatency + (1000*frameCount) / sampleRate; + mFrameCount = frameCount; // If IAudioTrack is re-created, don't let the requested frameCount // decrease. This can confuse clients that cache frameCount(). - if (mCblk->frameCount > mFrameCount) { - mFrameCount = mCblk->frameCount; + if (frameCount > mReqFrameCount) { + mReqFrameCount = frameCount; } return NO_ERROR; } @@ -952,10 +968,10 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) audioBuffer->frameCount = 0; audioBuffer->size = 0; - uint32_t framesAvail = cblk->framesAvailable(); + uint32_t framesAvail = cblk->framesAvailableOut(mFrameCount); cblk->lock.lock(); - if (cblk->flags & CBLK_INVALID_MSK) { + if (cblk->flags & CBLK_INVALID) { goto create_new_track; } cblk->lock.unlock(); @@ -974,18 +990,23 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) cblk->lock.unlock(); return WOULD_BLOCK; } - if (!(cblk->flags & CBLK_INVALID_MSK)) { + if (!(cblk->flags & CBLK_INVALID)) { mLock.unlock(); + // this condition is in shared memory, so if IAudioTrack and control block + // are replaced due to mediaserver death or IAudioTrack invalidation then + // cv won't be signalled, but fortunately the timeout will limit the wait result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); cblk->lock.unlock(); mLock.lock(); if (!mActive) { return status_t(STOPPED); } + // IAudioTrack may have been re-created while mLock was unlocked + cblk = mCblk; cblk->lock.lock(); } - if (cblk->flags & CBLK_INVALID_MSK) { + if (cblk->flags & CBLK_INVALID) { goto create_new_track; } if (CC_UNLIKELY(result != NO_ERROR)) { @@ -994,16 +1015,18 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) // 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) { - ALOGW( "obtainBuffer timed out (is the CPU pegged?) %p name=%#x" - "user=%08x, server=%08x", this, cblk->mName, cblk->user, cblk->server); + ALOGW("obtainBuffer timed out (is the CPU pegged?) %p name=%#x user=%08x, " + "server=%08x", this, cblk->mName, cblk->user, cblk->server); //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140) cblk->lock.unlock(); result = mAudioTrack->start(); cblk->lock.lock(); if (result == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID_ON, &cblk->flags); + android_atomic_or(CBLK_INVALID, &cblk->flags); create_new_track: - result = restoreTrack_l(cblk, false); + audio_track_cblk_t* temp = cblk; + result = restoreTrack_l(temp, false /*fromStart*/); + cblk = temp; } if (result != NO_ERROR) { ALOGW("obtainBuffer create Track error %d", result); @@ -1021,7 +1044,7 @@ create_new_track: } // read the server count again start_loop_here: - framesAvail = cblk->framesAvailable_l(); + framesAvail = cblk->framesAvailableOut_l(mFrameCount); } cblk->lock.unlock(); } @@ -1033,22 +1056,15 @@ create_new_track: } uint32_t u = cblk->user; - uint32_t bufferEnd = cblk->userBase + cblk->frameCount; + uint32_t bufferEnd = cblk->userBase + mFrameCount; if (framesReq > bufferEnd - u) { framesReq = bufferEnd - u; } - audioBuffer->flags = mMuted ? Buffer::MUTE : 0; - audioBuffer->channelCount = mChannelCount; audioBuffer->frameCount = framesReq; - audioBuffer->size = framesReq * cblk->frameSize; - if (audio_is_linear_pcm(mFormat)) { - audioBuffer->format = AUDIO_FORMAT_PCM_16_BIT; - } else { - audioBuffer->format = mFormat; - } - audioBuffer->raw = (int8_t *)cblk->buffer(u); + audioBuffer->size = framesReq * mFrameSizeAF; + audioBuffer->raw = cblk->buffer(mBuffers, mFrameSizeAF, u); active = mActive; return active ? status_t(NO_ERROR) : status_t(STOPPED); } @@ -1056,12 +1072,13 @@ create_new_track: void AudioTrack::releaseBuffer(Buffer* audioBuffer) { AutoMutex lock(mLock); - mCblk->stepUser(audioBuffer->frameCount); + audio_track_cblk_t* cblk = mCblk; + cblk->stepUserOut(audioBuffer->frameCount, mFrameCount); if (audioBuffer->frameCount > 0) { // restart track if it was disabled by audioflinger due to previous underrun - if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) { - android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags); - ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, mCblk->mName); + if (mActive && (cblk->flags & CBLK_DISABLED)) { + android_atomic_and(~CBLK_DISABLED, &cblk->flags); + ALOGW("releaseBuffer() track %p name=%#x disabled, restarting", this, cblk->mName); mAudioTrack->start(); } } @@ -1072,8 +1089,12 @@ void AudioTrack::releaseBuffer(Buffer* audioBuffer) ssize_t AudioTrack::write(const void* buffer, size_t userSize) { - if (mSharedBuffer != 0) return INVALID_OPERATION; - if (mIsTimed) return INVALID_OPERATION; + if (mSharedBuffer != 0) { + return INVALID_OPERATION; + } + if (mIsTimed) { + return INVALID_OPERATION; + } if (ssize_t(userSize) < 0) { // Sanity-check: user is most-likely passing an error code, and it would @@ -1096,6 +1117,9 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize) sp<IMemory> iMem = mCblkMemory; mLock.unlock(); + // since mLock is unlocked the IAudioTrack and shared memory may be re-created, + // so all cblk references might still refer to old shared memory, but that should be benign + ssize_t written = 0; const int8_t *src = (const int8_t *)buffer; Buffer audioBuffer; @@ -1107,8 +1131,9 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize) status_t err = obtainBuffer(&audioBuffer, -1); if (err < 0) { // out of buffers, return #bytes written - if (err == status_t(NO_MORE_BUFFERS)) + if (err == status_t(NO_MORE_BUFFERS)) { break; + } return ssize_t(err); } @@ -1121,8 +1146,8 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize) } else { toWrite = audioBuffer.size; memcpy(audioBuffer.i8, src, toWrite); - src += toWrite; } + src += toWrite; userSize -= toWrite; written += toWrite; @@ -1140,27 +1165,37 @@ TimedAudioTrack::TimedAudioTrack() { status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) { + AutoMutex lock(mLock); status_t result = UNKNOWN_ERROR; + // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed + // while we are accessing the cblk + sp<IAudioTrack> audioTrack = mAudioTrack; + sp<IMemory> iMem = mCblkMemory; + // 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. - if (!(mCblk->flags & CBLK_INVALID_MSK)) { + audio_track_cblk_t* cblk = mCblk; + if (!(cblk->flags & CBLK_INVALID)) { result = mAudioTrack->allocateTimedBuffer(size, buffer); if (result == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID_ON, &mCblk->flags); + android_atomic_or(CBLK_INVALID, &cblk->flags); } } // If the track is invalid at this point, attempt to restore it. and try the // allocation one more time. - if (mCblk->flags & CBLK_INVALID_MSK) { - mCblk->lock.lock(); - result = restoreTrack_l(mCblk, false); - mCblk->lock.unlock(); + if (cblk->flags & CBLK_INVALID) { + cblk->lock.lock(); + audio_track_cblk_t* temp = cblk; + result = restoreTrack_l(temp, false /*fromStart*/); + cblk = temp; + cblk->lock.unlock(); - if (result == OK) + if (result == OK) { result = mAudioTrack->allocateTimedBuffer(size, buffer); + } } return result; @@ -1172,10 +1207,11 @@ status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 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 && - mActive && (mCblk->flags & CBLK_DISABLED_MSK)) { - android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags); + mActive && (cblk->flags & CBLK_DISABLED)) { + android_atomic_and(~CBLK_DISABLED, &cblk->flags); ALOGW("queueTimedBuffer() track %p disabled, restarting", this); mAudioTrack->start(); } @@ -1206,15 +1242,20 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) bool active = mActive; mLock.unlock(); + // since mLock is unlocked the IAudioTrack and shared memory may be re-created, + // so all cblk references might still refer to old shared memory, but that should be benign + // Manage underrun callback - if (active && (cblk->framesAvailable() == cblk->frameCount)) { + if (active && (cblk->framesAvailableOut(mFrameCount) == mFrameCount)) { ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags); - if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) { + if (!(android_atomic_or(CBLK_UNDERRUN, &cblk->flags) & CBLK_UNDERRUN)) { mCbf(EVENT_UNDERRUN, mUserData, 0); - if (cblk->server == cblk->frameCount) { + if (cblk->server == mFrameCount) { mCbf(EVENT_BUFFER_END, mUserData, 0); } - if (mSharedBuffer != 0) return false; + if (mSharedBuffer != 0) { + return false; + } } } @@ -1265,12 +1306,15 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) status_t err = obtainBuffer(&audioBuffer, waitCount); if (err < NO_ERROR) { if (err != TIMED_OUT) { - ALOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up."); + ALOGE_IF(err != status_t(NO_MORE_BUFFERS), + "Error obtaining an audio buffer, giving up."); return false; } break; } - if (err == status_t(STOPPED)) return false; + 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 @@ -1293,7 +1337,9 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) break; } - if (writtenSize > reqSize) writtenSize = reqSize; + if (writtenSize > reqSize) { + writtenSize = reqSize; + } 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 @@ -1302,10 +1348,10 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) } 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 sample size of + // NOTE: cblk->frameSize is not equal to AudioTrack::frameSize() for + // 8 bit PCM data: in this case, cblk->frameSize is based on a sample size of // 16 bit. - audioBuffer.frameCount = writtenSize/mCblk->frameSize; + audioBuffer.frameCount = writtenSize / mFrameSizeAF; frames -= audioBuffer.frameCount; @@ -1321,112 +1367,91 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) return true; } -// must be called with mLock and cblk.lock held. Callers must also hold strong references on +// must be called with mLock and refCblk.lock held. Callers must also hold strong references on // the IAudioTrack and IMemory in case they are recreated here. -// If the IAudioTrack is successfully restored, the cblk pointer is updated -status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart) +// If the IAudioTrack is successfully restored, the refCblk pointer is updated +// FIXME Don't depend on caller to hold strong references. +status_t AudioTrack::restoreTrack_l(audio_track_cblk_t*& refCblk, bool fromStart) { status_t result; - if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) { - ALOGW("dead IAudioTrack, creating a new one from %s TID %d", - fromStart ? "start()" : "obtainBuffer()", gettid()); + audio_track_cblk_t* cblk = refCblk; + audio_track_cblk_t* newCblk = cblk; + ALOGW("dead IAudioTrack, creating a new one from %s", + fromStart ? "start()" : "obtainBuffer()"); - // signal old cblk condition so that other threads waiting for available buffers stop - // waiting now - cblk->cv.broadcast(); - cblk->lock.unlock(); + // signal old cblk condition so that other threads waiting for available buffers stop + // waiting now + cblk->cv.broadcast(); + cblk->lock.unlock(); - // refresh the audio configuration cache in this process to make sure we get new - // output parameters in getOutput_l() and createTrack_l() - AudioSystem::clearAudioConfigCache(); - - // 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 - result = createTrack_l(mStreamType, - cblk->sampleRate, - mFormat, - mChannelMask, - mFrameCount, - mFlags, - mSharedBuffer, - getOutput_l()); - - if (result == NO_ERROR) { - uint32_t user = cblk->user; - uint32_t server = cblk->server; - // restore write index and set other indexes to reflect empty buffer status - mCblk->user = user; - mCblk->server = user; - mCblk->userBase = user; - mCblk->serverBase = user; - // restore loop: this is not guaranteed to succeed if new frame count is not - // compatible with loop length - setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); - if (!fromStart) { - mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; - // 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) { - uint32_t frames = 0; - if (user > server) { - frames = ((user - server) > mCblk->frameCount) ? - mCblk->frameCount : (user - server); - memset(mCblk->buffers, 0, frames * mCblk->frameSize); - } - // restart playback even if buffer is not completely filled. - android_atomic_or(CBLK_FORCEREADY_ON, &mCblk->flags); - // stepUser() clears CBLK_UNDERRUN_ON flag enabling underrun callbacks to - // the client - mCblk->stepUser(frames); + // refresh the audio configuration cache in this process to make sure we get new + // output parameters in getOutput_l() and createTrack_l() + AudioSystem::clearAudioConfigCache(); + + // 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 + result = createTrack_l(mStreamType, + cblk->sampleRate, + mFormat, + mReqFrameCount, // so that frame count never goes down + mFlags, + mSharedBuffer, + getOutput_l()); + + if (result == NO_ERROR) { + uint32_t user = cblk->user; + uint32_t server = cblk->server; + // restore write index and set other indexes to reflect empty buffer status + newCblk = mCblk; + newCblk->user = user; + newCblk->server = user; + newCblk->userBase = user; + newCblk->serverBase = user; + // restore loop: this is not guaranteed to succeed if new frame count is not + // compatible with loop length + setLoop_l(cblk->loopStart, cblk->loopEnd, cblk->loopCount); + if (!fromStart) { + newCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; + // 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) { + uint32_t frames = 0; + if (user > server) { + frames = ((user - server) > mFrameCount) ? + mFrameCount : (user - server); + memset(mBuffers, 0, frames * mFrameSizeAF); } - } - if (mSharedBuffer != 0) { - mCblk->stepUser(mCblk->frameCount); - } - if (mActive) { - result = mAudioTrack->start(); - ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); - } - if (fromStart && result == NO_ERROR) { - mNewPosition = mCblk->server + mUpdatePeriod; + // restart playback even if buffer is not completely filled. + android_atomic_or(CBLK_FORCEREADY, &newCblk->flags); + // stepUser() clears CBLK_UNDERRUN flag enabling underrun callbacks to + // the client + newCblk->stepUserOut(frames, mFrameCount); } } - if (result != NO_ERROR) { - android_atomic_and(~CBLK_RESTORING_ON, &cblk->flags); - ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); + if (mSharedBuffer != 0) { + newCblk->stepUserOut(mFrameCount, mFrameCount); } - mRestoreStatus = result; - // signal old cblk condition for other threads waiting for restore completion - android_atomic_or(CBLK_RESTORED_ON, &cblk->flags); - cblk->cv.broadcast(); - } else { - if (!(cblk->flags & CBLK_RESTORED_MSK)) { - ALOGW("dead IAudioTrack, waiting for a new one TID %d", gettid()); - mLock.unlock(); - result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS)); - if (result == NO_ERROR) { - result = mRestoreStatus; - } - cblk->lock.unlock(); - mLock.lock(); - } else { - ALOGW("dead IAudioTrack, already restored TID %d", gettid()); - result = mRestoreStatus; - cblk->lock.unlock(); + if (mActive) { + result = mAudioTrack->start(); + ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result); + } + if (fromStart && result == NO_ERROR) { + mNewPosition = newCblk->server + mUpdatePeriod; } } + ALOGW_IF(result != NO_ERROR, "restoreTrack_l() failed status %d", result); ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x", - result, mActive, mCblk, cblk, mCblk->flags, cblk->flags); + result, mActive, newCblk, cblk, newCblk->flags, cblk->flags); if (result == NO_ERROR) { // from now on we switch to the newly created cblk - cblk = mCblk; + refCblk = newCblk; } - cblk->lock.lock(); + newCblk->lock.lock(); - ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d TID %d", result, gettid()); + ALOGW_IF(result != NO_ERROR, "restoreTrack_l() error %d", result); return result; } @@ -1438,12 +1463,16 @@ status_t AudioTrack::dump(int fd, const Vector<String16>& args) const char buffer[SIZE]; String8 result; + audio_track_cblk_t* cblk = mCblk; result.append(" AudioTrack::dump\n"); - snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]); + 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 == 0) ? 0 : mCblk->frameCount); + 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); + snprintf(buffer, 255, " sample rate(%u), status(%d), muted(%d)\n", + (cblk == 0) ? 0 : cblk->sampleRate, mStatus, mMuted); result.append(buffer); snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); result.append(buffer); @@ -1505,20 +1534,20 @@ void AudioTrack::AudioTrackThread::resume() audio_track_cblk_t::audio_track_cblk_t() : lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0), - userBase(0), serverBase(0), buffers(NULL), frameCount(0), + userBase(0), serverBase(0), frameCount_(0), loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000), mSendLevel(0), flags(0) { } -uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount) +uint32_t audio_track_cblk_t::stepUser(size_t stepCount, size_t frameCount, bool isOut) { - ALOGV("stepuser %08x %08x %d", user, server, frameCount); + ALOGV("stepuser %08x %08x %d", user, server, stepCount); uint32_t u = user; - u += frameCount; + u += stepCount; // Ensure that user is never ahead of server for AudioRecord - if (flags & CBLK_DIRECTION_MSK) { + if (isOut) { // If stepServer() has been called once, switch to normal obtainBuffer() timeout period if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) { bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; @@ -1528,30 +1557,29 @@ uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount) u = server; } - uint32_t fc = this->frameCount; - if (u >= fc) { + if (u >= frameCount) { // common case, user didn't just wrap - if (u - fc >= userBase ) { - userBase += fc; + if (u - frameCount >= userBase ) { + userBase += frameCount; } - } else if (u >= userBase + fc) { + } else if (u >= userBase + frameCount) { // user just wrapped - userBase += fc; + userBase += frameCount; } user = u; // Clear flow control error condition as new data has been written/read to/from buffer. - if (flags & CBLK_UNDERRUN_MSK) { - android_atomic_and(~CBLK_UNDERRUN_MSK, &flags); + if (flags & CBLK_UNDERRUN) { + android_atomic_and(~CBLK_UNDERRUN, &flags); } return u; } -bool audio_track_cblk_t::stepServer(uint32_t frameCount) +bool audio_track_cblk_t::stepServer(size_t stepCount, size_t frameCount, bool isOut) { - ALOGV("stepserver %08x %08x %d", user, server, frameCount); + ALOGV("stepserver %08x %08x %d", user, server, stepCount); if (!tryLock()) { ALOGW("stepServer() could not lock cblk"); @@ -1561,8 +1589,8 @@ bool audio_track_cblk_t::stepServer(uint32_t frameCount) uint32_t s = server; bool flushed = (s == user); - s += frameCount; - if (flags & CBLK_DIRECTION_MSK) { + s += stepCount; + if (isOut) { // 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) { @@ -1587,43 +1615,42 @@ bool audio_track_cblk_t::stepServer(uint32_t frameCount) } } - uint32_t fc = this->frameCount; - if (s >= fc) { + if (s >= frameCount) { // common case, server didn't just wrap - if (s - fc >= serverBase ) { - serverBase += fc; + if (s - frameCount >= serverBase ) { + serverBase += frameCount; } - } else if (s >= serverBase + fc) { + } else if (s >= serverBase + frameCount) { // server just wrapped - serverBase += fc; + serverBase += frameCount; } server = s; - if (!(flags & CBLK_INVALID_MSK)) { + if (!(flags & CBLK_INVALID)) { cv.signal(); } lock.unlock(); return true; } -void* audio_track_cblk_t::buffer(uint32_t offset) const +void* audio_track_cblk_t::buffer(void *buffers, size_t frameSize, uint32_t offset) const { return (int8_t *)buffers + (offset - userBase) * frameSize; } -uint32_t audio_track_cblk_t::framesAvailable() +uint32_t audio_track_cblk_t::framesAvailable(size_t frameCount, bool isOut) { Mutex::Autolock _l(lock); - return framesAvailable_l(); + return framesAvailable_l(frameCount, isOut); } -uint32_t audio_track_cblk_t::framesAvailable_l() +uint32_t audio_track_cblk_t::framesAvailable_l(size_t frameCount, bool isOut) { uint32_t u = user; uint32_t s = server; - if (flags & CBLK_DIRECTION_MSK) { + if (isOut) { uint32_t limit = (s < loopStart) ? s : loopStart; return limit + frameCount - u; } else { @@ -1631,12 +1658,12 @@ uint32_t audio_track_cblk_t::framesAvailable_l() } } -uint32_t audio_track_cblk_t::framesReady() +uint32_t audio_track_cblk_t::framesReady(bool isOut) { uint32_t u = user; uint32_t s = server; - if (flags & CBLK_DIRECTION_MSK) { + if (isOut) { if (u < loopEnd) { return u - s; } else { |