diff options
Diffstat (limited to 'media/libmedia')
| -rw-r--r-- | media/libmedia/Android.mk | 2 | ||||
| -rw-r--r-- | media/libmedia/AudioEffect.cpp | 21 | ||||
| -rw-r--r-- | media/libmedia/AudioRecord.cpp | 189 | ||||
| -rw-r--r-- | media/libmedia/AudioSystem.cpp | 44 | ||||
| -rw-r--r-- | media/libmedia/AudioTrack.cpp | 601 | ||||
| -rw-r--r-- | media/libmedia/IAudioFlinger.cpp | 40 | ||||
| -rw-r--r-- | media/libmedia/IAudioFlingerClient.cpp | 3 | ||||
| -rw-r--r-- | media/libmedia/IAudioPolicyService.cpp | 9 | ||||
| -rw-r--r-- | media/libmedia/SoundPool.cpp | 6 | ||||
| -rw-r--r-- | media/libmedia/ToneGenerator.cpp | 2 | ||||
| -rw-r--r-- | media/libmedia/Visualizer.cpp | 6 |
11 files changed, 481 insertions, 442 deletions
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk index 54666fb..f2b6441 100644 --- a/media/libmedia/Android.mk +++ b/media/libmedia/Android.mk @@ -54,7 +54,7 @@ LOCAL_SRC_FILES:= \ LOCAL_SHARED_LIBRARIES := \ libui libcutils libutils libbinder libsonivox libicuuc libexpat \ libcamera_client libstagefright_foundation \ - libgui libdl libaudioutils libmedia_native + libgui libdl libaudioutils LOCAL_WHOLE_STATIC_LIBRARY := libmedia_helper diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp index 680604b..3317d57 100644 --- a/media/libmedia/AudioEffect.cpp +++ b/media/libmedia/AudioEffect.cpp @@ -152,7 +152,8 @@ status_t AudioEffect::set(const effect_uuid_t *type, mCblk->buffer = (uint8_t *)mCblk + bufOffset; iEffect->asBinder()->linkToDeath(mIEffectClient); - ALOGV("set() %p OK effect: %s id: %d status %d enabled %d", this, mDescriptor.name, mId, mStatus, mEnabled); + ALOGV("set() %p OK effect: %s id: %d status %d enabled %d", this, mDescriptor.name, mId, + mStatus, mEnabled); return mStatus; } @@ -266,9 +267,11 @@ status_t AudioEffect::setParameter(effect_param_t *param) uint32_t size = sizeof(int); uint32_t psize = ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + param->vsize; - ALOGV("setParameter: param: %d, param2: %d", *(int *)param->data, (param->psize == 8) ? *((int *)param->data + 1): -1); + ALOGV("setParameter: param: %d, param2: %d", *(int *)param->data, + (param->psize == 8) ? *((int *)param->data + 1): -1); - return mIEffect->command(EFFECT_CMD_SET_PARAM, sizeof (effect_param_t) + psize, param, &size, ¶m->status); + return mIEffect->command(EFFECT_CMD_SET_PARAM, sizeof (effect_param_t) + psize, param, &size, + ¶m->status); } status_t AudioEffect::setParameterDeferred(effect_param_t *param) @@ -321,11 +324,14 @@ status_t AudioEffect::getParameter(effect_param_t *param) return BAD_VALUE; } - ALOGV("getParameter: param: %d, param2: %d", *(int *)param->data, (param->psize == 8) ? *((int *)param->data + 1): -1); + ALOGV("getParameter: param: %d, param2: %d", *(int *)param->data, + (param->psize == 8) ? *((int *)param->data + 1): -1); - uint32_t psize = sizeof(effect_param_t) + ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + param->vsize; + uint32_t psize = sizeof(effect_param_t) + ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + + param->vsize; - return mIEffect->command(EFFECT_CMD_GET_PARAM, sizeof(effect_param_t) + param->psize, param, &psize, param); + return mIEffect->command(EFFECT_CMD_GET_PARAM, sizeof(effect_param_t) + param->psize, param, + &psize, param); } @@ -346,7 +352,8 @@ void AudioEffect::binderDied() void AudioEffect::controlStatusChanged(bool controlGranted) { - ALOGV("controlStatusChanged %p control %d callback %p mUserData %p", this, controlGranted, mCbf, mUserData); + ALOGV("controlStatusChanged %p control %d callback %p mUserData %p", this, controlGranted, mCbf, + mUserData); if (controlGranted) { if (mStatus == ALREADY_EXISTS) { mStatus = NO_ERROR; diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp index 8ea6306..c2ef68c 100644 --- a/media/libmedia/AudioRecord.cpp +++ b/media/libmedia/AudioRecord.cpp @@ -36,7 +36,7 @@ namespace android { // static status_t AudioRecord::getMinFrameCount( - int* frameCount, + size_t* frameCount, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask) @@ -54,7 +54,7 @@ status_t AudioRecord::getMinFrameCount( } if (size == 0) { - ALOGE("Unsupported configuration: sampleRate %d, format %d, channelMask %#x", + ALOGE("Unsupported configuration: sampleRate %u, format %d, channelMask %#x", sampleRate, format, channelMask); return BAD_VALUE; } @@ -63,7 +63,7 @@ status_t AudioRecord::getMinFrameCount( size <<= 1; if (audio_is_linear_pcm(format)) { - int channelCount = popcount(channelMask); + uint32_t channelCount = popcount(channelMask); size /= channelCount * audio_bytes_per_sample(format); } @@ -119,15 +119,22 @@ status_t AudioRecord::set( uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + int frameCountInt, callback_t cbf, void* user, int notificationFrames, 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("set(): sampleRate %d, channelMask %#x, frameCount %d",sampleRate, channelMask, frameCount); + ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask, + frameCount); AutoMutex lock(mLock); @@ -155,8 +162,9 @@ status_t AudioRecord::set( if (!audio_is_input_channel(channelMask)) { return BAD_VALUE; } - - int channelCount = popcount(channelMask); + mChannelMask = channelMask; + uint32_t channelCount = popcount(channelMask); + mChannelCount = channelCount; if (sessionId == 0 ) { mSessionId = AudioSystem::newAudioSessionId(); @@ -176,7 +184,7 @@ status_t AudioRecord::set( } // validate framecount - int minFrameCount = 0; + size_t minFrameCount = 0; status_t status = getMinFrameCount(&minFrameCount, sampleRate, format, channelMask); if (status != NO_ERROR) { return status; @@ -194,8 +202,7 @@ status_t AudioRecord::set( } // create the IAudioRecord - status = openRecord_l(sampleRate, format, channelMask, - frameCount, input); + status = openRecord_l(sampleRate, format, frameCount, input); if (status != NO_ERROR) { return status; } @@ -209,9 +216,14 @@ status_t AudioRecord::set( mFormat = format; // Update buffer size in case it has been limited by AudioFlinger during track creation - mFrameCount = mCblk->frameCount; - mChannelCount = (uint8_t)channelCount; - mChannelMask = channelMask; + mFrameCount = mCblk->frameCount_; + + if (audio_is_linear_pcm(mFormat)) { + mFrameSize = channelCount * audio_bytes_per_sample(format); + } else { + mFrameSize = sizeof(uint8_t); + } + mActive = false; mCbf = cbf; mNotificationFrames = notificationFrames; @@ -247,25 +259,16 @@ audio_format_t AudioRecord::format() const return mFormat; } -int AudioRecord::channelCount() const +uint32_t AudioRecord::channelCount() const { return mChannelCount; } -uint32_t AudioRecord::frameCount() const +size_t AudioRecord::frameCount() const { return mFrameCount; } -size_t AudioRecord::frameSize() const -{ - if (audio_is_linear_pcm(mFormat)) { - return channelCount()*audio_bytes_per_sample(mFormat); - } else { - return sizeof(uint8_t); - } -} - audio_source_t AudioRecord::inputSource() const { return mInputSource; @@ -291,17 +294,19 @@ status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession) mActive = true; cblk->lock.lock(); - if (!(cblk->flags & CBLK_INVALID_MSK)) { + if (!(cblk->flags & CBLK_INVALID)) { cblk->lock.unlock(); ALOGV("mAudioRecord->start()"); ret = mAudioRecord->start(event, triggerSession); cblk->lock.lock(); if (ret == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID_ON, &cblk->flags); + android_atomic_or(CBLK_INVALID, &cblk->flags); } } - if (cblk->flags & CBLK_INVALID_MSK) { - ret = restoreRecord_l(cblk); + if (cblk->flags & CBLK_INVALID) { + audio_track_cblk_t* temp = cblk; + ret = restoreRecord_l(temp); + cblk = temp; } cblk->lock.unlock(); if (ret == NO_ERROR) { @@ -425,13 +430,13 @@ unsigned int AudioRecord::getInputFramesLost() const status_t AudioRecord::openRecord_l( uint32_t sampleRate, audio_format_t format, - audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, audio_io_handle_t input) { status_t status; const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); if (audioFlinger == 0) { + ALOGE("Could not get audioflinger"); return NO_INIT; } @@ -441,7 +446,7 @@ status_t AudioRecord::openRecord_l( int originalSessionId = mSessionId; sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), input, sampleRate, format, - channelMask, + mChannelMask, frameCount, IAudioFlinger::TRACK_DEFAULT, tid, @@ -454,20 +459,20 @@ status_t AudioRecord::openRecord_l( ALOGE("AudioFlinger could not create record track, status: %d", status); return status; } - sp<IMemory> cblk = record->getCblk(); - if (cblk == 0) { + sp<IMemory> iMem = record->getCblk(); + if (iMem == 0) { ALOGE("Could not get control block"); return NO_INIT; } mAudioRecord.clear(); mAudioRecord = record; mCblkMemory.clear(); - mCblkMemory = cblk; - mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer()); - mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); - android_atomic_and(~CBLK_DIRECTION_MSK, &mCblk->flags); - mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; - mCblk->waitTimeMs = 0; + mCblkMemory = iMem; + audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); + mCblk = cblk; + mBuffers = (char*)cblk + sizeof(audio_track_cblk_t); + cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS; + cblk->waitTimeMs = 0; return NO_ERROR; } @@ -483,7 +488,7 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) audioBuffer->frameCount = 0; audioBuffer->size = 0; - uint32_t framesReady = cblk->framesReady(); + uint32_t framesReady = cblk->framesReadyIn(); if (framesReady == 0) { cblk->lock.lock(); @@ -498,17 +503,22 @@ status_t AudioRecord::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 IAudioRecord and control block + // are replaced due to mediaserver death or IAudioRecord 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); } + // IAudioRecord 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_record; } if (CC_UNLIKELY(result != NO_ERROR)) { @@ -521,9 +531,11 @@ status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); cblk->lock.lock(); if (result == DEAD_OBJECT) { - android_atomic_or(CBLK_INVALID_ON, &cblk->flags); + android_atomic_or(CBLK_INVALID, &cblk->flags); create_new_record: - result = AudioRecord::restoreRecord_l(cblk); + audio_track_cblk_t* temp = cblk; + result = AudioRecord::restoreRecord_l(temp); + cblk = temp; } if (result != NO_ERROR) { ALOGW("obtainBuffer create Track error %d", result); @@ -539,7 +551,7 @@ create_new_record: } // read the server count again start_loop_here: - framesReady = cblk->framesReady(); + framesReady = cblk->framesReadyIn(); } cblk->lock.unlock(); } @@ -553,18 +565,15 @@ create_new_record: } 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 = 0; - audioBuffer->channelCount= mChannelCount; - audioBuffer->format = mFormat; audioBuffer->frameCount = framesReq; - audioBuffer->size = framesReq*cblk->frameSize; - audioBuffer->raw = (int8_t*)cblk->buffer(u); + audioBuffer->size = framesReq * mFrameSize; + audioBuffer->raw = cblk->buffer(mBuffers, mFrameSize, u); active = mActive; return active ? status_t(NO_ERROR) : status_t(STOPPED); } @@ -572,7 +581,7 @@ create_new_record: void AudioRecord::releaseBuffer(Buffer* audioBuffer) { AutoMutex lock(mLock); - mCblk->stepUser(audioBuffer->frameCount); + mCblk->stepUserIn(audioBuffer->frameCount, mFrameCount); } audio_io_handle_t AudioRecord::getInput() const @@ -631,10 +640,12 @@ ssize_t AudioRecord::read(void* buffer, size_t userSize) status_t err = obtainBuffer(&audioBuffer, ((2 * MAX_RUN_TIMEOUT_MS) / WAIT_PERIOD_MS)); if (err < 0) { // out of buffers, return #bytes written - if (err == status_t(NO_MORE_BUFFERS)) + if (err == status_t(NO_MORE_BUFFERS)) { break; - if (err == status_t(TIMED_OUT)) + } + if (err == status_t(TIMED_OUT)) { err = 0; + } return ssize_t(err); } @@ -701,7 +712,8 @@ bool AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread) status_t err = obtainBuffer(&audioBuffer, 1); 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; @@ -733,11 +745,11 @@ bool AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread) // Manage overrun callback - if (active && (cblk->framesAvailable() == 0)) { + if (active && (cblk->framesAvailableIn(mFrameCount) == 0)) { // The value of active is stale, but we are almost sure to be active here because // otherwise we would have exited when obtainBuffer returned STOPPED earlier. ALOGV("Overrun 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_OVERRUN, mUserData, NULL); } } @@ -753,57 +765,40 @@ bool AudioRecord::processAudioBuffer(const sp<AudioRecordThread>& thread) // must be called with mLock and cblk.lock held. Callers must also hold strong references on // the IAudioRecord and IMemory in case they are recreated here. // If the IAudioRecord is successfully restored, the cblk pointer is updated -status_t AudioRecord::restoreRecord_l(audio_track_cblk_t*& cblk) +status_t AudioRecord::restoreRecord_l(audio_track_cblk_t*& refCblk) { status_t result; - if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) { - ALOGW("dead IAudioRecord, creating a new one"); - // signal old cblk condition so that other threads waiting for available buffers stop - // waiting now - cblk->cv.broadcast(); - cblk->lock.unlock(); + audio_track_cblk_t* cblk = refCblk; + audio_track_cblk_t* newCblk = cblk; + ALOGW("dead IAudioRecord, creating a new one"); - // if the new IAudioRecord is created, openRecord_l() will modify the - // following member variables: mAudioRecord, mCblkMemory and mCblk. - // It will also delete the strong references on previous IAudioRecord and IMemory - result = openRecord_l(cblk->sampleRate, mFormat, mChannelMask, - mFrameCount, getInput_l()); - if (result == NO_ERROR) { - // callback thread or sync event hasn't changed - result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); - } - if (result != NO_ERROR) { - mActive = false; - } + // 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 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 IAudioRecord, waiting for a new one to be created"); - mLock.unlock(); - result = cblk->cv.waitRelative(cblk->lock, milliseconds(RESTORE_TIMEOUT_MS)); - cblk->lock.unlock(); - mLock.lock(); - } else { - ALOGW("dead IAudioRecord, already restored"); - result = NO_ERROR; - cblk->lock.unlock(); - } - if (result != NO_ERROR || !mActive) { - result = status_t(STOPPED); - } + // if the new IAudioRecord is created, openRecord_l() will modify the + // following member variables: mAudioRecord, mCblkMemory and mCblk. + // It will also delete the strong references on previous IAudioRecord and IMemory + result = openRecord_l(cblk->sampleRate, mFormat, mFrameCount, getInput_l()); + if (result == NO_ERROR) { + newCblk = mCblk; + // callback thread or sync event hasn't changed + result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0); + } + if (result != NO_ERROR) { + mActive = false; } + ALOGV("restoreRecord_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, "restoreRecord_l() error %d", result); diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp index 207f96f..028e4a3 100644 --- a/media/libmedia/AudioSystem.cpp +++ b/media/libmedia/AudioSystem.cpp @@ -205,12 +205,7 @@ int AudioSystem::logToLinear(float volume) return volume ? 100 - int(dBConvertInverse * log(volume) + 0.5) : 0; } -// DEPRECATED -status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType) { - return getOutputSamplingRate(samplingRate, (audio_stream_type_t)streamType); -} - -status_t AudioSystem::getOutputSamplingRate(int* samplingRate, audio_stream_type_t streamType) +status_t AudioSystem::getOutputSamplingRate(uint32_t* samplingRate, audio_stream_type_t streamType) { audio_io_handle_t output; @@ -228,7 +223,7 @@ status_t AudioSystem::getOutputSamplingRate(int* samplingRate, audio_stream_type status_t AudioSystem::getSamplingRate(audio_io_handle_t output, audio_stream_type_t streamType, - int* samplingRate) + uint32_t* samplingRate) { OutputDescriptor *outputDesc; @@ -246,17 +241,13 @@ status_t AudioSystem::getSamplingRate(audio_io_handle_t output, gLock.unlock(); } - ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %d", streamType, output, *samplingRate); + ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %u", streamType, output, + *samplingRate); return NO_ERROR; } -// DEPRECATED -status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType) { - return getOutputFrameCount(frameCount, (audio_stream_type_t)streamType); -} - -status_t AudioSystem::getOutputFrameCount(int* frameCount, audio_stream_type_t streamType) +status_t AudioSystem::getOutputFrameCount(size_t* frameCount, audio_stream_type_t streamType) { audio_io_handle_t output; @@ -274,7 +265,7 @@ status_t AudioSystem::getOutputFrameCount(int* frameCount, audio_stream_type_t s status_t AudioSystem::getFrameCount(audio_io_handle_t output, audio_stream_type_t streamType, - int* frameCount) + size_t* frameCount) { OutputDescriptor *outputDesc; @@ -290,7 +281,8 @@ status_t AudioSystem::getFrameCount(audio_io_handle_t output, gLock.unlock(); } - ALOGV("getFrameCount() streamType %d, output %d, frameCount %d", streamType, output, *frameCount); + ALOGV("getFrameCount() streamType %d, output %d, frameCount %d", streamType, output, + *frameCount); return NO_ERROR; } @@ -369,7 +361,8 @@ status_t AudioSystem::setVoiceVolume(float value) return af->setVoiceVolume(value); } -status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, audio_stream_type_t stream) +status_t AudioSystem::getRenderPosition(size_t *halFrames, size_t *dspFrames, + audio_stream_type_t stream) { const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger(); if (af == 0) return PERMISSION_DENIED; @@ -381,7 +374,7 @@ status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames return af->getRenderPosition(halFrames, dspFrames, getOutput(stream)); } -unsigned int AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) { +size_t AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) { const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger(); unsigned int result = 0; if (af == 0) return result; @@ -449,8 +442,10 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle OutputDescriptor *outputDesc = new OutputDescriptor(*desc); gOutputs.add(ioHandle, outputDesc); - ALOGV("ioConfigChanged() new output samplingRate %d, format %d channels %#x frameCount %d latency %d", - outputDesc->samplingRate, outputDesc->format, outputDesc->channels, outputDesc->frameCount, outputDesc->latency); + ALOGV("ioConfigChanged() new output samplingRate %u, format %d channels %#x frameCount %u " + "latency %d", + outputDesc->samplingRate, outputDesc->format, outputDesc->channels, + outputDesc->frameCount, outputDesc->latency); } break; case OUTPUT_CLOSED: { if (gOutputs.indexOfKey(ioHandle) < 0) { @@ -471,7 +466,8 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle if (param2 == NULL) break; desc = (const OutputDescriptor *)param2; - ALOGV("ioConfigChanged() new config for output %d samplingRate %d, format %d channels %#x frameCount %d latency %d", + ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %d channels %#x " + "frameCount %d latency %d", ioHandle, desc->samplingRate, desc->format, desc->channels, desc->frameCount, desc->latency); OutputDescriptor *outputDesc = gOutputs.valueAt(index); @@ -510,7 +506,7 @@ sp<IAudioPolicyService> AudioSystem::gAudioPolicyService; sp<AudioSystem::AudioPolicyServiceClient> AudioSystem::gAudioPolicyServiceClient; -// establish binder interface to AudioFlinger service +// establish binder interface to AudioPolicy service const sp<IAudioPolicyService>& AudioSystem::get_audio_policy_service() { gLock.lock(); @@ -744,14 +740,14 @@ status_t AudioSystem::isSourceActive(audio_source_t stream, bool* state) return NO_ERROR; } -int32_t AudioSystem::getPrimaryOutputSamplingRate() +uint32_t AudioSystem::getPrimaryOutputSamplingRate() { const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger(); if (af == 0) return 0; return af->getPrimaryOutputSamplingRate(); } -int32_t AudioSystem::getPrimaryOutputFrameCount() +size_t AudioSystem::getPrimaryOutputFrameCount() { const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger(); if (af == 0) return 0; 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 { diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp index ce8ffc4..a010bb6 100644 --- a/media/libmedia/IAudioFlinger.cpp +++ b/media/libmedia/IAudioFlinger.cpp @@ -89,8 +89,8 @@ public: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, - track_flags_t flags, + size_t frameCount, + track_flags_t *flags, const sp<IMemory>& sharedBuffer, audio_io_handle_t output, pid_t tid, @@ -106,7 +106,8 @@ public: data.writeInt32(format); data.writeInt32(channelMask); data.writeInt32(frameCount); - data.writeInt32((int32_t) flags); + track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT; + data.writeInt32(lFlags); data.writeStrongBinder(sharedBuffer->asBinder()); data.writeInt32((int32_t) output); data.writeInt32((int32_t) tid); @@ -119,6 +120,10 @@ public: if (lStatus != NO_ERROR) { ALOGE("createTrack error: %s", strerror(-lStatus)); } else { + lFlags = reply.readInt32(); + if (flags != NULL) { + *flags = lFlags; + } lSessionId = reply.readInt32(); if (sessionId != NULL) { *sessionId = lSessionId; @@ -138,7 +143,7 @@ public: uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, - int frameCount, + size_t frameCount, track_flags_t flags, pid_t tid, int *sessionId, @@ -501,7 +506,7 @@ public: return reply.readInt32(); } - virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, + virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames, audio_io_handle_t output) const { Parcel data, reply; @@ -522,7 +527,7 @@ public: return status; } - virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const + virtual size_t getInputFramesLost(audio_io_handle_t ioHandle) const { Parcel data, reply; data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor()); @@ -690,7 +695,7 @@ public: return (audio_module_handle_t) reply.readInt32(); } - virtual int32_t getPrimaryOutputSamplingRate() + virtual uint32_t getPrimaryOutputSamplingRate() { Parcel data, reply; data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor()); @@ -698,7 +703,7 @@ public: return reply.readInt32(); } - virtual int32_t getPrimaryOutputFrameCount() + virtual size_t getPrimaryOutputFrameCount() { Parcel data, reply; data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor()); @@ -723,7 +728,7 @@ status_t BnAudioFlinger::onTransact( uint32_t sampleRate = data.readInt32(); audio_format_t format = (audio_format_t) data.readInt32(); audio_channel_mask_t channelMask = data.readInt32(); - size_t bufferCount = data.readInt32(); + size_t frameCount = data.readInt32(); track_flags_t flags = (track_flags_t) data.readInt32(); sp<IMemory> buffer = interface_cast<IMemory>(data.readStrongBinder()); audio_io_handle_t output = (audio_io_handle_t) data.readInt32(); @@ -732,7 +737,8 @@ status_t BnAudioFlinger::onTransact( status_t status; sp<IAudioTrack> track = createTrack(pid, (audio_stream_type_t) streamType, sampleRate, format, - channelMask, bufferCount, flags, buffer, output, tid, &sessionId, &status); + channelMask, frameCount, &flags, buffer, output, tid, &sessionId, &status); + reply->writeInt32(flags); reply->writeInt32(sessionId); reply->writeInt32(status); reply->writeStrongBinder(track->asBinder()); @@ -745,13 +751,13 @@ status_t BnAudioFlinger::onTransact( uint32_t sampleRate = data.readInt32(); audio_format_t format = (audio_format_t) data.readInt32(); audio_channel_mask_t channelMask = data.readInt32(); - size_t bufferCount = data.readInt32(); + size_t frameCount = data.readInt32(); track_flags_t flags = (track_flags_t) data.readInt32(); pid_t tid = (pid_t) data.readInt32(); int sessionId = data.readInt32(); status_t status; sp<IAudioRecord> record = openRecord(pid, input, - sampleRate, format, channelMask, bufferCount, flags, tid, &sessionId, &status); + sampleRate, format, channelMask, frameCount, flags, tid, &sessionId, &status); reply->writeInt32(sessionId); reply->writeInt32(status); reply->writeStrongBinder(record->asBinder()); @@ -865,7 +871,8 @@ status_t BnAudioFlinger::onTransact( case REGISTER_CLIENT: { CHECK_INTERFACE(IAudioFlinger, data, reply); - sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient>(data.readStrongBinder()); + sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient>( + data.readStrongBinder()); registerClient(client); return NO_ERROR; } break; @@ -965,8 +972,8 @@ status_t BnAudioFlinger::onTransact( case GET_RENDER_POSITION: { CHECK_INTERFACE(IAudioFlinger, data, reply); audio_io_handle_t output = (audio_io_handle_t) data.readInt32(); - uint32_t halFrames; - uint32_t dspFrames; + size_t halFrames; + size_t dspFrames; status_t status = getRenderPosition(&halFrames, &dspFrames, output); reply->writeInt32(status); if (status == NO_ERROR) { @@ -1043,7 +1050,8 @@ status_t BnAudioFlinger::onTransact( int id; int enabled; - sp<IEffect> effect = createEffect(pid, &desc, client, priority, output, sessionId, &status, &id, &enabled); + sp<IEffect> effect = createEffect(pid, &desc, client, priority, output, sessionId, + &status, &id, &enabled); reply->writeInt32(status); reply->writeInt32(id); reply->writeInt32(enabled); diff --git a/media/libmedia/IAudioFlingerClient.cpp b/media/libmedia/IAudioFlingerClient.cpp index 4178b29..2d1e0f8 100644 --- a/media/libmedia/IAudioFlingerClient.cpp +++ b/media/libmedia/IAudioFlingerClient.cpp @@ -50,7 +50,8 @@ public: ALOGV("ioConfigChanged stream %d", stream); data.writeInt32(stream); } else if (event != AudioSystem::OUTPUT_CLOSED && event != AudioSystem::INPUT_CLOSED) { - const AudioSystem::OutputDescriptor *desc = (const AudioSystem::OutputDescriptor *)param2; + const AudioSystem::OutputDescriptor *desc = + (const AudioSystem::OutputDescriptor *)param2; data.writeInt32(desc->samplingRate); data.writeInt32(desc->format); data.writeInt32(desc->channels); diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp index 401437c..769deae 100644 --- a/media/libmedia/IAudioPolicyService.cpp +++ b/media/libmedia/IAudioPolicyService.cpp @@ -399,13 +399,15 @@ status_t BnAudioPolicyService::onTransact( case SET_PHONE_STATE: { CHECK_INTERFACE(IAudioPolicyService, data, reply); - reply->writeInt32(static_cast <uint32_t>(setPhoneState((audio_mode_t) data.readInt32()))); + reply->writeInt32(static_cast <uint32_t>(setPhoneState( + (audio_mode_t) data.readInt32()))); return NO_ERROR; } break; case SET_FORCE_USE: { CHECK_INTERFACE(IAudioPolicyService, data, reply); - audio_policy_force_use_t usage = static_cast <audio_policy_force_use_t>(data.readInt32()); + audio_policy_force_use_t usage = static_cast <audio_policy_force_use_t>( + data.readInt32()); audio_policy_forced_cfg_t config = static_cast <audio_policy_forced_cfg_t>(data.readInt32()); reply->writeInt32(static_cast <uint32_t>(setForceUse(usage, config))); @@ -414,7 +416,8 @@ status_t BnAudioPolicyService::onTransact( case GET_FORCE_USE: { CHECK_INTERFACE(IAudioPolicyService, data, reply); - audio_policy_force_use_t usage = static_cast <audio_policy_force_use_t>(data.readInt32()); + audio_policy_force_use_t usage = static_cast <audio_policy_force_use_t>( + data.readInt32()); reply->writeInt32(static_cast <uint32_t>(getForceUse(usage))); return NO_ERROR; } break; diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp index abc8899..ee70ef7 100644 --- a/media/libmedia/SoundPool.cpp +++ b/media/libmedia/SoundPool.cpp @@ -489,7 +489,7 @@ Sample::~Sample() ::close(mFd); } mData.clear(); - delete mUrl; + free(mUrl); } status_t Sample::doLoad() @@ -568,8 +568,8 @@ void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftV } // initialize track - int afFrameCount; - int afSampleRate; + size_t afFrameCount; + uint32_t afSampleRate; audio_stream_type_t streamType = mSoundPool->streamType(); if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { afFrameCount = kDefaultFrameCount; diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp index 253602d..42584fe 100644 --- a/media/libmedia/ToneGenerator.cpp +++ b/media/libmedia/ToneGenerator.cpp @@ -1036,7 +1036,7 @@ bool ToneGenerator::initAudioTrack() { goto initAudioTrack_exit; } - mpAudioTrack->setVolume(mVolume, mVolume); + mpAudioTrack->setVolume(mVolume); mState = TONE_INIT; diff --git a/media/libmedia/Visualizer.cpp b/media/libmedia/Visualizer.cpp index 8196e10..5b4071b 100644 --- a/media/libmedia/Visualizer.cpp +++ b/media/libmedia/Visualizer.cpp @@ -88,7 +88,8 @@ status_t Visualizer::setEnabled(bool enabled) return status; } -status_t Visualizer::setCaptureCallBack(capture_cbk_t cbk, void* user, uint32_t flags, uint32_t rate) +status_t Visualizer::setCaptureCallBack(capture_cbk_t cbk, void* user, uint32_t flags, + uint32_t rate) { if (rate > CAPTURE_RATE_MAX) { return BAD_VALUE; @@ -334,7 +335,8 @@ void Visualizer::controlStatusChanged(bool controlGranted) { //------------------------------------------------------------------------- -Visualizer::CaptureThread::CaptureThread(Visualizer& receiver, uint32_t captureRate, bool bCanCallJava) +Visualizer::CaptureThread::CaptureThread(Visualizer& receiver, uint32_t captureRate, + bool bCanCallJava) : Thread(bCanCallJava), mReceiver(receiver) { mSleepTimeUs = 1000000000 / captureRate; 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