diff options
22 files changed, 326 insertions, 182 deletions
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h index 49e1afc..2218fad 100644 --- a/include/media/AudioSystem.h +++ b/include/media/AudioSystem.h @@ -87,9 +87,12 @@ public: static float linearToLog(int volume); static int logToLinear(float volume); - static status_t getOutputSamplingRate(int* samplingRate, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); - static status_t getOutputFrameCount(int* frameCount, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); - static status_t getOutputLatency(uint32_t* latency, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); + static status_t getOutputSamplingRate(int* samplingRate, + audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); + static status_t getOutputFrameCount(int* frameCount, + audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); + static status_t getOutputLatency(uint32_t* latency, + audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); static status_t getSamplingRate(audio_io_handle_t output, audio_stream_type_t streamType, int* samplingRate); @@ -126,7 +129,8 @@ public: // - BAD_VALUE: invalid parameter // NOTE: this feature is not supported on all hardware platforms and it is // necessary to check returned status before using the returned values. - static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); + static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, + audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid static unsigned int getInputFramesLost(audio_io_handle_t ioHandle); @@ -147,8 +151,8 @@ public: NUM_CONFIG_EVENTS }; - // audio output descriptor used to cache output configurations in client process to avoid frequent calls - // through IAudioFlinger + // audio output descriptor used to cache output configurations in client process to avoid + // frequent calls through IAudioFlinger class OutputDescriptor { public: OutputDescriptor() @@ -162,8 +166,8 @@ public: }; // Events used to synchronize actions between audio sessions. - // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until playback - // is complete on another audio session. + // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until + // playback is complete on another audio session. // See definitions in MediaSyncEvent.java enum sync_event_t { SYNC_EVENT_SAME = -1, // used internally to indicate restart with same event @@ -183,8 +187,10 @@ public: // // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) // - static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, const char *device_address); - static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, const char *device_address); + static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, + const char *device_address); + static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, + const char *device_address); static status_t setPhoneState(audio_mode_t state); static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config); static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h index 34108b3..7dd22e8 100644 --- a/include/media/AudioTrack.h +++ b/include/media/AudioTrack.h @@ -53,9 +53,12 @@ public: enum event_type { EVENT_MORE_DATA = 0, // Request to write more data to PCM buffer. EVENT_UNDERRUN = 1, // PCM buffer underrun occured. - EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from loop start if loop count was not 0. - EVENT_MARKER = 3, // Playback head is at the specified marker position (See setMarkerPosition()). - EVENT_NEW_POS = 4, // Playback head is at a new position (See setPositionUpdatePeriod()). + EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from + // loop start if loop count was not 0. + EVENT_MARKER = 3, // Playback head is at the specified marker position + // (See setMarkerPosition()). + EVENT_NEW_POS = 4, // Playback head is at a new position + // (See setPositionUpdatePeriod()). EVENT_BUFFER_END = 5 // Playback head is at the end of the buffer. }; @@ -312,7 +315,8 @@ public: /* Sets marker position. When playback reaches the number of frames specified, a callback with * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker * notification callback. - * If the AudioTrack has been opened with no callback function associated, the operation will fail. + * If the AudioTrack has been opened with no callback function associated, the operation will + * fail. * * Parameters: * @@ -330,7 +334,8 @@ public: * a callback with event type EVENT_NEW_POS is called. * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification * callback. - * If the AudioTrack has been opened with no callback function associated, the operation will fail. + * If the AudioTrack has been opened with no callback function associated, the operation will + * fail. * * Parameters: * @@ -359,7 +364,8 @@ public: * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful operation * - INVALID_OPERATION: the AudioTrack is not stopped. - * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack buffer + * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack + * buffer */ status_t setPosition(uint32_t position); status_t getPosition(uint32_t *position); @@ -518,8 +524,10 @@ protected: callback_t mCbf; // callback handler for events, or NULL void* mUserData; - uint32_t mNotificationFramesReq; // requested number of frames between each notification callback - uint32_t mNotificationFramesAct; // actual number of frames between each notification callback + uint32_t mNotificationFramesReq; // requested number of frames between each + // notification callback + uint32_t mNotificationFramesAct; // actual number of frames between each + // notification callback sp<IMemory> mSharedBuffer; int mLoopCount; uint32_t mRemainingFrames; diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h index 65c26f4..b1ed7b0 100644 --- a/include/media/EffectsFactoryApi.h +++ b/include/media/EffectsFactoryApi.h @@ -74,7 +74,8 @@ int EffectQueryNumberEffects(uint32_t *pNumEffects); // -ENOENT no more effect available // -ENODEV factory failed to initialize // -EINVAL invalid pDescriptor -// -ENOSYS effect list has changed since last execution of EffectQueryNumberEffects() +// -ENOSYS effect list has changed since last execution of +// EffectQueryNumberEffects() // *pDescriptor: updated with the effect descriptor. // //////////////////////////////////////////////////////////////////////////////// @@ -91,12 +92,12 @@ int EffectQueryEffect(uint32_t index, effect_descriptor_t *pDescriptor); // // Input: // pEffectUuid: pointer to the effect uuid. -// sessionId: audio session to which this effect instance will be attached. All effects created -// with the same session ID are connected in series and process the same signal stream. -// Knowing that two effects are part of the same effect chain can help the library implement -// some kind of optimizations. -// ioId: identifies the output or input stream this effect is directed to at audio HAL. For future -// use especially with tunneled HW accelerated effects +// sessionId: audio session to which this effect instance will be attached. All effects +// created with the same session ID are connected in series and process the same signal +// stream. Knowing that two effects are part of the same effect chain can help the +// library implement some kind of optimizations. +// ioId: identifies the output or input stream this effect is directed to at audio HAL. +// For future use especially with tunneled HW accelerated effects // // Input/Output: // pHandle: address where to return the effect handle. @@ -109,7 +110,8 @@ int EffectQueryEffect(uint32_t index, effect_descriptor_t *pDescriptor); // *pHandle: updated with the effect handle. // //////////////////////////////////////////////////////////////////////////////// -int EffectCreate(const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle); +int EffectCreate(const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId, + effect_handle_t *pHandle); //////////////////////////////////////////////////////////////////////////////// // diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h index 5170a87..359780e 100644 --- a/include/media/IAudioFlinger.h +++ b/include/media/IAudioFlinger.h @@ -123,7 +123,8 @@ public: virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) = 0; - virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const = 0; + virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) + const = 0; // register a current process for audio output change notifications virtual void registerClient(const sp<IAudioFlingerClient>& client) = 0; diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h index cc2e069..f5b0604 100644 --- a/include/media/IAudioPolicyService.h +++ b/include/media/IAudioPolicyService.h @@ -44,9 +44,10 @@ public: audio_policy_dev_state_t state, const char *device_address) = 0; virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, - const char *device_address) = 0; + const char *device_address) = 0; virtual status_t setPhoneState(audio_mode_t state) = 0; - virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config) = 0; + virtual status_t setForceUse(audio_policy_force_use_t usage, + audio_policy_forced_cfg_t config) = 0; virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) = 0; virtual audio_io_handle_t getOutput(audio_stream_type_t stream, uint32_t samplingRate = 0, diff --git a/include/media/SoundPool.h b/include/media/SoundPool.h index 002b045..7bf3069 100644 --- a/include/media/SoundPool.h +++ b/include/media/SoundPool.h @@ -65,8 +65,10 @@ public: sp<IMemory> getIMemory() { return mData; } // hack - void init(int numChannels, int sampleRate, audio_format_t format, size_t size, sp<IMemory> data ) { - mNumChannels = numChannels; mSampleRate = sampleRate; mFormat = format; mSize = size; mData = data; } + void init(int numChannels, int sampleRate, audio_format_t format, size_t size, + sp<IMemory> data ) { + mNumChannels = numChannels; mSampleRate = sampleRate; mFormat = format; mSize = size; + mData = data; } private: void init(); diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h index 5b133f3..fe42afa 100644 --- a/include/private/media/AudioTrackShared.h +++ b/include/private/media/AudioTrackShared.h @@ -27,7 +27,8 @@ namespace android { // ---------------------------------------------------------------------------- // Maximum cumulated timeout milliseconds before restarting audioflinger thread -#define MAX_STARTUP_TIMEOUT_MS 3000 // Longer timeout period at startup to cope with A2DP init time +#define MAX_STARTUP_TIMEOUT_MS 3000 // Longer timeout period at startup to cope with A2DP + // init time #define MAX_RUN_TIMEOUT_MS 1000 #define WAIT_PERIOD_MS 10 #define RESTORE_TIMEOUT_MS 5000 // Maximum waiting time for a track to be restored @@ -100,7 +101,8 @@ public: uint8_t mName; // normal tracks: track name, fast tracks: track index // used by client only - uint16_t bufferTimeoutMs; // Maximum cumulated timeout before restarting audioflinger + uint16_t bufferTimeoutMs; // Maximum cumulated timeout before restarting + // audioflinger uint16_t waitTimeMs; // Cumulated wait time, used by client only private: 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..bdbee0d 100644 --- a/media/libmedia/AudioRecord.cpp +++ b/media/libmedia/AudioRecord.cpp @@ -127,7 +127,8 @@ status_t AudioRecord::set( int sessionId) { - ALOGV("set(): sampleRate %d, channelMask %#x, frameCount %d",sampleRate, channelMask, frameCount); + ALOGV("set(): sampleRate %d, channelMask %#x, frameCount %d",sampleRate, channelMask, + frameCount); AutoMutex lock(mLock); @@ -701,7 +702,8 @@ 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; diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp index 0e5c149..767c452 100644 --- a/media/libmedia/AudioSystem.cpp +++ b/media/libmedia/AudioSystem.cpp @@ -246,7 +246,8 @@ 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 %d", streamType, output, + *samplingRate); return NO_ERROR; } @@ -290,7 +291,8 @@ 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 +371,8 @@ 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(uint32_t *halFrames, uint32_t *dspFrames, + audio_stream_type_t stream) { const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger(); if (af == 0) return PERMISSION_DENIED; @@ -449,8 +452,10 @@ 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 %d, format %d channels %#x frameCount %d " + "latency %d", + outputDesc->samplingRate, outputDesc->format, outputDesc->channels, + outputDesc->frameCount, outputDesc->latency); } break; case OUTPUT_CLOSED: { if (gOutputs.indexOfKey(ioHandle) < 0) { @@ -471,7 +476,8 @@ 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 %d, format %d channels %#x " + "frameCount %d latency %d", ioHandle, desc->samplingRate, desc->format, desc->channels, desc->frameCount, desc->latency); OutputDescriptor *outputDesc = gOutputs.valueAt(index); diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index 362d022..5fc9b07 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -198,7 +198,8 @@ status_t AudioTrack::set( int sessionId) { - 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); @@ -617,12 +618,14 @@ status_t AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCou if (loopStart >= loopEnd || loopEnd - loopStart > cblk->frameCount || 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, cblk->frameCount, 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", + ALOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, " + "framecount %d", loopStart, loopEnd, cblk->frameCount); return BAD_VALUE; } @@ -924,7 +927,8 @@ status_t AudioTrack::createTrack_l( mCblk->stepUser(mCblk->frameCount); } - mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000)); + mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | + uint16_t(mVolume[LEFT] * 0x1000)); mCblk->setSendLevel(mSendLevel); mAudioTrack->attachAuxEffect(mAuxEffectId); mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS; @@ -994,8 +998,8 @@ 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(); @@ -1265,7 +1269,8 @@ 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; @@ -1439,11 +1444,14 @@ status_t AudioTrack::dump(int fd, const Vector<String16>& args) const String8 result; 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->frameCount); + snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, + mChannelCount, mCblk->frameCount); result.append(buffer); - snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted); + snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", + (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted); result.append(buffer); snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency); result.append(buffer); diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp index ce8ffc4..f412591 100644 --- a/media/libmedia/IAudioFlinger.cpp +++ b/media/libmedia/IAudioFlinger.cpp @@ -865,7 +865,8 @@ 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; @@ -1043,7 +1044,8 @@ 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/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; diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 9bdab2f..35bd431 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -1118,7 +1118,8 @@ void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, c // removeClient_l() must be called with AudioFlinger::mLock held void AudioFlinger::removeClient_l(pid_t pid) { - ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); + ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), + IPCThreadState::self()->getCallingPid()); mClients.removeItem(pid); } @@ -1221,7 +1222,8 @@ void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) { IoConfigEvent *ioEvent = new IoConfigEvent(event, param); mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); - ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); + ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, + param); mWaitWorkCV.signal(); } @@ -1250,7 +1252,8 @@ void AudioFlinger::ThreadBase::processConfigEvents() PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); if (err != 0) { - ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", + ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " + "error %d", prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); } } break; @@ -1667,7 +1670,8 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); result.append(buffer); - snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); + snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", + ns2ms(systemTime() - mLastWriteTime)); result.append(buffer); snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); result.append(buffer); @@ -1797,7 +1801,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac if (mType == DIRECT) { if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { - ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" + ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x " "for output %p with format %d", sampleRate, format, channelMask, mOutput, mFormat); lStatus = BAD_VALUE; @@ -1965,7 +1969,8 @@ status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) if (track->mainBuffer() != mMixBuffer) { sp<EffectChain> chain = getEffectChain_l(track->sessionId()); if (chain != 0) { - ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); + ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), + track->sessionId()); chain->incActiveTrackCnt(); } } @@ -2031,7 +2036,8 @@ void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { AudioSystem::OutputDescriptor desc; void *param2 = NULL; - ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); + ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, + param); switch (event) { case AudioSystem::OUTPUT_OPENED: @@ -2039,7 +2045,8 @@ void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { desc.channels = mChannelMask; desc.samplingRate = mSampleRate; desc.format = mFormat; - desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) + desc.frameCount = mNormalFrameCount; // FIXME see + // AudioFlinger::frameCount(audio_io_handle_t) desc.latency = latency(); param2 = &desc; break; @@ -2068,7 +2075,8 @@ void AudioFlinger::PlaybackThread::readOutputParameters() // Calculate size of normal mix buffer relative to the HAL output buffer size double multiplier = 1.0; - if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { + if (mType == MIXER && (kUseFastMixer == FastMixer_Static || + kUseFastMixer == FastMixer_Dynamic)) { size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer @@ -2087,9 +2095,10 @@ void AudioFlinger::PlaybackThread::readOutputParameters() multiplier = (double) maxNormalFrameCount / (double) mFrameCount; } } else { - // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC - // (it would be unusual for the normal mix buffer size to not be a multiple of fast - // track, but we sometimes have to do this to satisfy the maximum frame count constraint) + // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL + // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast + // track, but we sometimes have to do this to satisfy the maximum frame count + // constraint) // FIXME this rounding up should not be done if no HAL SRC uint32_t truncMult = (uint32_t) multiplier; if ((truncMult & 1)) { @@ -2103,7 +2112,8 @@ void AudioFlinger::PlaybackThread::readOutputParameters() mNormalFrameCount = multiplier * mFrameCount; // round up to nearest 16 frames to satisfy AudioMixer mNormalFrameCount = (mNormalFrameCount + 15) & ~15; - ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); + ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, + mNormalFrameCount); delete[] mMixBuffer; mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; @@ -2241,7 +2251,8 @@ bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; } -void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) +void AudioFlinger::PlaybackThread::threadLoop_removeTracks( + const Vector< sp<Track> >& tracksToRemove) { size_t count = tracksToRemove.size(); if (CC_UNLIKELY(count)) { @@ -2897,7 +2908,8 @@ void AudioFlinger::MixerThread::threadLoop_sleepTime() } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { memset (mMixBuffer, 0, mixBufferSize); sleepTime = 0; - ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); + ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), + "anticipated start"); } // TODO add standby time extension fct of effect tail } @@ -3131,7 +3143,8 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && !track->isTerminated()) { - ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); + ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, + this); mixedTracks++; @@ -3144,7 +3157,8 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac if (chain != 0) { tracksWithEffect++; } else { - ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", + ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " + "session %d", name, track->sessionId()); } } @@ -3274,7 +3288,8 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac chain->clearInputBuffer(); } - ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); + ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, + cblk->server, this); if ((track->sharedBuffer() != 0) || track->isTerminated() || track->isStopped() || track->isPaused()) { // We have consumed all the buffers of this track. @@ -3368,7 +3383,8 @@ track_is_ready: ; if (track->mainBuffer() != mMixBuffer) { chain = getEffectChain_l(track->sessionId()); if (chain != 0) { - ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); + ALOGV("stopping track on chain %p for session Id: %d", chain.get(), + track->sessionId()); chain->decActiveTrackCnt(); } } @@ -3381,7 +3397,8 @@ track_is_ready: ; // mix buffer must be cleared if all tracks are connected to an // effect chain as in this case the mixer will not write to // mix buffer and track effects will accumulate into it - if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { + if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || + (mixedTracks == 0 && fastTracks > 0)) { // FIXME as a performance optimization, should remember previous zero status memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); } @@ -3995,7 +4012,8 @@ void AudioFlinger::DirectOutputThread::cacheParameters_l() AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) - : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), DUPLICATING), + : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), + DUPLICATING), mWaitTimeMs(UINT_MAX) { addOutputTrack(mainThread); @@ -4116,18 +4134,21 @@ void AudioFlinger::DuplicatingThread::updateWaitTime_l() } -bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) +bool AudioFlinger::DuplicatingThread::outputsReady( + const SortedVector< sp<OutputTrack> > &outputTracks) { for (size_t i = 0; i < outputTracks.size(); i++) { sp<ThreadBase> thread = outputTracks[i]->thread().promote(); if (thread == 0) { - ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); + ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", + outputTracks[i].get()); return false; } PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); // see note at standby() declaration if (playbackThread->standby() && !playbackThread->isSuspended()) { - ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); + ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), + thread.get()); return false; } } @@ -4174,7 +4195,8 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase( // mChannelCount // mChannelMask { - ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); + ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), + sharedBuffer->size()); // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); size_t size = sizeof(audio_track_cblk_t); @@ -4335,7 +4357,8 @@ AudioFlinger::PlaybackThread::Track::Track( const sp<IMemory>& sharedBuffer, int sessionId, IAudioFlinger::track_flags_t flags) - : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), + : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, + sessionId), mMute(false), mFillingUpStatus(FS_INVALID), // mRetryCount initialized later when needed @@ -4354,7 +4377,8 @@ AudioFlinger::PlaybackThread::Track::Track( if (mCblk != NULL) { // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack - mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); + mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : + sizeof(uint8_t); // to avoid leaking a track name, do not allocate one unless there is an mCblk mName = thread->getTrackName_l(channelMask, sessionId); mCblk->mName = mName; @@ -4379,7 +4403,8 @@ AudioFlinger::PlaybackThread::Track::Track( thread->mFastTrackAvailMask &= ~(1 << i); } } - ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); + ALOGV("Track constructor name %d, calling pid %d", mName, + IPCThreadState::self()->getCallingPid()); } AudioFlinger::PlaybackThread::Track::~Track() @@ -4421,8 +4446,8 @@ void AudioFlinger::PlaybackThread::Track::destroy() /*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) { - result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " - " Server User Main buf Aux Buf Flags Underruns\n"); + result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate " + "L dB R dB Server User Main buf Aux Buf Flags Underruns\n"); } void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) @@ -4649,7 +4674,8 @@ void AudioFlinger::PlaybackThread::Track::stop() // and then to STOPPED and reset() when presentation is complete mState = STOPPING_1; } - ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); + ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, + playbackThread); } if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { thread->mLock.unlock(); @@ -5408,7 +5434,8 @@ AudioFlinger::RecordThread::RecordTrack::~RecordTrack() } // AudioBufferProvider interface -status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) +status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, + int64_t pts) { audio_track_cblk_t* cblk = this->cblk(); uint32_t framesAvail; @@ -5600,7 +5627,8 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr mOutBuffer.frameCount = pInBuffer->frameCount; nsecs_t startTime = systemTime(); if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { - ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); + ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, + mThread.unsafe_get()); outputBufferFull = true; break; } @@ -5612,7 +5640,8 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr } } - uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; + uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : + pInBuffer->frameCount; memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); mCblk->stepUser(outFrames); pInBuffer->frameCount -= outFrames; @@ -5625,7 +5654,8 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr mBufferQueue.removeAt(0); delete [] pInBuffer->mBuffer; delete pInBuffer; - ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); + ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, + mThread.unsafe_get(), mBufferQueue.size()); } else { break; } @@ -5641,11 +5671,14 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; pInBuffer->frameCount = inBuffer.frameCount; pInBuffer->i16 = pInBuffer->mBuffer; - memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); + memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * + sizeof(int16_t)); mBufferQueue.add(pInBuffer); - ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); + ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, + mThread.unsafe_get(), mBufferQueue.size()); } else { - ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); + ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", + mThread.unsafe_get(), this); } } } @@ -5670,7 +5703,8 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr return outputBufferFull; } -status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) +status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( + AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) { int active; status_t result; @@ -5934,13 +5968,14 @@ sp<IAudioRecord> AudioFlinger::openRecord( *sessionId = lSessionId; } } - // create new record track. The record track uses one track in mHardwareMixerThread by convention. + // create new record track. + // The record track uses one track in mHardwareMixerThread by convention. recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, frameCount, lSessionId, flags, tid, &lStatus); } if (lStatus != NO_ERROR) { - // remove local strong reference to Client before deleting the RecordTrack so that the Client - // destructor is called by the TrackBase destructor with mLock held + // remove local strong reference to Client before deleting the RecordTrack so that the + // Client destructor is called by the TrackBase destructor with mLock held client.clear(); recordTrack.clear(); goto Exit; @@ -5959,7 +5994,8 @@ Exit: // ---------------------------------------------------------------------------- -AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) +AudioFlinger::RecordHandle::RecordHandle( + const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) : BnAudioRecord(), mRecordTrack(recordTrack) { @@ -5974,7 +6010,8 @@ sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { return mRecordTrack->getCblk(); } -status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, int triggerSession) { +status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, + int triggerSession) { ALOGV("RecordHandle::start()"); return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); } @@ -6122,7 +6159,8 @@ bool AudioFlinger::RecordThread::threadLoop() size_t framesIn = mFrameCount - mRsmpInIndex; if (framesIn) { int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; - int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; + int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * + mActiveTrack->mCblk->frameSize; if (framesIn > framesOut) framesIn = framesOut; mRsmpInIndex += framesIn; @@ -6143,7 +6181,8 @@ bool AudioFlinger::RecordThread::threadLoop() if (framesOut && mFrameCount == mRsmpInIndex) { void *readInto; if (framesOut == mFrameCount && - ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { + ((int)mChannelCount == mReqChannelCount || + mFormat != AUDIO_FORMAT_PCM_16_BIT)) { readInto = buffer.raw; framesOut = 0; } else { @@ -6177,12 +6216,14 @@ bool AudioFlinger::RecordThread::threadLoop() if (mChannelCount == 1 && mReqChannelCount == 1) { framesOut >>= 1; } - mResampler->resample(mRsmpOutBuffer, framesOut, this /* AudioBufferProvider* */); - // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() - // are 32 bit aligned which should be always true. + mResampler->resample(mRsmpOutBuffer, framesOut, + this /* AudioBufferProvider* */); + // ditherAndClamp() works as long as all buffers returned by + // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. if (mChannelCount == 2 && mReqChannelCount == 1) { ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); - // the resampler always outputs stereo samples: do post stereo to mono conversion + // the resampler always outputs stereo samples: + // do post stereo to mono conversion downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, framesOut); } else { @@ -6656,7 +6697,8 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() status = BAD_VALUE; } else { mInDevice = value; - // disable AEC and NS if the device is a BT SCO headset supporting those pre processings + // disable AEC and NS if the device is a BT SCO headset supporting those + // pre processings if (mTracks.size() > 0) { bool suspend = audio_is_bluetooth_sco_device(mInDevice) && mAudioFlinger->btNrecIsOff(); @@ -6678,7 +6720,8 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() mAudioSource = (audio_source_t)value; } if (status == NO_ERROR) { - status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); + status = mInput->stream->common.set_parameters(&mInput->stream->common, + keyValuePair.string()); if (status == INVALID_OPERATION) { inputStandBy(); status = mInput->stream->common.set_parameters(&mInput->stream->common, @@ -6688,8 +6731,10 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() if (status == BAD_VALUE && reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && reqFormat == AUDIO_FORMAT_PCM_16_BIT && - ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && - popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && + ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) + <= (2 * reqSamplingRate)) && + popcount(mInput->stream->common.get_channels(&mInput->stream->common)) + <= FCC_2 && (reqChannelCount <= FCC_2)) { status = NO_ERROR; } @@ -6783,7 +6828,8 @@ void AudioFlinger::RecordThread::readInputParameters() mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); mRsmpOutBuffer = new int32_t[mFrameCount * 2]; - // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples + // optmization: if mono to mono, alter input frame count as if we were inputing + // stereo samples if (mChannelCount == 1 && mReqChannelCount == 1) { mFrameCount >>= 1; } @@ -7010,7 +7056,8 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, &outStream); mHardwareStatus = AUDIO_HW_IDLE; - ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", + ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, " + "Channels %x, status %d", outStream, config.sample_rate, config.format, @@ -7063,7 +7110,8 @@ audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, MixerThread *thread2 = checkMixerThread_l(output2); if (thread1 == NULL || thread2 == NULL) { - ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); + ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, + output2); return 0; } @@ -7098,7 +7146,8 @@ status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) if (thread->type() == ThreadBase::MIXER) { for (size_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { - DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); + DuplicatingThread *dupThread = + (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); dupThread->removeOutputTrack((MixerThread *)thread.get()); } } @@ -7185,16 +7234,17 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); - ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", + ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " + "status %d", inStream, config.sample_rate, config.format, config.channel_mask, status); - // If the input could not be opened with the requested parameters and we can handle the conversion internally, - // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo - // or stereo to mono conversions on 16 bit PCM inputs. + // If the input could not be opened with the requested parameters and we can handle the + // conversion internally, try to open again with the proposed parameters. The AudioFlinger can + // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. if (status == BAD_VALUE && reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && (config.sample_rate <= 2 * reqSamplingRate) && @@ -8072,7 +8122,8 @@ status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& c for (size_t i = 0; i < mTracks.size(); ++i) { sp<Track> track = mTracks[i]; if (session == track->sessionId()) { - ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); + ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), + buffer); track->setMainBuffer(buffer); chain->incTrackCnt(); } @@ -8914,12 +8965,15 @@ void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) result.append("\t\tDescriptor:\n"); snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, - mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], + mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1], + mDescriptor.uuid.node[2], mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); result.append(buffer); snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", - mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, - mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], + mDescriptor.type.timeLow, mDescriptor.type.timeMid, + mDescriptor.type.timeHiAndVersion, + mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1], + mDescriptor.type.node[2], mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); result.append(buffer); snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", @@ -9003,7 +9057,8 @@ AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, mBuffer = (uint8_t *)mCblk + bufOffset; } } else { - ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); + ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + + sizeof(effect_param_cblk_t)); return; } } @@ -9130,8 +9185,9 @@ status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, // handle commands that are not forwarded transparently to effect engine if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { - // No need to trylock() here as this function is executed in the binder thread serving a particular client process: - // no risk to block the whole media server process or mixer threads is we are stuck here + // No need to trylock() here as this function is executed in the binder thread serving a + // particular client process: no risk to block the whole media server process or mixer + // threads if we are stuck here Mutex::Autolock _l(mCblk->lock); if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { @@ -9271,7 +9327,8 @@ AudioFlinger::EffectChain::~EffectChain() } // getEffectFromDesc_l() must be called with ThreadBase::mLock held -sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) +sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l( + effect_descriptor_t *descriptor) { size_t size = mEffects.size(); @@ -9430,7 +9487,8 @@ status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) // check invalid effect chaining combinations if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { - ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); + ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", + desc.name, d.name); return INVALID_OPERATION; } // remember position of first insert effect and by default @@ -9481,7 +9539,8 @@ status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) } mEffects.insertAt(effect, idx_insert); - ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); + ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, + idx_insert); } effect->configure(); return NO_ERROR; @@ -9512,7 +9571,8 @@ size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) } } mEffects.removeAt(i); - ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); + ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), + this, i); break; } } @@ -9736,7 +9796,8 @@ void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) for (size_t i = 0; i < types.size(); i++) { setEffectSuspended_l(types[i], false); } - ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); + ALOGV("setEffectSuspendedAll_l() remove entry for %08x", + mSuspendedEffects.keyAt(index)); mSuspendedEffects.removeItem((int)kKeyForSuspendAll); } } @@ -9762,7 +9823,8 @@ bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descript return true; } -void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) +void AudioFlinger::EffectChain::getSuspendEligibleEffects( + Vector< sp<AudioFlinger::EffectModule> > &effects) { effects.clear(); for (size_t i = 0; i < mEffects.size(); i++) { diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h index 116820f..2251b45 100644 --- a/services/audioflinger/AudioFlinger.h +++ b/services/audioflinger/AudioFlinger.h @@ -269,12 +269,14 @@ private: virtual ~AudioFlinger(); // call in any IAudioFlinger method that accesses mPrimaryHardwareDev - status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; } + status_t initCheck() const { return mPrimaryHardwareDev == NULL ? + NO_INIT : NO_ERROR; } // RefBase virtual void onFirstRef(); - AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, audio_devices_t devices); + AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, + audio_devices_t devices); void purgeStaleEffects_l(); // standby delay for MIXER and DUPLICATING playback threads is read from property @@ -746,7 +748,8 @@ private: const sp<PMDeathRecipient> mDeathRecipient; // list of suspended effects per session and per type. The first vector is // keyed by session ID, the second by type UUID timeLow field - KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions; + KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > + mSuspendedSessions; }; struct stream_type_t { @@ -788,7 +791,8 @@ private: static void appendDumpHeader(String8& result); void dump(char* buffer, size_t size); - virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, + virtual status_t start(AudioSystem::sync_event_t event = + AudioSystem::SYNC_EVENT_NONE, int triggerSession = 0); virtual void stop(); void pause(); @@ -823,7 +827,8 @@ private: Track& operator = (const Track&); // AudioBufferProvider interface - virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); + virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, + int64_t pts = kInvalidPTS); // releaseBuffer() not overridden virtual size_t framesReady() const; @@ -877,8 +882,8 @@ private: int32_t *mAuxBuffer; int mAuxEffectId; bool mHasVolumeController; - size_t mPresentationCompleteFrames; // number of frames written to the audio HAL - // when this track will be fully rendered + size_t mPresentationCompleteFrames; // number of frames written to the + // audio HAL when this track will be fully rendered private: IAudioFlinger::track_flags_t mFlags; @@ -1000,7 +1005,8 @@ private: int frameCount); virtual ~OutputTrack(); - virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, + virtual status_t start(AudioSystem::sync_event_t event = + AudioSystem::SYNC_EVENT_NONE, int triggerSession = 0); virtual void stop(); bool write(int16_t* data, uint32_t frames); @@ -1014,7 +1020,8 @@ private: NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value }; - status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs); + status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, + uint32_t waitTimeMs); void clearBufferQueue(); // Maximum number of pending buffers allocated by OutputTrack::write() @@ -1186,7 +1193,8 @@ public: void dumpTracks(int fd, const Vector<String16>& args); SortedVector< sp<Track> > mTracks; - // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread + // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by + // DuplicatingThread stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; AudioStreamOut *mOutput; @@ -1454,7 +1462,8 @@ public: // clear the buffer overflow flag void clearOverflow() { mOverflow = false; } // set the buffer overflow flag and return previous value - bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } + bool setOverflow() { bool tmp = mOverflow; mOverflow = true; + return tmp; } static void appendDumpHeader(String8& result); void dump(char* buffer, size_t size); @@ -1466,7 +1475,8 @@ public: RecordTrack& operator = (const RecordTrack&); // AudioBufferProvider interface - virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); + virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, + int64_t pts = kInvalidPTS); // releaseBuffer() not overridden bool mOverflow; // overflow on most recent attempt to fill client buffer @@ -1786,7 +1796,8 @@ mutable Mutex mLock; // mutex for process, commands and handl sp<IEffectClient> mEffectClient; // callback interface for client notifications /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() sp<IMemory> mCblkMemory; // shared memory for control block - effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory + effect_param_cblk_t* mCblk; // control block for deferred parameter setting via + // shared memory uint8_t* mBuffer; // pointer to parameter area in shared memory int mPriority; // client application priority to control the effect bool mHasControl; // true if this handle is controlling the effect @@ -1799,10 +1810,10 @@ mutable Mutex mLock; // mutex for process, commands and handl // the EffectChain class represents a group of effects associated to one audio session. // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). // The EffecChain with session ID 0 contains global effects applied to the output mix. - // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks) - // are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding - // in the effect process order. When attached to a track (session ID != 0), it also provide it's own - // input buffer used by the track as accumulation buffer. + // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to + // tracks) are insert only. The EffectChain maintains an ordered list of effect module, the + // order corresponding in the effect process order. When attached to a track (session ID != 0), + // it also provide it's own input buffer used by the track as accumulation buffer. class EffectChain : public RefBase { public: EffectChain(const wp<ThreadBase>& wThread, int sessionId); diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp index a4ed445..b3ca877 100644 --- a/services/audioflinger/AudioMixer.cpp +++ b/services/audioflinger/AudioMixer.cpp @@ -765,7 +765,8 @@ void AudioMixer::process__validate(state_t* state, int64_t pts) } -void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux) +void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount, + int32_t* temp, int32_t* aux) { t->resampler->setSampleRate(t->sampleRate); @@ -798,11 +799,13 @@ void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFram } } -void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux) +void AudioMixer::track__nop(track_t* t, int32_t* out, size_t outFrameCount, int32_t* temp, + int32_t* aux) { } -void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) +void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, + int32_t* aux) { int32_t vl = t->prevVolume[0]; int32_t vr = t->prevVolume[1]; @@ -844,7 +847,8 @@ void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, i t->adjustVolumeRamp(aux != NULL); } -void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) +void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, + int32_t* aux) { const int16_t vl = t->volume[0]; const int16_t vr = t->volume[1]; @@ -872,7 +876,8 @@ void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32 } } -void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) +void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, + int32_t* aux) { const int16_t *in = static_cast<const int16_t *>(t->in); @@ -962,7 +967,8 @@ void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount t->in = in; } -void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) +void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, + int32_t* aux) { const int16_t *in = static_cast<int16_t const *>(t->in); @@ -1147,7 +1153,8 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) while (outFrames) { size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount; if (inFrames) { - t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux); + t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, + state->resampleTemp, aux); t.frameCount -= inFrames; outFrames -= inFrames; if (CC_UNLIKELY(aux != NULL)) { @@ -1156,7 +1163,8 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts) } if (t.frameCount == 0 && outFrames) { t.bufferProvider->releaseBuffer(&t.buffer); - t.buffer.frameCount = (state->frameCount - numFrames) - (BLOCKSIZE - outFrames); + t.buffer.frameCount = (state->frameCount - numFrames) - + (BLOCKSIZE - outFrames); int64_t outputPTS = calculateOutputPTS( t, pts, numFrames + (BLOCKSIZE - outFrames)); t.bufferProvider->getNextBuffer(&t.buffer, outputPTS); @@ -1246,7 +1254,8 @@ void AudioMixer::process__genericResampling(state_t* state, int64_t pts) if (CC_UNLIKELY(aux != NULL)) { aux += outFrames; } - t.hook(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp, aux); + t.hook(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, + state->resampleTemp, aux); outFrames += t.buffer.frameCount; t.bufferProvider->releaseBuffer(&t.buffer); } @@ -1286,7 +1295,8 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state, // been enabled for mixing. if (in == NULL || ((unsigned long)in & 3)) { memset(out, 0, numFrames*MAX_NUM_CHANNELS*sizeof(int16_t)); - ALOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: buffer %p track %d, channels %d, needs %08x", + ALOGE_IF(((unsigned long)in & 3), "process stereo track: input buffer alignment pb: " + "buffer %p track %d, channels %d, needs %08x", in, i, t.channelCount, t.needs); return; } diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h index e60a298..fd21fda 100644 --- a/services/audioflinger/AudioMixer.h +++ b/services/audioflinger/AudioMixer.h @@ -146,7 +146,8 @@ private: struct track_t; class DownmixerBufferProvider; - typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux); + typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, + int32_t* aux); static const int BLOCKSIZE = 16; // 4 cache lines struct track_t { @@ -261,12 +262,17 @@ private: static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum); static void unprepareTrackForDownmix(track_t* pTrack, int trackName); - static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); + static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, + int32_t* aux); static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); - static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); - static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux); - static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); - static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux); + static void track__16BitsStereo(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, + int32_t* aux); + static void track__16BitsMono(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, + int32_t* aux); + static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, + int32_t* aux); + static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, + int32_t* aux); static void process__validate(state_t* state, int64_t pts); static void process__nop(state_t* state, int64_t pts); diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp index 8b99bd2..ea130ba 100644 --- a/services/audioflinger/AudioPolicyService.cpp +++ b/services/audioflinger/AudioPolicyService.cpp @@ -227,7 +227,8 @@ audio_io_handle_t AudioPolicyService::getOutput(audio_stream_type_t stream, } ALOGV("getOutput() tid %d", gettid()); Mutex::Autolock _l(mLock); - return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate, format, channelMask, flags); + return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate, format, channelMask, + flags); } status_t AudioPolicyService::startOutput(audio_io_handle_t output, @@ -280,7 +281,7 @@ audio_io_handle_t AudioPolicyService::getInput(audio_source_t inputSource, Mutex::Autolock _l(mLock); // the audio_in_acoustics_t parameter is ignored by get_input() audio_io_handle_t input = mpAudioPolicy->get_input(mpAudioPolicy, inputSource, samplingRate, - format, channelMask, (audio_in_acoustics_t) 0); + format, channelMask, (audio_in_acoustics_t) 0); if (input == 0) { return input; diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h index 63f9549..92653c1 100644 --- a/services/audioflinger/AudioPolicyService.h +++ b/services/audioflinger/AudioPolicyService.h @@ -142,11 +142,11 @@ private: status_t dumpInternals(int fd); // Thread used for tone playback and to send audio config commands to audio flinger - // For tone playback, using a separate thread is necessary to avoid deadlock with mLock because startTone() - // and stopTone() are normally called with mLock locked and requesting a tone start or stop will cause - // calls to AudioPolicyService and an attempt to lock mLock. - // For audio config commands, it is necessary because audio flinger requires that the calling process (user) - // has permission to modify audio settings. + // For tone playback, using a separate thread is necessary to avoid deadlock with mLock because + // startTone() and stopTone() are normally called with mLock locked and requesting a tone start + // or stop will cause calls to AudioPolicyService and an attempt to lock mLock. + // For audio config commands, it is necessary because audio flinger requires that the calling + // process (user) has permission to modify audio settings. class AudioCommandThread : public Thread { class AudioCommand; public: diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp index a8e23e4..151313b 100644 --- a/services/audioflinger/test-resample.cpp +++ b/services/audioflinger/test-resample.cpp @@ -61,7 +61,8 @@ struct HeaderWav { }; static int usage(const char* name) { - fprintf(stderr,"Usage: %s [-p] [-h] [-q <dq|lq|mq|hq|vhq>] [-i <input-sample-rate>] [-o <output-sample-rate>] <input-file> <output-file>\n", name); + fprintf(stderr,"Usage: %s [-p] [-h] [-q <dq|lq|mq|hq|vhq>] [-i <input-sample-rate>] " + "[-o <output-sample-rate>] <input-file> <output-file>\n", name); fprintf(stderr,"-p - enable profiling\n"); fprintf(stderr,"-h - create wav file\n"); fprintf(stderr,"-q - resampler quality\n"); |