/* //device/include/server/AudioFlinger/AudioFlinger.cpp ** ** Copyright 2007, The Android Open Source Project ** Copyright (c) 2010-2011, The Linux Foundation. All rights reserved. ** ** Licensed under the Apache License, Version 2.0 (the "License"); ** you may not use this file except in compliance with the License. ** You may obtain a copy of the License at ** ** http://www.apache.org/licenses/LICENSE-2.0 ** ** Unless required by applicable law or agreed to in writing, software ** distributed under the License is distributed on an "AS IS" BASIS, ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. ** See the License for the specific language governing permissions and ** limitations under the License. */ #define LOG_TAG "AudioFlinger" //#define LOG_NDEBUG 0 #define LOG_NDDEBUG 0 #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include "AudioMixer.h" #include "AudioFlinger.h" #ifdef WITH_A2DP #include "A2dpAudioInterface.h" #endif #ifdef SRS_PROCESSING #include "srs_processing.h" #endif #ifdef LVMX #include "lifevibes.h" #endif #include #include // ---------------------------------------------------------------------------- // the sim build doesn't have gettid #ifndef HAVE_GETTID # define gettid getpid #endif // ---------------------------------------------------------------------------- extern const char * const gEffectLibPath; namespace android { static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n"; static const char* kHardwareLockedString = "Hardware lock is taken\n"; //static const nsecs_t kStandbyTimeInNsecs = seconds(3); static const float MAX_GAIN = 4096.0f; static const float MAX_GAIN_INT = 0x1000; // retry counts for buffer fill timeout // 50 * ~20msecs = 1 second static const int8_t kMaxTrackRetries = 50; static const int8_t kMaxTrackStartupRetries = 50; // allow less retry attempts on direct output thread. // direct outputs can be a scarce resource in audio hardware and should // be released as quickly as possible. static const int8_t kMaxTrackRetriesDirect = 2; static const int kDumpLockRetries = 50; static const int kDumpLockSleep = 20000; static const nsecs_t kWarningThrottle = seconds(5); #define AUDIOFLINGER_SECURITY_ENABLED 1 // ---------------------------------------------------------------------------- static bool recordingAllowed() { #ifndef HAVE_ANDROID_OS return true; #endif #if AUDIOFLINGER_SECURITY_ENABLED if (getpid() == IPCThreadState::self()->getCallingPid()) return true; bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO"); return ok; #else if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO"))) LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest"); return true; #endif } static bool settingsAllowed() { #ifndef HAVE_ANDROID_OS return true; #endif #if AUDIOFLINGER_SECURITY_ENABLED if (getpid() == IPCThreadState::self()->getCallingPid()) return true; bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); return ok; #else if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"))) LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest"); return true; #endif } static uint32_t getInputChannelCount(uint32_t channels) { // only mono or stereo is supported for input sources return AudioSystem::popCount((channels) & (AudioSystem::CHANNEL_IN_STEREO | AudioSystem::CHANNEL_IN_MONO)); } // ---------------------------------------------------------------------------- AudioFlinger::AudioFlinger() : BnAudioFlinger(), mAudioHardware(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), mFmOn(false) { mHardwareStatus = AUDIO_HW_IDLE; mLPAOutput = NULL; mLPAHandle = -1; mLPAStreamIsActive = false; mLPASessionId = -2; // -2 is invalid session ID mIsEffectConfigChanged = false; mLPAEffectChain = NULL; mAudioHardware = AudioHardwareInterface::create(); mHardwareStatus = AUDIO_HW_INIT; if (mAudioHardware->initCheck() == NO_ERROR) { // open 16-bit output stream for s/w mixer mMode = AudioSystem::MODE_NORMAL; setMode(mMode); setMasterVolume(1.0f); setMasterMute(false); } else { LOGE("Couldn't even initialize the stubbed audio hardware!"); } #ifdef LVMX LifeVibes::init(); mLifeVibesClientPid = -1; #endif } AudioFlinger::~AudioFlinger() { while (!mRecordThreads.isEmpty()) { // closeInput() will remove first entry from mRecordThreads closeInput(mRecordThreads.keyAt(0)); } while (!mPlaybackThreads.isEmpty()) { // closeOutput() will remove first entry from mPlaybackThreads closeOutput(mPlaybackThreads.keyAt(0)); } if (mLPAOutput) { // Close the Output closeSession(mLPAHandle); } if (mAudioHardware) { delete mAudioHardware; } } status_t AudioFlinger::dumpClients(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; result.append("Clients:\n"); for (size_t i = 0; i < mClients.size(); ++i) { wp wClient = mClients.valueAt(i); if (wClient != 0) { sp client = wClient.promote(); if (client != 0) { snprintf(buffer, SIZE, " pid: %d\n", client->pid()); result.append(buffer); } } } write(fd, result.string(), result.size()); return NO_ERROR; } status_t AudioFlinger::dumpInternals(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; int hardwareStatus = mHardwareStatus; snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); result.append(buffer); write(fd, result.string(), result.size()); return NO_ERROR; } status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "Permission Denial: " "can't dump AudioFlinger from pid=%d, uid=%d\n", IPCThreadState::self()->getCallingPid(), IPCThreadState::self()->getCallingUid()); result.append(buffer); write(fd, result.string(), result.size()); return NO_ERROR; } static bool tryLock(Mutex& mutex) { bool locked = false; for (int i = 0; i < kDumpLockRetries; ++i) { if (mutex.tryLock() == NO_ERROR) { locked = true; break; } usleep(kDumpLockSleep); } return locked; } status_t AudioFlinger::dump(int fd, const Vector& args) { if (checkCallingPermission(String16("android.permission.DUMP")) == false) { dumpPermissionDenial(fd, args); } else { // get state of hardware lock bool hardwareLocked = tryLock(mHardwareLock); if (!hardwareLocked) { String8 result(kHardwareLockedString); write(fd, result.string(), result.size()); } else { mHardwareLock.unlock(); } bool locked = tryLock(mLock); // failed to lock - AudioFlinger is probably deadlocked if (!locked) { String8 result(kDeadlockedString); write(fd, result.string(), result.size()); } dumpClients(fd, args); dumpInternals(fd, args); // dump playback threads for (size_t i = 0; i < mPlaybackThreads.size(); i++) { mPlaybackThreads.valueAt(i)->dump(fd, args); } // dump record threads for (size_t i = 0; i < mRecordThreads.size(); i++) { mRecordThreads.valueAt(i)->dump(fd, args); } if (mAudioHardware) { mAudioHardware->dumpState(fd, args); } if (locked) mLock.unlock(); } return NO_ERROR; } // IAudioFlinger interface sp AudioFlinger::createTrack( pid_t pid, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, const sp& sharedBuffer, int output, int *sessionId, status_t *status) { sp track; sp trackHandle; sp client; wp wclient; status_t lStatus; int lSessionId; if (streamType >= AudioSystem::NUM_STREAM_TYPES) { LOGE("invalid stream type"); lStatus = BAD_VALUE; goto Exit; } { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); PlaybackThread *effectThread = NULL; if (thread == NULL) { LOGE("unknown output thread"); lStatus = BAD_VALUE; goto Exit; } wclient = mClients.valueFor(pid); if (wclient != NULL) { client = wclient.promote(); } else { client = new Client(this, pid); mClients.add(pid, client); } LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) { for (size_t i = 0; i < mPlaybackThreads.size(); i++) { sp t = mPlaybackThreads.valueAt(i); if (mPlaybackThreads.keyAt(i) != output) { // prevent same audio session on different output threads uint32_t sessions = t->hasAudioSession(*sessionId); if (sessions & PlaybackThread::TRACK_SESSION) { lStatus = BAD_VALUE; goto Exit; } // check if an effect with same session ID is waiting for a track to be created if (sessions & PlaybackThread::EFFECT_SESSION) { effectThread = t.get(); } } } lSessionId = *sessionId; } else { // if no audio session id is provided, create one here lSessionId = nextUniqueId(); if (sessionId != NULL) { *sessionId = lSessionId; } } LOGD("createTrack() lSessionId: %d", lSessionId); track = thread->createTrack_l(client, streamType, sampleRate, format, channelCount, frameCount, sharedBuffer, lSessionId, &lStatus); // move effect chain to this output thread if an effect on same session was waiting // for a track to be created if (lStatus == NO_ERROR && effectThread != NULL) { Mutex::Autolock _dl(thread->mLock); Mutex::Autolock _sl(effectThread->mLock); moveEffectChain_l(lSessionId, effectThread, thread, true); } } if (lStatus == NO_ERROR) { trackHandle = new TrackHandle(track); } else { // remove local strong reference to Client before deleting the Track so that the Client // destructor is called by the TrackBase destructor with mLock held client.clear(); track.clear(); } Exit: if(status) { *status = lStatus; } return trackHandle; } void AudioFlinger::createSession( pid_t pid, uint32_t sampleRate, int channelCount, int *sessionId, status_t *status) { status_t lStatus = NO_ERROR; { // createSession can be called from same PID (mediaserver process) only if(pid != getpid()){ lStatus = BAD_VALUE; goto Exit; } Mutex::Autolock _l(mLock); LOGV("createSession() sessionId: %d sampleRate %d channelCount %d", *sessionId, sampleRate, channelCount); if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) { for (size_t i = 0; i < mPlaybackThreads.size(); i++) { sp t = mPlaybackThreads.valueAt(i); // Check if the session ID is already associated with a track uint32_t sessions = t->hasAudioSession(*sessionId); if (sessions & PlaybackThread::TRACK_SESSION) { LOGE("There is a track already associated with this session %d", *sessionId); lStatus = BAD_VALUE; goto Exit; } // check if an effect with same session ID is waiting for a ssession to be created if (sessions & PlaybackThread::EFFECT_SESSION) { // Clear reference to previous effect chain if any if(mLPAEffectChain.get()) { mLPAEffectChain.clear(); } mLPAEffectChain = t->getEffectChain_l(*sessionId); } } mLPASessionId = *sessionId; LOGV("createSession() lSessionId: %d", mLPASessionId); if (mLPAEffectChain != NULL) { mLPAEffectChain->setLPAFlag(true); // For LPA, the volume will be applied in DSP. No need for volume // control in the Effect chain, so setting it to unity. uint32_t volume = 0x1000000; // Equals to 1.0 in 8.24 format mLPAEffectChain->setVolume_l(&volume,&volume); } else { LOGW("There was no effectChain created for the sessionId(%d)", mLPASessionId); } } else { if(sessionId != NULL) { LOGE("Error: Invalid sessionID (%d) for LPA playback", *sessionId); } } mLPASampleRate = sampleRate; mLPANumChannels = channelCount; } #ifdef SRS_PROCESSING LOGD("SRS_Processing - CreateSession - OutNotify_Init: %p TID %d\n", this, gettid()); SRS_Processing::ProcessOutNotify(SRS_Processing::AUTO, this, true); #endif Exit: if(status) { *status = lStatus; } } void AudioFlinger::deleteSession() { Mutex::Autolock _l(mLock); LOGV("deleteSession"); // -2 is invalid session ID mLPASessionId = -2; if (mLPAEffectChain != NULL) { mLPAEffectChain->setLPAFlag(false); size_t i, numEffects = mLPAEffectChain->getNumEffects(); for(i = 0; i < numEffects; i++) { sp effect = mLPAEffectChain->getEffectFromIndex_l(i); effect->setInBuffer(mLPAEffectChain->inBuffer()); if (i == numEffects-1) { effect->setOutBuffer(mLPAEffectChain->outBuffer()); } else { effect->setOutBuffer(mLPAEffectChain->inBuffer()); } effect->configure(); } mLPAEffectChain.clear(); mLPAEffectChain = NULL; } #ifdef SRS_PROCESSING LOGD("SRS_Processing - deleteSession - OutNotify_Init: %p TID %d\n", this, gettid()); SRS_Processing::ProcessOutNotify(SRS_Processing::AUTO, this, false); #endif } // ToDo: Should we go ahead with this frameCount? #define DEAFULT_FRAME_COUNT 1200 void AudioFlinger::applyEffectsOn(int16_t *inBuffer, int16_t *outBuffer, int size) { LOGV("applyEffectsOn: inBuf %p outBuf %p size %d", inBuffer, outBuffer, size); // This might be the first buffer to apply effects after effect config change // should not skip effects processing mIsEffectConfigChanged = false; volatile size_t numEffects = 0; if(mLPAEffectChain != NULL) { numEffects = mLPAEffectChain->getNumEffects(); } if( numEffects > 0) { size_t i = 0; int16_t *pIn = inBuffer; int16_t *pOut = outBuffer; int frameCount = size / (sizeof(int16_t) * mLPANumChannels); while(frameCount > 0) { if(mLPAEffectChain == NULL) { LOGV("LPA Effect Chain is removed - No effects processing !!"); numEffects = 0; break; } mLPAEffectChain->lock(); numEffects = mLPAEffectChain->getNumEffects(); if(!numEffects) { LOGV("applyEffectsOn: All the effects are removed - nothing to process"); mLPAEffectChain->unlock(); break; } int outFrameCount = (frameCount > DEAFULT_FRAME_COUNT ? DEAFULT_FRAME_COUNT: frameCount); bool isEffectEnabled = false; for(i = 0; i < numEffects; i++) { // If effect configuration is changed while applying effects do not process further if(mIsEffectConfigChanged) { mLPAEffectChain->unlock(); LOGV("applyEffectsOn: mIsEffectConfigChanged is set - no further processing"); return; } sp effect = mLPAEffectChain->getEffectFromIndex_l(i); if(effect == NULL) { LOGE("getEffectFromIndex_l(%d) returned NULL ptr", i); mLPAEffectChain->unlock(); return; } if(i == 0) { // For the first set input and output buffers different isEffectEnabled = effect->isProcessEnabled(); effect->setInBuffer(pIn); effect->setOutBuffer(pOut); } else { // For the remaining use previous effect's output buffer as input buffer effect->setInBuffer(pOut); effect->setOutBuffer(pOut); } // true indicates that it is being applied on LPA output effect->configure(true, mLPASampleRate, mLPANumChannels, outFrameCount); } if(isEffectEnabled) { // Clear the output buffer memset(pOut, 0, (outFrameCount * mLPANumChannels * sizeof(int16_t))); } else { // Copy input buffer content to the output buffer memcpy(pOut, pIn, (outFrameCount * mLPANumChannels * sizeof(int16_t))); } mLPAEffectChain->process_l(); mLPAEffectChain->unlock(); // Update input and output buffer pointers pIn += (outFrameCount * mLPANumChannels); pOut += (outFrameCount * mLPANumChannels); frameCount -= outFrameCount; } } if (!numEffects) { LOGV("applyEffectsOn: There are no effects to be applied"); if(inBuffer != outBuffer) { // No effect applied so just copy input buffer to output buffer memcpy(outBuffer, inBuffer, size); } } #ifdef SRS_PROCESSING SRS_Processing::ProcessOut(SRS_Processing::AUTO, this, outBuffer, size, mLPASampleRate, mLPANumChannels); #endif } uint32_t AudioFlinger::sampleRate(int output) const { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { LOGW("sampleRate() unknown thread %d", output); return 0; } return thread->sampleRate(); } int AudioFlinger::channelCount(int output) const { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { LOGW("channelCount() unknown thread %d", output); return 0; } return thread->channelCount(); } int AudioFlinger::format(int output) const { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { LOGW("format() unknown thread %d", output); return 0; } return thread->format(); } size_t AudioFlinger::frameCount(int output) const { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { LOGW("frameCount() unknown thread %d", output); return 0; } return thread->frameCount(); } uint32_t AudioFlinger::latency(int output) const { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { LOGW("latency() unknown thread %d", output); return 0; } return thread->latency(); } status_t AudioFlinger::setMasterVolume(float value) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } // when hw supports master volume, don't scale in sw mixer AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; if (mAudioHardware->setMasterVolume(value) == NO_ERROR) { value = 1.0f; } mHardwareStatus = AUDIO_HW_IDLE; mA2DPHandle = -1; mMasterVolume = value; for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) mPlaybackThreads.valueAt(i)->setMasterVolume(value); return NO_ERROR; } status_t AudioFlinger::setMode(int mode) { status_t ret; LOGD("setMode(%d)", mode); // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) { LOGW("Illegal value: setMode(%d)", mode); return BAD_VALUE; } { // scope for the lock AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_SET_MODE; ret = mAudioHardware->setMode(mode); mHardwareStatus = AUDIO_HW_IDLE; } if (NO_ERROR == ret) { Mutex::Autolock _l(mLock); mMode = mode; for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) mPlaybackThreads.valueAt(i)->setMode(mode); #ifdef LVMX LifeVibes::setMode(mode); #endif } return ret; } status_t AudioFlinger::setMicMute(bool state) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; status_t ret = mAudioHardware->setMicMute(state); mHardwareStatus = AUDIO_HW_IDLE; return ret; } bool AudioFlinger::getMicMute() const { bool state = AudioSystem::MODE_INVALID; mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; mAudioHardware->getMicMute(&state); mHardwareStatus = AUDIO_HW_IDLE; return state; } status_t AudioFlinger::setMasterMute(bool muted) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } mMasterMute = muted; for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) mPlaybackThreads.valueAt(i)->setMasterMute(muted); return NO_ERROR; } float AudioFlinger::masterVolume() const { return mMasterVolume; } bool AudioFlinger::masterMute() const { return mMasterMute; } status_t AudioFlinger::setStreamVolume(int stream, float value, int output) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { return BAD_VALUE; } AutoMutex lock(mLock); if( (mLPAOutput != NULL) && (mLPAStreamType == stream) ) { mLPAOutput->setVolume(value, value); } PlaybackThread *thread = NULL; if (output) { thread = checkPlaybackThread_l(output); if (thread == NULL) { return BAD_VALUE; } } mStreamTypes[stream].volume = value; if (thread == NULL) { for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); } } else { thread->setStreamVolume(stream, value); } return NO_ERROR; } status_t AudioFlinger::setStreamMute(int stream, bool muted) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES || uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) { return BAD_VALUE; } mStreamTypes[stream].mute = muted; for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); return NO_ERROR; } float AudioFlinger::streamVolume(int stream, int output) const { if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { return 0.0f; } AutoMutex lock(mLock); float volume; if (output) { PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { return 0.0f; } volume = thread->streamVolume(stream); } else { volume = mStreamTypes[stream].volume; } return volume; } bool AudioFlinger::streamMute(int stream) const { if (stream < 0 || stream >= (int)AudioSystem::NUM_STREAM_TYPES) { return true; } return mStreamTypes[stream].mute; } bool AudioFlinger::isStreamActive(int stream) const { Mutex::Autolock _l(mLock); for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->isStreamActive(stream)) { return true; } } if (mLPAStreamIsActive && mLPAOutput && mLPAStreamType == stream) { return true; } if (stream == AudioSystem::FM) { String8 key ("Fm-radio"); AudioParameter result(mAudioHardware->getParameters(key)); int value; if(result.getInt(String8("isFMON"),value) == NO_ERROR){ return true; } } return false; } status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) { status_t result; LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } #ifdef LVMX AudioParameter param = AudioParameter(keyValuePairs); LifeVibes::setParameters(ioHandle,keyValuePairs); String8 key = String8(AudioParameter::keyRouting); int device; if (NO_ERROR != param.getInt(key, device)) { device = -1; } key = String8(LifevibesTag); String8 value; int musicEnabled = -1; if (NO_ERROR == param.get(key, value)) { if (value == LifevibesEnable) { mLifeVibesClientPid = IPCThreadState::self()->getCallingPid(); musicEnabled = 1; } else if (value == LifevibesDisable) { mLifeVibesClientPid = -1; musicEnabled = 0; } } #endif AudioParameter param = AudioParameter(keyValuePairs); String8 key = String8(AudioParameter::keyRouting); int device; if (param.getInt(key, device) == NO_ERROR) { #ifdef SRS_PROCESSING if (mLPAOutput && mLPAStreamIsActive) { LOGV("setParameters:: routing change to device %d", device); SRS_Processing::ProcessOutRoute(SRS_Processing::AUTO, this, device); if(ioHandle > 0) { audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL); } } #endif if((device & AudioSystem::DEVICE_OUT_FM) && mFmOn == false){ mFmOn=true; } else if (mFmOn == true && !(device & AudioSystem::DEVICE_OUT_FM)){ mFmOn=false; } } // ioHandle == 0 means the parameters are global to the audio hardware interface if (ioHandle == 0) { AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_SET_PARAMETER; #ifdef SRS_PROCESSING bool status = SRS_Processing::ParamsSet(SRS_Processing::AUTO, keyValuePairs); if(status && mLPAOutput && mLPAStreamIsActive) { LOGV("setParameters:: Notifying EFFECT_CONFIG_CHANGED"); audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL); } #endif result = mAudioHardware->setParameters(keyValuePairs); #ifdef LVMX if (musicEnabled != -1) { LifeVibes::enableMusic((bool) musicEnabled); } #endif mHardwareStatus = AUDIO_HW_IDLE; return result; } // Ensure that the routing to LPA is invoked only when the LPA stream is // active. Otherwise if there is a input routing request and if there is a // Valid LPA handle, routing gets applied for the output descriptor rather // than to the input descriptor. if ( mLPAOutput && mLPAStreamIsActive && mLPAHandle == ioHandle ) { if(mLPAEffectChain != NULL) { AudioParameter param = AudioParameter(keyValuePairs); String8 key = String8(AudioParameter::keyRouting); int device; if (param.getInt(key, device) == NO_ERROR) { // forward device change to effects that have requested to be // aware of attached audio device. LOGD("mLPAEffectChain->setDevice_l(device)"); mLPAEffectChain->setDevice_l(device); } } result = mLPAOutput->setParameters(keyValuePairs); return result; } // hold a strong ref on thread in case closeOutput() or closeInput() is called // and the thread is exited once the lock is released sp thread; { Mutex::Autolock _l(mLock); thread = checkPlaybackThread_l(ioHandle); if (thread == NULL) { thread = checkRecordThread_l(ioHandle); } } if (thread != NULL) { result = thread->setParameters(keyValuePairs); #ifdef LVMX if ((NO_ERROR == result) && (device != -1)) { LifeVibes::setDevice(LifeVibes::threadIdToAudioOutputType(thread->id()), device); } #endif return result; } return BAD_VALUE; } String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) { // LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", // ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); if (ioHandle == 0) { #ifdef SRS_PROCESSING String8 srs_params = SRS_Processing::ParamsGet(SRS_Processing::AUTO, keys); if (srs_params != "") srs_params += ";"; srs_params += mAudioHardware->getParameters(keys); return srs_params; #else return mAudioHardware->getParameters(keys); #endif } Mutex::Autolock _l(mLock); PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); if (playbackThread != NULL) { return playbackThread->getParameters(keys); } RecordThread *recordThread = checkRecordThread_l(ioHandle); if (recordThread != NULL) { return recordThread->getParameters(keys); } return String8(""); } size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) { return mAudioHardware->getInputBufferSize(sampleRate, format, channelCount); } unsigned int AudioFlinger::getInputFramesLost(int ioHandle) { if (ioHandle == 0) { return 0; } Mutex::Autolock _l(mLock); RecordThread *recordThread = checkRecordThread_l(ioHandle); if (recordThread != NULL) { return recordThread->getInputFramesLost(); } return 0; } status_t AudioFlinger::setVoiceVolume(float value) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_SET_VOICE_VOLUME; status_t ret = mAudioHardware->setVoiceVolume(value); mHardwareStatus = AUDIO_HW_IDLE; return ret; } status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) { status_t status; Mutex::Autolock _l(mLock); PlaybackThread *playbackThread = checkPlaybackThread_l(output); if (playbackThread != NULL) { return playbackThread->getRenderPosition(halFrames, dspFrames); } return BAD_VALUE; } status_t AudioFlinger::setFmVolume(float value) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_SET_FM_VOLUME; status_t ret = mAudioHardware->setFmVolume(value); mHardwareStatus = AUDIO_HW_IDLE; return ret; } void AudioFlinger::registerClient(const sp& client) { Mutex::Autolock _l(mLock); sp binder = client->asBinder(); if (mNotificationClients.indexOfKey(binder) < 0) { sp notificationClient = new NotificationClient(this, client, binder); LOGV("registerClient() client %p, binder %p", notificationClient.get(), binder.get()); mNotificationClients.add(binder, notificationClient); binder->linkToDeath(notificationClient); // the config change is always sent from playback or record threads to avoid deadlock // with AudioSystem::gLock for (size_t i = 0; i < mPlaybackThreads.size(); i++) { mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); } for (size_t i = 0; i < mRecordThreads.size(); i++) { mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); } } // Send the notification to the client only once. if (mA2DPHandle != -1) { LOGV("A2DP active. Notifying the registered client"); client->ioConfigChanged(AudioSystem::A2DP_OUTPUT_STATE, mA2DPHandle, NULL); } } status_t AudioFlinger::deregisterClient(const sp& client) { LOGV("deregisterClient() %p, tid %d, calling tid %d", client.get(), gettid(), IPCThreadState::self()->getCallingPid()); Mutex::Autolock _l(mLock); sp binder = client->asBinder(); int index = mNotificationClients.indexOfKey(binder); if (index >= 0) { mNotificationClients.removeItemsAt(index); return true; } return false; } void AudioFlinger::removeNotificationClient(sp binder) { Mutex::Autolock _l(mLock); int index = mNotificationClients.indexOfKey(binder); if (index >= 0) { sp client = mNotificationClients.valueFor(binder); LOGV("removeNotificationClient() %p, binder %p", client.get(), binder.get()); #ifdef LVMX if (pid == mLifeVibesClientPid) { LOGV("Disabling lifevibes"); LifeVibes::enableMusic(false); mLifeVibesClientPid = -1; } #endif mNotificationClients.removeItem(binder); } } // audioConfigChanged_l() must be called with AudioFlinger::mLock held void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) { LOGV("AudioFlinger::audioConfigChanged_l: event %d", event); if (event == AudioSystem::EFFECT_CONFIG_CHANGED) { mIsEffectConfigChanged = true; } size_t size = mNotificationClients.size(); for (size_t i = 0; i < size; i++) { mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); } } // removeClient_l() must be called with AudioFlinger::mLock held void AudioFlinger::removeClient_l(pid_t pid) { LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); mClients.removeItem(pid); } // ---------------------------------------------------------------------------- AudioFlinger::ThreadBase::ThreadBase(const sp& audioFlinger, int id) : Thread(false), mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false) { } AudioFlinger::ThreadBase::~ThreadBase() { mParamCond.broadcast(); mNewParameters.clear(); } void AudioFlinger::ThreadBase::exit() { // keep a strong ref on ourself so that we wont get // destroyed in the middle of requestExitAndWait() sp strongMe = this; LOGV("ThreadBase::exit"); { AutoMutex lock(&mLock); mExiting = true; requestExit(); mWaitWorkCV.signal(); } requestExitAndWait(); } uint32_t AudioFlinger::ThreadBase::sampleRate() const { return mSampleRate; } int AudioFlinger::ThreadBase::channelCount() const { return (int)mChannelCount; } int AudioFlinger::ThreadBase::format() const { return mFormat; } size_t AudioFlinger::ThreadBase::frameCount() const { return mFrameCount; } status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) { status_t status; LOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); Mutex::Autolock _l(mLock); mNewParameters.add(keyValuePairs); mWaitWorkCV.signal(); // wait condition with timeout in case the thread loop has exited // before the request could be processed if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) { status = mParamStatus; mWaitWorkCV.signal(); } else { status = TIMED_OUT; } return status; } void AudioFlinger::ThreadBase::effectConfigChanged() { mAudioFlinger->mLock.lock(); LOGV("New effect is being added to LPA chain, Notifying LPA Player"); mAudioFlinger->audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL); mAudioFlinger->mLock.unlock(); } void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) { Mutex::Autolock _l(mLock); sendConfigEvent_l(event, param); } // sendConfigEvent_l() must be called with ThreadBase::mLock held void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) { ConfigEvent *configEvent = new ConfigEvent(); configEvent->mEvent = event; configEvent->mParam = param; mConfigEvents.add(configEvent); LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); mWaitWorkCV.signal(); } void AudioFlinger::ThreadBase::processConfigEvents() { mLock.lock(); while(!mConfigEvents.isEmpty()) { LOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); ConfigEvent *configEvent = mConfigEvents[0]; mConfigEvents.removeAt(0); // release mLock before locking AudioFlinger mLock: lock order is always // AudioFlinger then ThreadBase to avoid cross deadlock mLock.unlock(); mAudioFlinger->mLock.lock(); audioConfigChanged_l(configEvent->mEvent, configEvent->mParam); mAudioFlinger->mLock.unlock(); delete configEvent; mLock.lock(); } mLock.unlock(); } status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; bool locked = tryLock(mLock); if (!locked) { snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); write(fd, buffer, strlen(buffer)); } snprintf(buffer, SIZE, "standby: %d\n", mStandby); result.append(buffer); snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); result.append(buffer); snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); result.append(buffer); snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); result.append(buffer); snprintf(buffer, SIZE, "Format: %d\n", mFormat); result.append(buffer); snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); result.append(buffer); snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); result.append(buffer); result.append(" Index Command"); for (size_t i = 0; i < mNewParameters.size(); ++i) { snprintf(buffer, SIZE, "\n %02d ", i); result.append(buffer); result.append(mNewParameters[i]); } snprintf(buffer, SIZE, "\n\nPending config events: \n"); result.append(buffer); snprintf(buffer, SIZE, " Index event param\n"); result.append(buffer); for (size_t i = 0; i < mConfigEvents.size(); i++) { snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam); result.append(buffer); } result.append("\n"); write(fd, result.string(), result.size()); if (locked) { mLock.unlock(); } return NO_ERROR; } // ---------------------------------------------------------------------------- AudioFlinger::PlaybackThread::PlaybackThread(const sp& audioFlinger, AudioStreamOut* output, int id, uint32_t device) : ThreadBase(audioFlinger, id), mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output), mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), mDevice(device) { readOutputParameters(); mMasterVolume = mAudioFlinger->masterVolume(); mMasterMute = mAudioFlinger->masterMute(); for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); } } AudioFlinger::PlaybackThread::~PlaybackThread() { delete [] mMixBuffer; } status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector& args) { dumpInternals(fd, args); dumpTracks(fd, args); dumpEffectChains(fd, args); return NO_ERROR; } status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "Output thread %p tracks\n", this); result.append(buffer); result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); for (size_t i = 0; i < mTracks.size(); ++i) { sp track = mTracks[i]; if (track != 0) { track->dump(buffer, SIZE); result.append(buffer); } } snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); result.append(buffer); result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); for (size_t i = 0; i < mActiveTracks.size(); ++i) { wp wTrack = mActiveTracks[i]; if (wTrack != 0) { sp track = wTrack.promote(); if (track != 0) { track->dump(buffer, SIZE); result.append(buffer); } } } write(fd, result.string(), result.size()); return NO_ERROR; } status_t AudioFlinger::PlaybackThread::dumpEffectChains(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); write(fd, buffer, strlen(buffer)); for (size_t i = 0; i < mEffectChains.size(); ++i) { sp chain = mEffectChains[i]; if (chain != 0) { chain->dump(fd, args); } } return NO_ERROR; } status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); result.append(buffer); 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); snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); result.append(buffer); snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); result.append(buffer); snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); result.append(buffer); snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); result.append(buffer); write(fd, result.string(), result.size()); dumpBase(fd, args); return NO_ERROR; } // Thread virtuals status_t AudioFlinger::PlaybackThread::readyToRun() { if (mSampleRate == 0) { LOGE("No working audio driver found."); return NO_INIT; } LOGI("AudioFlinger's thread %p ready to run", this); return NO_ERROR; } void AudioFlinger::PlaybackThread::onFirstRef() { const size_t SIZE = 256; char buffer[SIZE]; snprintf(buffer, SIZE, "Playback Thread %p", this); run(buffer, ANDROID_PRIORITY_URGENT_AUDIO); } // PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held sp AudioFlinger::PlaybackThread::createTrack_l( const sp& client, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, const sp& sharedBuffer, int sessionId, status_t *status) { sp track; status_t lStatus; LOGV("PlaybackThread::createTrack_l() sessionId %d mType %d", sessionId, mType); if (mType == DIRECT) { if (sampleRate != mSampleRate || format != mFormat || channelCount != (int)mChannelCount) { LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p", sampleRate, format, channelCount, mOutput); lStatus = BAD_VALUE; goto Exit; } } else { // Resampler implementation limits input sampling rate to 2 x output sampling rate. if (sampleRate > mSampleRate*2) { LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); lStatus = BAD_VALUE; goto Exit; } } if (mOutput == 0) { LOGE("Audio driver not initialized."); lStatus = NO_INIT; goto Exit; } { // scope for mLock Mutex::Autolock _l(mLock); // all tracks in same audio session must share the same routing strategy otherwise // conflicts will happen when tracks are moved from one output to another by audio policy // manager uint32_t strategy = AudioSystem::getStrategyForStream((AudioSystem::stream_type)streamType); for (size_t i = 0; i < mTracks.size(); ++i) { sp t = mTracks[i]; if (t != 0) { if (sessionId == t->sessionId() && strategy != AudioSystem::getStrategyForStream((AudioSystem::stream_type)t->type())) { lStatus = BAD_VALUE; goto Exit; } } } track = new Track(this, client, streamType, sampleRate, format, channelCount, frameCount, sharedBuffer, sessionId); if (track->getCblk() == NULL || track->name() < 0) { lStatus = NO_MEMORY; goto Exit; } mTracks.add(track); sp chain = getEffectChain_l(sessionId); if (chain != 0) { LOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); track->setMainBuffer(chain->inBuffer()); chain->setStrategy(AudioSystem::getStrategyForStream((AudioSystem::stream_type)track->type())); } } lStatus = NO_ERROR; Exit: if(status) { *status = lStatus; } return track; } uint32_t AudioFlinger::PlaybackThread::latency() const { if (mOutput) { return mOutput->latency(); } else { return 0; } } status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) { #ifdef LVMX int audioOutputType = LifeVibes::getMixerType(mId, mType); if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { LifeVibes::setMasterVolume(audioOutputType, value); } #endif mMasterVolume = value; return NO_ERROR; } status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) { #ifdef LVMX int audioOutputType = LifeVibes::getMixerType(mId, mType); if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { LifeVibes::setMasterMute(audioOutputType, muted); } #endif mMasterMute = muted; return NO_ERROR; } float AudioFlinger::PlaybackThread::masterVolume() const { return mMasterVolume; } bool AudioFlinger::PlaybackThread::masterMute() const { return mMasterMute; } status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) { #ifdef LVMX int audioOutputType = LifeVibes::getMixerType(mId, mType); if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { LifeVibes::setStreamVolume(audioOutputType, stream, value); } #endif mStreamTypes[stream].volume = value; return NO_ERROR; } status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) { #ifdef LVMX int audioOutputType = LifeVibes::getMixerType(mId, mType); if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { LifeVibes::setStreamMute(audioOutputType, stream, muted); } #endif mStreamTypes[stream].mute = muted; return NO_ERROR; } float AudioFlinger::PlaybackThread::streamVolume(int stream) const { return mStreamTypes[stream].volume; } bool AudioFlinger::PlaybackThread::streamMute(int stream) const { return mStreamTypes[stream].mute; } bool AudioFlinger::PlaybackThread::isStreamActive(int stream) const { Mutex::Autolock _l(mLock); size_t count = mActiveTracks.size(); for (size_t i = 0 ; i < count ; ++i) { sp t = mActiveTracks[i].promote(); if (t == 0) continue; Track* const track = t.get(); if (t->type() == stream) return true; } return false; } // addTrack_l() must be called with ThreadBase::mLock held status_t AudioFlinger::PlaybackThread::addTrack_l(const sp& track) { status_t status = ALREADY_EXISTS; // set retry count for buffer fill track->mRetryCount = kMaxTrackStartupRetries; if (mActiveTracks.indexOf(track) < 0) { // the track is newly added, make sure it fills up all its // buffers before playing. This is to ensure the client will // effectively get the latency it requested. track->mFillingUpStatus = Track::FS_FILLING; track->mResetDone = false; mActiveTracks.add(track); if (track->mainBuffer() != mMixBuffer) { sp chain = getEffectChain_l(track->sessionId()); if (chain != 0) { LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); chain->startTrack(); } } status = NO_ERROR; } LOGV("mWaitWorkCV.broadcast"); mWaitWorkCV.broadcast(); return status; } // destroyTrack_l() must be called with ThreadBase::mLock held void AudioFlinger::PlaybackThread::destroyTrack_l(const sp& track) { track->mState = TrackBase::TERMINATED; if (mActiveTracks.indexOf(track) < 0) { mTracks.remove(track); deleteTrackName_l(track->name()); } } String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) { return mOutput->getParameters(keys); } // destroyTrack_l() must be called with AudioFlinger::mLock held void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { AudioSystem::OutputDescriptor desc; void *param2 = 0; LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); switch (event) { case AudioSystem::OUTPUT_OPENED: case AudioSystem::OUTPUT_CONFIG_CHANGED: desc.channels = mChannels; desc.samplingRate = mSampleRate; desc.format = mFormat; desc.frameCount = mFrameCount; desc.latency = latency(); param2 = &desc; break; case AudioSystem::STREAM_CONFIG_CHANGED: param2 = ¶m; case AudioSystem::OUTPUT_CLOSED: default: break; } if (event != AudioSystem::A2DP_OUTPUT_STATE) { mAudioFlinger->audioConfigChanged_l(event, mId, param2); } else { mAudioFlinger->audioConfigChanged_l(event, param, NULL); } } void AudioFlinger::PlaybackThread::readOutputParameters() { mSampleRate = mOutput->sampleRate(); mChannels = mOutput->channels(); mChannelCount = (uint16_t)AudioSystem::popCount(mChannels); mFormat = mOutput->format(); mFrameSize = (uint16_t)mOutput->frameSize(); mFrameCount = mOutput->bufferSize() / mFrameSize; // FIXME - Current mixer implementation only supports stereo output: Always // Allocate a stereo buffer even if HW output is mono. if (mMixBuffer != NULL) delete[] mMixBuffer; mMixBuffer = new int16_t[mFrameCount * 2]; memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); // force reconfiguration of effect chains and engines to take new buffer size and audio // parameters into account // Note that mLock is not held when readOutputParameters() is called from the constructor // but in this case nothing is done below as no audio sessions have effect yet so it doesn't // matter. // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains Vector< sp > effectChains = mEffectChains; for (size_t i = 0; i < effectChains.size(); i ++) { mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); } } status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) { if (halFrames == 0 || dspFrames == 0) { return BAD_VALUE; } if (mOutput == 0) { return INVALID_OPERATION; } *halFrames = mBytesWritten/mOutput->frameSize(); return mOutput->getRenderPosition(dspFrames); } uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) { Mutex::Autolock _l(mLock); uint32_t result = 0; if (getEffectChain_l(sessionId) != 0) { result = EFFECT_SESSION; } for (size_t i = 0; i < mTracks.size(); ++i) { sp track = mTracks[i]; if (sessionId == track->sessionId() && !(track->mCblk->flags & CBLK_INVALID_MSK)) { result |= TRACK_SESSION; break; } } return result; } uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) { // session AudioSystem::SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that // it is moved to correct output by audio policy manager when A2DP is connected or disconnected if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) { return AudioSystem::getStrategyForStream(AudioSystem::MUSIC); } for (size_t i = 0; i < mTracks.size(); i++) { sp track = mTracks[i]; if (sessionId == track->sessionId() && !(track->mCblk->flags & CBLK_INVALID_MSK)) { return AudioSystem::getStrategyForStream((AudioSystem::stream_type) track->type()); } } return AudioSystem::getStrategyForStream(AudioSystem::MUSIC); } sp AudioFlinger::PlaybackThread::getEffectChain(int sessionId) { Mutex::Autolock _l(mLock); return getEffectChain_l(sessionId); } sp AudioFlinger::PlaybackThread::getEffectChain_l(int sessionId) { sp chain; size_t size = mEffectChains.size(); for (size_t i = 0; i < size; i++) { if (mEffectChains[i]->sessionId() == sessionId) { chain = mEffectChains[i]; break; } } return chain; } void AudioFlinger::PlaybackThread::setMode(uint32_t mode) { Mutex::Autolock _l(mLock); size_t size = mEffectChains.size(); for (size_t i = 0; i < size; i++) { mEffectChains[i]->setMode_l(mode); } } // ---------------------------------------------------------------------------- AudioFlinger::MixerThread::MixerThread(const sp& audioFlinger, AudioStreamOut* output, int id, uint32_t device) : PlaybackThread(audioFlinger, output, id, device), mAudioMixer(0) { mType = PlaybackThread::MIXER; mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); // FIXME - Current mixer implementation only supports stereo output if (mChannelCount == 1) { LOGE("Invalid audio hardware channel count"); } } AudioFlinger::MixerThread::~MixerThread() { delete mAudioMixer; } bool AudioFlinger::MixerThread::threadLoop() { Vector< sp > tracksToRemove; uint32_t mixerStatus = MIXER_IDLE; nsecs_t standbyTime = systemTime(); size_t mixBufferSize = mFrameCount * mFrameSize; // FIXME: Relaxed timing because of a certain device that can't meet latency // Should be reduced to 2x after the vendor fixes the driver issue nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3; nsecs_t lastWarning = 0; bool longStandbyExit = false; uint32_t activeSleepTime = activeSleepTimeUs(); uint32_t idleSleepTime = idleSleepTimeUs(); uint32_t sleepTime = idleSleepTime; Vector< sp > effectChains; #ifdef SRS_PROCESSING LOGD("SRS_Processing - MixerThread - OutNotify_Init: %p TID %d\n", this, gettid()); SRS_Processing::ProcessOutNotify(SRS_Processing::AUTO, this, true); #endif while (!exitPending()) { processConfigEvents(); mixerStatus = MIXER_IDLE; { // scope for mLock Mutex::Autolock _l(mLock); if (checkForNewParameters_l()) { mixBufferSize = mFrameCount * mFrameSize; // FIXME: Relaxed timing because of a certain device that can't meet latency // Should be reduced to 2x after the vendor fixes the driver issue maxPeriod = seconds(mFrameCount) / mSampleRate * 3; activeSleepTime = activeSleepTimeUs(); idleSleepTime = idleSleepTimeUs(); } const SortedVector< wp >& activeTracks = mActiveTracks; // put audio hardware into standby after short delay if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || mSuspended) { if (!mStandby) { LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); mOutput->standby(); mStandby = true; mBytesWritten = 0; } if (!activeTracks.size() && mConfigEvents.isEmpty()) { // we're about to wait, flush the binder command buffer IPCThreadState::self()->flushCommands(); if (exitPending()) break; // wait until we have something to do... LOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); mWaitWorkCV.wait(mLock); LOGV("MixerThread %p TID %d waking up\n", this, gettid()); if (mMasterMute == false) { char value[PROPERTY_VALUE_MAX]; property_get("ro.audio.silent", value, "0"); if (atoi(value)) { LOGD("Silence is golden"); setMasterMute(true); } } standbyTime = systemTime() + kStandbyTimeInNsecs; sleepTime = idleSleepTime; continue; } } mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); LOGV("MixerThread:: calling lockEffectChains_l()"); // prevent any changes in effect chain list and in each effect chain // during mixing and effect process as the audio buffers could be deleted // or modified if an effect is created or deleted lockEffectChains_l(effectChains); } if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { LOGV("MixerThread:: calling mAudioMixer->process()"); // mix buffers... mAudioMixer->process(); sleepTime = 0; standbyTime = systemTime() + kStandbyTimeInNsecs; //TODO: delay standby when effects have a tail } else { // If no tracks are ready, sleep once for the duration of an output // buffer size, then write 0s to the output if (sleepTime == 0) { if (mixerStatus == MIXER_TRACKS_ENABLED) { sleepTime = activeSleepTime; } else { sleepTime = idleSleepTime; } } else if (mBytesWritten != 0 || (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { memset (mMixBuffer, 0, mixBufferSize); sleepTime = 0; LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); } // TODO add standby time extension fct of effect tail } if (mSuspended) { sleepTime = suspendSleepTimeUs(); } // sleepTime == 0 means we must write to audio hardware if (sleepTime == 0) { for (size_t i = 0; i < effectChains.size(); i ++) { LOGV("MixerThread:: calling effectChains[%d]->process_l", i); effectChains[i]->process_l(); } LOGV("MixerThread:: calling unlockEffectChains()"); // enable changes in effect chain unlockEffectChains(effectChains); #ifdef SRS_PROCESSING if (mFormat == AudioSystem::PCM_16_BIT) { SRS_Processing::ProcessOut(SRS_Processing::AUTO, this, mMixBuffer, mixBufferSize, mSampleRate, mChannelCount); } #endif #ifdef LVMX LOGV("MixerThread:: calling LifeVibes::getMixerType()"); int audioOutputType = LifeVibes::getMixerType(mId, mType); if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { LOGV("MixerThread:: calling LifeVibes::process()"); LifeVibes::process(audioOutputType, mMixBuffer, mixBufferSize); } #endif mLastWriteTime = systemTime(); mInWrite = true; mBytesWritten += mixBufferSize; int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); if (bytesWritten < 0) mBytesWritten -= mixBufferSize; mNumWrites++; mInWrite = false; nsecs_t now = systemTime(); nsecs_t delta = now - mLastWriteTime; if (delta > maxPeriod) { mNumDelayedWrites++; if ((now - lastWarning) > kWarningThrottle) { LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", ns2ms(delta), mNumDelayedWrites, this); lastWarning = now; } if (mStandby) { longStandbyExit = true; } } mStandby = false; } else { // enable changes in effect chain unlockEffectChains(effectChains); usleep(sleepTime); } // finally let go of all our tracks, without the lock held // since we can't guarantee the destructors won't acquire that // same lock. tracksToRemove.clear(); // Effect chains will be actually deleted here if they were removed from // mEffectChains list during mixing or effects processing effectChains.clear(); } if (!mStandby) { mOutput->standby(); } #ifdef SRS_PROCESSING LOGD("SRS_Processing - MixerThread - OutNotify_Exit: %p TID %d\n", this, gettid()); SRS_Processing::ProcessOutNotify(SRS_Processing::AUTO, this, false); #endif LOGV("MixerThread %p exiting", this); return false; } // prepareTracks_l() must be called with ThreadBase::mLock held uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp >& activeTracks, Vector< sp > *tracksToRemove) { uint32_t mixerStatus = MIXER_IDLE; // find out which tracks need to be processed size_t count = activeTracks.size(); size_t mixedTracks = 0; size_t tracksWithEffect = 0; float masterVolume = mMasterVolume; bool masterMute = mMasterMute; if (masterMute) { masterVolume = 0; } #ifdef LVMX bool tracksConnectedChanged = false; bool stateChanged = false; int audioOutputType = LifeVibes::getMixerType(mId, mType); if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { int activeTypes = 0; for (size_t i=0 ; i t = activeTracks[i].promote(); if (t == 0) continue; Track* const track = t.get(); int iTracktype=track->type(); activeTypes |= 1<type(); } LifeVibes::computeVolumes(audioOutputType, activeTypes, tracksConnectedChanged, stateChanged, masterVolume, masterMute); } #endif // Delegate master volume control to effect in output mix effect chain if needed sp chain = getEffectChain_l(AudioSystem::SESSION_OUTPUT_MIX); if (chain != 0) { uint32_t v = (uint32_t)(masterVolume * (1 << 24)); chain->setVolume_l(&v, &v); masterVolume = (float)((v + (1 << 23)) >> 24); chain.clear(); } for (size_t i=0 ; i t = activeTracks[i].promote(); if (t == 0) continue; Track* const track = t.get(); audio_track_cblk_t* cblk = track->cblk(); // The first time a track is added we wait // for all its buffers to be filled before processing it mAudioMixer->setActiveTrack(track->name()); if (cblk->framesReady() && track->isReady() && !track->isPaused() && !track->isTerminated()) { //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); mixedTracks++; // track->mainBuffer() != mMixBuffer means there is an effect chain // connected to the track chain.clear(); if (track->mainBuffer() != mMixBuffer) { chain = getEffectChain_l(track->sessionId()); // Delegate volume control to effect in track effect chain if needed if (chain != 0) { tracksWithEffect++; } else { LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", track->name(), track->sessionId()); } } int param = AudioMixer::VOLUME; if (track->mFillingUpStatus == Track::FS_FILLED) { // no ramp for the first volume setting track->mFillingUpStatus = Track::FS_ACTIVE; if (track->mState == TrackBase::RESUMING) { track->mState = TrackBase::ACTIVE; param = AudioMixer::RAMP_VOLUME; } } else if (cblk->server != 0) { // If the track is stopped before the first frame was mixed, // do not apply ramp param = AudioMixer::RAMP_VOLUME; } // compute volume for this track uint32_t vl, vr, va; if (track->isMuted() || track->isPausing() || mStreamTypes[track->type()].mute) { vl = vr = va = 0; if (track->isPausing()) { track->setPaused(); } } else { // read original volumes with volume control float typeVolume = mStreamTypes[track->type()].volume; #ifdef LVMX bool streamMute=false; // read the volume from the LivesVibes audio engine. if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { LifeVibes::getStreamVolumes(audioOutputType, track->type(), &typeVolume, &streamMute); if (streamMute) { typeVolume = 0; } } #endif float v = masterVolume * typeVolume; vl = (uint32_t)(v * cblk->volume[0]) << 12; vr = (uint32_t)(v * cblk->volume[1]) << 12; va = (uint32_t)(v * cblk->sendLevel); } // Delegate volume control to effect in track effect chain if needed if (chain != 0 && chain->setVolume_l(&vl, &vr)) { // Do not ramp volume if volume is controlled by effect param = AudioMixer::VOLUME; track->mHasVolumeController = true; } else { // force no volume ramp when volume controller was just disabled or removed // from effect chain to avoid volume spike if (track->mHasVolumeController) { param = AudioMixer::VOLUME; } track->mHasVolumeController = false; } // Convert volumes from 8.24 to 4.12 format int16_t left, right, aux; uint32_t v_clamped = (vl + (1 << 11)) >> 12; if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; left = int16_t(v_clamped); v_clamped = (vr + (1 << 11)) >> 12; if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; right = int16_t(v_clamped); if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; aux = int16_t(va); #ifdef LVMX if ( tracksConnectedChanged || stateChanged ) { // only do the ramp when the volume is changed by the user / application param = AudioMixer::VOLUME; } #endif // XXX: these things DON'T need to be done each time mAudioMixer->setBufferProvider(track); mAudioMixer->enable(AudioMixer::MIXING); mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); mAudioMixer->setParameter( AudioMixer::TRACK, AudioMixer::FORMAT, (void *)track->format()); mAudioMixer->setParameter( AudioMixer::TRACK, AudioMixer::CHANNEL_COUNT, (void *)track->channelCount()); mAudioMixer->setParameter( AudioMixer::RESAMPLE, AudioMixer::SAMPLE_RATE, (void *)(cblk->sampleRate)); mAudioMixer->setParameter( AudioMixer::TRACK, AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); mAudioMixer->setParameter( AudioMixer::TRACK, AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); // reset retry count track->mRetryCount = kMaxTrackRetries; mixerStatus = MIXER_TRACKS_READY; } else { //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); if (track->isStopped()) { track->reset(); } if (track->isTerminated() || track->isStopped() || track->isPaused()) { // We have consumed all the buffers of this track. // Remove it from the list of active tracks. tracksToRemove->add(track); } else { // No buffers for this track. Give it a few chances to // fill a buffer, then remove it from active list. if (--(track->mRetryCount) <= 0) { LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); tracksToRemove->add(track); // indicate to client process that the track was disabled because of underrun cblk->flags |= CBLK_DISABLED_ON; } else if (mixerStatus != MIXER_TRACKS_READY) { mixerStatus = MIXER_TRACKS_ENABLED; } } mAudioMixer->disable(AudioMixer::MIXING); } } // remove all the tracks that need to be... count = tracksToRemove->size(); if (UNLIKELY(count)) { for (size_t i=0 ; i& track = tracksToRemove->itemAt(i); mActiveTracks.remove(track); if (track->mainBuffer() != mMixBuffer) { chain = getEffectChain_l(track->sessionId()); if (chain != 0) { LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); chain->stopTrack(); } } if (track->isTerminated()) { mTracks.remove(track); deleteTrackName_l(track->mName); } } } // 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) { memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); } return mixerStatus; } void AudioFlinger::MixerThread::invalidateTracks(int streamType) { LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", this, streamType, mTracks.size()); Mutex::Autolock _l(mLock); size_t size = mTracks.size(); for (size_t i = 0; i < size; i++) { sp t = mTracks[i]; if (t->type() == streamType) { t->mCblk->lock.lock(); t->mCblk->flags |= CBLK_INVALID_ON; t->mCblk->cv.signal(); t->mCblk->lock.unlock(); } } } // getTrackName_l() must be called with ThreadBase::mLock held int AudioFlinger::MixerThread::getTrackName_l() { return mAudioMixer->getTrackName(); } // deleteTrackName_l() must be called with ThreadBase::mLock held void AudioFlinger::MixerThread::deleteTrackName_l(int name) { LOGD("remove track (%d) and delete from mixer", name); mAudioMixer->deleteTrackName(name); } // checkForNewParameters_l() must be called with ThreadBase::mLock held bool AudioFlinger::MixerThread::checkForNewParameters_l() { bool reconfig = false; while (!mNewParameters.isEmpty()) { status_t status = NO_ERROR; String8 keyValuePair = mNewParameters[0]; AudioParameter param = AudioParameter(keyValuePair); int value; #ifdef SRS_PROCESSING if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR){ SRS_Processing::ProcessOutRoute(SRS_Processing::AUTO, this, value); } #endif if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { reconfig = true; } if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { if (value != AudioSystem::PCM_16_BIT) { status = BAD_VALUE; } else { reconfig = true; } } if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { if (value != AudioSystem::CHANNEL_OUT_STEREO) { status = BAD_VALUE; } else { reconfig = true; } } if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { // do not accept frame count changes if tracks are open as the track buffer // size depends on frame count and correct behavior would not be garantied // if frame count is changed after track creation if (!mTracks.isEmpty()) { status = INVALID_OPERATION; } else { reconfig = true; } } if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { // forward device change to effects that have requested to be // aware of attached audio device. mDevice = (uint32_t)value; for (size_t i = 0; i < mEffectChains.size(); i++) { mEffectChains[i]->setDevice_l(mDevice); } } if (status == NO_ERROR) { status = mOutput->setParameters(keyValuePair); if (!mStandby && status == INVALID_OPERATION) { mOutput->standby(); mStandby = true; mBytesWritten = 0; status = mOutput->setParameters(keyValuePair); } if (status == NO_ERROR && reconfig) { delete mAudioMixer; readOutputParameters(); mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); for (size_t i = 0; i < mTracks.size() ; i++) { int name = getTrackName_l(); if (name < 0) break; mTracks[i]->mName = name; // limit track sample rate to 2 x new output sample rate if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); } } sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); } } mNewParameters.removeAt(0); mParamStatus = status; mParamCond.signal(); mWaitWorkCV.wait(mLock); } return reconfig; } status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; PlaybackThread::dumpInternals(fd, args); snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); result.append(buffer); write(fd, result.string(), result.size()); return NO_ERROR; } uint32_t AudioFlinger::MixerThread::activeSleepTimeUs() { return (uint32_t)(mOutput->latency() * 1000) / 2; } uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() { return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; } uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() { return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); } // ---------------------------------------------------------------------------- AudioFlinger::DirectOutputThread::DirectOutputThread(const sp& audioFlinger, AudioStreamOut* output, int id, uint32_t device) : PlaybackThread(audioFlinger, output, id, device) { mType = PlaybackThread::DIRECT; } AudioFlinger::DirectOutputThread::~DirectOutputThread() { } static inline int16_t clamp16(int32_t sample) { if ((sample>>15) ^ (sample>>31)) sample = 0x7FFF ^ (sample>>31); return sample; } static inline int32_t mul(int16_t in, int16_t v) { #if defined(__arm__) && !defined(__thumb__) int32_t out; asm( "smulbb %[out], %[in], %[v] \n" : [out]"=r"(out) : [in]"%r"(in), [v]"r"(v) : ); return out; #else return in * int32_t(v); #endif } void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) { // Do not apply volume on compressed audio if (!AudioSystem::isLinearPCM(mFormat)) { return; } // convert to signed 16 bit before volume calculation if (mFormat == AudioSystem::PCM_8_BIT) { size_t count = mFrameCount * mChannelCount; uint8_t *src = (uint8_t *)mMixBuffer + count-1; int16_t *dst = mMixBuffer + count-1; while(count--) { *dst-- = (int16_t)(*src--^0x80) << 8; } } size_t frameCount = mFrameCount; int16_t *out = mMixBuffer; if (ramp) { if (mChannelCount == 1) { int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; int32_t vlInc = d / (int32_t)frameCount; int32_t vl = ((int32_t)mLeftVolShort << 16); do { out[0] = clamp16(mul(out[0], vl >> 16) >> 12); out++; vl += vlInc; } while (--frameCount); } else { int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; int32_t vlInc = d / (int32_t)frameCount; d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; int32_t vrInc = d / (int32_t)frameCount; int32_t vl = ((int32_t)mLeftVolShort << 16); int32_t vr = ((int32_t)mRightVolShort << 16); do { out[0] = clamp16(mul(out[0], vl >> 16) >> 12); out[1] = clamp16(mul(out[1], vr >> 16) >> 12); out += 2; vl += vlInc; vr += vrInc; } while (--frameCount); } } else { if (mChannelCount == 1) { do { out[0] = clamp16(mul(out[0], leftVol) >> 12); out++; } while (--frameCount); } else { do { out[0] = clamp16(mul(out[0], leftVol) >> 12); out[1] = clamp16(mul(out[1], rightVol) >> 12); out += 2; } while (--frameCount); } } // convert back to unsigned 8 bit after volume calculation if (mFormat == AudioSystem::PCM_8_BIT) { size_t count = mFrameCount * mChannelCount; int16_t *src = mMixBuffer; uint8_t *dst = (uint8_t *)mMixBuffer; while(count--) { *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; } } mLeftVolShort = leftVol; mRightVolShort = rightVol; } bool AudioFlinger::DirectOutputThread::threadLoop() { uint32_t mixerStatus = MIXER_IDLE; sp trackToRemove; sp activeTrack; nsecs_t standbyTime = systemTime(); int8_t *curBuf; size_t mixBufferSize = mFrameCount*mFrameSize; uint32_t activeSleepTime = activeSleepTimeUs(); uint32_t idleSleepTime = idleSleepTimeUs(); uint32_t sleepTime = idleSleepTime; // use shorter standby delay as on normal output to release // hardware resources as soon as possible nsecs_t standbyDelay = microseconds(activeSleepTime*2); while (!exitPending()) { bool rampVolume; uint16_t leftVol; uint16_t rightVol; Vector< sp > effectChains; processConfigEvents(); mixerStatus = MIXER_IDLE; { // scope for the mLock Mutex::Autolock _l(mLock); if (checkForNewParameters_l()) { mixBufferSize = mFrameCount*mFrameSize; activeSleepTime = activeSleepTimeUs(); idleSleepTime = idleSleepTimeUs(); standbyDelay = microseconds(activeSleepTime*2); } // put audio hardware into standby after short delay if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || mSuspended) { // wait until we have something to do... if (!mStandby) { LOGV("Audio hardware entering standby, mixer %p\n", this); mOutput->standby(); mStandby = true; mBytesWritten = 0; } if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { // we're about to wait, flush the binder command buffer IPCThreadState::self()->flushCommands(); if (exitPending()) break; LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); mWaitWorkCV.wait(mLock); LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); if (mMasterMute == false) { char value[PROPERTY_VALUE_MAX]; property_get("ro.audio.silent", value, "0"); if (atoi(value)) { LOGD("Silence is golden"); setMasterMute(true); } } standbyTime = systemTime() + standbyDelay; sleepTime = idleSleepTime; continue; } } effectChains = mEffectChains; // find out which tracks need to be processed if (mActiveTracks.size() != 0) { sp t = mActiveTracks[0].promote(); if (t == 0) continue; Track* const track = t.get(); audio_track_cblk_t* cblk = track->cblk(); // The first time a track is added we wait // for all its buffers to be filled before processing it if (cblk->framesReady() && track->isReady() && !track->isPaused() && !track->isTerminated()) { //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); if (track->mFillingUpStatus == Track::FS_FILLED) { track->mFillingUpStatus = Track::FS_ACTIVE; mLeftVolFloat = mRightVolFloat = 0; mLeftVolShort = mRightVolShort = 0; if (track->mState == TrackBase::RESUMING) { track->mState = TrackBase::ACTIVE; rampVolume = true; } } else if (cblk->server != 0) { // If the track is stopped before the first frame was mixed, // do not apply ramp rampVolume = true; } // compute volume for this track float left, right; if (track->isMuted() || mMasterMute || track->isPausing() || mStreamTypes[track->type()].mute) { left = right = 0; if (track->isPausing()) { track->setPaused(); } } else { float typeVolume = mStreamTypes[track->type()].volume; float v = mMasterVolume * typeVolume; float v_clamped = v * cblk->volume[0]; if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; left = v_clamped/MAX_GAIN; v_clamped = v * cblk->volume[1]; if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; right = v_clamped/MAX_GAIN; } if (left != mLeftVolFloat || right != mRightVolFloat) { mLeftVolFloat = left; mRightVolFloat = right; // If audio HAL implements volume control, // force software volume to nominal value if (mOutput->setVolume(left, right) == NO_ERROR) { left = 1.0f; right = 1.0f; } // Convert volumes from float to 8.24 uint32_t vl = (uint32_t)(left * (1 << 24)); uint32_t vr = (uint32_t)(right * (1 << 24)); // Delegate volume control to effect in track effect chain if needed // only one effect chain can be present on DirectOutputThread, so if // there is one, the track is connected to it if (!effectChains.isEmpty()) { // Do not ramp volume if volume is controlled by effect if(effectChains[0]->setVolume_l(&vl, &vr)) { rampVolume = false; } } // Convert volumes from 8.24 to 4.12 format uint32_t v_clamped = (vl + (1 << 11)) >> 12; if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; leftVol = (uint16_t)v_clamped; v_clamped = (vr + (1 << 11)) >> 12; if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; rightVol = (uint16_t)v_clamped; } else { leftVol = mLeftVolShort; rightVol = mRightVolShort; rampVolume = false; } // reset retry count track->mRetryCount = kMaxTrackRetriesDirect; activeTrack = t; mixerStatus = MIXER_TRACKS_READY; } else { //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); if (track->isStopped()) { track->reset(); } if (track->isTerminated() || track->isStopped() || track->isPaused()) { // We have consumed all the buffers of this track. // Remove it from the list of active tracks. trackToRemove = track; } else { // No buffers for this track. Give it a few chances to // fill a buffer, then remove it from active list. if (--(track->mRetryCount) <= 0) { LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); trackToRemove = track; } else { mixerStatus = MIXER_TRACKS_ENABLED; } } } } // remove all the tracks that need to be... if (UNLIKELY(trackToRemove != 0)) { mActiveTracks.remove(trackToRemove); if (!effectChains.isEmpty()) { LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), trackToRemove->sessionId()); effectChains[0]->stopTrack(); } if (trackToRemove->isTerminated()) { mTracks.remove(trackToRemove); deleteTrackName_l(trackToRemove->mName); } } lockEffectChains_l(effectChains); } if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { AudioBufferProvider::Buffer buffer; size_t frameCount = mFrameCount; curBuf = (int8_t *)mMixBuffer; // output audio to hardware while (frameCount) { buffer.frameCount = frameCount; activeTrack->getNextBuffer(&buffer); if (UNLIKELY(buffer.raw == 0)) { memset(curBuf, 0, frameCount * mFrameSize); break; } memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); frameCount -= buffer.frameCount; curBuf += buffer.frameCount * mFrameSize; activeTrack->releaseBuffer(&buffer); } sleepTime = 0; standbyTime = systemTime() + standbyDelay; } else { if (sleepTime == 0) { if (mixerStatus == MIXER_TRACKS_ENABLED) { sleepTime = activeSleepTime; } else { sleepTime = idleSleepTime; } } else if (mBytesWritten != 0 && AudioSystem::isLinearPCM(mFormat)) { memset (mMixBuffer, 0, mFrameCount * mFrameSize); sleepTime = 0; } } if (mSuspended) { sleepTime = suspendSleepTimeUs(); } // sleepTime == 0 means we must write to audio hardware if (sleepTime == 0) { if (mixerStatus == MIXER_TRACKS_READY) { applyVolume(leftVol, rightVol, rampVolume); } for (size_t i = 0; i < effectChains.size(); i ++) { effectChains[i]->process_l(); } unlockEffectChains(effectChains); mLastWriteTime = systemTime(); mInWrite = true; mBytesWritten += mixBufferSize; int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); if (bytesWritten < 0) mBytesWritten -= mixBufferSize; mNumWrites++; mInWrite = false; mStandby = false; } else { unlockEffectChains(effectChains); usleep(sleepTime); } // finally let go of removed track, without the lock held // since we can't guarantee the destructors won't acquire that // same lock. trackToRemove.clear(); activeTrack.clear(); // Effect chains will be actually deleted here if they were removed from // mEffectChains list during mixing or effects processing effectChains.clear(); } if (!mStandby) { mOutput->standby(); } LOGV("DirectOutputThread %p exiting", this); return false; } // getTrackName_l() must be called with ThreadBase::mLock held int AudioFlinger::DirectOutputThread::getTrackName_l() { return 0; } // deleteTrackName_l() must be called with ThreadBase::mLock held void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) { } // checkForNewParameters_l() must be called with ThreadBase::mLock held bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() { bool reconfig = false; while (!mNewParameters.isEmpty()) { status_t status = NO_ERROR; String8 keyValuePair = mNewParameters[0]; AudioParameter param = AudioParameter(keyValuePair); int value; if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { // do not accept frame count changes if tracks are open as the track buffer // size depends on frame count and correct behavior would not be garantied // if frame count is changed after track creation if (!mTracks.isEmpty()) { status = INVALID_OPERATION; } else { reconfig = true; } } if (status == NO_ERROR) { status = mOutput->setParameters(keyValuePair); if (!mStandby && status == INVALID_OPERATION) { mOutput->standby(); mStandby = true; mBytesWritten = 0; status = mOutput->setParameters(keyValuePair); } if (status == NO_ERROR && reconfig) { readOutputParameters(); sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); } } mNewParameters.removeAt(0); mParamStatus = status; mParamCond.signal(); mWaitWorkCV.wait(mLock); } return reconfig; } uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() { uint32_t time; if (AudioSystem::isLinearPCM(mFormat)) { time = (uint32_t)(mOutput->latency() * 1000) / 2; } else { time = 10000; } return time; } uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() { uint32_t time; if (AudioSystem::isLinearPCM(mFormat)) { time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; } else { time = 10000; } return time; } uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() { uint32_t time; if (AudioSystem::isLinearPCM(mFormat)) { time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); } else { time = 10000; } return time; } // ---------------------------------------------------------------------------- AudioFlinger::DuplicatingThread::DuplicatingThread(const sp& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) { mType = PlaybackThread::DUPLICATING; addOutputTrack(mainThread); } AudioFlinger::DuplicatingThread::~DuplicatingThread() { for (size_t i = 0; i < mOutputTracks.size(); i++) { mOutputTracks[i]->destroy(); } mOutputTracks.clear(); } bool AudioFlinger::DuplicatingThread::threadLoop() { Vector< sp > tracksToRemove; uint32_t mixerStatus = MIXER_IDLE; nsecs_t standbyTime = systemTime(); size_t mixBufferSize = mFrameCount*mFrameSize; SortedVector< sp > outputTracks; uint32_t writeFrames = 0; uint32_t activeSleepTime = activeSleepTimeUs(); uint32_t idleSleepTime = idleSleepTimeUs(); uint32_t sleepTime = idleSleepTime; Vector< sp > effectChains; #ifdef SRS_PROCESSING LOGD("SRS_Processing - DuplicatingThread - OutNotify_Init: %p TID %d\n", this, gettid()); SRS_Processing::ProcessOutNotify(SRS_Processing::AUTO, this, true); #endif while (!exitPending()) { processConfigEvents(); mixerStatus = MIXER_IDLE; { // scope for the mLock Mutex::Autolock _l(mLock); if (checkForNewParameters_l()) { mixBufferSize = mFrameCount*mFrameSize; updateWaitTime(); activeSleepTime = activeSleepTimeUs(); idleSleepTime = idleSleepTimeUs(); } const SortedVector< wp >& activeTracks = mActiveTracks; for (size_t i = 0; i < mOutputTracks.size(); i++) { outputTracks.add(mOutputTracks[i]); } // put audio hardware into standby after short delay if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || mSuspended) { if (!mStandby) { for (size_t i = 0; i < outputTracks.size(); i++) { outputTracks[i]->stop(); } mStandby = true; mBytesWritten = 0; } if (!activeTracks.size() && mConfigEvents.isEmpty()) { // we're about to wait, flush the binder command buffer IPCThreadState::self()->flushCommands(); outputTracks.clear(); if (exitPending()) break; LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); mWaitWorkCV.wait(mLock); LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); if (mMasterMute == false) { char value[PROPERTY_VALUE_MAX]; property_get("ro.audio.silent", value, "0"); if (atoi(value)) { LOGD("Silence is golden"); setMasterMute(true); } } standbyTime = systemTime() + kStandbyTimeInNsecs; sleepTime = idleSleepTime; continue; } } mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); // prevent any changes in effect chain list and in each effect chain // during mixing and effect process as the audio buffers could be deleted // or modified if an effect is created or deleted lockEffectChains_l(effectChains); } if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { // mix buffers... if (outputsReady(outputTracks)) { mAudioMixer->process(); } else { memset(mMixBuffer, 0, mixBufferSize); } sleepTime = 0; writeFrames = mFrameCount; } else { if (sleepTime == 0) { if (mixerStatus == MIXER_TRACKS_ENABLED) { sleepTime = activeSleepTime; } else { sleepTime = idleSleepTime; } } else if (mBytesWritten != 0) { // flush remaining overflow buffers in output tracks for (size_t i = 0; i < outputTracks.size(); i++) { if (outputTracks[i]->isActive()) { sleepTime = 0; writeFrames = 0; memset(mMixBuffer, 0, mixBufferSize); break; } } } } if (mSuspended) { sleepTime = suspendSleepTimeUs(); } // sleepTime == 0 means we must write to audio hardware if (sleepTime == 0) { for (size_t i = 0; i < effectChains.size(); i ++) { effectChains[i]->process_l(); } // enable changes in effect chain unlockEffectChains(effectChains); #ifdef SRS_PROCESSING if (mFormat == AudioSystem::PCM_16_BIT) { SRS_Processing::ProcessOut(SRS_Processing::AUTO, this, mMixBuffer, mixBufferSize, mSampleRate, mChannelCount); } #endif standbyTime = systemTime() + kStandbyTimeInNsecs; for (size_t i = 0; i < outputTracks.size(); i++) { outputTracks[i]->write(mMixBuffer, writeFrames); } mStandby = false; mBytesWritten += mixBufferSize; } else { // enable changes in effect chain unlockEffectChains(effectChains); usleep(sleepTime); } // finally let go of all our tracks, without the lock held // since we can't guarantee the destructors won't acquire that // same lock. tracksToRemove.clear(); outputTracks.clear(); // Effect chains will be actually deleted here if they were removed from // mEffectChains list during mixing or effects processing effectChains.clear(); } #ifdef SRS_PROCESSING LOGD("SRS_Processing - DuplicatingThread - OutNotify_Exit: %p TID %d\n", this, gettid()); SRS_Processing::ProcessOutNotify(SRS_Processing::AUTO, this, false); #endif return false; } void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) { int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, this, mSampleRate, mFormat, mChannelCount, frameCount); if (outputTrack->cblk() != NULL) { thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f); mOutputTracks.add(outputTrack); LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); updateWaitTime(); } } void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) { Mutex::Autolock _l(mLock); for (size_t i = 0; i < mOutputTracks.size(); i++) { if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { mOutputTracks[i]->destroy(); mOutputTracks.removeAt(i); updateWaitTime(); return; } } LOGV("removeOutputTrack(): unkonwn thread: %p", thread); } void AudioFlinger::DuplicatingThread::updateWaitTime() { mWaitTimeMs = UINT_MAX; for (size_t i = 0; i < mOutputTracks.size(); i++) { sp strong = mOutputTracks[i]->thread().promote(); if (strong != NULL) { uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); if (waitTimeMs < mWaitTimeMs) { mWaitTimeMs = waitTimeMs; } } } } bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp > &outputTracks) { for (size_t i = 0; i < outputTracks.size(); i++) { sp thread = outputTracks[i]->thread().promote(); if (thread == 0) { LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); return false; } PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); if (playbackThread->standby() && !playbackThread->isSuspended()) { LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); return false; } } return true; } uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() { return (mWaitTimeMs * 1000) / 2; } // ---------------------------------------------------------------------------- // TrackBase constructor must be called with AudioFlinger::mLock held AudioFlinger::ThreadBase::TrackBase::TrackBase( const wp& thread, const sp& client, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, const sp& sharedBuffer, int sessionId) : RefBase(), mThread(thread), mClient(client), mCblk(0), mFrameCount(0), mState(IDLE), mClientTid(-1), mFormat(format), mFlags(flags & ~SYSTEM_FLAGS_MASK), mSessionId(sessionId) { LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); size_t size = sizeof(audio_track_cblk_t); size_t bufferSize = 0; if ( (format == AudioSystem::PCM_16_BIT) || (format == AudioSystem::PCM_8_BIT)) { bufferSize = frameCount*channelCount*sizeof(int16_t); } else if (format == AudioSystem::AMR_NB) { bufferSize = frameCount*channelCount*32; // full rate frame size } else if (format == AudioSystem::EVRC) { bufferSize = frameCount*channelCount*23; // full rate frame size } else if (format == AudioSystem::QCELP) { bufferSize = frameCount*channelCount*35; // full rate frame size } else if (format == AudioSystem::AAC) { bufferSize = frameCount*2048; // full rate frame size } if (sharedBuffer == 0) { size += bufferSize; } if (client != NULL) { mCblkMemory = client->heap()->allocate(size); if (mCblkMemory != 0) { mCblk = static_cast(mCblkMemory->pointer()); if (mCblk) { // construct the shared structure in-place. new(mCblk) audio_track_cblk_t(); // clear all buffers mCblk->frameCount = frameCount; mCblk->sampleRate = sampleRate; mCblk->channelCount = (uint8_t)channelCount; if (sharedBuffer == 0) { mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); // Change for Codec type if ( (format == AudioSystem::PCM_16_BIT) || (format == AudioSystem::PCM_8_BIT)) { memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); } else if (format == AudioSystem::AMR_NB) { memset(mBuffer, 0, frameCount*channelCount*32); // full rate frame size } else if (format == AudioSystem::EVRC) { memset(mBuffer, 0, frameCount*channelCount*23); // full rate frame size } else if (format == AudioSystem::QCELP) { memset(mBuffer, 0, frameCount*channelCount*35); // full rate frame size } else if (format == AudioSystem::AAC) { memset(mBuffer, 0, frameCount*2048); // full rate frame size } // Force underrun condition to avoid false underrun callback until first data is // written to buffer (other flags are cleared) mCblk->flags = CBLK_UNDERRUN_ON; } else { mBuffer = sharedBuffer->pointer(); } mBufferEnd = (uint8_t *)mBuffer + bufferSize; } } else { LOGE("not enough memory for AudioTrack size=%u", size); client->heap()->dump("AudioTrack"); return; } } else { mCblk = (audio_track_cblk_t *)(new uint8_t[size]); if (mCblk) { // construct the shared structure in-place. new(mCblk) audio_track_cblk_t(); // clear all buffers mCblk->frameCount = frameCount; mCblk->sampleRate = sampleRate; mCblk->channelCount = (uint8_t)channelCount; mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); // Force underrun condition to avoid false underrun callback until first data is // written to buffer (other flags are cleared) mCblk->flags = CBLK_UNDERRUN_ON; mBufferEnd = (uint8_t *)mBuffer + bufferSize; } } } AudioFlinger::ThreadBase::TrackBase::~TrackBase() { if (mCblk) { mCblk->~audio_track_cblk_t(); // destroy our shared-structure. if (mClient == NULL) { delete mCblk; } } mCblkMemory.clear(); // and free the shared memory if (mClient != NULL) { Mutex::Autolock _l(mClient->audioFlinger()->mLock); mClient.clear(); } } void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) { buffer->raw = 0; mFrameCount = buffer->frameCount; step(); buffer->frameCount = 0; } bool AudioFlinger::ThreadBase::TrackBase::step() { bool result; audio_track_cblk_t* cblk = this->cblk(); result = cblk->stepServer(mFrameCount); if (!result) { LOGV("stepServer failed acquiring cblk mutex"); mFlags |= STEPSERVER_FAILED; } return result; } void AudioFlinger::ThreadBase::TrackBase::reset() { audio_track_cblk_t* cblk = this->cblk(); cblk->user = 0; cblk->server = 0; cblk->userBase = 0; cblk->serverBase = 0; mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); LOGV("TrackBase::reset"); } sp AudioFlinger::ThreadBase::TrackBase::getCblk() const { return mCblkMemory; } int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { return (int)mCblk->sampleRate; } int AudioFlinger::ThreadBase::TrackBase::channelCount() const { return (int)mCblk->channelCount; } void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { audio_track_cblk_t* cblk = this->cblk(); int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; // Check validity of returned pointer in case the track control block would have been corrupted. if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || (cblk->channelCount == 2 && ((unsigned long)bufferStart & 3))) { LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ server %d, serverBase %d, user %d, userBase %d, channelCount %d", bufferStart, bufferEnd, mBuffer, mBufferEnd, cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount); return 0; } return bufferStart; } // ---------------------------------------------------------------------------- // Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held AudioFlinger::PlaybackThread::Track::Track( const wp& thread, const sp& client, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, const sp& sharedBuffer, int sessionId) : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId), mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), mAuxEffectId(0), mHasVolumeController(false) { if (mCblk != NULL) { sp baseThread = thread.promote(); if (baseThread != 0) { PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); mName = playbackThread->getTrackName_l(); mMainBuffer = playbackThread->mixBuffer(); } LOGD("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); if (mName < 0) { LOGE("no more track names available"); } mVolume[0] = 1.0f; mVolume[1] = 1.0f; mStreamType = streamType; // 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 = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t); } } AudioFlinger::PlaybackThread::Track::~Track() { LOGV("PlaybackThread::Track destructor"); sp thread = mThread.promote(); if (thread != 0) { Mutex::Autolock _l(thread->mLock); mState = TERMINATED; } } void AudioFlinger::PlaybackThread::Track::destroy() { // NOTE: destroyTrack_l() can remove a strong reference to this Track // by removing it from mTracks vector, so there is a risk that this Tracks's // desctructor is called. As the destructor needs to lock mLock, // we must acquire a strong reference on this Track before locking mLock // here so that the destructor is called only when exiting this function. // On the other hand, as long as Track::destroy() is only called by // TrackHandle destructor, the TrackHandle still holds a strong ref on // this Track with its member mTrack. sp keep(this); { // scope for mLock sp thread = mThread.promote(); if (thread != 0) { if (!isOutputTrack()) { if (mState == ACTIVE || mState == RESUMING) { AudioSystem::stopOutput(thread->id(), (AudioSystem::stream_type)mStreamType, mSessionId); } AudioSystem::releaseOutput(thread->id()); } Mutex::Autolock _l(thread->mLock); PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); playbackThread->destroyTrack_l(this); } } } void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) { snprintf(buffer, size, " %05d %05d %03u %03u %03u %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n", mName - AudioMixer::TRACK0, (mClient == NULL) ? getpid() : mClient->pid(), mStreamType, mFormat, mCblk->channelCount, mSessionId, mFrameCount, mState, mMute, mFillingUpStatus, mCblk->sampleRate, mCblk->volume[0], mCblk->volume[1], mCblk->server, mCblk->user, (int)mMainBuffer, (int)mAuxBuffer); } status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) { audio_track_cblk_t* cblk = this->cblk(); uint32_t framesReady; uint32_t framesReq = buffer->frameCount; LOGV("PlaybackThread::Track::getNextBuffer() mName %d framesReq %d", mName, framesReq); // Check if last stepServer failed, try to step now if (mFlags & TrackBase::STEPSERVER_FAILED) { if (!step()) goto getNextBuffer_exit; LOGV("stepServer recovered"); mFlags &= ~TrackBase::STEPSERVER_FAILED; } framesReady = cblk->framesReady(); if (LIKELY(framesReady)) { uint32_t s = cblk->server; uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; if (framesReq > framesReady) { framesReq = framesReady; } if (s + framesReq > bufferEnd) { framesReq = bufferEnd - s; } buffer->raw = getBuffer(s, framesReq); if (buffer->raw == 0) goto getNextBuffer_exit; buffer->frameCount = framesReq; return NO_ERROR; } getNextBuffer_exit: buffer->raw = 0; buffer->frameCount = 0; LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); return NOT_ENOUGH_DATA; } bool AudioFlinger::PlaybackThread::Track::isReady() const { if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; if (mCblk->framesReady() >= mCblk->frameCount || (mCblk->flags & CBLK_FORCEREADY_MSK)) { mFillingUpStatus = FS_FILLED; mCblk->flags &= ~CBLK_FORCEREADY_MSK; return true; } return false; } status_t AudioFlinger::PlaybackThread::Track::start() { status_t status = NO_ERROR; LOGD("start(%d), calling thread %d session %d", mName, IPCThreadState::self()->getCallingPid(), mSessionId); sp thread = mThread.promote(); if (thread != 0) { Mutex::Autolock _l(thread->mLock); int state = mState; // here the track could be either new, or restarted // in both cases "unstop" the track if (mState == PAUSED) { mState = TrackBase::RESUMING; LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); } else { mState = TrackBase::ACTIVE; LOGV("? => ACTIVE (%d) on thread %p", mName, this); } if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { thread->mLock.unlock(); status = AudioSystem::startOutput(thread->id(), (AudioSystem::stream_type)mStreamType, mSessionId); thread->mLock.lock(); } if (status == NO_ERROR) { PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); playbackThread->addTrack_l(this); } else { mState = state; } } else { status = BAD_VALUE; } return status; } void AudioFlinger::PlaybackThread::Track::stop() { LOGD("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); sp thread = mThread.promote(); if (thread != 0) { Mutex::Autolock _l(thread->mLock); int state = mState; if (mState > STOPPED) { mState = STOPPED; // If the track is not active (PAUSED and buffers full), flush buffers PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); if (playbackThread->mActiveTracks.indexOf(this) < 0) { reset(); } LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); } if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { thread->mLock.unlock(); AudioSystem::stopOutput(thread->id(), (AudioSystem::stream_type)mStreamType, mSessionId); thread->mLock.lock(); } } } void AudioFlinger::PlaybackThread::Track::pause() { LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); sp thread = mThread.promote(); if (thread != 0) { Mutex::Autolock _l(thread->mLock); if (mState == ACTIVE || mState == RESUMING) { mState = PAUSING; LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); if (!isOutputTrack()) { thread->mLock.unlock(); AudioSystem::stopOutput(thread->id(), (AudioSystem::stream_type)mStreamType, mSessionId); thread->mLock.lock(); } } } } void AudioFlinger::PlaybackThread::Track::flush() { LOGV("flush(%d)", mName); sp thread = mThread.promote(); if (thread != 0) { Mutex::Autolock _l(thread->mLock); if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { return; } // No point remaining in PAUSED state after a flush => go to // STOPPED state mState = STOPPED; mCblk->lock.lock(); // NOTE: reset() will reset cblk->user and cblk->server with // the risk that at the same time, the AudioMixer is trying to read // data. In this case, getNextBuffer() would return a NULL pointer // as audio buffer => the AudioMixer code MUST always test that pointer // returned by getNextBuffer() is not NULL! reset(); mCblk->lock.unlock(); } } void AudioFlinger::PlaybackThread::Track::reset() { // Do not reset twice to avoid discarding data written just after a flush and before // the audioflinger thread detects the track is stopped. if (!mResetDone) { TrackBase::reset(); // Force underrun condition to avoid false underrun callback until first data is // written to buffer mCblk->flags |= CBLK_UNDERRUN_ON; mCblk->flags &= ~CBLK_FORCEREADY_MSK; mFillingUpStatus = FS_FILLING; mResetDone = true; } } void AudioFlinger::PlaybackThread::Track::mute(bool muted) { mMute = muted; } void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) { mVolume[0] = left; mVolume[1] = right; } status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) { status_t status = DEAD_OBJECT; sp thread = mThread.promote(); if (thread != 0) { PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); status = playbackThread->attachAuxEffect(this, EffectId); } return status; } void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) { mAuxEffectId = EffectId; mAuxBuffer = buffer; } // ---------------------------------------------------------------------------- // RecordTrack constructor must be called with AudioFlinger::mLock held AudioFlinger::RecordThread::RecordTrack::RecordTrack( const wp& thread, const sp& client, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, int sessionId) : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, flags, 0, sessionId), mOverflow(false) { if (mCblk != NULL) { LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); if (format == AudioSystem::AMR_NB) { mCblk->frameSize = channelCount * 32; } else if (format == AudioSystem::EVRC) { mCblk->frameSize = channelCount * 23; } else if (format == AudioSystem::QCELP) { mCblk->frameSize = channelCount * 35; } else if (format == AudioSystem::AAC) { mCblk->frameSize = 2048; } else if (format == AudioSystem::PCM_16_BIT) { mCblk->frameSize = channelCount * sizeof(int16_t); } else if (format == AudioSystem::PCM_8_BIT) { mCblk->frameSize = channelCount * sizeof(int8_t); } else { mCblk->frameSize = sizeof(int8_t); } } } AudioFlinger::RecordThread::RecordTrack::~RecordTrack() { sp thread = mThread.promote(); if (thread != 0) { AudioSystem::releaseInput(thread->id()); } } status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) { audio_track_cblk_t* cblk = this->cblk(); uint32_t framesAvail; uint32_t framesReq = buffer->frameCount; // Check if last stepServer failed, try to step now if (mFlags & TrackBase::STEPSERVER_FAILED) { if (!step()) goto getNextBuffer_exit; LOGV("stepServer recovered"); mFlags &= ~TrackBase::STEPSERVER_FAILED; } framesAvail = cblk->framesAvailable_l(); if (LIKELY(framesAvail)) { uint32_t s = cblk->server; uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; if (framesReq > framesAvail) { framesReq = framesAvail; } if (s + framesReq > bufferEnd) { framesReq = bufferEnd - s; } buffer->raw = getBuffer(s, framesReq); if (buffer->raw == 0) goto getNextBuffer_exit; buffer->frameCount = framesReq; return NO_ERROR; } getNextBuffer_exit: buffer->raw = 0; buffer->frameCount = 0; return NOT_ENOUGH_DATA; } status_t AudioFlinger::RecordThread::RecordTrack::start() { sp thread = mThread.promote(); if (thread != 0) { RecordThread *recordThread = (RecordThread *)thread.get(); return recordThread->start(this); } else { return BAD_VALUE; } } void AudioFlinger::RecordThread::RecordTrack::stop() { sp thread = mThread.promote(); if (thread != 0) { RecordThread *recordThread = (RecordThread *)thread.get(); recordThread->stop(this); TrackBase::reset(); // Force overerrun condition to avoid false overrun callback until first data is // read from buffer mCblk->flags |= CBLK_UNDERRUN_ON; } } void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) { snprintf(buffer, size, " %05d %03u %03u %05d %04u %01d %05u %08x %08x\n", (mClient == NULL) ? getpid() : mClient->pid(), mFormat, mCblk->channelCount, mSessionId, mFrameCount, mState, mCblk->sampleRate, mCblk->server, mCblk->user); } // ---------------------------------------------------------------------------- AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( const wp& thread, DuplicatingThread *sourceThread, uint32_t sampleRate, int format, int channelCount, int frameCount) : Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL, 0), mActive(false), mSourceThread(sourceThread) { PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); if (mCblk != NULL) { mCblk->flags |= CBLK_DIRECTION_OUT; mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); mCblk->volume[0] = mCblk->volume[1] = 0x1000; mOutBuffer.frameCount = 0; playbackThread->mTracks.add(this); LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p", mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd); } else { LOGW("Error creating output track on thread %p", playbackThread); } } AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() { clearBufferQueue(); } status_t AudioFlinger::PlaybackThread::OutputTrack::start() { status_t status = Track::start(); if (status != NO_ERROR) { return status; } mActive = true; mRetryCount = 127; return status; } void AudioFlinger::PlaybackThread::OutputTrack::stop() { Track::stop(); clearBufferQueue(); mOutBuffer.frameCount = 0; mActive = false; } bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) { Buffer *pInBuffer; Buffer inBuffer; uint32_t channelCount = mCblk->channelCount; bool outputBufferFull = false; inBuffer.frameCount = frames; inBuffer.i16 = data; uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); if (!mActive && frames != 0) { start(); sp thread = mThread.promote(); if (thread != 0) { MixerThread *mixerThread = (MixerThread *)thread.get(); if (mCblk->frameCount > frames){ if (mBufferQueue.size() < kMaxOverFlowBuffers) { uint32_t startFrames = (mCblk->frameCount - frames); pInBuffer = new Buffer; pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; pInBuffer->frameCount = startFrames; pInBuffer->i16 = pInBuffer->mBuffer; memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); mBufferQueue.add(pInBuffer); } else { LOGW ("OutputTrack::write() %p no more buffers in queue", this); } } } } while (waitTimeLeftMs) { // First write pending buffers, then new data if (mBufferQueue.size()) { pInBuffer = mBufferQueue.itemAt(0); } else { pInBuffer = &inBuffer; } if (pInBuffer->frameCount == 0) { break; } if (mOutBuffer.frameCount == 0) { mOutBuffer.frameCount = pInBuffer->frameCount; nsecs_t startTime = systemTime(); if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); outputBufferFull = true; break; } uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); if (waitTimeLeftMs >= waitTimeMs) { waitTimeLeftMs -= waitTimeMs; } else { waitTimeLeftMs = 0; } } 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; pInBuffer->i16 += outFrames * channelCount; mOutBuffer.frameCount -= outFrames; mOutBuffer.i16 += outFrames * channelCount; if (pInBuffer->frameCount == 0) { if (mBufferQueue.size()) { mBufferQueue.removeAt(0); delete [] pInBuffer->mBuffer; delete pInBuffer; LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); } else { break; } } } // If we could not write all frames, allocate a buffer and queue it for next time. if (inBuffer.frameCount) { sp thread = mThread.promote(); if (thread != 0 && !thread->standby()) { if (mBufferQueue.size() < kMaxOverFlowBuffers) { pInBuffer = new Buffer; 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)); mBufferQueue.add(pInBuffer); LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); } else { LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); } } } // Calling write() with a 0 length buffer, means that no more data will be written: // If no more buffers are pending, fill output track buffer to make sure it is started // by output mixer. if (frames == 0 && mBufferQueue.size() == 0) { if (mCblk->user < mCblk->frameCount) { frames = mCblk->frameCount - mCblk->user; pInBuffer = new Buffer; pInBuffer->mBuffer = new int16_t[frames * channelCount]; pInBuffer->frameCount = frames; pInBuffer->i16 = pInBuffer->mBuffer; memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); mBufferQueue.add(pInBuffer); } else if (mActive) { stop(); } } return outputBufferFull; } status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) { int active; status_t result; audio_track_cblk_t* cblk = mCblk; uint32_t framesReq = buffer->frameCount; // LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); buffer->frameCount = 0; uint32_t framesAvail = cblk->framesAvailable(); if (framesAvail == 0) { Mutex::Autolock _l(cblk->lock); goto start_loop_here; while (framesAvail == 0) { active = mActive; if (UNLIKELY(!active)) { LOGV("Not active and NO_MORE_BUFFERS"); return AudioTrack::NO_MORE_BUFFERS; } result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); if (result != NO_ERROR) { return AudioTrack::NO_MORE_BUFFERS; } // read the server count again start_loop_here: framesAvail = cblk->framesAvailable_l(); } } // if (framesAvail < framesReq) { // return AudioTrack::NO_MORE_BUFFERS; // } if (framesReq > framesAvail) { framesReq = framesAvail; } uint32_t u = cblk->user; uint32_t bufferEnd = cblk->userBase + cblk->frameCount; if (u + framesReq > bufferEnd) { framesReq = bufferEnd - u; } buffer->frameCount = framesReq; buffer->raw = (void *)cblk->buffer(u); return NO_ERROR; } void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() { size_t size = mBufferQueue.size(); Buffer *pBuffer; for (size_t i = 0; i < size; i++) { pBuffer = mBufferQueue.itemAt(i); delete [] pBuffer->mBuffer; delete pBuffer; } mBufferQueue.clear(); } // ---------------------------------------------------------------------------- AudioFlinger::Client::Client(const sp& audioFlinger, pid_t pid) : RefBase(), mAudioFlinger(audioFlinger), mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), mPid(pid) { // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer } // Client destructor must be called with AudioFlinger::mLock held AudioFlinger::Client::~Client() { mAudioFlinger->removeClient_l(mPid); } const sp& AudioFlinger::Client::heap() const { return mMemoryDealer; } // ---------------------------------------------------------------------------- AudioFlinger::NotificationClient::NotificationClient(const sp& audioFlinger, const sp& client, sp binder) : mAudioFlinger(audioFlinger), mBinder(binder), mClient(client) { } AudioFlinger::NotificationClient::~NotificationClient() { mClient.clear(); } void AudioFlinger::NotificationClient::binderDied(const wp& who) { sp keep(this); { mAudioFlinger->removeNotificationClient(mBinder); } } // ---------------------------------------------------------------------------- AudioFlinger::TrackHandle::TrackHandle(const sp& track) : BnAudioTrack(), mTrack(track) { } AudioFlinger::TrackHandle::~TrackHandle() { // just stop the track on deletion, associated resources // will be freed from the main thread once all pending buffers have // been played. Unless it's not in the active track list, in which // case we free everything now... mTrack->destroy(); } status_t AudioFlinger::TrackHandle::start() { return mTrack->start(); } void AudioFlinger::TrackHandle::stop() { mTrack->stop(); } void AudioFlinger::TrackHandle::flush() { mTrack->flush(); } void AudioFlinger::TrackHandle::mute(bool e) { mTrack->mute(e); } void AudioFlinger::TrackHandle::pause() { mTrack->pause(); } void AudioFlinger::TrackHandle::setVolume(float left, float right) { mTrack->setVolume(left, right); } sp AudioFlinger::TrackHandle::getCblk() const { return mTrack->getCblk(); } status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) { return mTrack->attachAuxEffect(EffectId); } status_t AudioFlinger::TrackHandle::onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { return BnAudioTrack::onTransact(code, data, reply, flags); } // ---------------------------------------------------------------------------- sp AudioFlinger::openRecord( pid_t pid, int input, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, int *sessionId, status_t *status) { sp recordTrack; sp recordHandle; sp client; wp wclient; size_t inputBufferSize = 0; status_t lStatus; RecordThread *thread; size_t inFrameCount; int lSessionId; // check calling permissions if (!recordingAllowed()) { lStatus = PERMISSION_DENIED; goto Exit; } // Check that audio input stream accepts requested audio parameters inputBufferSize = mAudioHardware->getInputBufferSize(sampleRate, format, channelCount); if (inputBufferSize == 0) { lStatus = BAD_VALUE; LOGE("Bad audio input parameters: sampling rate %u, format %d, channels %d", sampleRate, format, channelCount); goto Exit; } // add client to list { // scope for mLock Mutex::Autolock _l(mLock); thread = checkRecordThread_l(input); if (thread == NULL) { lStatus = BAD_VALUE; goto Exit; } wclient = mClients.valueFor(pid); if (wclient != NULL) { client = wclient.promote(); } else { client = new Client(this, pid); mClients.add(pid, client); } // If no audio session id is provided, create one here if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) { lSessionId = *sessionId; } else { lSessionId = nextUniqueId(); if (sessionId != NULL) { *sessionId = lSessionId; } } // frameCount must be a multiple of input buffer size // Change for Codec type if ((format == AudioSystem::PCM_16_BIT) || (format == AudioSystem::PCM_8_BIT)) { inFrameCount = inputBufferSize/channelCount/sizeof(short); } else if (format == AudioSystem::AMR_NB) { inFrameCount = inputBufferSize/channelCount/32; } else if (format == AudioSystem::EVRC) { inFrameCount = inputBufferSize/channelCount/23; } else if (format == AudioSystem::QCELP) { inFrameCount = inputBufferSize/channelCount/35; } else if (format == AudioSystem::AAC) { inFrameCount = inputBufferSize/2048; } frameCount = ((frameCount - 1)/inFrameCount + 1) * inFrameCount; // create new record track. The record track uses one track in mHardwareMixerThread by convention. recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate, format, channelCount, frameCount, flags, lSessionId); } if (recordTrack->getCblk() == NULL) { // 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(); lStatus = NO_MEMORY; goto Exit; } // return to handle to client recordHandle = new RecordHandle(recordTrack); lStatus = NO_ERROR; Exit: if (status) { *status = lStatus; } return recordHandle; } // ---------------------------------------------------------------------------- AudioFlinger::RecordHandle::RecordHandle(const sp& recordTrack) : BnAudioRecord(), mRecordTrack(recordTrack) { } AudioFlinger::RecordHandle::~RecordHandle() { stop(); } status_t AudioFlinger::RecordHandle::start() { LOGV("RecordHandle::start()"); return mRecordTrack->start(); } void AudioFlinger::RecordHandle::stop() { LOGV("RecordHandle::stop()"); mRecordTrack->stop(); } sp AudioFlinger::RecordHandle::getCblk() const { return mRecordTrack->getCblk(); } status_t AudioFlinger::RecordHandle::onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { return BnAudioRecord::onTransact(code, data, reply, flags); } // ---------------------------------------------------------------------------- AudioFlinger::RecordThread::RecordThread(const sp& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels, int id) : ThreadBase(audioFlinger, id), mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) { mReqChannelCount = getInputChannelCount(channels); mReqSampleRate = sampleRate; readInputParameters(); } AudioFlinger::RecordThread::~RecordThread() { delete[] mRsmpInBuffer; if (mResampler != 0) { delete mResampler; delete[] mRsmpOutBuffer; } } void AudioFlinger::RecordThread::onFirstRef() { const size_t SIZE = 256; char buffer[SIZE]; snprintf(buffer, SIZE, "Record Thread %p", this); run(buffer, PRIORITY_URGENT_AUDIO); } bool AudioFlinger::RecordThread::threadLoop() { AudioBufferProvider::Buffer buffer; sp activeTrack; nsecs_t lastWarning = 0; // start recording while (!exitPending()) { processConfigEvents(); { // scope for mLock Mutex::Autolock _l(mLock); checkForNewParameters_l(); if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { if (!mStandby) { mInput->standby(); mStandby = true; } if (exitPending()) break; LOGV("RecordThread: loop stopping"); // go to sleep mWaitWorkCV.wait(mLock); LOGV("RecordThread: loop starting"); continue; } if (mActiveTrack != 0) { if (mActiveTrack->mState == TrackBase::PAUSING) { if (!mStandby) { mInput->standby(); mStandby = true; } mActiveTrack.clear(); mStartStopCond.broadcast(); } else if (mActiveTrack->mState == TrackBase::RESUMING) { if (mReqChannelCount != mActiveTrack->channelCount()) { mActiveTrack.clear(); mStartStopCond.broadcast(); } else if (mBytesRead != 0) { // record start succeeds only if first read from audio input // succeeds if (mBytesRead > 0) { mActiveTrack->mState = TrackBase::ACTIVE; } else { mActiveTrack.clear(); } mStartStopCond.broadcast(); } mStandby = false; } } } if (mActiveTrack != 0) { if (mActiveTrack->mState != TrackBase::ACTIVE && mActiveTrack->mState != TrackBase::RESUMING) { usleep(5000); continue; } buffer.frameCount = mFrameCount; if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { size_t framesOut = buffer.frameCount; if (mResampler == 0) { // no resampling while (framesOut) { 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; if (framesIn > framesOut) framesIn = framesOut; mRsmpInIndex += framesIn; framesOut -= framesIn; if ((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT) { memcpy(dst, src, framesIn * mFrameSize); } else { int16_t *src16 = (int16_t *)src; int16_t *dst16 = (int16_t *)dst; if (mChannelCount == 1) { while (framesIn--) { *dst16++ = *src16; *dst16++ = *src16++; } } else { while (framesIn--) { *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); src16 += 2; } } } } if (framesOut && mFrameCount == mRsmpInIndex) { if (((int) framesOut != mFrameCount) && (mFormat != AudioSystem::PCM_16_BIT) ) { mBytesRead = mInput->read(buffer.raw, buffer.frameCount * mFrameSize); if(mBytesRead >= 0 ){ buffer.frameCount = mBytesRead/mFrameSize; } framesOut = 0; } else if (framesOut == mFrameCount && ((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) { mBytesRead = mInput->read(buffer.raw, mInputBytes); if( mBytesRead >= 0 ){ buffer.frameCount = mBytesRead/mFrameSize; } framesOut = 0; } else { mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); mRsmpInIndex = 0; } if (mBytesRead < 0) { LOGE("Error reading audio input"); if (mActiveTrack->mState == TrackBase::ACTIVE) { // Force input into standby so that it tries to // recover at next read attempt mInput->standby(); usleep(5000); } mRsmpInIndex = mFrameCount; framesOut = 0; buffer.frameCount = 0; } } } } else { // resampling memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); // alter output frame count as if we were expecting stereo samples if (mChannelCount == 1 && mReqChannelCount == 1) { framesOut >>= 1; } mResampler->resample(mRsmpOutBuffer, framesOut, this); // 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) { AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); // the resampler always outputs stereo samples: do post stereo to mono conversion int16_t *src = (int16_t *)mRsmpOutBuffer; int16_t *dst = buffer.i16; while (framesOut--) { *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); src += 2; } } else { AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); } } mActiveTrack->releaseBuffer(&buffer); mActiveTrack->overflow(); } // client isn't retrieving buffers fast enough else { if (!mActiveTrack->setOverflow()) { nsecs_t now = systemTime(); if ((now - lastWarning) > kWarningThrottle) { LOGW("RecordThread: buffer overflow"); lastWarning = now; } } // Release the processor for a while before asking for a new buffer. // This will give the application more chance to read from the buffer and // clear the overflow. usleep(5000); } } } if (!mStandby) { mInput->standby(); } mActiveTrack.clear(); mStartStopCond.broadcast(); LOGV("RecordThread %p exiting", this); return false; } status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) { LOGD("RecordThread::start"); sp strongMe = this; status_t status = NO_ERROR; { AutoMutex lock(&mLock); if (mActiveTrack != 0) { if (recordTrack != mActiveTrack.get()) { status = -EBUSY; } else if (mActiveTrack->mState == TrackBase::PAUSING) { mActiveTrack->mState = TrackBase::ACTIVE; } return status; } recordTrack->mState = TrackBase::IDLE; mActiveTrack = recordTrack; mLock.unlock(); status_t status = AudioSystem::startInput(mId); mLock.lock(); if (status != NO_ERROR) { mActiveTrack.clear(); return status; } mActiveTrack->mState = TrackBase::RESUMING; mRsmpInIndex = mFrameCount; mBytesRead = 0; // signal thread to start LOGV("Signal record thread"); mWaitWorkCV.signal(); // do not wait for mStartStopCond if exiting if (mExiting) { mActiveTrack.clear(); status = INVALID_OPERATION; goto startError; } mStartStopCond.wait(mLock); if (mActiveTrack == 0) { LOGV("Record failed to start"); status = BAD_VALUE; goto startError; } LOGV("Record started OK"); return status; } startError: AudioSystem::stopInput(mId); return status; } void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { LOGD("RecordThread::stop"); sp strongMe = this; { AutoMutex lock(&mLock); if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { mActiveTrack->mState = TrackBase::PAUSING; // do not wait for mStartStopCond if exiting if (mExiting) { return; } mStartStopCond.wait(mLock); // if we have been restarted, recordTrack == mActiveTrack.get() here if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { mLock.unlock(); AudioSystem::stopInput(mId); mLock.lock(); LOGV("Record stopped OK"); } } } } status_t AudioFlinger::RecordThread::dump(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; pid_t pid = 0; snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); result.append(buffer); if (mActiveTrack != 0) { result.append("Active Track:\n"); result.append(" Clien Fmt Chn Session Buf S SRate Serv User\n"); mActiveTrack->dump(buffer, SIZE); result.append(buffer); snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); result.append(buffer); snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); result.append(buffer); snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); result.append(buffer); snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); result.append(buffer); snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); result.append(buffer); } else { result.append("No record client\n"); } write(fd, result.string(), result.size()); dumpBase(fd, args); return NO_ERROR; } status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) { size_t framesReq = buffer->frameCount; size_t framesReady = mFrameCount - mRsmpInIndex; int channelCount; if (framesReady == 0) { mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); if (mBytesRead < 0) { LOGE("RecordThread::getNextBuffer() Error reading audio input"); if (mActiveTrack->mState == TrackBase::ACTIVE) { // Force input into standby so that it tries to // recover at next read attempt mInput->standby(); usleep(5000); } buffer->raw = 0; buffer->frameCount = 0; return NOT_ENOUGH_DATA; } mRsmpInIndex = 0; framesReady = mFrameCount; } if (framesReq > framesReady) { framesReq = framesReady; } if (mChannelCount == 1 && mReqChannelCount == 2) { channelCount = 1; } else { channelCount = 2; } buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; buffer->frameCount = framesReq; return NO_ERROR; } void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) { mRsmpInIndex += buffer->frameCount; buffer->frameCount = 0; } bool AudioFlinger::RecordThread::checkForNewParameters_l() { bool reconfig = false; while (!mNewParameters.isEmpty()) { status_t status = NO_ERROR; String8 keyValuePair = mNewParameters[0]; AudioParameter param = AudioParameter(keyValuePair); int value; int reqFormat = mFormat; int reqSamplingRate = mReqSampleRate; int reqChannelCount = mReqChannelCount; if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { reqSamplingRate = value; reconfig = true; } if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { reqFormat = value; reconfig = true; } if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { reqChannelCount = getInputChannelCount(value); reconfig = true; } if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { // do not accept frame count changes if tracks are open as the track buffer // size depends on frame count and correct behavior would not be garantied // if frame count is changed after track creation if (mActiveTrack != 0) { status = INVALID_OPERATION; } else { reconfig = true; } } if (status == NO_ERROR) { status = mInput->setParameters(keyValuePair); if (status == INVALID_OPERATION) { mInput->standby(); status = mInput->setParameters(keyValuePair); } if (reconfig) { if (status == BAD_VALUE && reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT && ((int)mInput->sampleRate() <= 2 * reqSamplingRate) && (getInputChannelCount(mInput->channels()) < 3) && (reqChannelCount < 3)) { status = NO_ERROR; } if (status == NO_ERROR) { readInputParameters(); sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); } } } mNewParameters.removeAt(0); mParamStatus = status; mParamCond.signal(); mWaitWorkCV.wait(mLock); } return reconfig; } String8 AudioFlinger::RecordThread::getParameters(const String8& keys) { return mInput->getParameters(keys); } void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { AudioSystem::OutputDescriptor desc; void *param2 = 0; switch (event) { case AudioSystem::INPUT_OPENED: case AudioSystem::INPUT_CONFIG_CHANGED: desc.channels = mChannels; desc.samplingRate = mSampleRate; desc.format = mFormat; desc.frameCount = mFrameCount; desc.latency = 0; param2 = &desc; break; case AudioSystem::INPUT_CLOSED: default: break; } mAudioFlinger->audioConfigChanged_l(event, mId, param2); } void AudioFlinger::RecordThread::readInputParameters() { if (mRsmpInBuffer) delete mRsmpInBuffer; if (mRsmpOutBuffer) delete mRsmpOutBuffer; if (mResampler) delete mResampler; mResampler = 0; mSampleRate = mInput->sampleRate(); mChannels = mInput->channels(); mChannelCount = getInputChannelCount(mInput->channels()); mFormat = mInput->format(); mFrameSize = (uint16_t)mInput->frameSize(); mInputBytes = mInput->bufferSize(); mFrameCount = mInputBytes / mFrameSize; mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) { int channelCount; // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid // stereo to mono post process as the resampler always outputs stereo. if (mChannelCount == 1 && mReqChannelCount == 2) { channelCount = 1; } else { channelCount = 2; } mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); mResampler->setSampleRate(mSampleRate); 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 if (mChannelCount == 1 && mReqChannelCount == 1) { mFrameCount >>= 1; } } mRsmpInIndex = mFrameCount; } unsigned int AudioFlinger::RecordThread::getInputFramesLost() { return mInput->getInputFramesLost(); } // ---------------------------------------------------------------------------- int AudioFlinger::openOutput(uint32_t *pDevices, uint32_t *pSamplingRate, uint32_t *pFormat, uint32_t *pChannels, uint32_t *pLatencyMs, uint32_t flags) { status_t status; PlaybackThread *thread = NULL; mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; uint32_t format = pFormat ? *pFormat : 0; uint32_t channels = pChannels ? *pChannels : 0; uint32_t latency = pLatencyMs ? *pLatencyMs : 0; LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", pDevices ? *pDevices : 0, samplingRate, format, channels, flags); if (pDevices == NULL || *pDevices == 0) { return 0; } Mutex::Autolock _l(mLock); AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices, (int *)&format, &channels, &samplingRate, &status); LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", output, samplingRate, format, channels, status); mHardwareStatus = AUDIO_HW_IDLE; if (output != 0) { int id = nextUniqueId(); if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) || (format != AudioSystem::PCM_16_BIT) || (channels != AudioSystem::CHANNEL_OUT_STEREO)) { thread = new DirectOutputThread(this, output, id, *pDevices); LOGV("openOutput() created direct output: ID %d thread %p", id, thread); } else { thread = new MixerThread(this, output, id, *pDevices); LOGV("openOutput() created mixer output: ID %d thread %p", id, thread); #ifdef LVMX unsigned bitsPerSample = (format == AudioSystem::PCM_16_BIT) ? 16 : ((format == AudioSystem::PCM_8_BIT) ? 8 : 0); unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1; int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id()); LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount); LifeVibes::setDevice(audioOutputType, *pDevices); #endif } mPlaybackThreads.add(id, thread); if (pSamplingRate) *pSamplingRate = samplingRate; if (pFormat) *pFormat = format; if (pChannels) *pChannels = channels; if (pLatencyMs) *pLatencyMs = thread->latency(); // if the device is a A2DP, then this is an A2DP Output if ( true == AudioSystem::isA2dpDevice((AudioSystem::audio_devices) *pDevices) ) { mA2DPHandle = id; LOGV("A2DP device activated. The handle is set to %d", mA2DPHandle); } // notify client processes of the new output creation thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); return id; } return 0; } int AudioFlinger::openSession(uint32_t *pDevices, uint32_t *pFormat, uint32_t flags, int32_t streamType, int32_t sessionId) { status_t status; mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; uint32_t format = pFormat ? *pFormat : 0; LOGV("openSession(), Device %x, Format %d, flags %x sessionId %x", pDevices ? *pDevices : 0, format, flags, sessionId); if (pDevices == NULL || *pDevices == 0) { return 0; } Mutex::Autolock _l(mLock); AudioStreamOut *output = mAudioHardware->openOutputSession(*pDevices, (int *)&format, &status,sessionId); LOGV("openSession() openOutputSession returned output %p, Format %d, status %d", output, format, status); mHardwareStatus = AUDIO_HW_IDLE; if (output != 0) { int id = nextUniqueId(); mLPAOutput = output; mLPAHandle = id; mLPAStreamType = streamType; mLPAStreamIsActive = true; if (pFormat) *pFormat = format; return id; } return 0; } status_t AudioFlinger::pauseSession(int output, int32_t streamType) { if (output == mLPAHandle && streamType == mLPAStreamType ) { mLPAStreamIsActive = false; } return NO_ERROR; } status_t AudioFlinger::resumeSession(int output, int32_t streamType) { if (output == mLPAHandle && streamType == mLPAStreamType ) { mLPAStreamIsActive = true; } return NO_ERROR; } status_t AudioFlinger::closeSession(int output) { Mutex::Autolock _l(mLock); LOGV("closeSession() %d", output); // Is this required? //AudioSystem::stopOutput(output, (AudioSystem::stream_type)mStreamType); // Delete the Audio session if(mLPAOutput) { mLPAOutput->standby(); delete mLPAOutput; mLPAOutput = NULL; mLPAHandle = -1; mLPAStreamIsActive = false; } return NO_ERROR; } int AudioFlinger::openDuplicateOutput(int output1, int output2) { Mutex::Autolock _l(mLock); MixerThread *thread1 = checkMixerThread_l(output1); MixerThread *thread2 = checkMixerThread_l(output2); if (thread1 == NULL || thread2 == NULL) { LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); return 0; } int id = nextUniqueId(); DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); thread->addOutputTrack(thread2); mPlaybackThreads.add(id, thread); // notify client processes of the new output creation thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); return id; } status_t AudioFlinger::closeOutput(int output) { // keep strong reference on the playback thread so that // it is not destroyed while exit() is executed sp thread; { Mutex::Autolock _l(mLock); thread = checkPlaybackThread_l(output); if (thread == NULL) { return BAD_VALUE; } LOGV("closeOutput() %d", output); if (thread->type() == PlaybackThread::MIXER) { for (size_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) { DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); dupThread->removeOutputTrack((MixerThread *)thread.get()); } } } void *param2 = 0; audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); mPlaybackThreads.removeItem(output); if (mA2DPHandle == output) { mA2DPHandle = -1; LOGV("A2DP OutputClosed Notifying Client"); audioConfigChanged_l(AudioSystem::A2DP_OUTPUT_STATE, mA2DPHandle, param2); } } thread->exit(); if (thread->type() != PlaybackThread::DUPLICATING) { mAudioHardware->closeOutputStream(thread->getOutput()); } return NO_ERROR; } status_t AudioFlinger::suspendOutput(int output) { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { return BAD_VALUE; } LOGV("suspendOutput() %d", output); thread->suspend(); return NO_ERROR; } status_t AudioFlinger::restoreOutput(int output) { Mutex::Autolock _l(mLock); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { return BAD_VALUE; } LOGV("restoreOutput() %d", output); thread->restore(); return NO_ERROR; } int AudioFlinger::openInput(uint32_t *pDevices, uint32_t *pSamplingRate, uint32_t *pFormat, uint32_t *pChannels, uint32_t acoustics) { status_t status; RecordThread *thread = NULL; uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; uint32_t format = pFormat ? *pFormat : 0; uint32_t channels = pChannels ? *pChannels : 0; uint32_t reqSamplingRate = samplingRate; uint32_t reqFormat = format; uint32_t reqChannels = channels; if (pDevices == NULL || *pDevices == 0) { return 0; } Mutex::Autolock _l(mLock); AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices, (int *)&format, &channels, &samplingRate, &status, (AudioSystem::audio_in_acoustics)acoustics); LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", input, samplingRate, format, channels, acoustics, 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 (input == 0 && status == BAD_VALUE && reqFormat == format && format == AudioSystem::PCM_16_BIT && (samplingRate <= 2 * reqSamplingRate) && (getInputChannelCount(channels) < 3) && (getInputChannelCount(reqChannels) < 3)) { LOGV("openInput() reopening with proposed sampling rate and channels"); input = mAudioHardware->openInputStream(*pDevices, (int *)&format, &channels, &samplingRate, &status, (AudioSystem::audio_in_acoustics)acoustics); } if (input != 0) { int id = nextUniqueId(); // Start record thread thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id); mRecordThreads.add(id, thread); LOGV("openInput() created record thread: ID %d thread %p", id, thread); if (pSamplingRate) *pSamplingRate = reqSamplingRate; if (pFormat) *pFormat = format; if (pChannels) *pChannels = reqChannels; input->standby(); // notify client processes of the new input creation thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); return id; } return 0; } status_t AudioFlinger::closeInput(int input) { // keep strong reference on the record thread so that // it is not destroyed while exit() is executed sp thread; { Mutex::Autolock _l(mLock); thread = checkRecordThread_l(input); if (thread == NULL) { return BAD_VALUE; } LOGV("closeInput() %d", input); void *param2 = 0; audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); mRecordThreads.removeItem(input); } thread->exit(); mAudioHardware->closeInputStream(thread->getInput()); return NO_ERROR; } status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) { Mutex::Autolock _l(mLock); MixerThread *dstThread = checkMixerThread_l(output); if (dstThread == NULL) { LOGW("setStreamOutput() bad output id %d", output); return BAD_VALUE; } LOGV("setStreamOutput() stream %d to output %d", stream, output); audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); for (size_t i = 0; i < mPlaybackThreads.size(); i++) { PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); if (thread != dstThread && thread->type() != PlaybackThread::DIRECT) { MixerThread *srcThread = (MixerThread *)thread; srcThread->invalidateTracks(stream); } } if ( mA2DPHandle == output ) { LOGV("A2DP Activated and hence notifying the client"); dstThread->sendConfigEvent(AudioSystem::A2DP_OUTPUT_STATE, mA2DPHandle); } return NO_ERROR; } int AudioFlinger::newAudioSessionId() { return nextUniqueId(); } // checkPlaybackThread_l() must be called with AudioFlinger::mLock held AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const { PlaybackThread *thread = NULL; if (mPlaybackThreads.indexOfKey(output) >= 0) { thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); } return thread; } // checkMixerThread_l() must be called with AudioFlinger::mLock held AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const { PlaybackThread *thread = checkPlaybackThread_l(output); if (thread != NULL) { if (thread->type() == PlaybackThread::DIRECT) { thread = NULL; } } return (MixerThread *)thread; } // checkRecordThread_l() must be called with AudioFlinger::mLock held AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const { RecordThread *thread = NULL; if (mRecordThreads.indexOfKey(input) >= 0) { thread = (RecordThread *)mRecordThreads.valueFor(input).get(); } return thread; } int AudioFlinger::nextUniqueId() { return android_atomic_inc(&mNextUniqueId); } // ---------------------------------------------------------------------------- // Effect management // ---------------------------------------------------------------------------- status_t AudioFlinger::loadEffectLibrary(const char *libPath, int *handle) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } // only allow libraries loaded from /system/lib/soundfx for now if (strncmp(gEffectLibPath, libPath, strlen(gEffectLibPath)) != 0) { return PERMISSION_DENIED; } Mutex::Autolock _l(mLock); return EffectLoadLibrary(libPath, handle); } status_t AudioFlinger::unloadEffectLibrary(int handle) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } Mutex::Autolock _l(mLock); return EffectUnloadLibrary(handle); } status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) { Mutex::Autolock _l(mLock); return EffectQueryNumberEffects(numEffects); } status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) { Mutex::Autolock _l(mLock); return EffectQueryEffect(index, descriptor); } status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) { Mutex::Autolock _l(mLock); return EffectGetDescriptor(pUuid, descriptor); } // this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp static const effect_uuid_t VISUALIZATION_UUID_ = {0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; sp AudioFlinger::createEffect(pid_t pid, effect_descriptor_t *pDesc, const sp& effectClient, int32_t priority, int output, int sessionId, status_t *status, int *id, int *enabled) { status_t lStatus = NO_ERROR; sp handle; effect_interface_t itfe; effect_descriptor_t desc; sp client; wp wclient; LOGV("createEffect %s pid %d, client %p, priority %d, sessionId %d, output %d", pDesc->name, pid, effectClient.get(), priority, sessionId, output); if (pDesc == NULL) { lStatus = BAD_VALUE; goto Exit; } // check audio settings permission for global effects if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && !settingsAllowed()) { lStatus = PERMISSION_DENIED; goto Exit; } // Session AudioSystem::SESSION_OUTPUT_STAGE is reserved for output stage effects // that can only be created by audio policy manager (running in same process) if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE && getpid() != pid) { lStatus = PERMISSION_DENIED; goto Exit; } // check recording permission for visualizer if ((memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 || memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) && !recordingAllowed()) { lStatus = PERMISSION_DENIED; goto Exit; } if (output == 0) { if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE) { // output must be specified by AudioPolicyManager when using session // AudioSystem::SESSION_OUTPUT_STAGE lStatus = BAD_VALUE; goto Exit; } else if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) { // if the output returned by getOutputForEffect() is removed before we lock the // mutex below, the call to checkPlaybackThread_l(output) below will detect it // and we will exit safely output = AudioSystem::getOutputForEffect(&desc); } } { Mutex::Autolock _l(mLock); if (!EffectIsNullUuid(&pDesc->uuid)) { // if uuid is specified, request effect descriptor lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); if (lStatus < 0) { LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); goto Exit; } } else { // if uuid is not specified, look for an available implementation // of the required type in effect factory if (EffectIsNullUuid(&pDesc->type)) { LOGW("createEffect() no effect type"); lStatus = BAD_VALUE; goto Exit; } uint32_t numEffects = 0; effect_descriptor_t d; bool found = false; lStatus = EffectQueryNumberEffects(&numEffects); if (lStatus < 0) { LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); goto Exit; } for (uint32_t i = 0; i < numEffects; i++) { lStatus = EffectQueryEffect(i, &desc); if (lStatus < 0) { LOGW("createEffect() error %d from EffectQueryEffect", lStatus); continue; } if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { // If matching type found save effect descriptor. If the session is // 0 and the effect is not auxiliary, continue enumeration in case // an auxiliary version of this effect type is available found = true; memcpy(&d, &desc, sizeof(effect_descriptor_t)); if (sessionId != AudioSystem::SESSION_OUTPUT_MIX || (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { break; } } } if (!found) { lStatus = BAD_VALUE; LOGW("createEffect() effect not found"); goto Exit; } // For same effect type, chose auxiliary version over insert version if // connect to output mix (Compliance to OpenSL ES) if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { memcpy(&desc, &d, sizeof(effect_descriptor_t)); } } // Do not allow auxiliary effects on a session different from 0 (output mix) if (sessionId != AudioSystem::SESSION_OUTPUT_MIX && (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { lStatus = INVALID_OPERATION; goto Exit; } // return effect descriptor memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); // If output is not specified try to find a matching audio session ID in one of the // output threads. // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX // because of code checking output when entering the function. if (output == 0) { // look for the thread where the specified audio session is present for (size_t i = 0; i < mPlaybackThreads.size(); i++) { if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { output = mPlaybackThreads.keyAt(i); break; } } // If no output thread contains the requested session ID, default to // first output. The effect chain will be moved to the correct output // thread when a track with the same session ID is created if (output == 0 && mPlaybackThreads.size()) { output = mPlaybackThreads.keyAt(0); } } LOGV("createEffect() got output %d for effect %s", output, desc.name); PlaybackThread *thread = checkPlaybackThread_l(output); if (thread == NULL) { LOGE("createEffect() unknown output thread"); lStatus = BAD_VALUE; goto Exit; } // TODO: allow attachment of effect to inputs wclient = mClients.valueFor(pid); if (wclient != NULL) { client = wclient.promote(); } else { client = new Client(this, pid); mClients.add(pid, client); } // create effect on selected output trhead handle = thread->createEffect_l(client, effectClient, priority, sessionId, &desc, enabled, &lStatus); if (handle != 0 && id != NULL) { *id = handle->id(); } } Exit: if(status) { *status = lStatus; } return handle; } status_t AudioFlinger::moveEffects(int session, int srcOutput, int dstOutput) { LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", session, srcOutput, dstOutput); Mutex::Autolock _l(mLock); if (srcOutput == dstOutput) { LOGW("moveEffects() same dst and src outputs %d", dstOutput); return NO_ERROR; } PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); if (srcThread == NULL) { LOGW("moveEffects() bad srcOutput %d", srcOutput); return BAD_VALUE; } PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); if (dstThread == NULL) { LOGW("moveEffects() bad dstOutput %d", dstOutput); return BAD_VALUE; } Mutex::Autolock _dl(dstThread->mLock); Mutex::Autolock _sl(srcThread->mLock); moveEffectChain_l(session, srcThread, dstThread, false); return NO_ERROR; } // moveEffectChain_l mustbe called with both srcThread and dstThread mLocks held status_t AudioFlinger::moveEffectChain_l(int session, AudioFlinger::PlaybackThread *srcThread, AudioFlinger::PlaybackThread *dstThread, bool reRegister) { LOGV("moveEffectChain_l() session %d from thread %p to thread %p", session, srcThread, dstThread); sp chain = srcThread->getEffectChain_l(session); if (chain == 0) { LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", session, srcThread); return INVALID_OPERATION; } // remove chain first. This is useful only if reconfiguring effect chain on same output thread, // so that a new chain is created with correct parameters when first effect is added. This is // otherwise unecessary as removeEffect_l() will remove the chain when last effect is // removed. srcThread->removeEffectChain_l(chain); // transfer all effects one by one so that new effect chain is created on new thread with // correct buffer sizes and audio parameters and effect engines reconfigured accordingly int dstOutput = dstThread->id(); sp dstChain; uint32_t strategy; sp effect = chain->getEffectFromId_l(0); while (effect != 0) { srcThread->removeEffect_l(effect); dstThread->addEffect_l(effect); // if the move request is not received from audio policy manager, the effect must be // re-registered with the new strategy and output if (dstChain == 0) { dstChain = effect->chain().promote(); if (dstChain == 0) { LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); srcThread->addEffect_l(effect); return NO_INIT; } strategy = dstChain->strategy(); } if (reRegister) { AudioSystem::unregisterEffect(effect->id()); AudioSystem::registerEffect(&effect->desc(), dstOutput, strategy, session, effect->id()); } effect = chain->getEffectFromId_l(0); } return NO_ERROR; } // PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held sp AudioFlinger::PlaybackThread::createEffect_l( const sp& client, const sp& effectClient, int32_t priority, int sessionId, effect_descriptor_t *desc, int *enabled, status_t *status ) { sp effect; sp handle; status_t lStatus; sp track; sp chain; bool chainCreated = false; bool effectCreated = false; bool effectRegistered = false; if (mOutput == 0) { LOGW("createEffect_l() Audio driver not initialized."); lStatus = NO_INIT; goto Exit; } // Do not allow auxiliary effect on session other than 0 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY && sessionId != AudioSystem::SESSION_OUTPUT_MIX) { LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", desc->name, sessionId); lStatus = BAD_VALUE; goto Exit; } // Do not allow effects with session ID 0 on direct output or duplicating threads // TODO: add rule for hw accelerated effects on direct outputs with non PCM format if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && mType != MIXER) { LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", desc->name, sessionId); lStatus = BAD_VALUE; goto Exit; } LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); { // scope for mLock Mutex::Autolock _l(mLock); // check for existing effect chain with the requested audio session chain = getEffectChain_l(sessionId); if (chain == 0) { // create a new chain for this session LOGV("createEffect_l() new effect chain for session %d", sessionId); chain = new EffectChain(this, sessionId); addEffectChain_l(chain); chain->setStrategy(getStrategyForSession_l(sessionId)); chainCreated = true; if(sessionId == mAudioFlinger->mLPASessionId) { // Clear reference to previous effect chain if any if(mAudioFlinger->mLPAEffectChain.get()) { mAudioFlinger->mLPAEffectChain.clear(); } LOGV("New EffectChain is created for LPA session ID %d", sessionId); mAudioFlinger->mLPAEffectChain = chain; chain->setLPAFlag(true); // For LPA, the volume will be applied in DSP. No need for volume // control in the Effect chain, so setting it to unity. uint32_t volume = 0x1000000; // Equals to 1.0 in 8.24 format chain->setVolume_l(&volume,&volume); } } else { effect = chain->getEffectFromDesc_l(desc); } LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); if (effect == 0) { int id = mAudioFlinger->nextUniqueId(); // Check CPU and memory usage lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); if (lStatus != NO_ERROR) { goto Exit; } effectRegistered = true; // create a new effect module if none present in the chain effect = new EffectModule(this, chain, desc, id, sessionId); lStatus = effect->status(); if (lStatus != NO_ERROR) { goto Exit; } lStatus = chain->addEffect_l(effect); if (lStatus != NO_ERROR) { goto Exit; } effectCreated = true; effect->setDevice(mDevice); effect->setMode(mAudioFlinger->getMode()); if(chain == mAudioFlinger->mLPAEffectChain) { effect->setLPAFlag(true); } } // create effect handle and connect it to effect module handle = new EffectHandle(effect, client, effectClient, priority); lStatus = effect->addHandle(handle); if (enabled) { *enabled = (int)effect->isEnabled(); } } Exit: if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { Mutex::Autolock _l(mLock); if (effectCreated) { chain->removeEffect_l(effect); } if (effectRegistered) { AudioSystem::unregisterEffect(effect->id()); } if (chainCreated) { removeEffectChain_l(chain); } handle.clear(); } if(status) { *status = lStatus; } return handle; } // PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and // PlaybackThread::mLock held status_t AudioFlinger::PlaybackThread::addEffect_l(const sp& effect) { // check for existing effect chain with the requested audio session int sessionId = effect->sessionId(); sp chain = getEffectChain_l(sessionId); bool chainCreated = false; if (chain == 0) { // create a new chain for this session LOGV("addEffect_l() new effect chain for session %d", sessionId); chain = new EffectChain(this, sessionId); addEffectChain_l(chain); chain->setStrategy(getStrategyForSession_l(sessionId)); chainCreated = true; } LOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); if (chain->getEffectFromId_l(effect->id()) != 0) { LOGW("addEffect_l() %p effect %s already present in chain %p", this, effect->desc().name, chain.get()); return BAD_VALUE; } status_t status = chain->addEffect_l(effect); if (status != NO_ERROR) { if (chainCreated) { removeEffectChain_l(chain); } return status; } effect->setDevice(mDevice); effect->setMode(mAudioFlinger->getMode()); return NO_ERROR; } void AudioFlinger::PlaybackThread::removeEffect_l(const sp& effect) { LOGV("removeEffect_l() %p effect %p", this, effect.get()); effect_descriptor_t desc = effect->desc(); if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { detachAuxEffect_l(effect->id()); } sp chain = effect->chain().promote(); if (chain != 0) { // remove effect chain if removing last effect if (chain->removeEffect_l(effect) == 0) { removeEffectChain_l(chain); } } else { LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); } } void AudioFlinger::PlaybackThread::disconnectEffect(const sp& effect, const wp& handle) { Mutex::Autolock _l(mLock); LOGV("disconnectEffect() %p effect %p", this, effect.get()); // delete the effect module if removing last handle on it if (effect->removeHandle(handle) == 0) { removeEffect_l(effect); AudioSystem::unregisterEffect(effect->id()); } } status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp& chain) { int session = chain->sessionId(); int16_t *buffer = mMixBuffer; bool ownsBuffer = false; LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); if (session > 0) { // Only one effect chain can be present in direct output thread and it uses // the mix buffer as input if (mType != DIRECT) { size_t numSamples = mFrameCount * mChannelCount; buffer = new int16_t[numSamples]; memset(buffer, 0, numSamples * sizeof(int16_t)); LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); ownsBuffer = true; } // Attach all tracks with same session ID to this chain. for (size_t i = 0; i < mTracks.size(); ++i) { sp track = mTracks[i]; if (session == track->sessionId()) { LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); track->setMainBuffer(buffer); } } // indicate all active tracks in the chain for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { sp track = mActiveTracks[i].promote(); if (track == 0) continue; if (session == track->sessionId()) { LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); chain->startTrack(); } } } chain->setInBuffer(buffer, ownsBuffer); chain->setOutBuffer(mMixBuffer); // Effect chain for session AudioSystem::SESSION_OUTPUT_STAGE is inserted at end of effect // chains list in order to be processed last as it contains output stage effects // Effect chain for session AudioSystem::SESSION_OUTPUT_MIX is inserted before // session AudioSystem::SESSION_OUTPUT_STAGE to be processed // after track specific effects and before output stage // It is therefore mandatory that AudioSystem::SESSION_OUTPUT_MIX == 0 and // that AudioSystem::SESSION_OUTPUT_STAGE < AudioSystem::SESSION_OUTPUT_MIX // Effect chain for other sessions are inserted at beginning of effect // chains list to be processed before output mix effects. Relative order between other // sessions is not important size_t size = mEffectChains.size(); size_t i = 0; for (i = 0; i < size; i++) { if (mEffectChains[i]->sessionId() < session) break; } mEffectChains.insertAt(chain, i); return NO_ERROR; } size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp& chain) { int session = chain->sessionId(); LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); for (size_t i = 0; i < mEffectChains.size(); i++) { if (chain == mEffectChains[i]) { mEffectChains.removeAt(i); // detach all tracks with same session ID from this chain for (size_t i = 0; i < mTracks.size(); ++i) { sp track = mTracks[i]; if (session == track->sessionId()) { track->setMainBuffer(mMixBuffer); } } break; } } return mEffectChains.size(); } void AudioFlinger::PlaybackThread::lockEffectChains_l( Vector >& effectChains) { effectChains = mEffectChains; for (size_t i = 0; i < mEffectChains.size(); i++) { // Do not LPA playback track's effect chain if(mEffectChains[i] != mAudioFlinger->mLPAEffectChain) { mEffectChains[i]->lock(); } } } void AudioFlinger::PlaybackThread::unlockEffectChains( Vector >& effectChains) { for (size_t i = 0; i < effectChains.size(); i++) { // LPA playback track's effect chain is not locked - do not unlock if(mEffectChains[i] != mAudioFlinger->mLPAEffectChain) { effectChains[i]->unlock(); } } } sp AudioFlinger::PlaybackThread::getEffect_l(int sessionId, int effectId) { sp effect; sp chain = getEffectChain_l(sessionId); if (chain != 0) { effect = chain->getEffectFromId_l(effectId); } return effect; } status_t AudioFlinger::PlaybackThread::attachAuxEffect( const sp track, int EffectId) { Mutex::Autolock _l(mLock); return attachAuxEffect_l(track, EffectId); } status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( const sp track, int EffectId) { status_t status = NO_ERROR; LOGV("PlaybackThread::attachAuxEffect_l: EffectId %d", EffectId); if (EffectId == 0) { track->setAuxBuffer(0, NULL); } else { // Auxiliary effects are always in audio session AudioSystem::SESSION_OUTPUT_MIX sp effect = getEffect_l(AudioSystem::SESSION_OUTPUT_MIX, EffectId); if (effect != 0) { if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); } else { status = INVALID_OPERATION; } } else { status = BAD_VALUE; } } return status; } void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) { for (size_t i = 0; i < mTracks.size(); ++i) { sp track = mTracks[i]; if (track->auxEffectId() == effectId) { attachAuxEffect_l(track, 0); } } } // ---------------------------------------------------------------------------- // EffectModule implementation // ---------------------------------------------------------------------------- #undef LOG_TAG #define LOG_TAG "AudioFlinger::EffectModule" AudioFlinger::EffectModule::EffectModule(const wp& wThread, const wp& chain, effect_descriptor_t *desc, int id, int sessionId) : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), mStatus(NO_INIT), mState(IDLE), mIsForLPA(false) { LOGV("Constructor %p sessionId %d", this, sessionId); int lStatus; sp thread = mThread.promote(); if (thread == 0) { return; } PlaybackThread *p = (PlaybackThread *)thread.get(); memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); // create effect engine from effect factory mStatus = EffectCreate(&desc->uuid, sessionId, p->id(), &mEffectInterface); if (mStatus != NO_ERROR) { return; } lStatus = init(); if (lStatus < 0) { mStatus = lStatus; goto Error; } LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); return; Error: EffectRelease(mEffectInterface); mEffectInterface = NULL; LOGV("Constructor Error %d", mStatus); } AudioFlinger::EffectModule::~EffectModule() { LOGV("Destructor %p", this); if (mEffectInterface != NULL) { // release effect engine EffectRelease(mEffectInterface); } } status_t AudioFlinger::EffectModule::addHandle(sp& handle) { status_t status; Mutex::Autolock _l(mLock); // First handle in mHandles has highest priority and controls the effect module int priority = handle->priority(); size_t size = mHandles.size(); sp h; size_t i; for (i = 0; i < size; i++) { h = mHandles[i].promote(); if (h == 0) continue; if (h->priority() <= priority) break; } // if inserted in first place, move effect control from previous owner to this handle if (i == 0) { if (h != 0) { h->setControl(false, true); } handle->setControl(true, false); status = NO_ERROR; } else { status = ALREADY_EXISTS; } mHandles.insertAt(handle, i); return status; } size_t AudioFlinger::EffectModule::removeHandle(const wp& handle) { Mutex::Autolock _l(mLock); size_t size = mHandles.size(); size_t i; for (i = 0; i < size; i++) { if (mHandles[i] == handle) break; } if (i == size) { return size; } mHandles.removeAt(i); size = mHandles.size(); // if removed from first place, move effect control from this handle to next in line if (i == 0 && size != 0) { sp h = mHandles[0].promote(); if (h != 0) { h->setControl(true, true); } } // Release effect engine here so that it is done immediately. Otherwise it will be released // by the destructor when the last strong reference on the this object is released which can // happen after next process is called on this effect. if (size == 0 && mEffectInterface != NULL) { // release effect engine EffectRelease(mEffectInterface); mEffectInterface = NULL; } return size; } void AudioFlinger::EffectModule::disconnect(const wp& handle) { setEnabled(false); // keep a strong reference on this EffectModule to avoid calling the // destructor before we exit sp keep(this); { sp thread = mThread.promote(); if (thread != 0) { PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); playbackThread->disconnectEffect(keep, handle); } } } void AudioFlinger::EffectModule::updateState() { Mutex::Autolock _l(mLock); switch (mState) { case RESTART: reset_l(); // FALL THROUGH case STARTING: // clear auxiliary effect input buffer for next accumulation if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { memset(mConfig.inputCfg.buffer.raw, 0, mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); } start_l(); mState = ACTIVE; break; case STOPPING: stop_l(); mDisableWaitCnt = mMaxDisableWaitCnt; mState = STOPPED; break; case STOPPED: // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the // turn off sequence. if (--mDisableWaitCnt == 0) { reset_l(); mState = IDLE; } break; default: //IDLE , ACTIVE break; } } void AudioFlinger::EffectModule::process() { Mutex::Autolock _l(mLock); LOGV("EffectModule::process()"); if (mEffectInterface == NULL || mConfig.inputCfg.buffer.raw == NULL || mConfig.outputCfg.buffer.raw == NULL) { return; } if (isProcessEnabled()) { // do 32 bit to 16 bit conversion for auxiliary effect input buffer if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32, mConfig.inputCfg.buffer.s32, mConfig.inputCfg.buffer.frameCount/2); } // do the actual processing in the effect engine int ret = (*mEffectInterface)->process(mEffectInterface, &mConfig.inputCfg.buffer, &mConfig.outputCfg.buffer); // force transition to IDLE state when engine is ready if (mState == STOPPED && ret == -ENODATA) { mDisableWaitCnt = 1; } // clear auxiliary effect input buffer for next accumulation if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { memset(mConfig.inputCfg.buffer.raw, 0, mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); } } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { // If an insert effect is idle and input buffer is different from output buffer, // accumulate input onto output sp chain = mChain.promote(); if (chain != 0 && chain->activeTracks() != 0) { size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here int16_t *in = mConfig.inputCfg.buffer.s16; int16_t *out = mConfig.outputCfg.buffer.s16; for (size_t i = 0; i < frameCnt; i++) { out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); } } } } void AudioFlinger::EffectModule::reset_l() { if (mEffectInterface == NULL) { return; } (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); } status_t AudioFlinger::EffectModule::configure(bool isForLPA, int sampleRate, int channelCount, int frameCount) { uint32_t channels; // Acquire lock here to make sure that any other thread does not delete // the effect handle and release the effect module. Mutex::Autolock _l(mLock); if (mEffectInterface == NULL) { return NO_INIT; } sp thread = mThread.promote(); if (thread == 0) { return DEAD_OBJECT; } // TODO: handle configuration of effects replacing track process mIsForLPA = isForLPA; if(isForLPA) { if (channelCount == 1) { channels = CHANNEL_MONO; } else { channels = CHANNEL_STEREO; } LOGV("%s: LPA ON - channels %d", __func__, channels); } else { if (thread->channelCount() == 1) { channels = CHANNEL_MONO; } else { channels = CHANNEL_STEREO; } } if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { mConfig.inputCfg.channels = CHANNEL_MONO; } else { mConfig.inputCfg.channels = channels; } mConfig.outputCfg.channels = channels; mConfig.inputCfg.format = SAMPLE_FORMAT_PCM_S15; mConfig.outputCfg.format = SAMPLE_FORMAT_PCM_S15; if(isForLPA){ mConfig.inputCfg.samplingRate = sampleRate; LOGV("%s: LPA ON - sampleRate %d", __func__, sampleRate); } else { mConfig.inputCfg.samplingRate = thread->sampleRate(); } mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; mConfig.inputCfg.bufferProvider.cookie = NULL; mConfig.inputCfg.bufferProvider.getBuffer = NULL; mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; mConfig.outputCfg.bufferProvider.cookie = NULL; mConfig.outputCfg.bufferProvider.getBuffer = NULL; mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; // Insert effect: // - in session AudioSystem::SESSION_OUTPUT_MIX or AudioSystem::SESSION_OUTPUT_STAGE, // always overwrites output buffer: input buffer == output buffer // - in other sessions: // last effect in the chain accumulates in output buffer: input buffer != output buffer // other effect: overwrites output buffer: input buffer == output buffer // Auxiliary effect: // accumulates in output buffer: input buffer != output buffer // Therefore: accumulate <=> input buffer != output buffer if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; } else { mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; } mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; if(isForLPA) { mConfig.inputCfg.buffer.frameCount = frameCount; LOGV("%s: LPA ON - frameCount %d", __func__, frameCount); } else { mConfig.inputCfg.buffer.frameCount = thread->frameCount(); } mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; LOGV("configure() %p thread %p buffer %p framecount %d", this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); status_t cmdStatus; uint32_t size = sizeof(int); status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_CONFIGURE, sizeof(effect_config_t), &mConfig, &size, &cmdStatus); if (status == 0) { status = cmdStatus; } mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / (1000 * mConfig.outputCfg.buffer.frameCount); return status; } status_t AudioFlinger::EffectModule::init() { Mutex::Autolock _l(mLock); if (mEffectInterface == NULL) { return NO_INIT; } status_t cmdStatus; uint32_t size = sizeof(status_t); status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_INIT, 0, NULL, &size, &cmdStatus); if (status == 0) { status = cmdStatus; } return status; } status_t AudioFlinger::EffectModule::start_l() { if (mEffectInterface == NULL) { return NO_INIT; } status_t cmdStatus; uint32_t size = sizeof(status_t); status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_ENABLE, 0, NULL, &size, &cmdStatus); if (status == 0) { if(cmdStatus == 0) { // Some of the effects are applied based on the output device. // For example, BASS_BOOST will not be enabled if output device is Speaker. // If the command to set device is called first and then the command // to enable the effect, it is not checking the previously set device. // The below command to set device after enabling the effect, ensures that // output device is considered to enable/disable the effect. status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_DEVICE, sizeof(uint32_t), &mDevice, &size, &cmdStatus); if(status == 0) { status = cmdStatus; } } else { status = cmdStatus; } } return status; } status_t AudioFlinger::EffectModule::stop_l() { if (mEffectInterface == NULL) { return NO_INIT; } status_t cmdStatus; uint32_t size = sizeof(status_t); status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_DISABLE, 0, NULL, &size, &cmdStatus); if (status == 0) { status = cmdStatus; } return status; } status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, uint32_t cmdSize, void *pCmdData, uint32_t *replySize, void *pReplyData) { Mutex::Autolock _l(mLock); LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); if (mEffectInterface == NULL) { return NO_INIT; } status_t status = (*mEffectInterface)->command(mEffectInterface, cmdCode, cmdSize, pCmdData, replySize, pReplyData); if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { uint32_t size = (replySize == NULL) ? 0 : *replySize; for (size_t i = 1; i < mHandles.size(); i++) { sp h = mHandles[i].promote(); if (h != 0) { h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); } } } return status; } status_t AudioFlinger::EffectModule::setEnabled(bool enabled) { bool effectStateChanged = false; { Mutex::Autolock _l(mLock); LOGV("setEnabled %p enabled %d", this, enabled); if (enabled != isEnabled()) { effectStateChanged = true; switch (mState) { // going from disabled to enabled case IDLE: mState = STARTING; break; case STOPPED: mState = RESTART; break; case STOPPING: mState = ACTIVE; break; // going from enabled to disabled case RESTART: mState = STOPPED; break; case STARTING: mState = IDLE; break; case ACTIVE: mState = STOPPING; break; } for (size_t i = 1; i < mHandles.size(); i++) { sp h = mHandles[i].promote(); if (h != 0) { h->setEnabled(enabled); } } } } /* Send notification event to LPA Player when an effect for LPA output is enabled or disabled. */ if (effectStateChanged && mIsForLPA) { sp thread = mThread.promote(); thread->effectConfigChanged(); } return NO_ERROR; } bool AudioFlinger::EffectModule::isEnabled() { switch (mState) { case RESTART: case STARTING: case ACTIVE: return true; case IDLE: case STOPPING: case STOPPED: default: return false; } } bool AudioFlinger::EffectModule::isProcessEnabled() { switch (mState) { case RESTART: case ACTIVE: case STOPPING: case STOPPED: return true; case IDLE: case STARTING: default: return false; } } status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) { Mutex::Autolock _l(mLock); status_t status = NO_ERROR; // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) if (isProcessEnabled() && ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { status_t cmdStatus; uint32_t volume[2]; uint32_t *pVolume = NULL; uint32_t size = sizeof(volume); volume[0] = *left; volume[1] = *right; if (controller) { pVolume = volume; } status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_VOLUME, size, volume, &size, pVolume); if (controller && status == NO_ERROR && size == sizeof(volume)) { *left = volume[0]; *right = volume[1]; } } return status; } status_t AudioFlinger::EffectModule::setDevice(uint32_t device) { Mutex::Autolock _l(mLock); status_t status = NO_ERROR; if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { // convert device bit field from AudioSystem to EffectApi format. device = deviceAudioSystemToEffectApi(device); if (device == 0) { return BAD_VALUE; } status_t cmdStatus; uint32_t size = sizeof(status_t); status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_DEVICE, sizeof(uint32_t), &device, &size, &cmdStatus); mDevice = device; if (status == NO_ERROR) { status = cmdStatus; } } return status; } status_t AudioFlinger::EffectModule::setMode(uint32_t mode) { Mutex::Autolock _l(mLock); status_t status = NO_ERROR; if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { // convert audio mode from AudioSystem to EffectApi format. int effectMode = modeAudioSystemToEffectApi(mode); if (effectMode < 0) { return BAD_VALUE; } status_t cmdStatus; uint32_t size = sizeof(status_t); status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_AUDIO_MODE, sizeof(int), &effectMode, &size, &cmdStatus); if (status == NO_ERROR) { status = cmdStatus; } } return status; } // update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = { DEVICE_EARPIECE, // AudioSystem::DEVICE_OUT_EARPIECE DEVICE_SPEAKER, // AudioSystem::DEVICE_OUT_SPEAKER DEVICE_WIRED_HEADSET, // case AudioSystem::DEVICE_OUT_WIRED_HEADSET DEVICE_WIRED_HEADPHONE, // AudioSystem::DEVICE_OUT_WIRED_HEADPHONE DEVICE_BLUETOOTH_SCO, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO DEVICE_BLUETOOTH_SCO_HEADSET, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET DEVICE_BLUETOOTH_SCO_CARKIT, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT DEVICE_BLUETOOTH_A2DP, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES DEVICE_BLUETOOTH_A2DP_SPEAKER, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER DEVICE_AUX_DIGITAL, // AudioSystem::DEVICE_OUT_AUX_DIGITAL DEVICE_AUX_HDMI, DEVICE_FM, DEVICE_ANC_HEADSET, DEVICE_ANC_HEADPHONE, DEVICE_FM_TX, }; uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device) { uint32_t deviceOut = 0; while (device) { const uint32_t i = 31 - __builtin_clz(device); device &= ~(1 << i); if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) { LOGE("device convertion error for AudioSystem device 0x%08x", device); return 0; } deviceOut |= (uint32_t)sDeviceConvTable[i]; } return deviceOut; } // update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = { AUDIO_MODE_NORMAL, // AudioSystem::MODE_NORMAL AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE AUDIO_MODE_IN_CALL, // AudioSystem::MODE_IN_CALL AUDIO_MODE_IN_CALL // AudioSystem::MODE_IN_COMMUNICATION, same conversion as for MODE_IN_CALL }; int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode) { int modeOut = -1; if (mode < sizeof(sModeConvTable) / sizeof(uint32_t)) { modeOut = (int)sModeConvTable[mode]; } return modeOut; } status_t AudioFlinger::EffectModule::dump(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); result.append(buffer); bool locked = tryLock(mLock); // failed to lock - AudioFlinger is probably deadlocked if (!locked) { result.append("\t\tCould not lock Fx mutex:\n"); } result.append("\t\tSession Status State Engine:\n"); snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", mSessionId, mStatus, mState, (uint32_t)mEffectInterface); result.append(buffer); 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.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.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); result.append(buffer); snprintf(buffer, SIZE, "\t\t- apiVersion: %04X\n\t\t- flags: %08X\n", mDescriptor.apiVersion, mDescriptor.flags); result.append(buffer); snprintf(buffer, SIZE, "\t\t- name: %s\n", mDescriptor.name); result.append(buffer); snprintf(buffer, SIZE, "\t\t- implementor: %s\n", mDescriptor.implementor); result.append(buffer); result.append("\t\t- Input configuration:\n"); result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", (uint32_t)mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount, mConfig.inputCfg.samplingRate, mConfig.inputCfg.channels, mConfig.inputCfg.format); result.append(buffer); result.append("\t\t- Output configuration:\n"); result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", (uint32_t)mConfig.outputCfg.buffer.raw, mConfig.outputCfg.buffer.frameCount, mConfig.outputCfg.samplingRate, mConfig.outputCfg.channels, mConfig.outputCfg.format); result.append(buffer); snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); result.append(buffer); result.append("\t\t\tPid Priority Ctrl Locked client server\n"); for (size_t i = 0; i < mHandles.size(); ++i) { sp handle = mHandles[i].promote(); if (handle != 0) { handle->dump(buffer, SIZE); result.append(buffer); } } result.append("\n"); write(fd, result.string(), result.length()); if (locked) { mLock.unlock(); } return NO_ERROR; } // ---------------------------------------------------------------------------- // EffectHandle implementation // ---------------------------------------------------------------------------- #undef LOG_TAG #define LOG_TAG "AudioFlinger::EffectHandle" AudioFlinger::EffectHandle::EffectHandle(const sp& effect, const sp& client, const sp& effectClient, int32_t priority) : BnEffect(), mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false) { LOGV("constructor %p", this); int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); if (mCblkMemory != 0) { mCblk = static_cast(mCblkMemory->pointer()); if (mCblk) { new(mCblk) effect_param_cblk_t(); mBuffer = (uint8_t *)mCblk + bufOffset; } } else { LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); return; } } AudioFlinger::EffectHandle::~EffectHandle() { LOGV("Destructor %p", this); disconnect(); } status_t AudioFlinger::EffectHandle::enable() { if (!mHasControl) return INVALID_OPERATION; if (mEffect == 0) return DEAD_OBJECT; return mEffect->setEnabled(true); } status_t AudioFlinger::EffectHandle::disable() { if (!mHasControl) return INVALID_OPERATION; if (mEffect == NULL) return DEAD_OBJECT; return mEffect->setEnabled(false); } void AudioFlinger::EffectHandle::disconnect() { if (mEffect == 0) { return; } mEffect->disconnect(this); // release sp on module => module destructor can be called now mEffect.clear(); if (mCblk) { mCblk->~effect_param_cblk_t(); // destroy our shared-structure. } mCblkMemory.clear(); // and free the shared memory if (mClient != 0) { Mutex::Autolock _l(mClient->audioFlinger()->mLock); mClient.clear(); } } status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, uint32_t cmdSize, void *pCmdData, uint32_t *replySize, void *pReplyData) { LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); // only get parameter command is permitted for applications not controlling the effect if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { return INVALID_OPERATION; } if (mEffect == 0) return DEAD_OBJECT; // 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 Mutex::Autolock _l(mCblk->lock); if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { mCblk->serverIndex = 0; mCblk->clientIndex = 0; return BAD_VALUE; } status_t status = NO_ERROR; while (mCblk->serverIndex < mCblk->clientIndex) { int reply; uint32_t rsize = sizeof(int); int *p = (int *)(mBuffer + mCblk->serverIndex); int size = *p++; if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { LOGW("command(): invalid parameter block size"); break; } effect_param_t *param = (effect_param_t *)p; if (param->psize == 0 || param->vsize == 0) { LOGW("command(): null parameter or value size"); mCblk->serverIndex += size; continue; } uint32_t psize = sizeof(effect_param_t) + ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + param->vsize; status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, psize, p, &rsize, &reply); // stop at first error encountered if (ret != NO_ERROR) { status = ret; *(int *)pReplyData = reply; break; } else if (reply != NO_ERROR) { *(int *)pReplyData = reply; break; } mCblk->serverIndex += size; } mCblk->serverIndex = 0; mCblk->clientIndex = 0; return status; } else if (cmdCode == EFFECT_CMD_ENABLE) { *(int *)pReplyData = NO_ERROR; return enable(); } else if (cmdCode == EFFECT_CMD_DISABLE) { *(int *)pReplyData = NO_ERROR; return disable(); } LOGV("EffectHandle::command: isOnLPA %d", mEffect->isOnLPA()); if(mEffect->isOnLPA() && ((cmdCode == EFFECT_CMD_SET_PARAM) || (cmdCode == EFFECT_CMD_SET_PARAM_DEFERRED) || (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) || (cmdCode == EFFECT_CMD_SET_DEVICE) || (cmdCode == EFFECT_CMD_SET_VOLUME) || (cmdCode == EFFECT_CMD_SET_AUDIO_MODE)) ) { // Notify LPA Player for the change in Effect module // TODO: check if it is required to send mLPAHandle LOGV("Notifying LPA player for the change in effect config"); mClient->audioFlinger()->audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL); } return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); } sp AudioFlinger::EffectHandle::getCblk() const { return mCblkMemory; } void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal) { LOGV("setControl %p control %d", this, hasControl); mHasControl = hasControl; if (signal && mEffectClient != 0) { mEffectClient->controlStatusChanged(hasControl); } } void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, uint32_t cmdSize, void *pCmdData, uint32_t replySize, void *pReplyData) { if (mEffectClient != 0) { mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); } } void AudioFlinger::EffectHandle::setEnabled(bool enabled) { if (mEffectClient != 0) { mEffectClient->enableStatusChanged(enabled); } } status_t AudioFlinger::EffectHandle::onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { return BnEffect::onTransact(code, data, reply, flags); } void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) { bool locked = tryLock(mCblk->lock); snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", (mClient == NULL) ? getpid() : mClient->pid(), mPriority, mHasControl, !locked, mCblk->clientIndex, mCblk->serverIndex ); if (locked) { mCblk->lock.unlock(); } } #undef LOG_TAG #define LOG_TAG "AudioFlinger::EffectChain" AudioFlinger::EffectChain::EffectChain(const wp& wThread, int sessionId) : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX), mIsForLPATrack(false) { LOGV("EffectChain::ctor sessionId %d", sessionId); mStrategy = AudioSystem::getStrategyForStream(AudioSystem::MUSIC); } AudioFlinger::EffectChain::~EffectChain() { LOGV("EffectChain::dtor mOwnInBuffer %d", mOwnInBuffer); if (mOwnInBuffer) { delete mInBuffer; } } // getEffectFromDesc_l() must be called with PlaybackThread::mLock held sp AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) { sp effect; size_t size = mEffects.size(); for (size_t i = 0; i < size; i++) { if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { effect = mEffects[i]; break; } } return effect; } // getEffectFromId_l() must be called with PlaybackThread::mLock held sp AudioFlinger::EffectChain::getEffectFromId_l(int id) { sp effect; size_t size = mEffects.size(); for (size_t i = 0; i < size; i++) { // by convention, return first effect if id provided is 0 (0 is never a valid id) if (id == 0 || mEffects[i]->id() == id) { effect = mEffects[i]; break; } } return effect; } sp AudioFlinger::EffectChain::getEffectFromIndex_l(int idx) { sp effect = NULL; if(idx < 0 || idx >= mEffects.size()) { LOGE("EffectChain::getEffectFromIndex_l: invalid index %d", idx); } if(mEffects.size() > 0){ effect = mEffects[idx]; } return effect; } // Must be called with EffectChain::mLock locked void AudioFlinger::EffectChain::process_l() { LOGV("EffectChain::process_l()"); sp thread = mThread.promote(); if (thread == 0) { LOGW("process_l(): cannot promote mixer thread"); return; } PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); bool isGlobalSession = (mSessionId == AudioSystem::SESSION_OUTPUT_MIX) || (mSessionId == AudioSystem::SESSION_OUTPUT_STAGE); bool tracksOnSession = false; if (!isGlobalSession && !isForLPATrack()) { tracksOnSession = playbackThread->hasAudioSession(mSessionId) & PlaybackThread::TRACK_SESSION; } size_t size = mEffects.size(); LOGV("process_l(): isGlobalSession %d tracksOnSession %d isForLPA %d numEffects %d", isGlobalSession, tracksOnSession, isForLPATrack(), size); // do not process effect if no track is present in same audio session if (isGlobalSession || tracksOnSession || isForLPATrack()) { for (size_t i = 0; i < size; i++) { mEffects[i]->process(); } } for (size_t i = 0; i < size; i++) { mEffects[i]->updateState(); } // if no track is active, input buffer must be cleared here as the mixer process // will not do it if (tracksOnSession && activeTracks() == 0) { size_t numSamples = playbackThread->frameCount() * playbackThread->channelCount(); memset(mInBuffer, 0, numSamples * sizeof(int16_t)); } } // addEffect_l() must be called with PlaybackThread::mLock held status_t AudioFlinger::EffectChain::addEffect_l(const sp& effect) { effect_descriptor_t desc = effect->desc(); uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; Mutex::Autolock _l(mLock); effect->setChain(this); sp thread = mThread.promote(); if (thread == 0) { return NO_INIT; } effect->setThread(thread); if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { // Auxiliary effects are inserted at the beginning of mEffects vector as // they are processed first and accumulated in chain input buffer mEffects.insertAt(effect, 0); // the input buffer for auxiliary effect contains mono samples in // 32 bit format. This is to avoid saturation in AudoMixer // accumulation stage. Saturation is done in EffectModule::process() before // calling the process in effect engine size_t numSamples = thread->frameCount(); int32_t *buffer = new int32_t[numSamples]; memset(buffer, 0, numSamples * sizeof(int32_t)); effect->setInBuffer((int16_t *)buffer); // auxiliary effects output samples to chain input buffer for further processing // by insert effects effect->setOutBuffer(mInBuffer); } else { // Insert effects are inserted at the end of mEffects vector as they are processed // after track and auxiliary effects. // Insert effect order as a function of indicated preference: // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if // another effect is present // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the // last effect claiming first position // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the // first effect claiming last position // else if EFFECT_FLAG_INSERT_ANY insert after first or before last // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is // already present int size = (int)mEffects.size(); int idx_insert = size; int idx_insert_first = -1; int idx_insert_last = -1; for (int i = 0; i < size; i++) { effect_descriptor_t d = mEffects[i]->desc(); uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; if (iMode == EFFECT_FLAG_TYPE_INSERT) { // check invalid effect chaining combinations if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { LOGW("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 // select this as insert position for new effect if (idx_insert == size) { idx_insert = i; } // remember position of last insert effect claiming // first position if (iPref == EFFECT_FLAG_INSERT_FIRST) { idx_insert_first = i; } // remember position of first insert effect claiming // last position if (iPref == EFFECT_FLAG_INSERT_LAST && idx_insert_last == -1) { idx_insert_last = i; } } } // modify idx_insert from first position if needed if (insertPref == EFFECT_FLAG_INSERT_LAST) { if (idx_insert_last != -1) { idx_insert = idx_insert_last; } else { idx_insert = size; } } else { if (idx_insert_first != -1) { idx_insert = idx_insert_first + 1; } } // always read samples from chain input buffer effect->setInBuffer(mInBuffer); // if last effect in the chain, output samples to chain // output buffer, otherwise to chain input buffer if (idx_insert == size) { if (idx_insert != 0) { mEffects[idx_insert-1]->setOutBuffer(mInBuffer); mEffects[idx_insert-1]->configure(); } effect->setOutBuffer(mOutBuffer); } else { effect->setOutBuffer(mInBuffer); } mEffects.insertAt(effect, idx_insert); LOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); } effect->configure(); return NO_ERROR; } // removeEffect_l() must be called with PlaybackThread::mLock held size_t AudioFlinger::EffectChain::removeEffect_l(const sp& effect) { Mutex::Autolock _l(mLock); int size = (int)mEffects.size(); int i; uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; for (i = 0; i < size; i++) { if (effect == mEffects[i]) { if (type == EFFECT_FLAG_TYPE_AUXILIARY) { delete[] effect->inBuffer(); } else { if (i == size - 1 && i != 0) { mEffects[i - 1]->setOutBuffer(mOutBuffer); mEffects[i - 1]->configure(); } } mEffects.removeAt(i); LOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); break; } } return mEffects.size(); } // setDevice_l() must be called with PlaybackThread::mLock held void AudioFlinger::EffectChain::setDevice_l(uint32_t device) { size_t size = mEffects.size(); for (size_t i = 0; i < size; i++) { mEffects[i]->setDevice(device); } } // setMode_l() must be called with PlaybackThread::mLock held void AudioFlinger::EffectChain::setMode_l(uint32_t mode) { size_t size = mEffects.size(); for (size_t i = 0; i < size; i++) { mEffects[i]->setMode(mode); } } // setVolume_l() must be called with PlaybackThread::mLock held bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) { uint32_t newLeft = *left; uint32_t newRight = *right; bool hasControl = false; int ctrlIdx = -1; size_t size = mEffects.size(); // first update volume controller for (size_t i = size; i > 0; i--) { if (mEffects[i - 1]->isProcessEnabled() && (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { ctrlIdx = i - 1; hasControl = true; break; } } if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { if (hasControl) { *left = mNewLeftVolume; *right = mNewRightVolume; } return hasControl; } mVolumeCtrlIdx = ctrlIdx; mLeftVolume = newLeft; mRightVolume = newRight; // second get volume update from volume controller if (ctrlIdx >= 0) { mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); mNewLeftVolume = newLeft; mNewRightVolume = newRight; } // then indicate volume to all other effects in chain. // Pass altered volume to effects before volume controller // and requested volume to effects after controller uint32_t lVol = newLeft; uint32_t rVol = newRight; for (size_t i = 0; i < size; i++) { if ((int)i == ctrlIdx) continue; // this also works for ctrlIdx == -1 when there is no volume controller if ((int)i > ctrlIdx) { lVol = *left; rVol = *right; } mEffects[i]->setVolume(&lVol, &rVol, false); } *left = newLeft; *right = newRight; return hasControl; } status_t AudioFlinger::EffectChain::dump(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); result.append(buffer); bool locked = tryLock(mLock); // failed to lock - AudioFlinger is probably deadlocked if (!locked) { result.append("\tCould not lock mutex:\n"); } result.append("\tNum fx In buffer Out buffer Active tracks:\n"); snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", mEffects.size(), (uint32_t)mInBuffer, (uint32_t)mOutBuffer, mActiveTrackCnt); result.append(buffer); write(fd, result.string(), result.size()); for (size_t i = 0; i < mEffects.size(); ++i) { sp effect = mEffects[i]; if (effect != 0) { effect->dump(fd, args); } } if (locked) { mLock.unlock(); } return NO_ERROR; } #undef LOG_TAG #define LOG_TAG "AudioFlinger" // ---------------------------------------------------------------------------- status_t AudioFlinger::onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { return BnAudioFlinger::onTransact(code, data, reply, flags); } }; // namespace android