/* //device/include/server/AudioFlinger/AudioFlinger.cpp ** ** Copyright 2007, The Android Open Source Project ** ** Licensed under the Apache License, Version 2.0 (the "License"); ** you may not use this file except in compliance with the License. ** You may obtain a copy of the License at ** ** http://www.apache.org/licenses/LICENSE-2.0 ** ** Unless required by applicable law or agreed to in writing, software ** distributed under the License is distributed on an "AS IS" BASIS, ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. ** See the License for the specific language governing permissions and ** limitations under the License. */ #define LOG_TAG "AudioFlinger" //#define LOG_NDEBUG 0 #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 // ---------------------------------------------------------------------------- // the sim build doesn't have gettid #ifndef HAVE_GETTID # define gettid getpid #endif // ---------------------------------------------------------------------------- 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 unsigned long kBufferRecoveryInUsecs = 2000; static const unsigned long kMaxBufferRecoveryInUsecs = 20000; static const float MAX_GAIN = 4096.0f; // retry counts for buffer fill timeout // 50 * ~20msecs = 1 second static const int8_t kMaxTrackRetries = 50; static const int8_t kMaxTrackStartupRetries = 50; static const int kStartSleepTime = 30000; static const int kStopSleepTime = 30000; static const int kDumpLockRetries = 50; static const int kDumpLockSleep = 20000; // Maximum number of pending buffers allocated by OutputTrack::write() static const uint8_t kMaxOutputTrackBuffers = 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 } // ---------------------------------------------------------------------------- AudioFlinger::AudioFlinger() : BnAudioFlinger(), mAudioHardware(0), mA2dpAudioInterface(0), mA2dpEnabled(false), mNotifyA2dpChange(false), mForcedSpeakerCount(0), mA2dpDisableCount(0), mA2dpSuppressed(false), mForcedRoute(0), mRouteRestoreTime(0), mMusicMuteSaved(false) { mHardwareStatus = AUDIO_HW_IDLE; mAudioHardware = AudioHardwareInterface::create(); mHardwareStatus = AUDIO_HW_INIT; if (mAudioHardware->initCheck() == NO_ERROR) { // open 16-bit output stream for s/w mixer mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; status_t status; AudioStreamOut *hwOutput = mAudioHardware->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status); mHardwareStatus = AUDIO_HW_IDLE; if (hwOutput) { mHardwareMixerThread = new MixerThread(this, hwOutput, AudioSystem::AUDIO_OUTPUT_HARDWARE); } else { LOGE("Failed to initialize hardware output stream, status: %d", status); } #ifdef WITH_A2DP // Create A2DP interface mA2dpAudioInterface = new A2dpAudioInterface(); AudioStreamOut *a2dpOutput = mA2dpAudioInterface->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status); if (a2dpOutput) { mA2dpMixerThread = new MixerThread(this, a2dpOutput, AudioSystem::AUDIO_OUTPUT_A2DP); if (hwOutput) { uint32_t frameCount = ((a2dpOutput->bufferSize()/a2dpOutput->frameSize()) * hwOutput->sampleRate()) / a2dpOutput->sampleRate(); MixerThread::OutputTrack *a2dpOutTrack = new MixerThread::OutputTrack(mA2dpMixerThread, hwOutput->sampleRate(), AudioSystem::PCM_16_BIT, hwOutput->channelCount(), frameCount); mHardwareMixerThread->setOuputTrack(a2dpOutTrack); } } else { LOGE("Failed to initialize A2DP output stream, status: %d", status); } #endif // FIXME - this should come from settings setRouting(AudioSystem::MODE_NORMAL, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL); setRouting(AudioSystem::MODE_RINGTONE, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL); setRouting(AudioSystem::MODE_IN_CALL, AudioSystem::ROUTE_EARPIECE, AudioSystem::ROUTE_ALL); setMode(AudioSystem::MODE_NORMAL); setMasterVolume(1.0f); setMasterMute(false); // Start record thread mAudioRecordThread = new AudioRecordThread(mAudioHardware, this); if (mAudioRecordThread != 0) { mAudioRecordThread->run("AudioRecordThread", PRIORITY_URGENT_AUDIO); } } else { LOGE("Couldn't even initialize the stubbed audio hardware!"); } } AudioFlinger::~AudioFlinger() { if (mAudioRecordThread != 0) { mAudioRecordThread->exit(); mAudioRecordThread.clear(); } mHardwareMixerThread.clear(); delete mAudioHardware; // deleting mA2dpAudioInterface also deletes mA2dpOutput; #ifdef WITH_A2DP mA2dpMixerThread.clear(); delete mA2dpAudioInterface; #endif } #ifdef WITH_A2DP // setA2dpEnabled_l() must be called with AudioFlinger::mLock held void AudioFlinger::setA2dpEnabled_l(bool enable) { SortedVector < sp > tracks; SortedVector < wp > activeTracks; LOGV_IF(enable, "set output to A2DP\n"); LOGV_IF(!enable, "set output to hardware audio\n"); // Transfer tracks playing on MUSIC stream from one mixer to the other if (enable) { mHardwareMixerThread->getTracks_l(tracks, activeTracks); mA2dpMixerThread->putTracks_l(tracks, activeTracks); } else { mA2dpMixerThread->getTracks_l(tracks, activeTracks); mHardwareMixerThread->putTracks_l(tracks, activeTracks); } mA2dpEnabled = enable; mNotifyA2dpChange = true; mWaitWorkCV.broadcast(); } // checkA2dpEnabledChange_l() must be called with AudioFlinger::mLock held void AudioFlinger::checkA2dpEnabledChange_l() { if (mNotifyA2dpChange) { // Notify AudioSystem of the A2DP activation/deactivation size_t size = mNotificationClients.size(); for (size_t i = 0; i < size; i++) { sp binder = mNotificationClients.itemAt(i).promote(); if (binder != NULL) { LOGV("Notifying output change to client %p", binder.get()); sp client = interface_cast (binder); client->a2dpEnabledChanged(mA2dpEnabled); } } mNotifyA2dpChange = false; } } #endif // WITH_A2DP bool AudioFlinger::streamForcedToSpeaker(int streamType) { // NOTE that streams listed here must not be routed to A2DP by default: // AudioSystem::routedToA2dpOutput(streamType) == false return (streamType == AudioSystem::RING || streamType == AudioSystem::ALARM || streamType == AudioSystem::NOTIFICATION || streamType == AudioSystem::ENFORCED_AUDIBLE); } 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; if (hardwareStatus == AUDIO_HW_IDLE && mHardwareMixerThread->mStandby) { hardwareStatus = AUDIO_HW_STANDBY; } 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); mHardwareMixerThread->dump(fd, args); #ifdef WITH_A2DP mA2dpMixerThread->dump(fd, args); #endif // dump record client if (mAudioRecordThread != 0) mAudioRecordThread->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, status_t *status) { sp track; sp trackHandle; sp client; wp wclient; status_t lStatus; if (streamType >= AudioSystem::NUM_STREAM_TYPES) { LOGE("invalid stream type"); lStatus = BAD_VALUE; goto Exit; } { Mutex::Autolock _l(mLock); wclient = mClients.valueFor(pid); if (wclient != NULL) { client = wclient.promote(); } else { client = new Client(this, pid); mClients.add(pid, client); } #ifdef WITH_A2DP if (isA2dpEnabled() && AudioSystem::routedToA2dpOutput(streamType)) { track = mA2dpMixerThread->createTrack_l(client, streamType, sampleRate, format, channelCount, frameCount, sharedBuffer, &lStatus); } else #endif { track = mHardwareMixerThread->createTrack_l(client, streamType, sampleRate, format, channelCount, frameCount, sharedBuffer, &lStatus); } } if (lStatus == NO_ERROR) { trackHandle = new TrackHandle(track); } else { track.clear(); } Exit: if(status) { *status = lStatus; } return trackHandle; } uint32_t AudioFlinger::sampleRate(int output) const { #ifdef WITH_A2DP if (output == AudioSystem::AUDIO_OUTPUT_A2DP) { return mA2dpMixerThread->sampleRate(); } #endif return mHardwareMixerThread->sampleRate(); } int AudioFlinger::channelCount(int output) const { #ifdef WITH_A2DP if (output == AudioSystem::AUDIO_OUTPUT_A2DP) { return mA2dpMixerThread->channelCount(); } #endif return mHardwareMixerThread->channelCount(); } int AudioFlinger::format(int output) const { #ifdef WITH_A2DP if (output == AudioSystem::AUDIO_OUTPUT_A2DP) { return mA2dpMixerThread->format(); } #endif return mHardwareMixerThread->format(); } size_t AudioFlinger::frameCount(int output) const { #ifdef WITH_A2DP if (output == AudioSystem::AUDIO_OUTPUT_A2DP) { return mA2dpMixerThread->frameCount(); } #endif return mHardwareMixerThread->frameCount(); } uint32_t AudioFlinger::latency(int output) const { #ifdef WITH_A2DP if (output == AudioSystem::AUDIO_OUTPUT_A2DP) { return mA2dpMixerThread->latency(); } #endif return mHardwareMixerThread->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; mHardwareMixerThread->setMasterVolume(value); #ifdef WITH_A2DP mA2dpMixerThread->setMasterVolume(value); #endif return NO_ERROR; } status_t AudioFlinger::setRouting(int mode, uint32_t routes, uint32_t mask) { status_t err = NO_ERROR; // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } if ((mode < AudioSystem::MODE_CURRENT) || (mode >= AudioSystem::NUM_MODES)) { LOGW("Illegal value: setRouting(%d, %u, %u)", mode, routes, mask); return BAD_VALUE; } #ifdef WITH_A2DP LOGD("setRouting %d %d %d, tid %d, calling tid %d\n", mode, routes, mask, gettid(), IPCThreadState::self()->getCallingPid()); if (mode == AudioSystem::MODE_NORMAL && (mask & AudioSystem::ROUTE_BLUETOOTH_A2DP)) { AutoMutex lock(&mLock); bool enableA2dp = false; if (routes & AudioSystem::ROUTE_BLUETOOTH_A2DP) { enableA2dp = true; } if (mA2dpDisableCount > 0) { mA2dpSuppressed = enableA2dp; } else { setA2dpEnabled_l(enableA2dp); } LOGV("setOutput done\n"); } // setRouting() is always called at least for mode == AudioSystem::MODE_IN_CALL when // SCO is enabled, whatever current mode is so we can safely handle A2DP disabling only // in this case to avoid doing it several times. if (mode == AudioSystem::MODE_IN_CALL && (mask & AudioSystem::ROUTE_BLUETOOTH_SCO)) { AutoMutex lock(&mLock); handleRouteDisablesA2dp_l(routes); } #endif // do nothing if only A2DP routing is affected mask &= ~AudioSystem::ROUTE_BLUETOOTH_A2DP; if (mask) { AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_GET_ROUTING; uint32_t r; err = mAudioHardware->getRouting(mode, &r); if (err == NO_ERROR) { r = (r & ~mask) | (routes & mask); if (mode == AudioSystem::MODE_NORMAL || (mode == AudioSystem::MODE_CURRENT && getMode() == AudioSystem::MODE_NORMAL)) { mSavedRoute = r; r |= mForcedRoute; LOGV("setRouting mSavedRoute %08x mForcedRoute %08x\n", mSavedRoute, mForcedRoute); } mHardwareStatus = AUDIO_HW_SET_ROUTING; err = mAudioHardware->setRouting(mode, r); } mHardwareStatus = AUDIO_HW_IDLE; } return err; } uint32_t AudioFlinger::getRouting(int mode) const { uint32_t routes = 0; if ((mode >= AudioSystem::MODE_CURRENT) && (mode < AudioSystem::NUM_MODES)) { if (mode == AudioSystem::MODE_NORMAL || (mode == AudioSystem::MODE_CURRENT && getMode() == AudioSystem::MODE_NORMAL)) { routes = mSavedRoute; } else { mHardwareStatus = AUDIO_HW_GET_ROUTING; mAudioHardware->getRouting(mode, &routes); mHardwareStatus = AUDIO_HW_IDLE; } } else { LOGW("Illegal value: getRouting(%d)", mode); } return routes; } status_t AudioFlinger::setMode(int 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; } AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_SET_MODE; status_t ret = mAudioHardware->setMode(mode); mHardwareStatus = AUDIO_HW_IDLE; return ret; } int AudioFlinger::getMode() const { int mode = AudioSystem::MODE_INVALID; mHardwareStatus = AUDIO_HW_SET_MODE; mAudioHardware->getMode(&mode); mHardwareStatus = AUDIO_HW_IDLE; return mode; } 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; } mHardwareMixerThread->setMasterMute(muted); #ifdef WITH_A2DP mA2dpMixerThread->setMasterMute(muted); #endif return NO_ERROR; } float AudioFlinger::masterVolume() const { return mHardwareMixerThread->masterVolume(); } bool AudioFlinger::masterMute() const { return mHardwareMixerThread->masterMute(); } status_t AudioFlinger::setStreamVolume(int stream, float value) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES || uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) { return BAD_VALUE; } status_t ret = NO_ERROR; if (stream == AudioSystem::VOICE_CALL || stream == AudioSystem::BLUETOOTH_SCO) { float hwValue; if (stream == AudioSystem::VOICE_CALL) { hwValue = (float)AudioSystem::logToLinear(value)/100.0f; // offset value to reflect actual hardware volume that never reaches 0 // 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java) value = 0.01 + 0.99 * value; } else { // (type == AudioSystem::BLUETOOTH_SCO) hwValue = 1.0f; } AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_SET_VOICE_VOLUME; ret = mAudioHardware->setVoiceVolume(hwValue); mHardwareStatus = AUDIO_HW_IDLE; } mHardwareMixerThread->setStreamVolume(stream, value); #ifdef WITH_A2DP mA2dpMixerThread->setStreamVolume(stream, value); #endif return ret; } status_t AudioFlinger::setStreamMute(int stream, bool muted) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES || uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) { return BAD_VALUE; } #ifdef WITH_A2DP mA2dpMixerThread->setStreamMute(stream, muted); #endif if (stream == AudioSystem::MUSIC) { AutoMutex lock(&mHardwareLock); if (mForcedRoute != 0) mMusicMuteSaved = muted; else mHardwareMixerThread->setStreamMute(stream, muted); } else { mHardwareMixerThread->setStreamMute(stream, muted); } return NO_ERROR; } float AudioFlinger::streamVolume(int stream) const { if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { return 0.0f; } float volume = mHardwareMixerThread->streamVolume(stream); // remove correction applied by setStreamVolume() if (stream == AudioSystem::VOICE_CALL) { volume = (volume - 0.01) / 0.99 ; } return volume; } bool AudioFlinger::streamMute(int stream) const { if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { return true; } if (stream == AudioSystem::MUSIC && mForcedRoute != 0) { return mMusicMuteSaved; } return mHardwareMixerThread->streamMute(stream); } bool AudioFlinger::isMusicActive() const { #ifdef WITH_A2DP if (isA2dpEnabled()) { return mA2dpMixerThread->isMusicActive(); } #endif return mHardwareMixerThread->isMusicActive(); } status_t AudioFlinger::setParameter(const char* key, const char* value) { status_t result, result2; AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_SET_PARAMETER; LOGV("setParameter() key %s, value %s, tid %d, calling tid %d", key, value, gettid(), IPCThreadState::self()->getCallingPid()); result = mAudioHardware->setParameter(key, value); if (mA2dpAudioInterface) { result2 = mA2dpAudioInterface->setParameter(key, value); if (result2) result = result2; } mHardwareStatus = AUDIO_HW_IDLE; return result; } size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) { return mAudioHardware->getInputBufferSize(sampleRate, format, channelCount); } void AudioFlinger::registerClient(const sp& client) { LOGV("registerClient() %p, tid %d, calling tid %d", client.get(), gettid(), IPCThreadState::self()->getCallingPid()); Mutex::Autolock _l(mLock); sp binder = client->asBinder(); if (mNotificationClients.indexOf(binder) < 0) { LOGV("Adding notification client %p", binder.get()); binder->linkToDeath(this); mNotificationClients.add(binder); client->a2dpEnabledChanged(isA2dpEnabled()); } } void AudioFlinger::binderDied(const wp& who) { LOGV("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(), IPCThreadState::self()->getCallingPid()); Mutex::Autolock _l(mLock); IBinder *binder = who.unsafe_get(); if (binder != NULL) { int index = mNotificationClients.indexOf(binder); if (index >= 0) { LOGV("Removing notification client %p", binder); mNotificationClients.removeAt(index); } } } void AudioFlinger::removeClient(pid_t pid) { LOGV("removeClient() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); Mutex::Autolock _l(mLock); mClients.removeItem(pid); } bool AudioFlinger::isA2dpEnabled() const { return mA2dpEnabled; } void AudioFlinger::handleForcedSpeakerRoute(int command) { switch(command) { case ACTIVE_TRACK_ADDED: { AutoMutex lock(mHardwareLock); if (mForcedSpeakerCount++ == 0) { mRouteRestoreTime = 0; mMusicMuteSaved = mHardwareMixerThread->streamMute(AudioSystem::MUSIC); if (mForcedRoute == 0 && !(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) { LOGV("Route forced to Speaker ON %08x", mSavedRoute | AudioSystem::ROUTE_SPEAKER); mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, true); usleep(mHardwareMixerThread->latency()*1000); mHardwareStatus = AUDIO_HW_SET_ROUTING; mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute | AudioSystem::ROUTE_SPEAKER); mHardwareStatus = AUDIO_HW_IDLE; // delay track start so that audio hardware has time to siwtch routes usleep(kStartSleepTime); } mForcedRoute = AudioSystem::ROUTE_SPEAKER; } LOGV("mForcedSpeakerCount incremented to %d", mForcedSpeakerCount); } break; case ACTIVE_TRACK_REMOVED: { AutoMutex lock(mHardwareLock); if (mForcedSpeakerCount > 0){ if (--mForcedSpeakerCount == 0) { mRouteRestoreTime = systemTime() + milliseconds(kStopSleepTime/1000); } LOGV("mForcedSpeakerCount decremented to %d", mForcedSpeakerCount); } else { LOGE("mForcedSpeakerCount is already zero"); } } break; case CHECK_ROUTE_RESTORE_TIME: case FORCE_ROUTE_RESTORE: if (mRouteRestoreTime) { AutoMutex lock(mHardwareLock); if (mRouteRestoreTime && (systemTime() > mRouteRestoreTime || command == FORCE_ROUTE_RESTORE)) { mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, mMusicMuteSaved); mForcedRoute = 0; if (!(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) { mHardwareStatus = AUDIO_HW_SET_ROUTING; mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute); mHardwareStatus = AUDIO_HW_IDLE; LOGV("Route forced to Speaker OFF %08x", mSavedRoute); } mRouteRestoreTime = 0; } } break; } } #ifdef WITH_A2DP // handleRouteDisablesA2dp_l() must be called with AudioFlinger::mLock held void AudioFlinger::handleRouteDisablesA2dp_l(int routes) { if (routes & AudioSystem::ROUTE_BLUETOOTH_SCO) { if (mA2dpDisableCount++ == 0) { if (mA2dpEnabled) { setA2dpEnabled_l(false); mA2dpSuppressed = true; } } LOGV("mA2dpDisableCount incremented to %d", mA2dpDisableCount); } else { if (mA2dpDisableCount > 0) { if (--mA2dpDisableCount == 0) { if (mA2dpSuppressed) { setA2dpEnabled_l(true); mA2dpSuppressed = false; } } LOGV("mA2dpDisableCount decremented to %d", mA2dpDisableCount); } else { LOGE("mA2dpDisableCount is already zero"); } } } #endif // ---------------------------------------------------------------------------- AudioFlinger::MixerThread::MixerThread(const sp& audioFlinger, AudioStreamOut* output, int outputType) : Thread(false), mAudioFlinger(audioFlinger), mAudioMixer(0), mOutput(output), mOutputType(outputType), mSampleRate(0), mFrameCount(0), mChannelCount(0), mFormat(0), mMixBuffer(0), mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mStandby(false), mInWrite(false) { mSampleRate = output->sampleRate(); mChannelCount = output->channelCount(); // FIXME - Current mixer implementation only supports stereo output if (mChannelCount == 1) { LOGE("Invalid audio hardware channel count"); } mFormat = output->format(); mFrameCount = output->bufferSize() / output->channelCount() / sizeof(int16_t); mAudioMixer = new AudioMixer(mFrameCount, output->sampleRate()); // FIXME - Current mixer implementation only supports stereo output: Always // Allocate a stereo buffer even if HW output is mono. mMixBuffer = new int16_t[mFrameCount * 2]; memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); } AudioFlinger::MixerThread::~MixerThread() { delete [] mMixBuffer; delete mAudioMixer; } status_t AudioFlinger::MixerThread::dump(int fd, const Vector& args) { dumpInternals(fd, args); dumpTracks(fd, args); return NO_ERROR; } status_t AudioFlinger::MixerThread::dumpTracks(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "Output %d mixer thread tracks\n", mOutputType); result.append(buffer); result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\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 %d mixer thread active tracks\n", mOutputType); result.append(buffer); result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\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::MixerThread::dumpInternals(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "Output %d mixer thread internals\n", mOutputType); result.append(buffer); snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 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, "standby: %d\n", mStandby); result.append(buffer); write(fd, result.string(), result.size()); return NO_ERROR; } // Thread virtuals bool AudioFlinger::MixerThread::threadLoop() { unsigned long sleepTime = kBufferRecoveryInUsecs; int16_t* curBuf = mMixBuffer; Vector< sp > tracksToRemove; size_t enabledTracks = 0; nsecs_t standbyTime = systemTime(); size_t mixBufferSize = mFrameCount*mChannelCount*sizeof(int16_t); nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 2; #ifdef WITH_A2DP bool outputTrackActive = false; #endif do { enabledTracks = 0; { // scope for the AudioFlinger::mLock Mutex::Autolock _l(mAudioFlinger->mLock); #ifdef WITH_A2DP if (mOutputTrack != NULL && !mAudioFlinger->isA2dpEnabled()) { if (outputTrackActive) { mAudioFlinger->mLock.unlock(); mOutputTrack->stop(); mAudioFlinger->mLock.lock(); outputTrackActive = false; } } mAudioFlinger->checkA2dpEnabledChange_l(); #endif const SortedVector< wp >& activeTracks = mActiveTracks; // put audio hardware into standby after short delay if UNLIKELY(!activeTracks.size() && systemTime() > standbyTime) { // wait until we have something to do... LOGV("Audio hardware entering standby, output %d\n", mOutputType); if (!mStandby) { mOutput->standby(); mStandby = true; } #ifdef WITH_A2DP if (outputTrackActive) { mAudioFlinger->mLock.unlock(); mOutputTrack->stop(); mAudioFlinger->mLock.lock(); outputTrackActive = false; } #endif if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) { mAudioFlinger->handleForcedSpeakerRoute(FORCE_ROUTE_RESTORE); } // we're about to wait, flush the binder command buffer IPCThreadState::self()->flushCommands(); mAudioFlinger->mWaitWorkCV.wait(mAudioFlinger->mLock); LOGV("Audio hardware exiting standby, output %d\n", mOutputType); 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; continue; } // Forced route to speaker is handled by hardware mixer thread if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) { mAudioFlinger->handleForcedSpeakerRoute(CHECK_ROUTE_RESTORE_TIME); } // find out which tracks need to be processed size_t count = activeTracks.size(); 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->isStopped()) && !track->isPaused()) { //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); // compute volume for this track int16_t left, right; if (track->isMuted() || mMasterMute || track->isPausing()) { left = right = 0; if (track->isPausing()) { LOGV("paused(%d)", track->name()); 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 = int16_t(v_clamped); v_clamped = v * cblk->volume[1]; if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; right = int16_t(v_clamped); } // XXX: these things DON'T need to be done each time mAudioMixer->setBufferProvider(track); mAudioMixer->enable(AudioMixer::MIXING); int param; 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 { param = AudioMixer::VOLUME; } } else { param = AudioMixer::RAMP_VOLUME; } mAudioMixer->setParameter(param, AudioMixer::VOLUME0, left); mAudioMixer->setParameter(param, AudioMixer::VOLUME1, right); mAudioMixer->setParameter( AudioMixer::TRACK, AudioMixer::FORMAT, track->format()); mAudioMixer->setParameter( AudioMixer::TRACK, AudioMixer::CHANNEL_COUNT, track->channelCount()); mAudioMixer->setParameter( AudioMixer::RESAMPLE, AudioMixer::SAMPLE_RATE, int(cblk->sampleRate)); // reset retry count track->mRetryCount = kMaxTrackRetries; enabledTracks++; } 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. LOGV("remove(%d) from active list", track->name()); 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", track->name()); tracksToRemove.add(track); } } // LOGV("disable(%d)", track->name()); 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[i]; removeActiveTrack_l(track); if (track->isTerminated()) { mTracks.remove(track); deleteTrackName_l(track->mName); } } } } if (LIKELY(enabledTracks)) { // mix buffers... mAudioMixer->process(curBuf); #ifdef WITH_A2DP if (mOutputTrack != NULL && mAudioFlinger->isA2dpEnabled()) { if (!outputTrackActive) { LOGV("starting output track in mixer for output %d", mOutputType); mOutputTrack->start(); outputTrackActive = true; } mOutputTrack->write(curBuf, mFrameCount); } #endif // output audio to hardware mLastWriteTime = systemTime(); mInWrite = true; mOutput->write(curBuf, mixBufferSize); mNumWrites++; mInWrite = false; mStandby = false; nsecs_t temp = systemTime(); standbyTime = temp + kStandbyTimeInNsecs; nsecs_t delta = temp - mLastWriteTime; if (delta > maxPeriod) { LOGW("write blocked for %llu msecs", ns2ms(delta)); mNumDelayedWrites++; } sleepTime = kBufferRecoveryInUsecs; } else { #ifdef WITH_A2DP if (mOutputTrack != NULL && mAudioFlinger->isA2dpEnabled()) { if (outputTrackActive) { mOutputTrack->write(curBuf, 0); if (mOutputTrack->bufferQueueEmpty()) { mOutputTrack->stop(); outputTrackActive = false; } else { standbyTime = systemTime() + kStandbyTimeInNsecs; } } } #endif // There was nothing to mix this round, which means all // active tracks were late. Sleep a little bit to give // them another chance. If we're too late, the audio // hardware will zero-fill for us. //LOGV("no buffers - usleep(%lu)", sleepTime); usleep(sleepTime); if (sleepTime < kMaxBufferRecoveryInUsecs) { sleepTime += kBufferRecoveryInUsecs; } } // 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(); } while (true); return false; } status_t AudioFlinger::MixerThread::readyToRun() { if (mSampleRate == 0) { LOGE("No working audio driver found."); return NO_INIT; } LOGI("AudioFlinger's thread ready to run for output %d", mOutputType); return NO_ERROR; } void AudioFlinger::MixerThread::onFirstRef() { const size_t SIZE = 256; char buffer[SIZE]; snprintf(buffer, SIZE, "Mixer Thread for output %d", mOutputType); run(buffer, ANDROID_PRIORITY_URGENT_AUDIO); } // MixerThread::createTrack_l() must be called with AudioFlinger::mLock held sp AudioFlinger::MixerThread::createTrack_l( const sp& client, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, const sp& sharedBuffer, status_t *status) { sp track; status_t lStatus; // Resampler implementation limits input sampling rate to 2 x output sampling rate. if (sampleRate > MAX_SAMPLE_RATE || sampleRate > mSampleRate*2) { LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); lStatus = BAD_VALUE; goto Exit; } if (mSampleRate == 0) { LOGE("Audio driver not initialized."); lStatus = NO_INIT; goto Exit; } track = new Track(this, client, streamType, sampleRate, format, channelCount, frameCount, sharedBuffer); if (track->getCblk() == NULL) { lStatus = NO_MEMORY; goto Exit; } mTracks.add(track); lStatus = NO_ERROR; Exit: if(status) { *status = lStatus; } return track; } // getTracks_l() must be called with AudioFlinger::mLock held void AudioFlinger::MixerThread::getTracks_l( SortedVector < sp >& tracks, SortedVector < wp >& activeTracks) { size_t size = mTracks.size(); LOGV ("MixerThread::getTracks_l() for output %d, mTracks.size %d, mActiveTracks.size %d", mOutputType, mTracks.size(), mActiveTracks.size()); for (size_t i = 0; i < size; i++) { sp t = mTracks[i]; if (AudioSystem::routedToA2dpOutput(t->mStreamType)) { tracks.add(t); int j = mActiveTracks.indexOf(t); if (j >= 0) { t = mActiveTracks[j].promote(); if (t != NULL) { activeTracks.add(t); } } } } size = activeTracks.size(); for (size_t i = 0; i < size; i++) { removeActiveTrack_l(activeTracks[i]); } size = tracks.size(); for (size_t i = 0; i < size; i++) { sp t = tracks[i]; mTracks.remove(t); deleteTrackName_l(t->name()); } } // putTracks_l() must be called with AudioFlinger::mLock held void AudioFlinger::MixerThread::putTracks_l( SortedVector < sp >& tracks, SortedVector < wp >& activeTracks) { LOGV ("MixerThread::putTracks_l() for output %d, tracks.size %d, activeTracks.size %d", mOutputType, tracks.size(), activeTracks.size()); size_t size = tracks.size(); for (size_t i = 0; i < size ; i++) { sp t = tracks[i]; int name = getTrackName_l(); if (name < 0) return; t->mName = name; t->mMixerThread = this; mTracks.add(t); int j = activeTracks.indexOf(t); if (j >= 0) { addActiveTrack_l(t); } } } uint32_t AudioFlinger::MixerThread::sampleRate() const { return mSampleRate; } int AudioFlinger::MixerThread::channelCount() const { return mChannelCount; } int AudioFlinger::MixerThread::format() const { return mFormat; } size_t AudioFlinger::MixerThread::frameCount() const { return mFrameCount; } uint32_t AudioFlinger::MixerThread::latency() const { if (mOutput) { return mOutput->latency(); } else { return 0; } } status_t AudioFlinger::MixerThread::setMasterVolume(float value) { mMasterVolume = value; return NO_ERROR; } status_t AudioFlinger::MixerThread::setMasterMute(bool muted) { mMasterMute = muted; return NO_ERROR; } float AudioFlinger::MixerThread::masterVolume() const { return mMasterVolume; } bool AudioFlinger::MixerThread::masterMute() const { return mMasterMute; } status_t AudioFlinger::MixerThread::setStreamVolume(int stream, float value) { mStreamTypes[stream].volume = value; return NO_ERROR; } status_t AudioFlinger::MixerThread::setStreamMute(int stream, bool muted) { mStreamTypes[stream].mute = muted; return NO_ERROR; } float AudioFlinger::MixerThread::streamVolume(int stream) const { return mStreamTypes[stream].volume; } bool AudioFlinger::MixerThread::streamMute(int stream) const { return mStreamTypes[stream].mute; } bool AudioFlinger::MixerThread::isMusicActive() const { 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->mStreamType == AudioSystem::MUSIC) return true; } return false; } // addTrack_l() must be called with AudioFlinger::mLock held status_t AudioFlinger::MixerThread::addTrack_l(const sp& track) { status_t status = ALREADY_EXISTS; // here the track could be either new, or restarted // in both cases "unstop" the track if (track->isPaused()) { track->mState = TrackBase::RESUMING; LOGV("PAUSED => RESUMING (%d)", track->name()); } else { track->mState = TrackBase::ACTIVE; LOGV("? => ACTIVE (%d)", track->name()); } // 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; addActiveTrack_l(track); status = NO_ERROR; } LOGV("mWaitWorkCV.broadcast"); mAudioFlinger->mWaitWorkCV.broadcast(); return status; } // destroyTrack_l() must be called with AudioFlinger::mLock held void AudioFlinger::MixerThread::destroyTrack_l(const sp& track) { track->mState = TrackBase::TERMINATED; if (mActiveTracks.indexOf(track) < 0) { LOGV("remove track (%d) and delete from mixer", track->name()); mTracks.remove(track); deleteTrackName_l(track->name()); } } // addActiveTrack_l() must be called with AudioFlinger::mLock held void AudioFlinger::MixerThread::addActiveTrack_l(const wp& t) { mActiveTracks.add(t); // Force routing to speaker for certain stream types // The forced routing to speaker is managed by hardware mixer if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) { sp track = t.promote(); if (track == NULL) return; if (streamForcedToSpeaker(track->type())) { mAudioFlinger->handleForcedSpeakerRoute(ACTIVE_TRACK_ADDED); } } } // removeActiveTrack_l() must be called with AudioFlinger::mLock held void AudioFlinger::MixerThread::removeActiveTrack_l(const wp& t) { mActiveTracks.remove(t); // Force routing to speaker for certain stream types // The forced routing to speaker is managed by hardware mixer if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) { sp track = t.promote(); if (track == NULL) return; if (streamForcedToSpeaker(track->type())) { mAudioFlinger->handleForcedSpeakerRoute(ACTIVE_TRACK_REMOVED); } } } // getTrackName_l() must be called with AudioFlinger::mLock held int AudioFlinger::MixerThread::getTrackName_l() { return mAudioMixer->getTrackName(); } // deleteTrackName_l() must be called with AudioFlinger::mLock held void AudioFlinger::MixerThread::deleteTrackName_l(int name) { mAudioMixer->deleteTrackName(name); } size_t AudioFlinger::MixerThread::getOutputFrameCount() { return mOutput->bufferSize() / mOutput->channelCount() / sizeof(int16_t); } // ---------------------------------------------------------------------------- // TrackBase constructor must be called with AudioFlinger::mLock held AudioFlinger::MixerThread::TrackBase::TrackBase( const sp& mixerThread, const sp& client, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, const sp& sharedBuffer) : RefBase(), mMixerThread(mixerThread), mClient(client), mStreamType(streamType), mFrameCount(0), mState(IDLE), mClientTid(-1), mFormat(format), mFlags(flags & ~SYSTEM_FLAGS_MASK) { mName = mixerThread->getTrackName_l(); LOGV("TrackBase contructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); if (mName < 0) { LOGE("no more track names availlable"); return; } 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 = frameCount*channelCount*sizeof(int16_t); 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 = (uint16_t)sampleRate; mCblk->channels = (uint16_t)channelCount; if (sharedBuffer == 0) { 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 mCblk->flowControlFlag = 1; } 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 = (uint16_t)sampleRate; mCblk->channels = (uint16_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 mCblk->flowControlFlag = 1; mBufferEnd = (uint8_t *)mBuffer + bufferSize; } } } AudioFlinger::MixerThread::TrackBase::~TrackBase() { if (mCblk) { mCblk->~audio_track_cblk_t(); // destroy our shared-structure. } mCblkMemory.clear(); // and free the shared memory mClient.clear(); } void AudioFlinger::MixerThread::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) { buffer->raw = 0; mFrameCount = buffer->frameCount; step(); buffer->frameCount = 0; } bool AudioFlinger::MixerThread::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::MixerThread::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::MixerThread::TrackBase::getCblk() const { return mCblkMemory; } int AudioFlinger::MixerThread::TrackBase::sampleRate() const { return (int)mCblk->sampleRate; } int AudioFlinger::MixerThread::TrackBase::channelCount() const { return mCblk->channels; } void* AudioFlinger::MixerThread::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { audio_track_cblk_t* cblk = this->cblk(); int16_t *bufferStart = (int16_t *)mBuffer + (offset-cblk->serverBase)*cblk->channels; int16_t *bufferEnd = bufferStart + frames * cblk->channels; // Check validity of returned pointer in case the track control block would have been corrupted. if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || (cblk->channels == 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, channels %d", bufferStart, bufferEnd, mBuffer, mBufferEnd, cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channels); return 0; } return bufferStart; } // ---------------------------------------------------------------------------- // Track constructor must be called with AudioFlinger::mLock held AudioFlinger::MixerThread::Track::Track( const sp& mixerThread, const sp& client, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, const sp& sharedBuffer) : TrackBase(mixerThread, client, streamType, sampleRate, format, channelCount, frameCount, 0, sharedBuffer) { mVolume[0] = 1.0f; mVolume[1] = 1.0f; mMute = false; mSharedBuffer = sharedBuffer; } AudioFlinger::MixerThread::Track::~Track() { wp weak(this); // never create a strong ref from the dtor Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock); mState = TERMINATED; } void AudioFlinger::MixerThread::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 AudioFlinger::mLock, // we must acquire a strong reference on this Track before locking AudioFlinger::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 AudioFlinger::mLock Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock); mMixerThread->destroyTrack_l(this); } } void AudioFlinger::MixerThread::Track::dump(char* buffer, size_t size) { snprintf(buffer, size, " %5d %5d %3u %3u %3u %3u %1d %1d %1d %5u %5u %5u %04x %04x\n", mName - AudioMixer::TRACK0, (mClient == NULL) ? getpid() : mClient->pid(), mStreamType, mFormat, mCblk->channels, mFrameCount, mState, mMute, mFillingUpStatus, mCblk->sampleRate, mCblk->volume[0], mCblk->volume[1], mCblk->server, mCblk->user); } status_t AudioFlinger::MixerThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) { audio_track_cblk_t* cblk = this->cblk(); uint32_t framesReady; 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; } 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; return NOT_ENOUGH_DATA; } bool AudioFlinger::MixerThread::Track::isReady() const { if (mFillingUpStatus != FS_FILLING) return true; if (mCblk->framesReady() >= mCblk->frameCount || mCblk->forceReady) { mFillingUpStatus = FS_FILLED; mCblk->forceReady = 0; LOGV("Track::isReady() track %d for output %d", mName, mMixerThread->mOutputType); return true; } return false; } status_t AudioFlinger::MixerThread::Track::start() { LOGV("start(%d), calling thread %d for output %d", mName, IPCThreadState::self()->getCallingPid(), mMixerThread->mOutputType); Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock); mMixerThread->addTrack_l(this); return NO_ERROR; } void AudioFlinger::MixerThread::Track::stop() { LOGV("stop(%d), calling thread %d for output %d", mName, IPCThreadState::self()->getCallingPid(), mMixerThread->mOutputType); Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock); if (mState > STOPPED) { mState = STOPPED; // If the track is not active (PAUSED and buffers full), flush buffers if (mMixerThread->mActiveTracks.indexOf(this) < 0) { reset(); } LOGV("(> STOPPED) => STOPPED (%d)", mName); } } void AudioFlinger::MixerThread::Track::pause() { LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock); if (mState == ACTIVE || mState == RESUMING) { mState = PAUSING; LOGV("ACTIVE/RESUMING => PAUSING (%d)", mName); } } void AudioFlinger::MixerThread::Track::flush() { LOGV("flush(%d)", mName); Mutex::Autolock _l(mMixerThread->mAudioFlinger->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::MixerThread::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->flowControlFlag = 1; mCblk->forceReady = 0; mFillingUpStatus = FS_FILLING; mResetDone = true; } } void AudioFlinger::MixerThread::Track::mute(bool muted) { mMute = muted; } void AudioFlinger::MixerThread::Track::setVolume(float left, float right) { mVolume[0] = left; mVolume[1] = right; } // ---------------------------------------------------------------------------- // RecordTrack constructor must be called with AudioFlinger::mLock held AudioFlinger::MixerThread::RecordTrack::RecordTrack( const sp& mixerThread, const sp& client, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags) : TrackBase(mixerThread, client, streamType, sampleRate, format, channelCount, frameCount, flags, 0), mOverflow(false) { } AudioFlinger::MixerThread::RecordTrack::~RecordTrack() { Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock); mMixerThread->deleteTrackName_l(mName); } status_t AudioFlinger::MixerThread::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::MixerThread::RecordTrack::start() { return mMixerThread->mAudioFlinger->startRecord(this); } void AudioFlinger::MixerThread::RecordTrack::stop() { mMixerThread->mAudioFlinger->stopRecord(this); TrackBase::reset(); // Force overerrun condition to avoid false overrun callback until first data is // read from buffer mCblk->flowControlFlag = 1; } // ---------------------------------------------------------------------------- AudioFlinger::MixerThread::OutputTrack::OutputTrack( const sp& mixerThread, uint32_t sampleRate, int format, int channelCount, int frameCount) : Track(mixerThread, NULL, AudioSystem::SYSTEM, sampleRate, format, channelCount, frameCount, NULL), mOutputMixerThread(mixerThread) { mCblk->out = 1; mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); mCblk->volume[0] = mCblk->volume[1] = 0x1000; mOutBuffer.frameCount = 0; mCblk->bufferTimeoutMs = 10; LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channels %d mBufferEnd %p", mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channels, mBufferEnd); } AudioFlinger::MixerThread::OutputTrack::~OutputTrack() { stop(); } status_t AudioFlinger::MixerThread::OutputTrack::start() { status_t status = Track::start(); mRetryCount = 127; return status; } void AudioFlinger::MixerThread::OutputTrack::stop() { Track::stop(); clearBufferQueue(); mOutBuffer.frameCount = 0; } void AudioFlinger::MixerThread::OutputTrack::write(int16_t* data, uint32_t frames) { Buffer *pInBuffer; Buffer inBuffer; uint32_t channels = mCblk->channels; inBuffer.frameCount = frames; inBuffer.i16 = data; if (mCblk->user == 0) { if (mOutputMixerThread->isMusicActive()) { mCblk->forceReady = 1; LOGV("OutputTrack::start() force ready"); } else if (mCblk->frameCount > frames){ if (mBufferQueue.size() < kMaxOutputTrackBuffers) { uint32_t startFrames = (mCblk->frameCount - frames); LOGV("OutputTrack::start() write %d frames", startFrames); pInBuffer = new Buffer; pInBuffer->mBuffer = new int16_t[startFrames * channels]; pInBuffer->frameCount = startFrames; pInBuffer->i16 = pInBuffer->mBuffer; memset(pInBuffer->raw, 0, startFrames * channels * sizeof(int16_t)); mBufferQueue.add(pInBuffer); } else { LOGW ("OutputTrack::write() no more buffers"); } } } while (1) { // 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; if (obtainBuffer(&mOutBuffer) == (status_t)AudioTrack::NO_MORE_BUFFERS) { break; } } uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channels * sizeof(int16_t)); mCblk->stepUser(outFrames); pInBuffer->frameCount -= outFrames; pInBuffer->i16 += outFrames * channels; mOutBuffer.frameCount -= outFrames; mOutBuffer.i16 += outFrames * channels; if (pInBuffer->frameCount == 0) { if (mBufferQueue.size()) { mBufferQueue.removeAt(0); delete [] pInBuffer->mBuffer; delete pInBuffer; } else { break; } } } // If we could not write all frames, allocate a buffer and queue it for next time. if (inBuffer.frameCount) { if (mBufferQueue.size() < kMaxOutputTrackBuffers) { pInBuffer = new Buffer; pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channels]; pInBuffer->frameCount = inBuffer.frameCount; pInBuffer->i16 = pInBuffer->mBuffer; memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channels * sizeof(int16_t)); mBufferQueue.add(pInBuffer); } else { LOGW("OutputTrack::write() no more buffers"); } } // 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 && mCblk->user < mCblk->frameCount) { frames = mCblk->frameCount - mCblk->user; pInBuffer = new Buffer; pInBuffer->mBuffer = new int16_t[frames * channels]; pInBuffer->frameCount = frames; pInBuffer->i16 = pInBuffer->mBuffer; memset(pInBuffer->raw, 0, frames * channels * sizeof(int16_t)); mBufferQueue.add(pInBuffer); } } status_t AudioFlinger::MixerThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer) { int active; int timeout = 0; 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) { 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::MixerThread::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)), mPid(pid) { // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer } AudioFlinger::Client::~Client() { mAudioFlinger->removeClient(mPid); } const sp& AudioFlinger::Client::heap() const { return mMemoryDealer; } // ---------------------------------------------------------------------------- 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::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 streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, status_t *status) { sp recordTrack; sp recordHandle; sp client; wp wclient; AudioStreamIn* input = 0; int inFrameCount; size_t inputBufferSize; status_t lStatus; // check calling permissions if (!recordingAllowed()) { lStatus = PERMISSION_DENIED; goto Exit; } if (uint32_t(streamType) >= AudioRecord::NUM_STREAM_TYPES) { LOGE("invalid stream type"); lStatus = BAD_VALUE; goto Exit; } if (sampleRate > MAX_SAMPLE_RATE) { LOGE("Sample rate out of range"); lStatus = BAD_VALUE; goto Exit; } if (mAudioRecordThread == 0) { LOGE("Audio record thread not started"); lStatus = NO_INIT; 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); wclient = mClients.valueFor(pid); if (wclient != NULL) { client = wclient.promote(); } else { client = new Client(this, pid); mClients.add(pid, client); } // frameCount must be a multiple of input buffer size inFrameCount = inputBufferSize/channelCount/sizeof(short); frameCount = ((frameCount - 1)/inFrameCount + 1) * inFrameCount; // create new record track. The record track uses one track in mHardwareMixerThread by convention. recordTrack = new MixerThread::RecordTrack(mHardwareMixerThread, client, streamType, sampleRate, format, channelCount, frameCount, flags); } if (recordTrack->getCblk() == NULL) { 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; } status_t AudioFlinger::startRecord(MixerThread::RecordTrack* recordTrack) { if (mAudioRecordThread != 0) { return mAudioRecordThread->start(recordTrack); } return NO_INIT; } void AudioFlinger::stopRecord(MixerThread::RecordTrack* recordTrack) { if (mAudioRecordThread != 0) { mAudioRecordThread->stop(recordTrack); } } // ---------------------------------------------------------------------------- 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::AudioRecordThread::AudioRecordThread(AudioHardwareInterface* audioHardware, const sp& audioFlinger) : mAudioHardware(audioHardware), mAudioFlinger(audioFlinger), mActive(false) { } AudioFlinger::AudioRecordThread::~AudioRecordThread() { } bool AudioFlinger::AudioRecordThread::threadLoop() { LOGV("AudioRecordThread: start record loop"); AudioBufferProvider::Buffer buffer; int inBufferSize = 0; int inFrameCount = 0; AudioStreamIn* input = 0; mActive = 0; // start recording while (!exitPending()) { if (!mActive) { mLock.lock(); if (!mActive && !exitPending()) { LOGV("AudioRecordThread: loop stopping"); if (input) { delete input; input = 0; } mRecordTrack.clear(); mStopped.signal(); mWaitWorkCV.wait(mLock); LOGV("AudioRecordThread: loop starting"); if (mRecordTrack != 0) { input = mAudioHardware->openInputStream(mRecordTrack->format(), mRecordTrack->channelCount(), mRecordTrack->sampleRate(), &mStartStatus, (AudioSystem::audio_in_acoustics)(mRecordTrack->mFlags >> 16)); if (input != 0) { inBufferSize = input->bufferSize(); inFrameCount = inBufferSize/input->frameSize(); } } else { mStartStatus = NO_INIT; } if (mStartStatus !=NO_ERROR) { LOGW("record start failed, status %d", mStartStatus); mActive = false; mRecordTrack.clear(); } mWaitWorkCV.signal(); } mLock.unlock(); } else if (mRecordTrack != 0) { buffer.frameCount = inFrameCount; if (LIKELY(mRecordTrack->getNextBuffer(&buffer) == NO_ERROR && (int)buffer.frameCount == inFrameCount)) { LOGV("AudioRecordThread read: %d frames", buffer.frameCount); ssize_t bytesRead = input->read(buffer.raw, inBufferSize); if (bytesRead < 0) { LOGE("Error reading audio input"); sleep(1); } mRecordTrack->releaseBuffer(&buffer); mRecordTrack->overflow(); } // client isn't retrieving buffers fast enough else { if (!mRecordTrack->setOverflow()) LOGW("AudioRecordThread: buffer overflow"); // 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 (input) { delete input; } mRecordTrack.clear(); return false; } status_t AudioFlinger::AudioRecordThread::start(MixerThread::RecordTrack* recordTrack) { LOGV("AudioRecordThread::start"); AutoMutex lock(&mLock); mActive = true; // If starting the active track, just reset mActive in case a stop // was pending and exit if (recordTrack == mRecordTrack.get()) return NO_ERROR; if (mRecordTrack != 0) return -EBUSY; mRecordTrack = recordTrack; // signal thread to start LOGV("Signal record thread"); mWaitWorkCV.signal(); mWaitWorkCV.wait(mLock); LOGV("Record started, status %d", mStartStatus); return mStartStatus; } void AudioFlinger::AudioRecordThread::stop(MixerThread::RecordTrack* recordTrack) { LOGV("AudioRecordThread::stop"); AutoMutex lock(&mLock); if (mActive && (recordTrack == mRecordTrack.get())) { mActive = false; mStopped.wait(mLock); } } void AudioFlinger::AudioRecordThread::exit() { LOGV("AudioRecordThread::exit"); { AutoMutex lock(&mLock); requestExit(); mWaitWorkCV.signal(); } requestExitAndWait(); } status_t AudioFlinger::AudioRecordThread::dump(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; pid_t pid = 0; if (mRecordTrack != 0 && mRecordTrack->mClient != 0) { snprintf(buffer, SIZE, "Record client pid: %d\n", mRecordTrack->mClient->pid()); result.append(buffer); } else { result.append("No record client\n"); } write(fd, result.string(), result.size()); return NO_ERROR; } status_t AudioFlinger::onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { return BnAudioFlinger::onTransact(code, data, reply, flags); } // ---------------------------------------------------------------------------- void AudioFlinger::instantiate() { defaultServiceManager()->addService( String16("media.audio_flinger"), new AudioFlinger()); } }; // namespace android