/* //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 #ifdef LVMX #include "lifevibes.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 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; // 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 } // ---------------------------------------------------------------------------- AudioFlinger::AudioFlinger() : BnAudioFlinger(), mAudioHardware(0), mMasterVolume(1.0f), mMasterMute(false), mNextThreadId(0) { 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 setMode(AudioSystem::MODE_NORMAL); 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 (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, 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); PlaybackThread *thread = checkPlaybackThread_l(output); 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); } track = thread->createTrack_l(client, streamType, sampleRate, format, channelCount, frameCount, sharedBuffer, &lStatus); } 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; } 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; 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) { // 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); #ifdef LVMX if (NO_ERROR == ret) { LifeVibes::setMode(mode); } #endif mHardwareStatus = AUDIO_HW_IDLE; 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); 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; } } 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 // ioHandle == 0 means the parameters are global to the audio hardware interface if (ioHandle == 0) { AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_SET_PARAMETER; result = mAudioHardware->setParameters(keyValuePairs); #ifdef LVMX if (musicEnabled != -1) { LifeVibes::enableMusic((bool) musicEnabled); } #endif mHardwareStatus = AUDIO_HW_IDLE; 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) { return mAudioHardware->getParameters(keys); } 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; } void AudioFlinger::registerClient(const sp& client) { Mutex::Autolock _l(mLock); int pid = IPCThreadState::self()->getCallingPid(); if (mNotificationClients.indexOfKey(pid) < 0) { sp notificationClient = new NotificationClient(this, client, pid); LOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); mNotificationClients.add(pid, notificationClient); sp binder = client->asBinder(); 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); } } } void AudioFlinger::removeNotificationClient(pid_t pid) { Mutex::Autolock _l(mLock); int index = mNotificationClients.indexOfKey(pid); if (index >= 0) { sp client = mNotificationClients.valueFor(pid); LOGV("removeNotificationClient() %p, pid %d", client.get(), pid); #ifdef LVMX if (pid == mLifeVibesClientPid) { LOGV("Disabling lifevibes"); LifeVibes::enableMusic(false); mLifeVibesClientPid = -1; } #endif mNotificationClients.removeItem(pid); } } // audioConfigChanged_l() must be called with AudioFlinger::mLock held void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) { 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::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) : ThreadBase(audioFlinger, id), mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output), mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) { 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); 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 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 thread %p active tracks\n", this); 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::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); 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, status_t *status) { sp track; status_t lStatus; 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); track = new Track(this, client, streamType, sampleRate, format, channelCount, frameCount, sharedBuffer); if (track->getCblk() == NULL || track->name() < 0) { lStatus = NO_MEMORY; goto Exit; } mTracks.add(track); } 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); 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; } mAudioFlinger->audioConfigChanged_l(event, mId, param2); } 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)); } 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); } // ---------------------------------------------------------------------------- AudioFlinger::MixerThread::MixerThread(const sp& audioFlinger, AudioStreamOut* output, int id) : PlaybackThread(audioFlinger, output, id), 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() { int16_t* curBuf = mMixBuffer; 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; 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); } if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { // mix buffers... mAudioMixer->process(curBuf); sleepTime = 0; standbyTime = systemTime() + kStandbyTimeInNsecs; } 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 (curBuf, 0, mixBufferSize); sleepTime = 0; LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); } } if (mSuspended) { sleepTime = idleSleepTime; } // sleepTime == 0 means we must write to audio hardware if (sleepTime == 0) { mLastWriteTime = systemTime(); mInWrite = true; mBytesWritten += mixBufferSize; #ifdef LVMX int audioOutputType = LifeVibes::getMixerType(mId, mType); if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { LifeVibes::process(audioOutputType, curBuf, mixBufferSize); } #endif int bytesWritten = (int)mOutput->write(curBuf, 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 { 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(); } if (!mStandby) { mOutput->standby(); } 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(); float masterVolume = mMasterVolume; bool masterMute = mMasterMute; #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 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() && !track->isTerminated()) { //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); // compute volume for this track int16_t left, right; if (track->isMuted() || masterMute || track->isPausing() || mStreamTypes[track->type()].mute) { left = right = 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; 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 = 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; } #ifdef LVMX if ( tracksConnectedChanged || stateChanged ) { // only do the ramp when the volume is changed by the user / application param = AudioMixer::VOLUME; } #endif 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; 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); mAudioMixer->disable(AudioMixer::MIXING); } 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); } 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->isTerminated()) { mTracks.remove(track); deleteTrackName_l(track->mName); } } } 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) { LOGV("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; 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 (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; } // ---------------------------------------------------------------------------- AudioFlinger::DirectOutputThread::DirectOutputThread(const sp& audioFlinger, AudioStreamOut* output, int id) : PlaybackThread(audioFlinger, output, id), mLeftVolume (1.0), mRightVolume(1.0) { mType = PlaybackThread::DIRECT; } AudioFlinger::DirectOutputThread::~DirectOutputThread() { } 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()) { 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; } } // 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->isStopped()) && !track->isPaused() && !track->isTerminated()) { //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); // 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 != mLeftVolume || right != mRightVolume) { mOutput->setVolume(left, right); left = mLeftVolume; right = mRightVolume; } if (track->mFillingUpStatus == Track::FS_FILLED) { track->mFillingUpStatus = Track::FS_ACTIVE; if (track->mState == TrackBase::RESUMING) { track->mState = TrackBase::ACTIVE; } } // 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 (trackToRemove->isTerminated()) { mTracks.remove(trackToRemove); deleteTrackName_l(trackToRemove->mName); } } } 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 = idleSleepTime; } // sleepTime == 0 means we must write to audio hardware if (sleepTime == 0) { mLastWriteTime = systemTime(); mInWrite = true; mBytesWritten += mixBufferSize; int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); if (bytesWritten < 0) mBytesWritten -= mixBufferSize; mNumWrites++; mInWrite = false; mStandby = false; } else { 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(); } 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; } else { time = 10000; } return time; } // ---------------------------------------------------------------------------- AudioFlinger::DuplicatingThread::DuplicatingThread(const sp& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) : MixerThread(audioFlinger, mainThread->getOutput(), id), 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() { int16_t* curBuf = mMixBuffer; 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; 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); } if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { // mix buffers... if (outputsReady(outputTracks)) { mAudioMixer->process(curBuf); } else { memset(curBuf, 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; break; } } } } if (mSuspended) { sleepTime = idleSleepTime; } // sleepTime == 0 means we must write to audio hardware if (sleepTime == 0) { standbyTime = systemTime() + kStandbyTimeInNsecs; for (size_t i = 0; i < outputTracks.size(); i++) { outputTracks[i]->write(curBuf, writeFrames); } mStandby = false; mBytesWritten += mixBufferSize; } else { 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(); } 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) : RefBase(), mThread(thread), mClient(client), mCblk(0), mFrameCount(0), mState(IDLE), mClientTid(-1), mFormat(format), mFlags(flags & ~SYSTEM_FLAGS_MASK) { 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 = sampleRate; mCblk->channelCount = (uint8_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->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 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 || ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 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) : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer), mMute(false), mSharedBuffer(sharedBuffer), mName(-1) { if (mCblk != NULL) { sp baseThread = thread.promote(); if (baseThread != 0) { PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); mName = playbackThread->getTrackName_l(); } LOGV("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); } 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, " %5d %5d %3u %3u %3u %04u %1d %1d %1d %5u %5u %5u %08x %08x\n", mName - AudioMixer::TRACK0, (mClient == NULL) ? getpid() : mClient->pid(), mStreamType, mFormat, mCblk->channelCount, mFrameCount, mState, mMute, mFillingUpStatus, mCblk->sampleRate, mCblk->volume[0], mCblk->volume[1], mCblk->server, mCblk->user); } status_t AudioFlinger::PlaybackThread::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; 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) 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; LOGV("start(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 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); 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() { LOGV("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); 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); 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; } // ---------------------------------------------------------------------------- // 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) : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, flags, 0), mOverflow(false) { if (mCblk != NULL) { LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 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 %04u %01d %05u %08x %08x\n", (mClient == NULL) ? getpid() : mClient->pid(), mFormat, mCblk->channelCount, 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), 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, pid_t pid) : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) { } AudioFlinger::NotificationClient::~NotificationClient() { mClient.clear(); } void AudioFlinger::NotificationClient::binderDied(const wp& who) { sp keep(this); { mAudioFlinger->removeNotificationClient(mPid); } } // ---------------------------------------------------------------------------- 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 input, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, status_t *status) { sp recordTrack; sp recordHandle; sp client; wp wclient; status_t lStatus; RecordThread *thread; size_t inFrameCount; // check calling permissions if (!recordingAllowed()) { lStatus = PERMISSION_DENIED; 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); } // 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); } 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 = AudioSystem::popCount(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; // 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 (framesOut == mFrameCount && ((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) { mBytesRead = mInput->read(buffer.raw, mInputBytes); 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()) LOGW("RecordThread: 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 (!mStandby) { mInput->standby(); } mActiveTrack.clear(); mStartStopCond.broadcast(); LOGV("RecordThread %p exiting", this); return false; } status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) { LOGV("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) { LOGV("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 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 = AudioSystem::popCount(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) && (AudioSystem::popCount(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 = (uint16_t)AudioSystem::popCount(mChannels); 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) { if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) || (format != AudioSystem::PCM_16_BIT) || (channels != AudioSystem::CHANNEL_OUT_STEREO)) { thread = new DirectOutputThread(this, output, ++mNextThreadId); LOGV("openOutput() created direct output: ID %d thread %p", mNextThreadId, thread); } else { thread = new MixerThread(this, output, ++mNextThreadId); LOGV("openOutput() created mixer output: ID %d thread %p", mNextThreadId, 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(mNextThreadId, thread); if (pSamplingRate) *pSamplingRate = samplingRate; if (pFormat) *pFormat = format; if (pChannels) *pChannels = channels; if (pLatencyMs) *pLatencyMs = thread->latency(); // notify client processes of the new output creation thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); return mNextThreadId; } return 0; } 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; } DuplicatingThread *thread = new DuplicatingThread(this, thread1, ++mNextThreadId); thread->addOutputTrack(thread2); mPlaybackThreads.add(mNextThreadId, thread); // notify client processes of the new output creation thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); return mNextThreadId; } 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); } 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) && (AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(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) { // Start record thread thread = new RecordThread(this, input, reqSamplingRate, reqChannels, ++mNextThreadId); mRecordThreads.add(mNextThreadId, thread); LOGV("openInput() created record thread: ID %d thread %p", mNextThreadId, 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 mNextThreadId; } 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); } } return NO_ERROR; } // 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; } // ---------------------------------------------------------------------------- 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