/* //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 namespace android { //static const nsecs_t kStandbyTimeInNsecs = seconds(3); static const unsigned long kBufferRecoveryInUsecs = 2000; static const unsigned long kMaxBufferRecoveryInUsecs = 20000; static const float MAX_GAIN = 4096.0f; // retry counts for buffer fill timeout // 50 * ~20msecs = 1 second static const int8_t kMaxTrackRetries = 50; static const int8_t kMaxTrackStartupRetries = 50; #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(), Thread(false), mMasterVolume(0), mMasterMute(true), mHardwareAudioMixer(0), mA2dpAudioMixer(0), mAudioMixer(0), mAudioHardware(0), mA2dpAudioInterface(0), mHardwareOutput(0), mA2dpOutput(0), mOutput(0), mRequestedOutput(0), mAudioRecordThread(0), mSampleRate(0), mFrameCount(0), mChannelCount(0), mFormat(0), mMixBuffer(0), mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mStandby(false), mInWrite(false) { mHardwareStatus = AUDIO_HW_IDLE; mAudioHardware = AudioHardwareInterface::create(); mHardwareStatus = AUDIO_HW_INIT; if (mAudioHardware->initCheck() == NO_ERROR) { // open 16-bit output stream for s/w mixer mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; status_t status; mHardwareOutput = mAudioHardware->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status); mHardwareStatus = AUDIO_HW_IDLE; if (mHardwareOutput) { mHardwareAudioMixer = new AudioMixer(getOutputFrameCount(mHardwareOutput), mHardwareOutput->sampleRate()); mRequestedOutput = mHardwareOutput; doSetOutput(mHardwareOutput); // FIXME - this should come from settings setMasterVolume(1.0f); setRouting(AudioSystem::MODE_NORMAL, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL); setRouting(AudioSystem::MODE_RINGTONE, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL); setRouting(AudioSystem::MODE_IN_CALL, AudioSystem::ROUTE_EARPIECE, AudioSystem::ROUTE_ALL); setMode(AudioSystem::MODE_NORMAL); mMasterMute = false; } else { LOGE("Failed to initialize output stream, status: %d", status); } #ifdef WITH_A2DP // Create A2DP interface mA2dpAudioInterface = new A2dpAudioInterface(); mA2dpOutput = mA2dpAudioInterface->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status); mA2dpAudioMixer = new AudioMixer(getOutputFrameCount(mA2dpOutput), mA2dpOutput->sampleRate()); // create a buffer big enough for both hardware and A2DP audio output. size_t hwFrameCount = getOutputFrameCount(mHardwareOutput); size_t a2dpFrameCount = getOutputFrameCount(mA2dpOutput); size_t frameCount = (hwFrameCount > a2dpFrameCount ? hwFrameCount : a2dpFrameCount); #else size_t frameCount = getOutputFrameCount(mHardwareOutput); #endif // FIXME - Current mixer implementation only supports stereo output: Always // Allocate a stereo buffer even if HW output is mono. mMixBuffer = new int16_t[frameCount * 2]; memset(mMixBuffer, 0, frameCount * 2 * sizeof(int16_t)); // Start record thread mAudioRecordThread = new AudioRecordThread(mAudioHardware); if (mAudioRecordThread != 0) { mAudioRecordThread->run("AudioRecordThread", PRIORITY_URGENT_AUDIO); } } else { LOGE("Couldn't even initialize the stubbed audio hardware!"); } char value[PROPERTY_VALUE_MAX]; property_get("ro.audio.silent", value, "0"); if (atoi(value)) { LOGD("Silence is golden"); mMasterMute = true; } } AudioFlinger::~AudioFlinger() { if (mAudioRecordThread != 0) { mAudioRecordThread->exit(); mAudioRecordThread.clear(); } delete mAudioHardware; // deleting mA2dpAudioInterface also deletes mA2dpOutput; delete mA2dpAudioInterface; delete [] mMixBuffer; delete mHardwareAudioMixer; delete mA2dpAudioMixer; } void AudioFlinger::setOutput(AudioStreamOut* output) { mRequestedOutput = output; } void AudioFlinger::doSetOutput(AudioStreamOut* output) { mSampleRate = output->sampleRate(); mChannelCount = output->channelCount(); // FIXME - Current mixer implementation only supports stereo output if (mChannelCount == 1) { LOGE("Invalid audio hardware channel count"); } mFormat = output->format(); mFrameCount = getOutputFrameCount(output); mAudioMixer = (output == mA2dpOutput ? mA2dpAudioMixer : mHardwareAudioMixer); mOutput = output; } size_t AudioFlinger::getOutputFrameCount(AudioStreamOut* output) { return output->bufferSize() / output->channelCount() / sizeof(int16_t); } 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::dumpTracks(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; result.append("Tracks:\n"); 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) { wp wTrack = mTracks[i]; if (wTrack != 0) { sp track = wTrack.promote(); if (track != 0) { track->dump(buffer, SIZE); result.append(buffer); } } } result.append("Active Tracks:\n"); 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 = mTracks[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::dumpInternals(int fd, const Vector& args) { const size_t SIZE = 256; char buffer[SIZE]; String8 result; snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", audioMixer()->trackNames()); result.append(buffer); snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); result.append(buffer); snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); result.append(buffer); snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); result.append(buffer); snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); result.append(buffer); snprintf(buffer, SIZE, "standby: %d\n", mStandby); result.append(buffer); snprintf(buffer, SIZE, "Hardware status: %d\n", mHardwareStatus); 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; } status_t AudioFlinger::dump(int fd, const Vector& args) { if (checkCallingPermission(String16("android.permission.DUMP")) == false) { dumpPermissionDenial(fd, args); } else { AutoMutex lock(&mLock); dumpClients(fd, args); dumpTracks(fd, args); dumpInternals(fd, args); if (mAudioHardware) { mAudioHardware->dumpState(fd, args); } } return NO_ERROR; } // Thread virtuals bool AudioFlinger::threadLoop() { unsigned long sleepTime = kBufferRecoveryInUsecs; int16_t* curBuf = mMixBuffer; Vector< sp > tracksToRemove; size_t enabledTracks = 0; nsecs_t standbyTime = systemTime(); do { enabledTracks = 0; { // scope for the mLock Mutex::Autolock _l(mLock); const SortedVector< wp >& activeTracks = mActiveTracks; // put audio hardware into standby after short delay if UNLIKELY(!activeTracks.size() && systemTime() > standbyTime) { // wait until we have something to do... LOGV("Audio hardware entering standby\n"); mHardwareStatus = AUDIO_HW_STANDBY; if (!mStandby) { mOutput->standby(); mStandby = true; } mHardwareStatus = AUDIO_HW_IDLE; // we're about to wait, flush the binder command buffer IPCThreadState::self()->flushCommands(); mWaitWorkCV.wait(mLock); LOGV("Audio hardware exiting standby\n"); standbyTime = systemTime() + kStandbyTimeInNsecs; continue; } // check for change in output if (mRequestedOutput != mOutput) { // put current output into standby mode if (mOutput) mOutput->standby(); // change output doSetOutput(mRequestedOutput); } // find out which tracks need to be processed size_t count = activeTracks.size(); for (size_t i=0 ; i t = activeTracks[i].promote(); if (t == 0) continue; Track* const track = t.get(); audio_track_cblk_t* cblk = track->cblk(); // The first time a track is added we wait // for all its buffers to be filled before processing it mAudioMixer->setActiveTrack(track->name()); if (cblk->framesReady() && (track->isReady() || track->isStopped()) && !track->isPaused()) { //LOGD("u=%08x, s=%08x [OK]", u, s); // compute volume for this track int16_t left, right; if (track->isMuted() || mMasterMute || track->isPausing()) { left = right = 0; if (track->isPausing()) { LOGV("paused(%d)", track->name()); track->setPaused(); } } else { float typeVolume = mStreamTypes[track->type()].volume; float v = mMasterVolume * typeVolume; float v_clamped = v * cblk->volume[0]; if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; left = int16_t(v_clamped); v_clamped = v * cblk->volume[1]; if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; right = int16_t(v_clamped); } // XXX: these things DON'T need to be done each time mAudioMixer->setBufferProvider(track); mAudioMixer->enable(AudioMixer::MIXING); int param; if ( track->mFillingUpStatus == Track::FS_FILLED) { // no ramp for the first volume setting track->mFillingUpStatus = Track::FS_ACTIVE; if (track->mState == TrackBase::RESUMING) { track->mState = TrackBase::ACTIVE; param = AudioMixer::RAMP_VOLUME; } else { param = AudioMixer::VOLUME; } } else { param = AudioMixer::RAMP_VOLUME; } mAudioMixer->setParameter(param, AudioMixer::VOLUME0, left); mAudioMixer->setParameter(param, AudioMixer::VOLUME1, right); mAudioMixer->setParameter( AudioMixer::TRACK, AudioMixer::FORMAT, track->format()); mAudioMixer->setParameter( AudioMixer::TRACK, AudioMixer::CHANNEL_COUNT, track->channelCount()); mAudioMixer->setParameter( AudioMixer::RESAMPLE, AudioMixer::SAMPLE_RATE, int(cblk->sampleRate)); // reset retry count track->mRetryCount = kMaxTrackRetries; enabledTracks++; } else { //LOGD("u=%08x, s=%08x [NOT READY]", u, s); if (track->isStopped()) { track->reset(); } if (track->isTerminated() || track->isStopped() || track->isPaused()) { // We have consumed all the buffers of this track. // Remove it from the list of active tracks. LOGV("remove(%d) from active list", track->name()); tracksToRemove.add(track); } else { // No buffers for this track. Give it a few chances to // fill a buffer, then remove it from active list. if (--(track->mRetryCount) <= 0) { LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); tracksToRemove.add(track); } } // LOGV("disable(%d)", track->name()); mAudioMixer->disable(AudioMixer::MIXING); } } // remove all the tracks that need to be... count = tracksToRemove.size(); if (UNLIKELY(count)) { for (size_t i=0 ; i& track = tracksToRemove[i]; mActiveTracks.remove(track); if (track->isTerminated()) { mTracks.remove(track); mAudioMixer->deleteTrackName(track->mName); } } } } if (LIKELY(enabledTracks)) { // mix buffers... mAudioMixer->process(curBuf); // output audio to hardware mLastWriteTime = systemTime(); mInWrite = true; size_t mixBufferSize = mFrameCount*mChannelCount*sizeof(int16_t); mOutput->write(curBuf, mixBufferSize); mNumWrites++; mInWrite = false; mStandby = false; nsecs_t temp = systemTime(); standbyTime = temp + kStandbyTimeInNsecs; nsecs_t delta = temp - mLastWriteTime; nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 2; if (delta > maxPeriod) { LOGW("write blocked for %llu msecs", ns2ms(delta)); mNumDelayedWrites++; } sleepTime = kBufferRecoveryInUsecs; } else { // There was nothing to mix this round, which means all // active tracks were late. Sleep a little bit to give // them another chance. If we're too late, the audio // hardware will zero-fill for us. LOGV("no buffers - usleep(%lu)", sleepTime); usleep(sleepTime); if (sleepTime < kMaxBufferRecoveryInUsecs) { sleepTime += kBufferRecoveryInUsecs; } } // finally let go of all our tracks, without the lock held // since we can't guarantee the destructors won't acquire that // same lock. tracksToRemove.clear(); } while (true); return false; } status_t AudioFlinger::readyToRun() { if (mSampleRate == 0) { LOGE("No working audio driver found."); return NO_INIT; } LOGI("AudioFlinger's main thread ready to run."); return NO_ERROR; } void AudioFlinger::onFirstRef() { run("AudioFlinger", ANDROID_PRIORITY_URGENT_AUDIO); } // IAudioFlinger interface sp AudioFlinger::createTrack( pid_t pid, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, const sp& sharedBuffer, status_t *status) { sp track; sp trackHandle; sp client; wp wclient; status_t lStatus; if (streamType >= AudioTrack::NUM_STREAM_TYPES) { LOGE("invalid stream type"); lStatus = BAD_VALUE; goto Exit; } // Resampler implementation limits input sampling rate to 2 x output sampling rate. if (sampleRate > MAX_SAMPLE_RATE || sampleRate > mSampleRate*2) { LOGE("Sample rate out of range: %d", sampleRate); lStatus = BAD_VALUE; goto Exit; } { Mutex::Autolock _l(mLock); if (mSampleRate == 0) { LOGE("Audio driver not initialized."); lStatus = NO_INIT; goto Exit; } wclient = mClients.valueFor(pid); if (wclient != NULL) { client = wclient.promote(); } else { client = new Client(this, pid); mClients.add(pid, client); } track = new Track(this, client, streamType, sampleRate, format, channelCount, frameCount, sharedBuffer); mTracks.add(track); trackHandle = new TrackHandle(track); lStatus = NO_ERROR; } Exit: if(status) { *status = lStatus; } return trackHandle; } uint32_t AudioFlinger::sampleRate() const { return mSampleRate; } int AudioFlinger::channelCount() const { return mChannelCount; } int AudioFlinger::format() const { return mFormat; } size_t AudioFlinger::frameCount() const { return mFrameCount; } uint32_t AudioFlinger::latency() const { if (mOutput) { return mOutput->latency(); } else { return 0; } } 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) { mMasterVolume = 1.0f; } else { mMasterVolume = value; } mHardwareStatus = AUDIO_HW_IDLE; return NO_ERROR; } status_t AudioFlinger::setRouting(int mode, uint32_t routes, uint32_t mask) { status_t err = NO_ERROR; // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } if ((mode < AudioSystem::MODE_CURRENT) || (mode >= AudioSystem::NUM_MODES)) { LOGW("Illegal value: setRouting(%d, %u, %u)", mode, routes, mask); return BAD_VALUE; } #ifdef WITH_A2DP LOGD("setRouting %d %d %d\n", mode, routes, mask); if (mode == AudioSystem::MODE_NORMAL && (mask & AudioSystem::ROUTE_BLUETOOTH_A2DP)) { if (routes & AudioSystem::ROUTE_BLUETOOTH_A2DP) { LOGD("set output to A2DP\n"); setOutput(mA2dpOutput); } else { LOGD("set output to hardware audio\n"); setOutput(mHardwareOutput); } LOGD("setOutput done\n"); } #endif // do nothing if only A2DP routing is affected mask &= ~AudioSystem::ROUTE_BLUETOOTH_A2DP; if (mask) { AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_GET_ROUTING; uint32_t r; err = mAudioHardware->getRouting(mode, &r); if (err == NO_ERROR) { r = (r & ~mask) | (routes & mask); mHardwareStatus = AUDIO_HW_SET_ROUTING; err = mAudioHardware->setRouting(mode, r); } mHardwareStatus = AUDIO_HW_IDLE; } return err; } uint32_t AudioFlinger::getRouting(int mode) const { uint32_t routes = 0; if ((mode >= AudioSystem::MODE_CURRENT) && (mode < AudioSystem::NUM_MODES)) { mHardwareStatus = AUDIO_HW_GET_ROUTING; mAudioHardware->getRouting(mode, &routes); mHardwareStatus = AUDIO_HW_IDLE; } else { LOGW("Illegal value: getRouting(%d)", mode); } return routes; } status_t AudioFlinger::setMode(int mode) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) { LOGW("Illegal value: setMode(%d)", mode); return BAD_VALUE; } AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_SET_MODE; status_t ret = mAudioHardware->setMode(mode); mHardwareStatus = AUDIO_HW_IDLE; return ret; } int AudioFlinger::getMode() const { int mode = AudioSystem::MODE_INVALID; mHardwareStatus = AUDIO_HW_SET_MODE; mAudioHardware->getMode(&mode); mHardwareStatus = AUDIO_HW_IDLE; return mode; } status_t AudioFlinger::setMicMute(bool state) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; status_t ret = mAudioHardware->setMicMute(state); mHardwareStatus = AUDIO_HW_IDLE; return ret; } bool AudioFlinger::getMicMute() const { bool state = AudioSystem::MODE_INVALID; mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; mAudioHardware->getMicMute(&state); mHardwareStatus = AUDIO_HW_IDLE; return state; } status_t AudioFlinger::setMasterMute(bool muted) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } mMasterMute = muted; return NO_ERROR; } float AudioFlinger::masterVolume() const { return mMasterVolume; } bool AudioFlinger::masterMute() const { return mMasterMute; } status_t AudioFlinger::setStreamVolume(int stream, float value) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) { return BAD_VALUE; } mStreamTypes[stream].volume = value; status_t ret = NO_ERROR; if (stream == AudioTrack::VOICE_CALL) { AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_SET_VOICE_VOLUME; ret = mAudioHardware->setVoiceVolume(value); mHardwareStatus = AUDIO_HW_IDLE; } return ret; } status_t AudioFlinger::setStreamMute(int stream, bool muted) { // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; } if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) { return BAD_VALUE; } mStreamTypes[stream].mute = muted; return NO_ERROR; } float AudioFlinger::streamVolume(int stream) const { if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) { return 0.0f; } return mStreamTypes[stream].volume; } bool AudioFlinger::streamMute(int stream) const { if (uint32_t(stream) >= AudioTrack::NUM_STREAM_TYPES) { return true; } return mStreamTypes[stream].mute; } bool AudioFlinger::isMusicActive() const { size_t count = mActiveTracks.size(); for (size_t i = 0 ; i < count ; ++i) { sp t = mActiveTracks[i].promote(); if (t == 0) continue; Track* const track = t.get(); if (t->mStreamType == AudioTrack::MUSIC) return true; } return false; } status_t AudioFlinger::setParameter(const char* key, const char* value) { status_t result, result2; AutoMutex lock(mHardwareLock); mHardwareStatus = AUDIO_SET_PARAMETER; result = mAudioHardware->setParameter(key, value); if (mA2dpAudioInterface) { result2 = mA2dpAudioInterface->setParameter(key, value); if (result2) result = result2; } mHardwareStatus = AUDIO_HW_IDLE; return result; } void AudioFlinger::removeClient(pid_t pid) { Mutex::Autolock _l(mLock); mClients.removeItem(pid); } status_t AudioFlinger::addTrack(const sp& track) { Mutex::Autolock _l(mLock); // here the track could be either new, or restarted // in both cases "unstop" the track if (track->isPaused()) { track->mState = TrackBase::RESUMING; LOGV("PAUSED => RESUMING (%d)", track->name()); } else { track->mState = TrackBase::ACTIVE; LOGV("? => ACTIVE (%d)", track->name()); } // set retry count for buffer fill track->mRetryCount = kMaxTrackStartupRetries; LOGV("mWaitWorkCV.broadcast"); mWaitWorkCV.broadcast(); 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); return NO_ERROR; } return ALREADY_EXISTS; } void AudioFlinger::removeTrack(wp track, int name) { Mutex::Autolock _l(mLock); sp t = track.promote(); if (t!=NULL && (t->mState <= TrackBase::STOPPED)) { remove_track_l(track, name); } } void AudioFlinger::remove_track_l(wp track, int name) { sp t = track.promote(); if (t!=NULL) { t->reset(); } audioMixer()->deleteTrackName(name); mActiveTracks.remove(track); mWaitWorkCV.broadcast(); } void AudioFlinger::destroyTrack(const sp& track) { // NOTE: We're acquiring a strong reference on the track before // acquiring the lock, this is to make sure removing it from // mTracks won't cause the destructor to be called while the lock is // held (note that technically, 'track' could be a reference to an item // in mTracks, which is why we need to do this). sp keep(track); Mutex::Autolock _l(mLock); track->mState = TrackBase::TERMINATED; if (mActiveTracks.indexOf(track) < 0) { LOGV("remove track (%d) and delete from mixer", track->name()); mTracks.remove(track); audioMixer()->deleteTrackName(keep->name()); } } // ---------------------------------------------------------------------------- AudioFlinger::Client::Client(const sp& audioFlinger, pid_t pid) : RefBase(), mAudioFlinger(audioFlinger), mMemoryDealer(new MemoryDealer(1024*1024)), mPid(pid) { // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer } AudioFlinger::Client::~Client() { mAudioFlinger->removeClient(mPid); } const sp& AudioFlinger::Client::heap() const { return mMemoryDealer; } // ---------------------------------------------------------------------------- AudioFlinger::TrackBase::TrackBase( const sp& audioFlinger, const sp& client, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, const sp& sharedBuffer) : RefBase(), mAudioFlinger(audioFlinger), mClient(client), mStreamType(streamType), mFrameCount(0), mState(IDLE), mClientTid(-1), mFormat(format), mFlags(0) { mName = audioFlinger->audioMixer()->getTrackName(); if (mName < 0) { LOGE("no more track names availlable"); return; } LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); size_t size = sizeof(audio_track_cblk_t); size_t bufferSize = frameCount*channelCount*sizeof(int16_t); if (sharedBuffer == 0) { size += bufferSize; } 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->channels = channelCount; if (sharedBuffer == 0) { mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); // Force underrun condition to avoid false underrun callback until first data is // written to buffer mCblk->flowControlFlag = 1; } else { mBuffer = sharedBuffer->pointer(); } mBufferEnd = (uint8_t *)mBuffer + bufferSize; } } else { LOGE("not enough memory for AudioTrack size=%u", size); client->heap()->dump("AudioTrack"); return; } } AudioFlinger::TrackBase::~TrackBase() { mCblk->~audio_track_cblk_t(); // destroy our shared-structure. mCblkMemory.clear(); // and free the shared memory mClient.clear(); } void AudioFlinger::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) { buffer->raw = 0; mFrameCount = buffer->frameCount; step(); buffer->frameCount = 0; } bool AudioFlinger::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::TrackBase::reset() { audio_track_cblk_t* cblk = this->cblk(); cblk->user = 0; cblk->server = 0; cblk->userBase = 0; cblk->serverBase = 0; mFlags = 0; LOGV("TrackBase::reset"); } sp AudioFlinger::TrackBase::getCblk() const { return mCblkMemory; } int AudioFlinger::TrackBase::sampleRate() const { return mCblk->sampleRate; } int AudioFlinger::TrackBase::channelCount() const { return mCblk->channels; } void* AudioFlinger::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { audio_track_cblk_t* cblk = this->cblk(); int16_t *bufferStart = (int16_t *)mBuffer + (offset-cblk->serverBase)*cblk->channels; int16_t *bufferEnd = bufferStart + frames * cblk->channels; // Check validity of returned pointer in case the track control block would have been corrupted. if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd) { LOGW("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ server %d, serverBase %d, user %d, userBase %d", bufferStart, bufferEnd, mBuffer, mBufferEnd, cblk->server, cblk->serverBase, cblk->user, cblk->userBase); return 0; } return bufferStart; } // ---------------------------------------------------------------------------- AudioFlinger::Track::Track( const sp& audioFlinger, const sp& client, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, const sp& sharedBuffer) : TrackBase(audioFlinger, client, streamType, sampleRate, format, channelCount, frameCount, sharedBuffer) { mVolume[0] = 1.0f; mVolume[1] = 1.0f; mMute = false; mSharedBuffer = sharedBuffer; } AudioFlinger::Track::~Track() { wp weak(this); // never create a strong ref from the dtor mState = TERMINATED; mAudioFlinger->removeTrack(weak, mName); } void AudioFlinger::Track::destroy() { mAudioFlinger->destroyTrack(this); } void AudioFlinger::Track::dump(char* buffer, size_t size) { snprintf(buffer, size, " %5d %5d %3u %3u %3u %3u %1d %1d %1d %5u %5u %5u %04x %04x\n", mName - AudioMixer::TRACK0, mClient->pid(), mStreamType, mFormat, mCblk->channels, mFrameCount, mState, mMute, mFillingUpStatus, mCblk->sampleRate, mCblk->volume[0], mCblk->volume[1], mCblk->server, mCblk->user); } status_t AudioFlinger::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) { audio_track_cblk_t* cblk = this->cblk(); uint32_t framesReady; uint32_t framesReq = buffer->frameCount; // Check if last stepServer failed, try to step now if (mFlags & TrackBase::STEPSERVER_FAILED) { if (!step()) goto getNextBuffer_exit; LOGV("stepServer recovered"); mFlags &= ~TrackBase::STEPSERVER_FAILED; } framesReady = cblk->framesReady(); if (LIKELY(framesReady)) { uint32_t s = cblk->server; uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; if (framesReq > framesReady) { framesReq = framesReady; } if (s + framesReq > bufferEnd) { framesReq = bufferEnd - s; } buffer->raw = getBuffer(s, framesReq); if (buffer->raw == 0) goto getNextBuffer_exit; buffer->frameCount = framesReq; return NO_ERROR; } getNextBuffer_exit: buffer->raw = 0; buffer->frameCount = 0; return NOT_ENOUGH_DATA; } bool AudioFlinger::Track::isReady() const { if (mFillingUpStatus != FS_FILLING) return true; if (mCblk->framesReady() >= mCblk->frameCount || mCblk->forceReady) { mFillingUpStatus = FS_FILLED; mCblk->forceReady = 0; return true; } return false; } status_t AudioFlinger::Track::start() { LOGV("start(%d)", mName); mAudioFlinger->addTrack(this); return NO_ERROR; } void AudioFlinger::Track::stop() { LOGV("stop(%d)", mName); Mutex::Autolock _l(mAudioFlinger->mLock); if (mState > STOPPED) { mState = STOPPED; // If the track is not active (PAUSED and buffers full), flush buffers if (mAudioFlinger->mActiveTracks.indexOf(this) < 0) { reset(); } LOGV("(> STOPPED) => STOPPED (%d)", mName); } } void AudioFlinger::Track::pause() { LOGV("pause(%d)", mName); Mutex::Autolock _l(mAudioFlinger->mLock); if (mState == ACTIVE || mState == RESUMING) { mState = PAUSING; LOGV("ACTIVE/RESUMING => PAUSING (%d)", mName); } } void AudioFlinger::Track::flush() { LOGV("flush(%d)", mName); Mutex::Autolock _l(mAudioFlinger->mLock); if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { return; } // No point remaining in PAUSED state after a flush => go to // STOPPED state mState = STOPPED; // 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(); } void AudioFlinger::Track::reset() { // Do not reset twice to avoid discarding data written just after a flush and before // the audioflinger thread detects the track is stopped. if (!mResetDone) { TrackBase::reset(); // Force underrun condition to avoid false underrun callback until first data is // written to buffer mCblk->flowControlFlag = 1; mCblk->forceReady = 0; mFillingUpStatus = FS_FILLING; mResetDone = true; } } void AudioFlinger::Track::mute(bool muted) { mMute = muted; } void AudioFlinger::Track::setVolume(float left, float right) { mVolume[0] = left; mVolume[1] = right; } // ---------------------------------------------------------------------------- AudioFlinger::TrackHandle::TrackHandle(const sp& track) : BnAudioTrack(), mTrack(track) { } AudioFlinger::TrackHandle::~TrackHandle() { // just stop the track on deletion, associated resources // will be freed from the main thread once all pending buffers have // been played. Unless it's not in the active track list, in which // case we free everything now... mTrack->destroy(); } status_t AudioFlinger::TrackHandle::start() { return mTrack->start(); } void AudioFlinger::TrackHandle::stop() { mTrack->stop(); } void AudioFlinger::TrackHandle::flush() { mTrack->flush(); } void AudioFlinger::TrackHandle::mute(bool e) { mTrack->mute(e); } void AudioFlinger::TrackHandle::pause() { mTrack->pause(); } void AudioFlinger::TrackHandle::setVolume(float left, float right) { mTrack->setVolume(left, right); } sp AudioFlinger::TrackHandle::getCblk() const { return mTrack->getCblk(); } status_t AudioFlinger::TrackHandle::onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { return BnAudioTrack::onTransact(code, data, reply, flags); } // ---------------------------------------------------------------------------- sp AudioFlinger::openRecord( pid_t pid, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, status_t *status) { sp thread; sp recordTrack; sp recordHandle; sp client; wp wclient; AudioStreamIn* input = 0; int inFrameCount; size_t inputBufferSize; status_t lStatus; // check calling permissions if (!recordingAllowed()) { lStatus = PERMISSION_DENIED; goto Exit; } if (uint32_t(streamType) >= AudioRecord::NUM_STREAM_TYPES) { LOGE("invalid stream type"); lStatus = BAD_VALUE; goto Exit; } if (sampleRate > MAX_SAMPLE_RATE) { LOGE("Sample rate out of range"); lStatus = BAD_VALUE; goto Exit; } if (mSampleRate == 0) { LOGE("Audio driver not initialized"); lStatus = NO_INIT; goto Exit; } if (mAudioRecordThread == 0) { LOGE("Audio record thread not started"); lStatus = NO_INIT; goto Exit; } // Check that audio input stream accepts requested audio parameters inputBufferSize = mAudioHardware->getInputBufferSize(sampleRate, format, channelCount); if (inputBufferSize == 0) { lStatus = BAD_VALUE; LOGE("Bad audio input parameters: sampling rate %u, format %d, channels %d", sampleRate, format, channelCount); goto Exit; } // add client to list { Mutex::Autolock _l(mLock); wclient = mClients.valueFor(pid); if (wclient != NULL) { client = wclient.promote(); } else { client = new Client(this, pid); mClients.add(pid, client); } } // frameCount must be a multiple of input buffer size inFrameCount = inputBufferSize/channelCount/sizeof(short); frameCount = ((frameCount - 1)/inFrameCount + 1) * inFrameCount; // create new record track and pass to record thread recordTrack = new RecordTrack(this, client, streamType, sampleRate, format, channelCount, frameCount); // return to handle to client recordHandle = new RecordHandle(recordTrack); lStatus = NO_ERROR; Exit: if (status) { *status = lStatus; } return recordHandle; } status_t AudioFlinger::startRecord(RecordTrack* recordTrack) { if (mAudioRecordThread != 0) { return mAudioRecordThread->start(recordTrack); } return NO_INIT; } void AudioFlinger::stopRecord(RecordTrack* recordTrack) { if (mAudioRecordThread != 0) { mAudioRecordThread->stop(recordTrack); } } // ---------------------------------------------------------------------------- AudioFlinger::RecordTrack::RecordTrack( const sp& audioFlinger, const sp& client, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount) : TrackBase(audioFlinger, client, streamType, sampleRate, format, channelCount, frameCount, 0), mOverflow(false) { } AudioFlinger::RecordTrack::~RecordTrack() { mAudioFlinger->audioMixer()->deleteTrackName(mName); } status_t AudioFlinger::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::RecordTrack::start() { return mAudioFlinger->startRecord(this); } void AudioFlinger::RecordTrack::stop() { mAudioFlinger->stopRecord(this); TrackBase::reset(); // Force overerrun condition to avoid false overrun callback until first data is // read from buffer mCblk->flowControlFlag = 1; } // ---------------------------------------------------------------------------- AudioFlinger::RecordHandle::RecordHandle(const sp& recordTrack) : BnAudioRecord(), mRecordTrack(recordTrack) { } AudioFlinger::RecordHandle::~RecordHandle() { stop(); } status_t AudioFlinger::RecordHandle::start() { LOGV("RecordHandle::start()"); return mRecordTrack->start(); } void AudioFlinger::RecordHandle::stop() { LOGV("RecordHandle::stop()"); mRecordTrack->stop(); } sp AudioFlinger::RecordHandle::getCblk() const { return mRecordTrack->getCblk(); } status_t AudioFlinger::RecordHandle::onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { return BnAudioRecord::onTransact(code, data, reply, flags); } // ---------------------------------------------------------------------------- AudioFlinger::AudioRecordThread::AudioRecordThread(AudioHardwareInterface* audioHardware) : mAudioHardware(audioHardware), mActive(false) { } AudioFlinger::AudioRecordThread::~AudioRecordThread() { } bool AudioFlinger::AudioRecordThread::threadLoop() { LOGV("AudioRecordThread: start record loop"); AudioBufferProvider::Buffer buffer; int inBufferSize = 0; int inFrameCount = 0; AudioStreamIn* input = 0; mActive = 0; // start recording while (!exitPending()) { if (!mActive) { mLock.lock(); if (!mActive && !exitPending()) { LOGV("AudioRecordThread: loop stopping"); if (input) { delete input; input = 0; } mRecordTrack.clear(); mWaitWorkCV.wait(mLock); LOGV("AudioRecordThread: loop starting"); if (mRecordTrack != 0) { input = mAudioHardware->openInputStream(mRecordTrack->format(), mRecordTrack->channelCount(), mRecordTrack->sampleRate(), &mStartStatus); if (input != 0) { inBufferSize = input->bufferSize(); inFrameCount = inBufferSize/input->frameSize(); } } else { mStartStatus = NO_INIT; } if (mStartStatus !=NO_ERROR) { LOGW("record start failed, status %d", mStartStatus); mActive = false; mRecordTrack.clear(); } mWaitWorkCV.signal(); } mLock.unlock(); } else if (mRecordTrack != 0){ buffer.frameCount = inFrameCount; if (LIKELY(mRecordTrack->getNextBuffer(&buffer) == NO_ERROR)) { LOGV("AudioRecordThread read: %d frames", buffer.frameCount); if (input->read(buffer.raw, inBufferSize) < 0) { LOGE("Error reading audio input"); sleep(1); } mRecordTrack->releaseBuffer(&buffer); mRecordTrack->overflow(); } // client isn't retrieving buffers fast enough else { if (!mRecordTrack->setOverflow()) LOGW("AudioRecordThread: buffer overflow"); // Release the processor for a while before asking for a new buffer. // This will give the application more chance to read from the buffer and // clear the overflow. usleep(5000); } } } if (input) { delete input; } mRecordTrack.clear(); return false; } status_t AudioFlinger::AudioRecordThread::start(RecordTrack* recordTrack) { LOGV("AudioRecordThread::start"); AutoMutex lock(&mLock); mActive = true; // If starting the active track, just reset mActive in case a stop // was pending and exit if (recordTrack == mRecordTrack.get()) return NO_ERROR; if (mRecordTrack != 0) return -EBUSY; mRecordTrack = recordTrack; // signal thread to start LOGV("Signal record thread"); mWaitWorkCV.signal(); mWaitWorkCV.wait(mLock); LOGV("Record started, status %d", mStartStatus); return mStartStatus; } void AudioFlinger::AudioRecordThread::stop(RecordTrack* recordTrack) { LOGV("AudioRecordThread::stop"); AutoMutex lock(&mLock); if (mActive && (recordTrack == mRecordTrack.get())) { mActive = false; } } void AudioFlinger::AudioRecordThread::exit() { LOGV("AudioRecordThread::exit"); { AutoMutex lock(&mLock); requestExit(); mWaitWorkCV.signal(); } requestExitAndWait(); } 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