| /* //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 <math.h> |
| #include <signal.h> |
| #include <sys/time.h> |
| #include <sys/resource.h> |
| |
| #include <binder/IServiceManager.h> |
| #include <utils/Log.h> |
| #include <binder/Parcel.h> |
| #include <binder/IPCThreadState.h> |
| #include <utils/String16.h> |
| #include <utils/threads.h> |
| |
| #include <cutils/properties.h> |
| |
| #include <media/AudioTrack.h> |
| #include <media/AudioRecord.h> |
| |
| #include <private/media/AudioTrackShared.h> |
| |
| #include <hardware_legacy/AudioHardwareInterface.h> |
| |
| #include "AudioMixer.h" |
| #include "AudioFlinger.h" |
| |
| #ifdef WITH_A2DP |
| #include "A2dpAudioInterface.h" |
| #endif |
| |
| // ---------------------------------------------------------------------------- |
| // the sim build doesn't have gettid |
| |
| #ifndef HAVE_GETTID |
| # define gettid getpid |
| #endif |
| |
| // ---------------------------------------------------------------------------- |
| |
| namespace android { |
| |
| static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n"; |
| static const char* kHardwareLockedString = "Hardware lock is taken\n"; |
| |
| //static const nsecs_t kStandbyTimeInNsecs = seconds(3); |
| static const unsigned long kBufferRecoveryInUsecs = 2000; |
| static const unsigned long kMaxBufferRecoveryInUsecs = 20000; |
| static const float MAX_GAIN = 4096.0f; |
| |
| // retry counts for buffer fill timeout |
| // 50 * ~20msecs = 1 second |
| static const int8_t kMaxTrackRetries = 50; |
| static const int8_t kMaxTrackStartupRetries = 50; |
| |
| static const int kDumpLockRetries = 50; |
| static const int kDumpLockSleep = 20000; |
| |
| |
| #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) |
| { |
| 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!"); |
| } |
| } |
| |
| AudioFlinger::~AudioFlinger() |
| { |
| mRecordThreads.clear(); |
| mPlaybackThreads.clear(); |
| } |
| |
| |
| |
| status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& 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<Client> wClient = mClients.valueAt(i); |
| if (wClient != 0) { |
| sp<Client> 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<String16>& 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<String16>& 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<String16>& 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[i]->dump(fd, args); |
| } |
| |
| // dump record threads |
| for (size_t i = 0; i < mPlaybackThreads.size(); i++) { |
| mRecordThreads[i]->dump(fd, args); |
| } |
| |
| if (mAudioHardware) { |
| mAudioHardware->dumpState(fd, args); |
| } |
| if (locked) mLock.unlock(); |
| } |
| return NO_ERROR; |
| } |
| |
| |
| // IAudioFlinger interface |
| |
| |
| sp<IAudioTrack> AudioFlinger::createTrack( |
| pid_t pid, |
| int streamType, |
| uint32_t sampleRate, |
| int format, |
| int channelCount, |
| int frameCount, |
| uint32_t flags, |
| const sp<IMemory>& sharedBuffer, |
| void *output, |
| status_t *status) |
| { |
| sp<PlaybackThread::Track> track; |
| sp<TrackHandle> trackHandle; |
| sp<Client> client; |
| wp<Client> 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 { |
| track.clear(); |
| } |
| |
| Exit: |
| if(status) { |
| *status = lStatus; |
| } |
| return trackHandle; |
| } |
| |
| uint32_t AudioFlinger::sampleRate(void *output) const |
| { |
| Mutex::Autolock _l(mLock); |
| PlaybackThread *thread = checkPlaybackThread_l(output); |
| if (thread == NULL) { |
| LOGW("sampleRate() unknown thread %p", output); |
| return 0; |
| } |
| return thread->sampleRate(); |
| } |
| |
| int AudioFlinger::channelCount(void *output) const |
| { |
| Mutex::Autolock _l(mLock); |
| PlaybackThread *thread = checkPlaybackThread_l(output); |
| if (thread == NULL) { |
| LOGW("channelCount() unknown thread %p", output); |
| return 0; |
| } |
| return thread->channelCount(); |
| } |
| |
| int AudioFlinger::format(void *output) const |
| { |
| Mutex::Autolock _l(mLock); |
| PlaybackThread *thread = checkPlaybackThread_l(output); |
| if (thread == NULL) { |
| LOGW("format() unknown thread %p", output); |
| return 0; |
| } |
| return thread->format(); |
| } |
| |
| size_t AudioFlinger::frameCount(void *output) const |
| { |
| Mutex::Autolock _l(mLock); |
| PlaybackThread *thread = checkPlaybackThread_l(output); |
| if (thread == NULL) { |
| LOGW("frameCount() unknown thread %p", output); |
| return 0; |
| } |
| return thread->frameCount(); |
| } |
| |
| uint32_t AudioFlinger::latency(void *output) const |
| { |
| Mutex::Autolock _l(mLock); |
| PlaybackThread *thread = checkPlaybackThread_l(output); |
| if (thread == NULL) { |
| LOGW("latency() unknown thread %p", 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[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); |
| 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[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, void *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; |
| } |
| } |
| |
| status_t ret = NO_ERROR; |
| |
| if (stream == AudioSystem::VOICE_CALL || |
| stream == AudioSystem::BLUETOOTH_SCO) { |
| float hwValue; |
| if (stream == AudioSystem::VOICE_CALL) { |
| hwValue = (float)AudioSystem::logToLinear(value)/100.0f; |
| // offset value to reflect actual hardware volume that never reaches 0 |
| // 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java) |
| value = 0.01 + 0.99 * value; |
| } else { // (type == AudioSystem::BLUETOOTH_SCO) |
| hwValue = 1.0f; |
| } |
| |
| AutoMutex lock(mHardwareLock); |
| mHardwareStatus = AUDIO_SET_VOICE_VOLUME; |
| ret = mAudioHardware->setVoiceVolume(hwValue); |
| mHardwareStatus = AUDIO_HW_IDLE; |
| |
| } |
| |
| mStreamTypes[stream].volume = value; |
| |
| if (thread == NULL) { |
| for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) |
| mPlaybackThreads[i]->setStreamVolume(stream, value); |
| |
| } else { |
| thread->setStreamVolume(stream, value); |
| } |
| |
| return ret; |
| } |
| |
| 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[i]->setStreamMute(stream, muted); |
| |
| return NO_ERROR; |
| } |
| |
| float AudioFlinger::streamVolume(int stream, void *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; |
| } |
| |
| // remove correction applied by setStreamVolume() |
| if (stream == AudioSystem::VOICE_CALL) { |
| volume = (volume - 0.01) / 0.99 ; |
| } |
| |
| return volume; |
| } |
| |
| bool AudioFlinger::streamMute(int stream) const |
| { |
| if (stream < 0 || stream >= (int)AudioSystem::NUM_STREAM_TYPES) { |
| return true; |
| } |
| |
| return mStreamTypes[stream].mute; |
| } |
| |
| bool AudioFlinger::isMusicActive() const |
| { |
| Mutex::Autolock _l(mLock); |
| for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { |
| if (mPlaybackThreads[i]->isMusicActive()) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| status_t AudioFlinger::setParameters(void *ioHandle, const String8& keyValuePairs) |
| { |
| status_t result; |
| |
| LOGV("setParameters(): io %p, keyvalue %s, tid %d, calling tid %d", |
| ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); |
| // check calling permissions |
| if (!settingsAllowed()) { |
| return PERMISSION_DENIED; |
| } |
| |
| // 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); |
| mHardwareStatus = AUDIO_HW_IDLE; |
| return result; |
| } |
| |
| // Check if parameters are for an output |
| PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); |
| if (playbackThread != NULL) { |
| return playbackThread->setParameters(keyValuePairs); |
| } |
| |
| // Check if parameters are for an input |
| RecordThread *recordThread = checkRecordThread_l(ioHandle); |
| if (recordThread != NULL) { |
| return recordThread->setParameters(keyValuePairs); |
| } |
| |
| return BAD_VALUE; |
| } |
| |
| String8 AudioFlinger::getParameters(void *ioHandle, const String8& keys) |
| { |
| // LOGV("getParameters() io %p, keys %s, tid %d, calling tid %d", |
| // ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); |
| |
| if (ioHandle == 0) { |
| return mAudioHardware->getParameters(keys); |
| } |
| 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); |
| } |
| |
| void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) |
| { |
| |
| LOGV("registerClient() %p, tid %d, calling tid %d", client.get(), gettid(), IPCThreadState::self()->getCallingPid()); |
| Mutex::Autolock _l(mLock); |
| |
| sp<IBinder> binder = client->asBinder(); |
| if (mNotificationClients.indexOf(binder) < 0) { |
| LOGV("Adding notification client %p", binder.get()); |
| binder->linkToDeath(this); |
| mNotificationClients.add(binder); |
| } |
| |
| // 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[i]->sendConfigEvent(AudioSystem::OUTPUT_OPENED); |
| } |
| |
| for (size_t i = 0; i < mRecordThreads.size(); i++) { |
| mRecordThreads[i]->sendConfigEvent(AudioSystem::INPUT_OPENED); |
| } |
| } |
| |
| void AudioFlinger::binderDied(const wp<IBinder>& who) { |
| |
| LOGV("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(), IPCThreadState::self()->getCallingPid()); |
| Mutex::Autolock _l(mLock); |
| |
| IBinder *binder = who.unsafe_get(); |
| |
| if (binder != NULL) { |
| int index = mNotificationClients.indexOf(binder); |
| if (index >= 0) { |
| LOGV("Removing notification client %p", binder); |
| mNotificationClients.removeAt(index); |
| } |
| } |
| } |
| |
| void AudioFlinger::audioConfigChanged(int event, void *param1, void *param2) { |
| Mutex::Autolock _l(mLock); |
| size_t size = mNotificationClients.size(); |
| for (size_t i = 0; i < size; i++) { |
| sp<IBinder> binder = mNotificationClients.itemAt(i); |
| LOGV("audioConfigChanged() Notifying change to client %p", binder.get()); |
| sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient> (binder); |
| client->ioConfigChanged(event, param1, param2); |
| } |
| } |
| |
| void AudioFlinger::removeClient(pid_t pid) |
| { |
| LOGV("removeClient() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); |
| Mutex::Autolock _l(mLock); |
| mClients.removeItem(pid); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger) |
| : Thread(false), |
| mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), |
| mFormat(0), mFrameSize(1), mNewParameters(String8("")), mStandby(false) |
| { |
| } |
| |
| AudioFlinger::ThreadBase::~ThreadBase() |
| { |
| } |
| |
| void AudioFlinger::ThreadBase::exit() |
| { |
| // keep a strong ref on ourself so that we want get |
| // destroyed in the middle of requestExitAndWait() |
| sp <ThreadBase> strongMe = this; |
| |
| LOGV("ThreadBase::exit"); |
| { |
| AutoMutex lock(&mLock); |
| requestExit(); |
| mWaitWorkCV.signal(); |
| } |
| requestExitAndWait(); |
| } |
| |
| uint32_t AudioFlinger::ThreadBase::sampleRate() const |
| { |
| return mSampleRate; |
| } |
| |
| int AudioFlinger::ThreadBase::channelCount() const |
| { |
| return 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 result; |
| |
| Mutex::Autolock _l(mLock); |
| mNewParameters = keyValuePairs; |
| |
| mWaitWorkCV.signal(); |
| mParamCond.wait(mLock); |
| |
| return mParamStatus; |
| } |
| |
| void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) |
| { |
| Mutex::Autolock _l(mLock); |
| 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 because audioConfigChanged() will call |
| // Audioflinger::audioConfigChanged() which locks AudioFlinger mLock thus creating |
| // potential cross deadlock between AudioFlinger::mLock and mLock |
| mLock.unlock(); |
| audioConfigChanged(configEvent->mEvent, configEvent->mParam); |
| delete configEvent; |
| mLock.lock(); |
| } |
| mLock.unlock(); |
| } |
| |
| |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output) |
| : ThreadBase(audioFlinger), |
| mOutput(output), |
| mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), |
| mInWrite(false), mMixBuffer(0), mSuspended(false), mBytesWritten(0) |
| { |
| 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); |
| } |
| // notify client processes that a new input has been opened |
| sendConfigEvent(AudioSystem::OUTPUT_OPENED); |
| } |
| |
| AudioFlinger::PlaybackThread::~PlaybackThread() |
| { |
| delete [] mMixBuffer; |
| if (mType != DUPLICATING) { |
| mAudioFlinger->mAudioHardware->closeOutputStream(mOutput); |
| } |
| } |
| |
| status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) |
| { |
| dumpInternals(fd, args); |
| dumpTracks(fd, args); |
| return NO_ERROR; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& 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> 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<Track> wTrack = mActiveTracks[i]; |
| if (wTrack != 0) { |
| sp<Track> 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<String16>& args) |
| { |
| const size_t SIZE = 256; |
| char buffer[SIZE]; |
| String8 result; |
| |
| snprintf(buffer, SIZE, "Output 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, "standby: %d\n", mStandby); |
| result.append(buffer); |
| write(fd, result.string(), result.size()); |
| 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::Track> AudioFlinger::PlaybackThread::createTrack_l( |
| const sp<AudioFlinger::Client>& client, |
| int streamType, |
| uint32_t sampleRate, |
| int format, |
| int channelCount, |
| int frameCount, |
| const sp<IMemory>& sharedBuffer, |
| status_t *status) |
| { |
| sp<Track> track; |
| status_t lStatus; |
| |
| if (mType == DIRECT) { |
| if (sampleRate != mSampleRate || format != mFormat || channelCount != 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) { |
| 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) |
| { |
| mMasterVolume = value; |
| return NO_ERROR; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) |
| { |
| 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) |
| { |
| mStreamTypes[stream].volume = value; |
| return NO_ERROR; |
| } |
| |
| status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) |
| { |
| 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::isMusicActive() const |
| { |
| Mutex::Autolock _l(mLock); |
| size_t count = mActiveTracks.size(); |
| for (size_t i = 0 ; i < count ; ++i) { |
| sp<Track> t = mActiveTracks[i].promote(); |
| if (t == 0) continue; |
| Track* const track = t.get(); |
| if (t->mStreamType == AudioSystem::MUSIC) |
| return true; |
| } |
| return false; |
| } |
| |
| // addTrack_l() must be called with ThreadBase::mLock held |
| status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) |
| { |
| status_t status = ALREADY_EXISTS; |
| |
| // here the track could be either new, or restarted |
| // in both cases "unstop" the track |
| if (track->isPaused()) { |
| track->mState = TrackBase::RESUMING; |
| LOGV("PAUSED => RESUMING (%d)", track->name()); |
| } else { |
| track->mState = TrackBase::ACTIVE; |
| LOGV("? => ACTIVE (%d)", track->name()); |
| } |
| // set retry count for buffer fill |
| track->mRetryCount = kMaxTrackStartupRetries; |
| if (mActiveTracks.indexOf(track) < 0) { |
| // the track is newly added, make sure it fills up all its |
| // buffers before playing. This is to ensure the client will |
| // effectively get the latency it requested. |
| track->mFillingUpStatus = Track::FS_FILLING; |
| track->mResetDone = false; |
| 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) |
| { |
| track->mState = TrackBase::TERMINATED; |
| if (mActiveTracks.indexOf(track) < 0) { |
| LOGV("remove track (%d) and delete from mixer", track->name()); |
| mTracks.remove(track); |
| deleteTrackName_l(track->name()); |
| } |
| } |
| |
| String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) |
| { |
| return mOutput->getParameters(keys); |
| } |
| |
| void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { |
| AudioSystem::OutputDescriptor desc; |
| void *param2 = 0; |
| |
| LOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, param); |
| |
| switch (event) { |
| case AudioSystem::OUTPUT_OPENED: |
| case AudioSystem::OUTPUT_CONFIG_CHANGED: |
| desc.channels = mChannelCount; |
| 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(event, this, param2); |
| } |
| |
| void AudioFlinger::PlaybackThread::readOutputParameters() |
| { |
| mSampleRate = mOutput->sampleRate(); |
| mChannelCount = AudioSystem::popCount(mOutput->channels()); |
| |
| mFormat = mOutput->format(); |
| mFrameSize = mOutput->frameSize(); |
| mFrameCount = mOutput->bufferSize() / mFrameSize; |
| |
| mMinBytesToWrite = (mOutput->latency() * mSampleRate * mFrameSize) / 1000; |
| // 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)); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output) |
| : PlaybackThread(audioFlinger, output), |
| 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() |
| { |
| unsigned long sleepTime = kBufferRecoveryInUsecs; |
| int16_t* curBuf = mMixBuffer; |
| Vector< sp<Track> > tracksToRemove; |
| size_t enabledTracks = 0; |
| nsecs_t standbyTime = systemTime(); |
| size_t mixBufferSize = mFrameCount * mFrameSize; |
| nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 2; |
| |
| while (!exitPending()) |
| { |
| processConfigEvents(); |
| |
| enabledTracks = 0; |
| { // scope for mLock |
| |
| Mutex::Autolock _l(mLock); |
| |
| if (checkForNewParameters_l()) { |
| mixBufferSize = mFrameCount * mFrameSize; |
| maxPeriod = seconds(mFrameCount) / mSampleRate * 2; |
| } |
| |
| const SortedVector< wp<Track> >& 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; |
| continue; |
| } |
| } |
| |
| enabledTracks = prepareTracks_l(activeTracks, &tracksToRemove); |
| } |
| |
| if (LIKELY(enabledTracks)) { |
| // mix buffers... |
| mAudioMixer->process(curBuf); |
| |
| // output audio to hardware |
| if (mSuspended) { |
| usleep(kMaxBufferRecoveryInUsecs); |
| } else { |
| mLastWriteTime = systemTime(); |
| mInWrite = true; |
| int bytesWritten = (int)mOutput->write(curBuf, mixBufferSize); |
| if (bytesWritten > 0) mBytesWritten += bytesWritten; |
| mNumWrites++; |
| mInWrite = false; |
| mStandby = false; |
| nsecs_t temp = systemTime(); |
| standbyTime = temp + kStandbyTimeInNsecs; |
| nsecs_t delta = temp - mLastWriteTime; |
| if (delta > maxPeriod) { |
| LOGW("write blocked for %llu msecs", ns2ms(delta)); |
| mNumDelayedWrites++; |
| } |
| sleepTime = kBufferRecoveryInUsecs; |
| } |
| } else { |
| // 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("thread %p no buffers - usleep(%lu)", this, 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(); |
| } |
| |
| if (!mStandby) { |
| mOutput->standby(); |
| } |
| sendConfigEvent(AudioSystem::OUTPUT_CLOSED); |
| processConfigEvents(); |
| |
| LOGV("MixerThread %p exiting", this); |
| return false; |
| } |
| |
| // prepareTracks_l() must be called with ThreadBase::mLock held |
| size_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) |
| { |
| |
| size_t enabledTracks = 0; |
| // find out which tracks need to be processed |
| size_t count = activeTracks.size(); |
| for (size_t i=0 ; i<count ; i++) { |
| sp<Track> t = activeTracks[i].promote(); |
| if (t == 0) continue; |
| |
| Track* const track = t.get(); |
| audio_track_cblk_t* cblk = track->cblk(); |
| |
| // The first time a track is added we wait |
| // for all its buffers to be filled before processing it |
| mAudioMixer->setActiveTrack(track->name()); |
| if (cblk->framesReady() && (track->isReady() || track->isStopped()) && |
| !track->isPaused()) |
| { |
| //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); |
| |
| // compute volume for this track |
| int16_t left, right; |
| if (track->isMuted() || mMasterMute || track->isPausing() || |
| 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 = int16_t(v_clamped); |
| v_clamped = v * cblk->volume[1]; |
| if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; |
| right = int16_t(v_clamped); |
| } |
| |
| // XXX: these things DON'T need to be done each time |
| mAudioMixer->setBufferProvider(track); |
| mAudioMixer->enable(AudioMixer::MIXING); |
| |
| int param; |
| if ( track->mFillingUpStatus == Track::FS_FILLED) { |
| // no ramp for the first volume setting |
| track->mFillingUpStatus = Track::FS_ACTIVE; |
| if (track->mState == TrackBase::RESUMING) { |
| track->mState = TrackBase::ACTIVE; |
| param = AudioMixer::RAMP_VOLUME; |
| } else { |
| param = AudioMixer::VOLUME; |
| } |
| } else { |
| param = AudioMixer::RAMP_VOLUME; |
| } |
| mAudioMixer->setParameter(param, AudioMixer::VOLUME0, left); |
| mAudioMixer->setParameter(param, AudioMixer::VOLUME1, right); |
| mAudioMixer->setParameter( |
| AudioMixer::TRACK, |
| AudioMixer::FORMAT, track->format()); |
| mAudioMixer->setParameter( |
| AudioMixer::TRACK, |
| AudioMixer::CHANNEL_COUNT, track->channelCount()); |
| mAudioMixer->setParameter( |
| AudioMixer::RESAMPLE, |
| AudioMixer::SAMPLE_RATE, |
| int(cblk->sampleRate)); |
| |
| // reset retry count |
| track->mRetryCount = kMaxTrackRetries; |
| enabledTracks++; |
| } else { |
| //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); |
| if (track->isStopped()) { |
| track->reset(); |
| } |
| if (track->isTerminated() || track->isStopped() || track->isPaused()) { |
| // We have consumed all the buffers of this track. |
| // Remove it from the list of active tracks. |
| 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", track->name()); |
| tracksToRemove->add(track); |
| } |
| // For tracks using static shared memry buffer, make sure that we have |
| // written enough data to audio hardware before disabling the track |
| // NOTE: this condition with arrive before track->mRetryCount <= 0 so we |
| // don't care about code removing track from active list above. |
| if ((track->mSharedBuffer == 0) || (mBytesWritten >= mMinBytesToWrite)) { |
| mAudioMixer->disable(AudioMixer::MIXING); |
| } else { |
| enabledTracks++; |
| } |
| } |
| } |
| } |
| |
| // remove all the tracks that need to be... |
| count = tracksToRemove->size(); |
| if (UNLIKELY(count)) { |
| for (size_t i=0 ; i<count ; i++) { |
| const sp<Track>& track = tracksToRemove->itemAt(i); |
| mActiveTracks.remove(track); |
| if (track->isTerminated()) { |
| mTracks.remove(track); |
| deleteTrackName_l(track->mName); |
| } |
| } |
| } |
| |
| return enabledTracks; |
| } |
| |
| void AudioFlinger::MixerThread::getTracks( |
| SortedVector < sp<Track> >& tracks, |
| SortedVector < wp<Track> >& activeTracks, |
| int streamType) |
| { |
| LOGV ("MixerThread::getTracks() mixer %p, mTracks.size %d, mActiveTracks.size %d", this, mTracks.size(), mActiveTracks.size()); |
| Mutex::Autolock _l(mLock); |
| size_t size = mTracks.size(); |
| for (size_t i = 0; i < size; i++) { |
| sp<Track> t = mTracks[i]; |
| if (t->type() == streamType) { |
| tracks.add(t); |
| int j = mActiveTracks.indexOf(t); |
| if (j >= 0) { |
| t = mActiveTracks[j].promote(); |
| if (t != NULL) { |
| activeTracks.add(t); |
| } |
| } |
| } |
| } |
| |
| size = activeTracks.size(); |
| for (size_t i = 0; i < size; i++) { |
| mActiveTracks.remove(activeTracks[i]); |
| } |
| |
| size = tracks.size(); |
| for (size_t i = 0; i < size; i++) { |
| sp<Track> t = tracks[i]; |
| mTracks.remove(t); |
| deleteTrackName_l(t->name()); |
| } |
| } |
| |
| void AudioFlinger::MixerThread::putTracks( |
| SortedVector < sp<Track> >& tracks, |
| SortedVector < wp<Track> >& activeTracks) |
| { |
| LOGV ("MixerThread::putTracks() mixer %p, tracks.size %d, activeTracks.size %d", this, tracks.size(), activeTracks.size()); |
| Mutex::Autolock _l(mLock); |
| size_t size = tracks.size(); |
| for (size_t i = 0; i < size ; i++) { |
| sp<Track> t = tracks[i]; |
| int name = getTrackName_l(); |
| |
| if (name < 0) return; |
| |
| t->mName = name; |
| t->mThread = this; |
| mTracks.add(t); |
| |
| int j = activeTracks.indexOf(t); |
| if (j >= 0) { |
| mActiveTracks.add(t); |
| } |
| } |
| } |
| |
| // 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) |
| { |
| mAudioMixer->deleteTrackName(name); |
| } |
| |
| // checkForNewParameters_l() must be called with ThreadBase::mLock held |
| bool AudioFlinger::MixerThread::checkForNewParameters_l() |
| { |
| bool reconfig = false; |
| |
| if (mNewParameters != "") { |
| status_t status = NO_ERROR; |
| AudioParameter param = AudioParameter(mNewParameters); |
| 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(mNewParameters); |
| if (!mStandby && status == INVALID_OPERATION) { |
| mOutput->standby(); |
| mStandby = true; |
| mBytesWritten = 0; |
| status = mOutput->setParameters(mNewParameters); |
| } |
| 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; |
| } |
| sendConfigEvent(AudioSystem::OUTPUT_CONFIG_CHANGED); |
| } |
| } |
| mParamStatus = status; |
| mNewParameters = ""; |
| mParamCond.signal(); |
| } |
| return reconfig; |
| } |
| |
| status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& 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; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output) |
| : PlaybackThread(audioFlinger, output), |
| mLeftVolume (1.0), mRightVolume(1.0) |
| { |
| mType = PlaybackThread::DIRECT; |
| } |
| |
| AudioFlinger::DirectOutputThread::~DirectOutputThread() |
| { |
| } |
| |
| |
| bool AudioFlinger::DirectOutputThread::threadLoop() |
| { |
| unsigned long sleepTime = kBufferRecoveryInUsecs; |
| sp<Track> trackToRemove; |
| sp<Track> activeTrack; |
| nsecs_t standbyTime = systemTime(); |
| int8_t *curBuf; |
| size_t mixBufferSize = mFrameCount*mFrameSize; |
| |
| while (!exitPending()) |
| { |
| processConfigEvents(); |
| |
| { // scope for the mLock |
| |
| Mutex::Autolock _l(mLock); |
| |
| if (checkForNewParameters_l()) { |
| mixBufferSize = mFrameCount*mFrameSize; |
| } |
| |
| // 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() + kStandbyTimeInNsecs; |
| continue; |
| } |
| } |
| |
| // find out which tracks need to be processed |
| if (mActiveTracks.size() != 0) { |
| sp<Track> 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()) |
| { |
| //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 = kMaxTrackRetries; |
| activeTrack = t; |
| } 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; |
| } |
| |
| // For tracks using static shared memry buffer, make sure that we have |
| // written enough data to audio hardware before disabling the track |
| // NOTE: this condition with arrive before track->mRetryCount <= 0 so we |
| // don't care about code removing track from active list above. |
| if ((track->mSharedBuffer != 0) && (mBytesWritten < mMinBytesToWrite)) { |
| activeTrack = t; |
| } |
| } |
| } |
| } |
| |
| // 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 (activeTrack != 0) { |
| AudioBufferProvider::Buffer buffer; |
| size_t frameCount = mFrameCount; |
| curBuf = (int8_t *)mMixBuffer; |
| // output audio to hardware |
| mLastWriteTime = systemTime(); |
| mInWrite = true; |
| 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); |
| } |
| if (mSuspended) { |
| usleep(kMaxBufferRecoveryInUsecs); |
| } else { |
| int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); |
| if (bytesWritten) mBytesWritten += bytesWritten; |
| mNumWrites++; |
| mInWrite = false; |
| mStandby = false; |
| nsecs_t temp = systemTime(); |
| standbyTime = temp + kStandbyTimeInNsecs; |
| 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 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(); |
| } |
| sendConfigEvent(AudioSystem::OUTPUT_CLOSED); |
| processConfigEvents(); |
| |
| 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; |
| |
| if (mNewParameters != "") { |
| status_t status = NO_ERROR; |
| AudioParameter param = AudioParameter(mNewParameters); |
| 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(mNewParameters); |
| if (!mStandby && status == INVALID_OPERATION) { |
| mOutput->standby(); |
| mStandby = true; |
| mBytesWritten = 0; |
| status = mOutput->setParameters(mNewParameters); |
| } |
| if (status == NO_ERROR && reconfig) { |
| readOutputParameters(); |
| sendConfigEvent(AudioSystem::OUTPUT_CONFIG_CHANGED); |
| } |
| } |
| mParamStatus = status; |
| mNewParameters = ""; |
| mParamCond.signal(); |
| } |
| return reconfig; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread) |
| : MixerThread(audioFlinger, mainThread->getOutput()) |
| { |
| mType = PlaybackThread::DUPLICATING; |
| addOutputTrack(mainThread); |
| } |
| |
| AudioFlinger::DuplicatingThread::~DuplicatingThread() |
| { |
| mOutputTracks.clear(); |
| } |
| |
| bool AudioFlinger::DuplicatingThread::threadLoop() |
| { |
| unsigned long sleepTime = kBufferRecoveryInUsecs; |
| int16_t* curBuf = mMixBuffer; |
| Vector< sp<Track> > tracksToRemove; |
| size_t enabledTracks = 0; |
| nsecs_t standbyTime = systemTime(); |
| size_t mixBufferSize = mFrameCount*mFrameSize; |
| SortedVector< sp<OutputTrack> > outputTracks; |
| |
| while (!exitPending()) |
| { |
| processConfigEvents(); |
| |
| enabledTracks = 0; |
| { // scope for the mLock |
| |
| Mutex::Autolock _l(mLock); |
| |
| if (checkForNewParameters_l()) { |
| mixBufferSize = mFrameCount*mFrameSize; |
| } |
| |
| const SortedVector< wp<Track> >& 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++) { |
| mLock.unlock(); |
| outputTracks[i]->stop(); |
| mLock.lock(); |
| } |
| 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 = kBufferRecoveryInUsecs; |
| continue; |
| } |
| } |
| |
| enabledTracks = prepareTracks_l(activeTracks, &tracksToRemove); |
| } |
| |
| bool mustSleep = true; |
| if (LIKELY(enabledTracks)) { |
| // mix buffers... |
| mAudioMixer->process(curBuf); |
| if (!mSuspended) { |
| for (size_t i = 0; i < outputTracks.size(); i++) { |
| outputTracks[i]->write(curBuf, mFrameCount); |
| } |
| mStandby = false; |
| mustSleep = false; |
| mBytesWritten += mixBufferSize; |
| } |
| } else { |
| // flush remaining overflow buffers in output tracks |
| for (size_t i = 0; i < outputTracks.size(); i++) { |
| if (outputTracks[i]->isActive()) { |
| outputTracks[i]->write(curBuf, 0); |
| standbyTime = systemTime() + kStandbyTimeInNsecs; |
| mustSleep = false; |
| } |
| } |
| } |
| if (mustSleep) { |
| // LOGV("threadLoop() sleeping %d", sleepTime); |
| usleep(sleepTime); |
| if (sleepTime < kMaxBufferRecoveryInUsecs) { |
| sleepTime += kBufferRecoveryInUsecs; |
| } |
| } else { |
| 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(); |
| outputTracks.clear(); |
| } |
| |
| if (!mStandby) { |
| for (size_t i = 0; i < outputTracks.size(); i++) { |
| mLock.unlock(); |
| outputTracks[i]->stop(); |
| mLock.lock(); |
| } |
| } |
| |
| sendConfigEvent(AudioSystem::OUTPUT_CLOSED); |
| processConfigEvents(); |
| |
| return false; |
| } |
| |
| void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) |
| { |
| int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); |
| OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, |
| mSampleRate, |
| mFormat, |
| mChannelCount, |
| frameCount); |
| thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f); |
| mOutputTracks.add(outputTrack); |
| LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); |
| } |
| |
| 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.removeAt(i); |
| return; |
| } |
| } |
| LOGV("removeOutputTrack(): unkonwn thread: %p", thread); |
| } |
| |
| |
| // ---------------------------------------------------------------------------- |
| |
| // TrackBase constructor must be called with AudioFlinger::mLock held |
| AudioFlinger::ThreadBase::TrackBase::TrackBase( |
| const wp<ThreadBase>& thread, |
| const sp<Client>& client, |
| uint32_t sampleRate, |
| int format, |
| int channelCount, |
| int frameCount, |
| uint32_t flags, |
| const sp<IMemory>& sharedBuffer) |
| : RefBase(), |
| mThread(thread), |
| mClient(client), |
| 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<audio_track_cblk_t *>(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 = (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->flowControlFlag = 1; |
| } else { |
| mBuffer = sharedBuffer->pointer(); |
| } |
| mBufferEnd = (uint8_t *)mBuffer + bufferSize; |
| } |
| } else { |
| LOGE("not enough memory for AudioTrack size=%u", size); |
| client->heap()->dump("AudioTrack"); |
| return; |
| } |
| } else { |
| mCblk = (audio_track_cblk_t *)(new uint8_t[size]); |
| if (mCblk) { // construct the shared structure in-place. |
| new(mCblk) audio_track_cblk_t(); |
| // clear all buffers |
| mCblk->frameCount = frameCount; |
| mCblk->sampleRate = sampleRate; |
| mCblk->channels = (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->flowControlFlag = 1; |
| mBufferEnd = (uint8_t *)mBuffer + bufferSize; |
| } |
| } |
| } |
| |
| AudioFlinger::PlaybackThread::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 |
| mClient.clear(); |
| } |
| |
| void AudioFlinger::PlaybackThread::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) |
| { |
| buffer->raw = 0; |
| mFrameCount = buffer->frameCount; |
| step(); |
| buffer->frameCount = 0; |
| } |
| |
| bool AudioFlinger::PlaybackThread::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::PlaybackThread::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<IMemory> AudioFlinger::PlaybackThread::TrackBase::getCblk() const |
| { |
| return mCblkMemory; |
| } |
| |
| int AudioFlinger::PlaybackThread::TrackBase::sampleRate() const { |
| return (int)mCblk->sampleRate; |
| } |
| |
| int AudioFlinger::PlaybackThread::TrackBase::channelCount() const { |
| return (int)mCblk->channels; |
| } |
| |
| void* AudioFlinger::PlaybackThread::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, channels %d", |
| bufferStart, bufferEnd, mBuffer, mBufferEnd, |
| cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channels); |
| return 0; |
| } |
| |
| return bufferStart; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| // Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held |
| AudioFlinger::PlaybackThread::Track::Track( |
| const wp<ThreadBase>& thread, |
| const sp<Client>& client, |
| int streamType, |
| uint32_t sampleRate, |
| int format, |
| int channelCount, |
| int frameCount, |
| const sp<IMemory>& sharedBuffer) |
| : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer), |
| mMute(false), mSharedBuffer(sharedBuffer), mName(-1) |
| { |
| sp<ThreadBase> 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<ThreadBase> 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<Track> keep(this); |
| { // scope for mLock |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| 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 %3u %1d %1d %1d %5u %5u %5u %04x %04x\n", |
| mName - AudioMixer::TRACK0, |
| (mClient == NULL) ? getpid() : mClient->pid(), |
| mStreamType, |
| mFormat, |
| mCblk->channels, |
| mFrameCount, |
| mState, |
| mMute, |
| mFillingUpStatus, |
| mCblk->sampleRate, |
| mCblk->volume[0], |
| mCblk->volume[1], |
| mCblk->server, |
| mCblk->user); |
| } |
| |
| status_t AudioFlinger::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"); |
| return NOT_ENOUGH_DATA; |
| } |
| |
| bool AudioFlinger::PlaybackThread::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::PlaybackThread::Track::start() |
| { |
| LOGV("start(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| Mutex::Autolock _l(thread->mLock); |
| PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); |
| playbackThread->addTrack_l(this); |
| } |
| return NO_ERROR; |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::stop() |
| { |
| LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| Mutex::Autolock _l(thread->mLock); |
| 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)", mName); |
| } |
| } |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::pause() |
| { |
| LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); |
| sp<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| Mutex::Autolock _l(thread->mLock); |
| if (mState == ACTIVE || mState == RESUMING) { |
| mState = PAUSING; |
| LOGV("ACTIVE/RESUMING => PAUSING (%d)", mName); |
| } |
| } |
| } |
| |
| void AudioFlinger::PlaybackThread::Track::flush() |
| { |
| LOGV("flush(%d)", mName); |
| sp<ThreadBase> 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->flowControlFlag = 1; |
| mCblk->forceReady = 0; |
| 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<ThreadBase>& thread, |
| const sp<Client>& client, |
| uint32_t sampleRate, |
| int format, |
| int channelCount, |
| int frameCount, |
| uint32_t flags) |
| : TrackBase(thread, client, sampleRate, format, |
| channelCount, frameCount, flags, 0), |
| mOverflow(false) |
| { |
| 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() |
| { |
| } |
| |
| 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<ThreadBase> thread = mThread.promote(); |
| if (thread != 0) { |
| RecordThread *recordThread = (RecordThread *)thread.get(); |
| return recordThread->start(this); |
| } |
| return NO_INIT; |
| } |
| |
| void AudioFlinger::RecordThread::RecordTrack::stop() |
| { |
| sp<ThreadBase> 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->flowControlFlag = 1; |
| } |
| } |
| |
| |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( |
| const wp<ThreadBase>& thread, |
| uint32_t sampleRate, |
| int format, |
| int channelCount, |
| int frameCount) |
| : Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL), |
| mActive(false) |
| { |
| |
| PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); |
| mCblk->out = 1; |
| mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); |
| mCblk->volume[0] = mCblk->volume[1] = 0x1000; |
| mOutBuffer.frameCount = 0; |
| mWaitTimeMs = (playbackThread->frameCount() * 2 * 1000) / playbackThread->sampleRate(); |
| |
| LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channels %d mBufferEnd %p mWaitTimeMs %d", |
| mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channels, mBufferEnd, mWaitTimeMs); |
| |
| } |
| |
| AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() |
| { |
| stop(); |
| } |
| |
| 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 channels = mCblk->channels; |
| bool outputBufferFull = false; |
| inBuffer.frameCount = frames; |
| inBuffer.i16 = data; |
| |
| uint32_t waitTimeLeftMs = mWaitTimeMs; |
| |
| if (!mActive) { |
| start(); |
| sp<ThreadBase> 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 * channels]; |
| pInBuffer->frameCount = startFrames; |
| pInBuffer->i16 = pInBuffer->mBuffer; |
| memset(pInBuffer->raw, 0, startFrames * channels * 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 no more output buffers", this); |
| outputBufferFull = true; |
| break; |
| } |
| uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); |
| // LOGV("OutputTrack::write() waitTimeMs %d waitTimeLeftMs %d", waitTimeMs, waitTimeLeftMs) |
| 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 * channels * sizeof(int16_t)); |
| mCblk->stepUser(outFrames); |
| pInBuffer->frameCount -= outFrames; |
| pInBuffer->i16 += outFrames * channels; |
| mOutBuffer.frameCount -= outFrames; |
| mOutBuffer.i16 += outFrames * channels; |
| |
| if (pInBuffer->frameCount == 0) { |
| if (mBufferQueue.size()) { |
| mBufferQueue.removeAt(0); |
| delete [] pInBuffer->mBuffer; |
| delete pInBuffer; |
| LOGV("OutputTrack::write() %p released overflow buffer %d", this, mBufferQueue.size()); |
| } else { |
| break; |
| } |
| } |
| } |
| |
| // If we could not write all frames, allocate a buffer and queue it for next time. |
| if (inBuffer.frameCount) { |
| if (mBufferQueue.size() < kMaxOverFlowBuffers) { |
| pInBuffer = new Buffer; |
| pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channels]; |
| pInBuffer->frameCount = inBuffer.frameCount; |
| pInBuffer->i16 = pInBuffer->mBuffer; |
| memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channels * sizeof(int16_t)); |
| mBufferQueue.add(pInBuffer); |
| LOGV("OutputTrack::write() %p adding overflow buffer %d", this, mBufferQueue.size()); |
| } else { |
| LOGW("OutputTrack::write() %p no more overflow buffers", 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 * channels]; |
| pInBuffer->frameCount = frames; |
| pInBuffer->i16 = pInBuffer->mBuffer; |
| memset(pInBuffer->raw, 0, frames * channels * sizeof(int16_t)); |
| mBufferQueue.add(pInBuffer); |
| } else { |
| 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>& 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<MemoryDealer>& AudioFlinger::Client::heap() const |
| { |
| return mMemoryDealer; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& 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<IMemory> 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<IAudioRecord> AudioFlinger::openRecord( |
| pid_t pid, |
| void *input, |
| uint32_t sampleRate, |
| int format, |
| int channelCount, |
| int frameCount, |
| uint32_t flags, |
| status_t *status) |
| { |
| sp<RecordThread::RecordTrack> recordTrack; |
| sp<RecordHandle> recordHandle; |
| sp<Client> client; |
| wp<Client> 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) { |
| 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<AudioFlinger::RecordThread::RecordTrack>& 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<IMemory> 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>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels) : |
| ThreadBase(audioFlinger), |
| mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) |
| { |
| mReqChannelCount = AudioSystem::popCount(channels); |
| mReqSampleRate = sampleRate; |
| readInputParameters(); |
| sendConfigEvent(AudioSystem::INPUT_OPENED); |
| } |
| |
| |
| AudioFlinger::RecordThread::~RecordThread() |
| { |
| mAudioFlinger->mAudioHardware->closeInputStream(mInput); |
| 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<RecordTrack> 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) { |
| mActiveTrack.clear(); |
| mStartStopCond.broadcast(); |
| } else if (mActiveTrack->mState == TrackBase::RESUMING) { |
| mRsmpInIndex = mFrameCount; |
| if (mReqChannelCount != mActiveTrack->channelCount()) { |
| mActiveTrack.clear(); |
| } else { |
| mActiveTrack->mState == TrackBase::ACTIVE; |
| } |
| mStartStopCond.broadcast(); |
| } |
| mStandby = false; |
| } |
| } |
| |
| if (mActiveTrack != 0) { |
| 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 (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) { |
| ssize_t bytesRead; |
| if (framesOut == mFrameCount && |
| (mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) { |
| bytesRead = mInput->read(buffer.raw, mInputBytes); |
| framesOut = 0; |
| } else { |
| bytesRead = mInput->read(mRsmpInBuffer, mInputBytes); |
| mRsmpInIndex = 0; |
| } |
| if (bytesRead < 0) { |
| LOGE("Error reading audio input"); |
| sleep(1); |
| 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(); |
| |
| sendConfigEvent(AudioSystem::INPUT_CLOSED); |
| processConfigEvents(); |
| |
| LOGV("RecordThread %p exiting", this); |
| return false; |
| } |
| |
| status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) |
| { |
| LOGV("RecordThread::start"); |
| AutoMutex lock(&mLock); |
| |
| if (mActiveTrack != 0) { |
| if (recordTrack != mActiveTrack.get()) return -EBUSY; |
| |
| if (mActiveTrack->mState == TrackBase::PAUSING) mActiveTrack->mState = TrackBase::RESUMING; |
| |
| return NO_ERROR; |
| } |
| |
| mActiveTrack = recordTrack; |
| mActiveTrack->mState = TrackBase::RESUMING; |
| // signal thread to start |
| LOGV("Signal record thread"); |
| mWaitWorkCV.signal(); |
| mStartStopCond.wait(mLock); |
| if (mActiveTrack != 0) { |
| LOGV("Record started OK"); |
| return NO_ERROR; |
| } else { |
| LOGV("Record failed to start"); |
| return BAD_VALUE; |
| } |
| } |
| |
| void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { |
| LOGV("RecordThread::stop"); |
| AutoMutex lock(&mLock); |
| if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { |
| mActiveTrack->mState = TrackBase::PAUSING; |
| mStartStopCond.wait(mLock); |
| } |
| } |
| |
| status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) |
| { |
| const size_t SIZE = 256; |
| char buffer[SIZE]; |
| String8 result; |
| pid_t pid = 0; |
| |
| if (mActiveTrack != 0 && mActiveTrack->mClient != 0) { |
| snprintf(buffer, SIZE, "Record client pid: %d\n", mActiveTrack->mClient->pid()); |
| result.append(buffer); |
| } else { |
| result.append("No record client\n"); |
| } |
| write(fd, result.string(), result.size()); |
| 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) { |
| ssize_t bytesRead = mInput->read(mRsmpInBuffer, mInputBytes); |
| if (bytesRead < 0) { |
| LOGE("RecordThread::getNextBuffer() Error reading audio input"); |
| sleep(1); |
| 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; |
| |
| if (mNewParameters != "") { |
| status_t status = NO_ERROR; |
| AudioParameter param = AudioParameter(mNewParameters); |
| 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(mNewParameters); |
| if (status == INVALID_OPERATION) { |
| mInput->standby(); |
| status = mInput->setParameters(mNewParameters); |
| } |
| 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(AudioSystem::INPUT_CONFIG_CHANGED); |
| } |
| } |
| } |
| mNewParameters = ""; |
| mParamStatus = status; |
| mParamCond.signal(); |
| } |
| return reconfig; |
| } |
| |
| String8 AudioFlinger::RecordThread::getParameters(const String8& keys) |
| { |
| return mInput->getParameters(keys); |
| } |
| |
| void AudioFlinger::RecordThread::audioConfigChanged(int event, int param) { |
| AudioSystem::OutputDescriptor desc; |
| void *param2 = 0; |
| |
| switch (event) { |
| case AudioSystem::INPUT_OPENED: |
| case AudioSystem::INPUT_CONFIG_CHANGED: |
| desc.channels = mChannelCount; |
| desc.samplingRate = mSampleRate; |
| desc.format = mFormat; |
| desc.frameCount = mFrameCount; |
| desc.latency = 0; |
| param2 = &desc; |
| break; |
| |
| case AudioSystem::INPUT_CLOSED: |
| default: |
| break; |
| } |
| mAudioFlinger->audioConfigChanged(event, this, param2); |
| } |
| |
| void AudioFlinger::RecordThread::readInputParameters() |
| { |
| if (mRsmpInBuffer) delete mRsmpInBuffer; |
| if (mRsmpOutBuffer) delete mRsmpOutBuffer; |
| if (mResampler) delete mResampler; |
| mResampler = 0; |
| |
| mSampleRate = mInput->sampleRate(); |
| mChannelCount = AudioSystem::popCount(mInput->channels()); |
| mFormat = mInput->format(); |
| mFrameSize = 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; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| void *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 NULL; |
| } |
| 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); |
| LOGV("openOutput() created direct output %p", thread); |
| } else { |
| thread = new MixerThread(this, output); |
| LOGV("openOutput() created mixer output %p", thread); |
| } |
| mPlaybackThreads.add(thread); |
| |
| if (pSamplingRate) *pSamplingRate = samplingRate; |
| if (pFormat) *pFormat = format; |
| if (pChannels) *pChannels = channels; |
| if (pLatencyMs) *pLatencyMs = thread->latency(); |
| } |
| |
| return thread; |
| } |
| |
| void *AudioFlinger::openDuplicateOutput(void *output1, void *output2) |
| { |
| Mutex::Autolock _l(mLock); |
| |
| if (checkMixerThread_l(output1) == NULL || |
| checkMixerThread_l(output2) == NULL) { |
| LOGW("openDuplicateOutput() wrong output mixer type %p or %p", output1, output2); |
| return NULL; |
| } |
| |
| DuplicatingThread *thread = new DuplicatingThread(this, (MixerThread *)output1); |
| thread->addOutputTrack( (MixerThread *)output2); |
| mPlaybackThreads.add(thread); |
| return thread; |
| } |
| |
| status_t AudioFlinger::closeOutput(void *output) |
| { |
| PlaybackThread *thread; |
| { |
| Mutex::Autolock _l(mLock); |
| thread = checkPlaybackThread_l(output); |
| if (thread == NULL) { |
| return BAD_VALUE; |
| } |
| |
| LOGV("closeOutput() %p", thread); |
| |
| if (thread->type() == PlaybackThread::MIXER) { |
| for (size_t i = 0; i < mPlaybackThreads.size(); i++) { |
| if (mPlaybackThreads[i]->type() == PlaybackThread::DUPLICATING) { |
| DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads[i].get(); |
| dupThread->removeOutputTrack((MixerThread *)thread); |
| } |
| } |
| } |
| mPlaybackThreads.remove(thread); |
| } |
| thread->exit(); |
| |
| return NO_ERROR; |
| } |
| |
| status_t AudioFlinger::suspendOutput(void *output) |
| { |
| Mutex::Autolock _l(mLock); |
| PlaybackThread *thread = checkPlaybackThread_l(output); |
| |
| if (thread == NULL) { |
| return BAD_VALUE; |
| } |
| |
| LOGV("suspendOutput() %p", output); |
| thread->suspend(); |
| |
| return NO_ERROR; |
| } |
| |
| status_t AudioFlinger::restoreOutput(void *output) |
| { |
| Mutex::Autolock _l(mLock); |
| PlaybackThread *thread = checkPlaybackThread_l(output); |
| |
| if (thread == NULL) { |
| return BAD_VALUE; |
| } |
| |
| LOGV("restoreOutput() %p", output); |
| |
| thread->restore(); |
| |
| return NO_ERROR; |
| } |
| |
| void *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 NULL; |
| } |
| 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); |
| mRecordThreads.add(thread); |
| |
| if (pSamplingRate) *pSamplingRate = reqSamplingRate; |
| if (pFormat) *pFormat = format; |
| if (pChannels) *pChannels = reqChannels; |
| |
| input->standby(); |
| } |
| |
| return thread; |
| } |
| |
| status_t AudioFlinger::closeInput(void *input) |
| { |
| RecordThread *thread; |
| { |
| Mutex::Autolock _l(mLock); |
| thread = checkRecordThread_l(input); |
| if (thread == NULL) { |
| return BAD_VALUE; |
| } |
| |
| LOGV("closeInput() %p", thread); |
| mRecordThreads.remove(thread); |
| } |
| thread->exit(); |
| |
| return NO_ERROR; |
| } |
| |
| status_t AudioFlinger::setStreamOutput(uint32_t stream, void *output) |
| { |
| Mutex::Autolock _l(mLock); |
| MixerThread *dstThread = checkMixerThread_l(output); |
| if (dstThread == NULL) { |
| LOGW("setStreamOutput() bad output thread %p", output); |
| return BAD_VALUE; |
| } |
| |
| LOGV("setStreamOutput() stream %d to output %p", stream, dstThread); |
| |
| for (size_t i = 0; i < mPlaybackThreads.size(); i++) { |
| PlaybackThread *thread = mPlaybackThreads[i].get(); |
| if (thread != dstThread && |
| thread->type() != PlaybackThread::DIRECT) { |
| MixerThread *srcThread = (MixerThread *)thread; |
| SortedVector < sp<MixerThread::Track> > tracks; |
| SortedVector < wp<MixerThread::Track> > activeTracks; |
| srcThread->getTracks(tracks, activeTracks, stream); |
| if (tracks.size()) { |
| dstThread->putTracks(tracks, activeTracks); |
| dstThread->sendConfigEvent(AudioSystem::STREAM_CONFIG_CHANGED, stream); |
| } |
| } |
| } |
| |
| return NO_ERROR; |
| } |
| |
| // checkPlaybackThread_l() must be called with AudioFlinger::mLock held |
| AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(void *output) const |
| { |
| PlaybackThread *thread = NULL; |
| |
| for (size_t i = 0; i < mPlaybackThreads.size(); i++) { |
| if (mPlaybackThreads[i] == output) { |
| thread = (PlaybackThread *)output; |
| break; |
| } |
| } |
| |
| return thread; |
| } |
| |
| // checkMixerThread_l() must be called with AudioFlinger::mLock held |
| AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(void *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(void *input) const |
| { |
| RecordThread *thread = NULL; |
| |
| for (size_t i = 0; i < mRecordThreads.size(); i++) { |
| if (mRecordThreads[i] == input) { |
| thread = (RecordThread *)input; |
| break; |
| } |
| } |
| |
| 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 |