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Diffstat (limited to 'services/audioflinger/AudioFlinger.cpp')
| -rw-r--r-- | services/audioflinger/AudioFlinger.cpp | 6078 |
1 files changed, 6078 insertions, 0 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp new file mode 100644 index 0000000..97eb6c0 --- /dev/null +++ b/services/audioflinger/AudioFlinger.cpp @@ -0,0 +1,6078 @@ +/* //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 <private/media/AudioEffectShared.h> +#include <hardware_legacy/AudioHardwareInterface.h> + +#include "AudioMixer.h" +#include "AudioFlinger.h" + +#ifdef WITH_A2DP +#include "A2dpAudioInterface.h" +#endif + +#ifdef LVMX +#include "lifevibes.h" +#endif + +#include <media/EffectsFactoryApi.h> +#include <media/EffectVisualizerApi.h> + +// ---------------------------------------------------------------------------- +// the sim build doesn't have gettid + +#ifndef HAVE_GETTID +# define gettid getpid +#endif + +// ---------------------------------------------------------------------------- + +namespace android { + +static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n"; +static const char* kHardwareLockedString = "Hardware lock is taken\n"; + +//static const nsecs_t kStandbyTimeInNsecs = seconds(3); +static const float MAX_GAIN = 4096.0f; +static const float MAX_GAIN_INT = 0x1000; + +// retry counts for buffer fill timeout +// 50 * ~20msecs = 1 second +static const int8_t kMaxTrackRetries = 50; +static const int8_t kMaxTrackStartupRetries = 50; +// allow less retry attempts on direct output thread. +// direct outputs can be a scarce resource in audio hardware and should +// be released as quickly as possible. +static const int8_t kMaxTrackRetriesDirect = 2; + +static const int kDumpLockRetries = 50; +static const int kDumpLockSleep = 20000; + +static const nsecs_t kWarningThrottle = seconds(5); + + +#define AUDIOFLINGER_SECURITY_ENABLED 1 + +// ---------------------------------------------------------------------------- + +static bool recordingAllowed() { +#ifndef HAVE_ANDROID_OS + return true; +#endif +#if AUDIOFLINGER_SECURITY_ENABLED + if (getpid() == IPCThreadState::self()->getCallingPid()) return true; + bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); + if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO"); + return ok; +#else + if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO"))) + LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest"); + return true; +#endif +} + +static bool settingsAllowed() { +#ifndef HAVE_ANDROID_OS + return true; +#endif +#if AUDIOFLINGER_SECURITY_ENABLED + if (getpid() == IPCThreadState::self()->getCallingPid()) return true; + bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); + if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); + return ok; +#else + if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"))) + LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest"); + return true; +#endif +} + +// ---------------------------------------------------------------------------- + +AudioFlinger::AudioFlinger() + : BnAudioFlinger(), + mAudioHardware(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), + mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0) +{ + mHardwareStatus = AUDIO_HW_IDLE; + + mAudioHardware = AudioHardwareInterface::create(); + + mHardwareStatus = AUDIO_HW_INIT; + if (mAudioHardware->initCheck() == NO_ERROR) { + // open 16-bit output stream for s/w mixer + mMode = AudioSystem::MODE_NORMAL; + setMode(mMode); + + setMasterVolume(1.0f); + setMasterMute(false); + } else { + LOGE("Couldn't even initialize the stubbed audio hardware!"); + } +#ifdef LVMX + LifeVibes::init(); + mLifeVibesClientPid = -1; +#endif +} + +AudioFlinger::~AudioFlinger() +{ + while (!mRecordThreads.isEmpty()) { + // closeInput() will remove first entry from mRecordThreads + closeInput(mRecordThreads.keyAt(0)); + } + while (!mPlaybackThreads.isEmpty()) { + // closeOutput() will remove first entry from mPlaybackThreads + closeOutput(mPlaybackThreads.keyAt(0)); + } + if (mAudioHardware) { + delete mAudioHardware; + } +} + + + +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.valueAt(i)->dump(fd, args); + } + + // dump record threads + for (size_t i = 0; i < mRecordThreads.size(); i++) { + mRecordThreads.valueAt(i)->dump(fd, args); + } + + if (mAudioHardware) { + mAudioHardware->dumpState(fd, args); + } + if (locked) mLock.unlock(); + } + return NO_ERROR; +} + + +// IAudioFlinger interface + + +sp<IAudioTrack> AudioFlinger::createTrack( + pid_t pid, + int streamType, + uint32_t sampleRate, + int format, + int channelCount, + int frameCount, + uint32_t flags, + const sp<IMemory>& sharedBuffer, + int output, + int *sessionId, + status_t *status) +{ + sp<PlaybackThread::Track> track; + sp<TrackHandle> trackHandle; + sp<Client> client; + wp<Client> wclient; + status_t lStatus; + int lSessionId; + + if (streamType >= AudioSystem::NUM_STREAM_TYPES) { + LOGE("invalid stream type"); + lStatus = BAD_VALUE; + goto Exit; + } + + { + Mutex::Autolock _l(mLock); + PlaybackThread *thread = checkPlaybackThread_l(output); + 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); + } + + // If no audio session id is provided, create one here + // TODO: enforce same stream type for all tracks in same audio session? + // TODO: prevent same audio session on different output threads + LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); + if (sessionId != NULL && *sessionId != 0) { + lSessionId = *sessionId; + } else { + lSessionId = nextUniqueId(); + if (sessionId != NULL) { + *sessionId = lSessionId; + } + } + LOGV("createTrack() lSessionId: %d", lSessionId); + + track = thread->createTrack_l(client, streamType, sampleRate, format, + channelCount, frameCount, sharedBuffer, lSessionId, &lStatus); + } + if (lStatus == NO_ERROR) { + trackHandle = new TrackHandle(track); + } else { + // remove local strong reference to Client before deleting the Track so that the Client + // destructor is called by the TrackBase destructor with mLock held + client.clear(); + track.clear(); + } + +Exit: + if(status) { + *status = lStatus; + } + return trackHandle; +} + +uint32_t AudioFlinger::sampleRate(int output) const +{ + Mutex::Autolock _l(mLock); + PlaybackThread *thread = checkPlaybackThread_l(output); + if (thread == NULL) { + LOGW("sampleRate() unknown thread %d", output); + return 0; + } + return thread->sampleRate(); +} + +int AudioFlinger::channelCount(int output) const +{ + Mutex::Autolock _l(mLock); + PlaybackThread *thread = checkPlaybackThread_l(output); + if (thread == NULL) { + LOGW("channelCount() unknown thread %d", output); + return 0; + } + return thread->channelCount(); +} + +int AudioFlinger::format(int output) const +{ + Mutex::Autolock _l(mLock); + PlaybackThread *thread = checkPlaybackThread_l(output); + if (thread == NULL) { + LOGW("format() unknown thread %d", output); + return 0; + } + return thread->format(); +} + +size_t AudioFlinger::frameCount(int output) const +{ + Mutex::Autolock _l(mLock); + PlaybackThread *thread = checkPlaybackThread_l(output); + if (thread == NULL) { + LOGW("frameCount() unknown thread %d", output); + return 0; + } + return thread->frameCount(); +} + +uint32_t AudioFlinger::latency(int output) const +{ + Mutex::Autolock _l(mLock); + PlaybackThread *thread = checkPlaybackThread_l(output); + if (thread == NULL) { + LOGW("latency() unknown thread %d", output); + return 0; + } + return thread->latency(); +} + +status_t AudioFlinger::setMasterVolume(float value) +{ + // check calling permissions + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + + // when hw supports master volume, don't scale in sw mixer + AutoMutex lock(mHardwareLock); + mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; + if (mAudioHardware->setMasterVolume(value) == NO_ERROR) { + value = 1.0f; + } + mHardwareStatus = AUDIO_HW_IDLE; + + mMasterVolume = value; + for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) + mPlaybackThreads.valueAt(i)->setMasterVolume(value); + + return NO_ERROR; +} + +status_t AudioFlinger::setMode(int mode) +{ + status_t ret; + + // check calling permissions + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) { + LOGW("Illegal value: setMode(%d)", mode); + return BAD_VALUE; + } + + { // scope for the lock + AutoMutex lock(mHardwareLock); + mHardwareStatus = AUDIO_HW_SET_MODE; + ret = mAudioHardware->setMode(mode); + mHardwareStatus = AUDIO_HW_IDLE; + } + + if (NO_ERROR == ret) { + Mutex::Autolock _l(mLock); + mMode = mode; + for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) + mPlaybackThreads.valueAt(i)->setMode(mode); +#ifdef LVMX + LifeVibes::setMode(mode); +#endif + } + + return ret; +} + +status_t AudioFlinger::setMicMute(bool state) +{ + // check calling permissions + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + + AutoMutex lock(mHardwareLock); + mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; + status_t ret = mAudioHardware->setMicMute(state); + mHardwareStatus = AUDIO_HW_IDLE; + return ret; +} + +bool AudioFlinger::getMicMute() const +{ + bool state = AudioSystem::MODE_INVALID; + mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; + mAudioHardware->getMicMute(&state); + mHardwareStatus = AUDIO_HW_IDLE; + return state; +} + +status_t AudioFlinger::setMasterMute(bool muted) +{ + // check calling permissions + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + + mMasterMute = muted; + for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) + mPlaybackThreads.valueAt(i)->setMasterMute(muted); + + return NO_ERROR; +} + +float AudioFlinger::masterVolume() const +{ + return mMasterVolume; +} + +bool AudioFlinger::masterMute() const +{ + return mMasterMute; +} + +status_t AudioFlinger::setStreamVolume(int stream, float value, int output) +{ + // check calling permissions + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + + if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { + return BAD_VALUE; + } + + AutoMutex lock(mLock); + PlaybackThread *thread = NULL; + if (output) { + thread = checkPlaybackThread_l(output); + if (thread == NULL) { + return BAD_VALUE; + } + } + + mStreamTypes[stream].volume = value; + + if (thread == NULL) { + for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { + mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); + } + } else { + thread->setStreamVolume(stream, value); + } + + return NO_ERROR; +} + +status_t AudioFlinger::setStreamMute(int stream, bool muted) +{ + // check calling permissions + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + + if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES || + uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) { + return BAD_VALUE; + } + + mStreamTypes[stream].mute = muted; + for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) + mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); + + return NO_ERROR; +} + +float AudioFlinger::streamVolume(int stream, int output) const +{ + if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { + return 0.0f; + } + + AutoMutex lock(mLock); + float volume; + if (output) { + PlaybackThread *thread = checkPlaybackThread_l(output); + if (thread == NULL) { + return 0.0f; + } + volume = thread->streamVolume(stream); + } else { + volume = mStreamTypes[stream].volume; + } + + return volume; +} + +bool AudioFlinger::streamMute(int stream) const +{ + if (stream < 0 || stream >= (int)AudioSystem::NUM_STREAM_TYPES) { + return true; + } + + return mStreamTypes[stream].mute; +} + +bool AudioFlinger::isStreamActive(int stream) const +{ + Mutex::Autolock _l(mLock); + for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { + if (mPlaybackThreads.valueAt(i)->isStreamActive(stream)) { + return true; + } + } + return false; +} + +status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) +{ + status_t result; + + LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", + ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); + // check calling permissions + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + +#ifdef LVMX + AudioParameter param = AudioParameter(keyValuePairs); + LifeVibes::setParameters(ioHandle,keyValuePairs); + String8 key = String8(AudioParameter::keyRouting); + int device; + if (NO_ERROR != param.getInt(key, device)) { + device = -1; + } + + key = String8(LifevibesTag); + String8 value; + int musicEnabled = -1; + if (NO_ERROR == param.get(key, value)) { + if (value == LifevibesEnable) { + mLifeVibesClientPid = IPCThreadState::self()->getCallingPid(); + musicEnabled = 1; + } else if (value == LifevibesDisable) { + mLifeVibesClientPid = -1; + musicEnabled = 0; + } + } +#endif + + // ioHandle == 0 means the parameters are global to the audio hardware interface + if (ioHandle == 0) { + AutoMutex lock(mHardwareLock); + mHardwareStatus = AUDIO_SET_PARAMETER; + result = mAudioHardware->setParameters(keyValuePairs); +#ifdef LVMX + if (musicEnabled != -1) { + LifeVibes::enableMusic((bool) musicEnabled); + } +#endif + mHardwareStatus = AUDIO_HW_IDLE; + return result; + } + + // hold a strong ref on thread in case closeOutput() or closeInput() is called + // and the thread is exited once the lock is released + sp<ThreadBase> thread; + { + Mutex::Autolock _l(mLock); + thread = checkPlaybackThread_l(ioHandle); + if (thread == NULL) { + thread = checkRecordThread_l(ioHandle); + } + } + if (thread != NULL) { + result = thread->setParameters(keyValuePairs); +#ifdef LVMX + if ((NO_ERROR == result) && (device != -1)) { + LifeVibes::setDevice(LifeVibes::threadIdToAudioOutputType(thread->id()), device); + } +#endif + return result; + } + return BAD_VALUE; +} + +String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) +{ +// LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", +// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); + + if (ioHandle == 0) { + return mAudioHardware->getParameters(keys); + } + + Mutex::Autolock _l(mLock); + + PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); + if (playbackThread != NULL) { + return playbackThread->getParameters(keys); + } + RecordThread *recordThread = checkRecordThread_l(ioHandle); + if (recordThread != NULL) { + return recordThread->getParameters(keys); + } + return String8(""); +} + +size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) +{ + return mAudioHardware->getInputBufferSize(sampleRate, format, channelCount); +} + +unsigned int AudioFlinger::getInputFramesLost(int ioHandle) +{ + if (ioHandle == 0) { + return 0; + } + + Mutex::Autolock _l(mLock); + + RecordThread *recordThread = checkRecordThread_l(ioHandle); + if (recordThread != NULL) { + return recordThread->getInputFramesLost(); + } + return 0; +} + +status_t AudioFlinger::setVoiceVolume(float value) +{ + // check calling permissions + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + + AutoMutex lock(mHardwareLock); + mHardwareStatus = AUDIO_SET_VOICE_VOLUME; + status_t ret = mAudioHardware->setVoiceVolume(value); + mHardwareStatus = AUDIO_HW_IDLE; + + return ret; +} + +status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) +{ + status_t status; + + Mutex::Autolock _l(mLock); + + PlaybackThread *playbackThread = checkPlaybackThread_l(output); + if (playbackThread != NULL) { + return playbackThread->getRenderPosition(halFrames, dspFrames); + } + + return BAD_VALUE; +} + +void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) +{ + + Mutex::Autolock _l(mLock); + + int pid = IPCThreadState::self()->getCallingPid(); + if (mNotificationClients.indexOfKey(pid) < 0) { + sp<NotificationClient> notificationClient = new NotificationClient(this, + client, + pid); + LOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); + + mNotificationClients.add(pid, notificationClient); + + sp<IBinder> binder = client->asBinder(); + binder->linkToDeath(notificationClient); + + // the config change is always sent from playback or record threads to avoid deadlock + // with AudioSystem::gLock + for (size_t i = 0; i < mPlaybackThreads.size(); i++) { + mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); + } + + for (size_t i = 0; i < mRecordThreads.size(); i++) { + mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); + } + } +} + +void AudioFlinger::removeNotificationClient(pid_t pid) +{ + Mutex::Autolock _l(mLock); + + int index = mNotificationClients.indexOfKey(pid); + if (index >= 0) { + sp <NotificationClient> client = mNotificationClients.valueFor(pid); + LOGV("removeNotificationClient() %p, pid %d", client.get(), pid); +#ifdef LVMX + if (pid == mLifeVibesClientPid) { + LOGV("Disabling lifevibes"); + LifeVibes::enableMusic(false); + mLifeVibesClientPid = -1; + } +#endif + mNotificationClients.removeItem(pid); + } +} + +// audioConfigChanged_l() must be called with AudioFlinger::mLock held +void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) +{ + size_t size = mNotificationClients.size(); + for (size_t i = 0; i < size; i++) { + mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); + } +} + +// removeClient_l() must be called with AudioFlinger::mLock held +void AudioFlinger::removeClient_l(pid_t pid) +{ + LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); + mClients.removeItem(pid); +} + + +// ---------------------------------------------------------------------------- + +AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id) + : Thread(false), + mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), + mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false) +{ +} + +AudioFlinger::ThreadBase::~ThreadBase() +{ + mParamCond.broadcast(); + mNewParameters.clear(); +} + +void AudioFlinger::ThreadBase::exit() +{ + // keep a strong ref on ourself so that we wont get + // destroyed in the middle of requestExitAndWait() + sp <ThreadBase> strongMe = this; + + LOGV("ThreadBase::exit"); + { + AutoMutex lock(&mLock); + mExiting = true; + requestExit(); + mWaitWorkCV.signal(); + } + requestExitAndWait(); +} + +uint32_t AudioFlinger::ThreadBase::sampleRate() const +{ + return mSampleRate; +} + +int AudioFlinger::ThreadBase::channelCount() const +{ + return (int)mChannelCount; +} + +int AudioFlinger::ThreadBase::format() const +{ + return mFormat; +} + +size_t AudioFlinger::ThreadBase::frameCount() const +{ + return mFrameCount; +} + +status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) +{ + status_t status; + + LOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); + Mutex::Autolock _l(mLock); + + mNewParameters.add(keyValuePairs); + mWaitWorkCV.signal(); + // wait condition with timeout in case the thread loop has exited + // before the request could be processed + if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) { + status = mParamStatus; + mWaitWorkCV.signal(); + } else { + status = TIMED_OUT; + } + return status; +} + +void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) +{ + Mutex::Autolock _l(mLock); + sendConfigEvent_l(event, param); +} + +// sendConfigEvent_l() must be called with ThreadBase::mLock held +void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) +{ + ConfigEvent *configEvent = new ConfigEvent(); + configEvent->mEvent = event; + configEvent->mParam = param; + mConfigEvents.add(configEvent); + LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); + mWaitWorkCV.signal(); +} + +void AudioFlinger::ThreadBase::processConfigEvents() +{ + mLock.lock(); + while(!mConfigEvents.isEmpty()) { + LOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); + ConfigEvent *configEvent = mConfigEvents[0]; + mConfigEvents.removeAt(0); + // release mLock before locking AudioFlinger mLock: lock order is always + // AudioFlinger then ThreadBase to avoid cross deadlock + mLock.unlock(); + mAudioFlinger->mLock.lock(); + audioConfigChanged_l(configEvent->mEvent, configEvent->mParam); + mAudioFlinger->mLock.unlock(); + delete configEvent; + mLock.lock(); + } + mLock.unlock(); +} + +status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + bool locked = tryLock(mLock); + if (!locked) { + snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); + write(fd, buffer, strlen(buffer)); + } + + snprintf(buffer, SIZE, "standby: %d\n", mStandby); + result.append(buffer); + snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); + result.append(buffer); + snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); + result.append(buffer); + snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); + result.append(buffer); + snprintf(buffer, SIZE, "Format: %d\n", mFormat); + result.append(buffer); + snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); + result.append(buffer); + + snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); + result.append(buffer); + result.append(" Index Command"); + for (size_t i = 0; i < mNewParameters.size(); ++i) { + snprintf(buffer, SIZE, "\n %02d ", i); + result.append(buffer); + result.append(mNewParameters[i]); + } + + snprintf(buffer, SIZE, "\n\nPending config events: \n"); + result.append(buffer); + snprintf(buffer, SIZE, " Index event param\n"); + result.append(buffer); + for (size_t i = 0; i < mConfigEvents.size(); i++) { + snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam); + result.append(buffer); + } + result.append("\n"); + + write(fd, result.string(), result.size()); + + if (locked) { + mLock.unlock(); + } + return NO_ERROR; +} + + +// ---------------------------------------------------------------------------- + +AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) + : ThreadBase(audioFlinger, id), + mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output), + mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), + mDevice(device) +{ + readOutputParameters(); + + mMasterVolume = mAudioFlinger->masterVolume(); + mMasterMute = mAudioFlinger->masterMute(); + + for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { + mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); + mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); + } +} + +AudioFlinger::PlaybackThread::~PlaybackThread() +{ + delete [] mMixBuffer; +} + +status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) +{ + dumpInternals(fd, args); + dumpTracks(fd, args); + dumpEffectChains(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 Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); + for (size_t i = 0; i < mTracks.size(); ++i) { + sp<Track> track = mTracks[i]; + if (track != 0) { + track->dump(buffer, SIZE); + result.append(buffer); + } + } + + snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); + result.append(buffer); + result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); + for (size_t i = 0; i < mActiveTracks.size(); ++i) { + wp<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::dumpEffectChains(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); + write(fd, buffer, strlen(buffer)); + + for (size_t i = 0; i < mEffectChains.size(); ++i) { + sp<EffectChain> chain = mEffectChains[i]; + if (chain != 0) { + chain->dump(fd, args); + } + } + 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, "\nOutput thread %p internals\n", this); + result.append(buffer); + snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); + result.append(buffer); + snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); + result.append(buffer); + snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); + result.append(buffer); + snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); + result.append(buffer); + snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); + result.append(buffer); + snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); + result.append(buffer); + write(fd, result.string(), result.size()); + + dumpBase(fd, args); + + return NO_ERROR; +} + +// Thread virtuals +status_t AudioFlinger::PlaybackThread::readyToRun() +{ + if (mSampleRate == 0) { + LOGE("No working audio driver found."); + return NO_INIT; + } + LOGI("AudioFlinger's thread %p ready to run", this); + return NO_ERROR; +} + +void AudioFlinger::PlaybackThread::onFirstRef() +{ + const size_t SIZE = 256; + char buffer[SIZE]; + + snprintf(buffer, SIZE, "Playback Thread %p", this); + + run(buffer, ANDROID_PRIORITY_URGENT_AUDIO); +} + +// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held +sp<AudioFlinger::PlaybackThread::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, + int sessionId, + status_t *status) +{ + sp<Track> track; + status_t lStatus; + + if (mType == DIRECT) { + if (sampleRate != mSampleRate || format != mFormat || channelCount != (int)mChannelCount) { + LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p", + sampleRate, format, channelCount, mOutput); + lStatus = BAD_VALUE; + goto Exit; + } + } else { + // Resampler implementation limits input sampling rate to 2 x output sampling rate. + if (sampleRate > mSampleRate*2) { + LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); + lStatus = BAD_VALUE; + goto Exit; + } + } + + if (mOutput == 0) { + LOGE("Audio driver not initialized."); + lStatus = NO_INIT; + goto Exit; + } + + { // scope for mLock + Mutex::Autolock _l(mLock); + track = new Track(this, client, streamType, sampleRate, format, + channelCount, frameCount, sharedBuffer, sessionId); + if (track->getCblk() == NULL || track->name() < 0) { + lStatus = NO_MEMORY; + goto Exit; + } + mTracks.add(track); + + sp<EffectChain> chain = getEffectChain_l(sessionId); + if (chain != 0) { + LOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); + track->setMainBuffer(chain->inBuffer()); + } + } + lStatus = NO_ERROR; + +Exit: + if(status) { + *status = lStatus; + } + return track; +} + +uint32_t AudioFlinger::PlaybackThread::latency() const +{ + if (mOutput) { + return mOutput->latency(); + } + else { + return 0; + } +} + +status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) +{ +#ifdef LVMX + int audioOutputType = LifeVibes::getMixerType(mId, mType); + if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { + LifeVibes::setMasterVolume(audioOutputType, value); + } +#endif + mMasterVolume = value; + return NO_ERROR; +} + +status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) +{ +#ifdef LVMX + int audioOutputType = LifeVibes::getMixerType(mId, mType); + if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { + LifeVibes::setMasterMute(audioOutputType, muted); + } +#endif + mMasterMute = muted; + return NO_ERROR; +} + +float AudioFlinger::PlaybackThread::masterVolume() const +{ + return mMasterVolume; +} + +bool AudioFlinger::PlaybackThread::masterMute() const +{ + return mMasterMute; +} + +status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) +{ +#ifdef LVMX + int audioOutputType = LifeVibes::getMixerType(mId, mType); + if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { + LifeVibes::setStreamVolume(audioOutputType, stream, value); + } +#endif + mStreamTypes[stream].volume = value; + return NO_ERROR; +} + +status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) +{ +#ifdef LVMX + int audioOutputType = LifeVibes::getMixerType(mId, mType); + if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { + LifeVibes::setStreamMute(audioOutputType, stream, muted); + } +#endif + mStreamTypes[stream].mute = muted; + return NO_ERROR; +} + +float AudioFlinger::PlaybackThread::streamVolume(int stream) const +{ + return mStreamTypes[stream].volume; +} + +bool AudioFlinger::PlaybackThread::streamMute(int stream) const +{ + return mStreamTypes[stream].mute; +} + +bool AudioFlinger::PlaybackThread::isStreamActive(int stream) const +{ + Mutex::Autolock _l(mLock); + size_t count = mActiveTracks.size(); + for (size_t i = 0 ; i < count ; ++i) { + sp<Track> t = mActiveTracks[i].promote(); + if (t == 0) continue; + Track* const track = t.get(); + if (t->type() == stream) + return true; + } + return false; +} + +// addTrack_l() must be called with ThreadBase::mLock held +status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) +{ + status_t status = ALREADY_EXISTS; + + // set retry count for buffer fill + track->mRetryCount = kMaxTrackStartupRetries; + if (mActiveTracks.indexOf(track) < 0) { + // the track is newly added, make sure it fills up all its + // buffers before playing. This is to ensure the client will + // effectively get the latency it requested. + track->mFillingUpStatus = Track::FS_FILLING; + track->mResetDone = false; + mActiveTracks.add(track); + if (track->mainBuffer() != mMixBuffer) { + sp<EffectChain> chain = getEffectChain_l(track->sessionId()); + if (chain != 0) { + LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); + chain->startTrack(); + } + } + + status = NO_ERROR; + } + + LOGV("mWaitWorkCV.broadcast"); + mWaitWorkCV.broadcast(); + + return status; +} + +// destroyTrack_l() must be called with ThreadBase::mLock held +void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) +{ + track->mState = TrackBase::TERMINATED; + if (mActiveTracks.indexOf(track) < 0) { + mTracks.remove(track); + deleteTrackName_l(track->name()); + } +} + +String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) +{ + return mOutput->getParameters(keys); +} + +// destroyTrack_l() must be called with AudioFlinger::mLock held +void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { + AudioSystem::OutputDescriptor desc; + void *param2 = 0; + + LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); + + switch (event) { + case AudioSystem::OUTPUT_OPENED: + case AudioSystem::OUTPUT_CONFIG_CHANGED: + desc.channels = mChannels; + desc.samplingRate = mSampleRate; + desc.format = mFormat; + desc.frameCount = mFrameCount; + desc.latency = latency(); + param2 = &desc; + break; + + case AudioSystem::STREAM_CONFIG_CHANGED: + param2 = ¶m; + case AudioSystem::OUTPUT_CLOSED: + default: + break; + } + mAudioFlinger->audioConfigChanged_l(event, mId, param2); +} + +void AudioFlinger::PlaybackThread::readOutputParameters() +{ + mSampleRate = mOutput->sampleRate(); + mChannels = mOutput->channels(); + mChannelCount = (uint16_t)AudioSystem::popCount(mChannels); + mFormat = mOutput->format(); + mFrameSize = (uint16_t)mOutput->frameSize(); + mFrameCount = mOutput->bufferSize() / mFrameSize; + + // FIXME - Current mixer implementation only supports stereo output: Always + // Allocate a stereo buffer even if HW output is mono. + if (mMixBuffer != NULL) delete[] mMixBuffer; + mMixBuffer = new int16_t[mFrameCount * 2]; + memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); + + //TODO handle effects reconfig +} + +status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) +{ + if (halFrames == 0 || dspFrames == 0) { + return BAD_VALUE; + } + if (mOutput == 0) { + return INVALID_OPERATION; + } + *halFrames = mBytesWritten/mOutput->frameSize(); + + return mOutput->getRenderPosition(dspFrames); +} + +bool AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) +{ + Mutex::Autolock _l(mLock); + if (getEffectChain_l(sessionId) != 0) { + return true; + } + + for (size_t i = 0; i < mTracks.size(); ++i) { + sp<Track> track = mTracks[i]; + if (sessionId == track->sessionId()) { + return true; + } + } + + return false; +} + +sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain(int sessionId) +{ + Mutex::Autolock _l(mLock); + return getEffectChain_l(sessionId); +} + +sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain_l(int sessionId) +{ + sp<EffectChain> chain; + + size_t size = mEffectChains.size(); + for (size_t i = 0; i < size; i++) { + if (mEffectChains[i]->sessionId() == sessionId) { + chain = mEffectChains[i]; + break; + } + } + return chain; +} + +void AudioFlinger::PlaybackThread::setMode(uint32_t mode) +{ + Mutex::Autolock _l(mLock); + size_t size = mEffectChains.size(); + for (size_t i = 0; i < size; i++) { + mEffectChains[i]->setMode(mode); + } +} + +// ---------------------------------------------------------------------------- + +AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) + : PlaybackThread(audioFlinger, output, id, device), + mAudioMixer(0) +{ + mType = PlaybackThread::MIXER; + mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); + + // FIXME - Current mixer implementation only supports stereo output + if (mChannelCount == 1) { + LOGE("Invalid audio hardware channel count"); + } +} + +AudioFlinger::MixerThread::~MixerThread() +{ + delete mAudioMixer; +} + +bool AudioFlinger::MixerThread::threadLoop() +{ + Vector< sp<Track> > tracksToRemove; + uint32_t mixerStatus = MIXER_IDLE; + nsecs_t standbyTime = systemTime(); + size_t mixBufferSize = mFrameCount * mFrameSize; + // FIXME: Relaxed timing because of a certain device that can't meet latency + // Should be reduced to 2x after the vendor fixes the driver issue + nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3; + nsecs_t lastWarning = 0; + bool longStandbyExit = false; + uint32_t activeSleepTime = activeSleepTimeUs(); + uint32_t idleSleepTime = idleSleepTimeUs(); + uint32_t sleepTime = idleSleepTime; + Vector< sp<EffectChain> > effectChains; + + while (!exitPending()) + { + processConfigEvents(); + + mixerStatus = MIXER_IDLE; + { // scope for mLock + + Mutex::Autolock _l(mLock); + + if (checkForNewParameters_l()) { + mixBufferSize = mFrameCount * mFrameSize; + // FIXME: Relaxed timing because of a certain device that can't meet latency + // Should be reduced to 2x after the vendor fixes the driver issue + maxPeriod = seconds(mFrameCount) / mSampleRate * 3; + activeSleepTime = activeSleepTimeUs(); + idleSleepTime = idleSleepTimeUs(); + } + + const SortedVector< wp<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; + sleepTime = idleSleepTime; + continue; + } + } + + mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); + + // prevent any changes in effect chain list and in each effect chain + // during mixing and effect process as the audio buffers could be deleted + // or modified if an effect is created or deleted + effectChains = mEffectChains; + lockEffectChains_l(); + } + + if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { + // mix buffers... + mAudioMixer->process(); + sleepTime = 0; + standbyTime = systemTime() + kStandbyTimeInNsecs; + //TODO: delay standby when effects have a tail + } else { + // If no tracks are ready, sleep once for the duration of an output + // buffer size, then write 0s to the output + if (sleepTime == 0) { + if (mixerStatus == MIXER_TRACKS_ENABLED) { + sleepTime = activeSleepTime; + } else { + sleepTime = idleSleepTime; + } + } else if (mBytesWritten != 0 || + (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { + memset (mMixBuffer, 0, mixBufferSize); + sleepTime = 0; + LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); + } + // TODO add standby time extension fct of effect tail + } + + if (mSuspended) { + sleepTime = idleSleepTime; + } + // sleepTime == 0 means we must write to audio hardware + if (sleepTime == 0) { + for (size_t i = 0; i < effectChains.size(); i ++) { + effectChains[i]->process_l(); + } + // enable changes in effect chain + unlockEffectChains(); +#ifdef LVMX + int audioOutputType = LifeVibes::getMixerType(mId, mType); + if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { + LifeVibes::process(audioOutputType, mMixBuffer, mixBufferSize); + } +#endif + mLastWriteTime = systemTime(); + mInWrite = true; + mBytesWritten += mixBufferSize; + + int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); + if (bytesWritten < 0) mBytesWritten -= mixBufferSize; + mNumWrites++; + mInWrite = false; + nsecs_t now = systemTime(); + nsecs_t delta = now - mLastWriteTime; + if (delta > maxPeriod) { + mNumDelayedWrites++; + if ((now - lastWarning) > kWarningThrottle) { + LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", + ns2ms(delta), mNumDelayedWrites, this); + lastWarning = now; + } + if (mStandby) { + longStandbyExit = true; + } + } + mStandby = false; + } else { + // enable changes in effect chain + unlockEffectChains(); + usleep(sleepTime); + } + + // finally let go of all our tracks, without the lock held + // since we can't guarantee the destructors won't acquire that + // same lock. + tracksToRemove.clear(); + + // Effect chains will be actually deleted here if they were removed from + // mEffectChains list during mixing or effects processing + effectChains.clear(); + } + + if (!mStandby) { + mOutput->standby(); + } + + LOGV("MixerThread %p exiting", this); + return false; +} + +// prepareTracks_l() must be called with ThreadBase::mLock held +uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) +{ + + uint32_t mixerStatus = MIXER_IDLE; + // find out which tracks need to be processed + size_t count = activeTracks.size(); + size_t mixedTracks = 0; + size_t tracksWithEffect = 0; + + float masterVolume = mMasterVolume; + bool masterMute = mMasterMute; + +#ifdef LVMX + bool tracksConnectedChanged = false; + bool stateChanged = false; + + int audioOutputType = LifeVibes::getMixerType(mId, mType); + if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) + { + int activeTypes = 0; + for (size_t i=0 ; i<count ; i++) { + sp<Track> t = activeTracks[i].promote(); + if (t == 0) continue; + Track* const track = t.get(); + int iTracktype=track->type(); + activeTypes |= 1<<track->type(); + } + LifeVibes::computeVolumes(audioOutputType, activeTypes, tracksConnectedChanged, stateChanged, masterVolume, masterMute); + } +#endif + // Delegate master volume control to effect in output mix effect chain if needed + sp<EffectChain> chain = getEffectChain_l(0); + if (chain != 0) { + uint32_t v = (uint32_t)(masterVolume * (1 << 24)); + chain->setVolume(&v, &v); + masterVolume = (float)((v + (1 << 23)) >> 24); + chain.clear(); + } + + 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() && !track->isTerminated()) + { + //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); + + mixedTracks++; + + // track->mainBuffer() != mMixBuffer means there is an effect chain + // connected to the track + chain.clear(); + if (track->mainBuffer() != mMixBuffer) { + chain = getEffectChain_l(track->sessionId()); + // Delegate volume control to effect in track effect chain if needed + if (chain != 0) { + tracksWithEffect++; + } else { + LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", + track->name(), track->sessionId()); + } + } + + + int param = AudioMixer::VOLUME; + if (track->mFillingUpStatus == Track::FS_FILLED) { + // no ramp for the first volume setting + track->mFillingUpStatus = Track::FS_ACTIVE; + if (track->mState == TrackBase::RESUMING) { + track->mState = TrackBase::ACTIVE; + param = AudioMixer::RAMP_VOLUME; + } + } else if (cblk->server != 0) { + // If the track is stopped before the first frame was mixed, + // do not apply ramp + param = AudioMixer::RAMP_VOLUME; + } + + // compute volume for this track + int16_t left, right, aux; + if (track->isMuted() || masterMute || track->isPausing() || + mStreamTypes[track->type()].mute) { + left = right = aux = 0; + if (track->isPausing()) { + track->setPaused(); + } + } else { + // read original volumes with volume control + float typeVolume = mStreamTypes[track->type()].volume; +#ifdef LVMX + bool streamMute=false; + // read the volume from the LivesVibes audio engine. + if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) + { + LifeVibes::getStreamVolumes(audioOutputType, track->type(), &typeVolume, &streamMute); + if (streamMute) { + typeVolume = 0; + } + } +#endif + float v = masterVolume * typeVolume; + uint32_t vl = (uint32_t)(v * cblk->volume[0]) << 12; + uint32_t vr = (uint32_t)(v * cblk->volume[1]) << 12; + + // Delegate volume control to effect in track effect chain if needed + if (chain != 0 && chain->setVolume(&vl, &vr)) { + // Do not ramp volume is volume is controlled by effect + param = AudioMixer::VOLUME; + } + + // Convert volumes from 8.24 to 4.12 format + uint32_t v_clamped = (vl + (1 << 11)) >> 12; + if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; + left = int16_t(v_clamped); + v_clamped = (vr + (1 << 11)) >> 12; + if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; + right = int16_t(v_clamped); + + v_clamped = (uint32_t)(v * cblk->sendLevel); + if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; + aux = int16_t(v_clamped); + } + +#ifdef LVMX + if ( tracksConnectedChanged || stateChanged ) + { + // only do the ramp when the volume is changed by the user / application + param = AudioMixer::VOLUME; + } +#endif + + // XXX: these things DON'T need to be done each time + mAudioMixer->setBufferProvider(track); + mAudioMixer->enable(AudioMixer::MIXING); + + mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); + mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); + mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); + mAudioMixer->setParameter( + AudioMixer::TRACK, + AudioMixer::FORMAT, (void *)track->format()); + mAudioMixer->setParameter( + AudioMixer::TRACK, + AudioMixer::CHANNEL_COUNT, (void *)track->channelCount()); + mAudioMixer->setParameter( + AudioMixer::RESAMPLE, + AudioMixer::SAMPLE_RATE, + (void *)(cblk->sampleRate)); + mAudioMixer->setParameter( + AudioMixer::TRACK, + AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); + mAudioMixer->setParameter( + AudioMixer::TRACK, + AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); + + // reset retry count + track->mRetryCount = kMaxTrackRetries; + mixerStatus = MIXER_TRACKS_READY; + } else { + //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); + if (track->isStopped()) { + track->reset(); + } + if (track->isTerminated() || track->isStopped() || track->isPaused()) { + // We have consumed all the buffers of this track. + // Remove it from the list of active tracks. + tracksToRemove->add(track); + } else { + // No buffers for this track. Give it a few chances to + // fill a buffer, then remove it from active list. + if (--(track->mRetryCount) <= 0) { + LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); + tracksToRemove->add(track); + } else if (mixerStatus != MIXER_TRACKS_READY) { + mixerStatus = MIXER_TRACKS_ENABLED; + } + } + mAudioMixer->disable(AudioMixer::MIXING); + } + } + + // remove all the tracks that need to be... + count = tracksToRemove->size(); + if (UNLIKELY(count)) { + for (size_t i=0 ; i<count ; i++) { + const sp<Track>& track = tracksToRemove->itemAt(i); + mActiveTracks.remove(track); + if (track->mainBuffer() != mMixBuffer) { + chain = getEffectChain_l(track->sessionId()); + if (chain != 0) { + LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); + chain->stopTrack(); + } + } + if (track->isTerminated()) { + mTracks.remove(track); + deleteTrackName_l(track->mName); + } + } + } + + // mix buffer must be cleared if all tracks are connected to an + // effect chain as in this case the mixer will not write to + // mix buffer and track effects will accumulate into it + if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { + memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); + } + + return mixerStatus; +} + +void AudioFlinger::MixerThread::invalidateTracks(int streamType) +{ + LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", this, streamType, mTracks.size()); + Mutex::Autolock _l(mLock); + size_t size = mTracks.size(); + for (size_t i = 0; i < size; i++) { + sp<Track> t = mTracks[i]; + if (t->type() == streamType) { + t->mCblk->lock.lock(); + t->mCblk->flags |= CBLK_INVALID_ON; + t->mCblk->cv.signal(); + t->mCblk->lock.unlock(); + } + } +} + + +// getTrackName_l() must be called with ThreadBase::mLock held +int AudioFlinger::MixerThread::getTrackName_l() +{ + return mAudioMixer->getTrackName(); +} + +// deleteTrackName_l() must be called with ThreadBase::mLock held +void AudioFlinger::MixerThread::deleteTrackName_l(int name) +{ + LOGV("remove track (%d) and delete from mixer", name); + mAudioMixer->deleteTrackName(name); +} + +// checkForNewParameters_l() must be called with ThreadBase::mLock held +bool AudioFlinger::MixerThread::checkForNewParameters_l() +{ + bool reconfig = false; + + while (!mNewParameters.isEmpty()) { + status_t status = NO_ERROR; + String8 keyValuePair = mNewParameters[0]; + AudioParameter param = AudioParameter(keyValuePair); + int value; + + if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { + reconfig = true; + } + if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { + if (value != AudioSystem::PCM_16_BIT) { + status = BAD_VALUE; + } else { + reconfig = true; + } + } + if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { + if (value != AudioSystem::CHANNEL_OUT_STEREO) { + status = BAD_VALUE; + } else { + reconfig = true; + } + } + if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { + // do not accept frame count changes if tracks are open as the track buffer + // size depends on frame count and correct behavior would not be garantied + // if frame count is changed after track creation + if (!mTracks.isEmpty()) { + status = INVALID_OPERATION; + } else { + reconfig = true; + } + } + if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { + // forward device change to effects that have requested to be + // aware of attached audio device. + mDevice = (uint32_t)value; + for (size_t i = 0; i < mEffectChains.size(); i++) { + mEffectChains[i]->setDevice(mDevice); + } + } + + if (status == NO_ERROR) { + status = mOutput->setParameters(keyValuePair); + if (!mStandby && status == INVALID_OPERATION) { + mOutput->standby(); + mStandby = true; + mBytesWritten = 0; + status = mOutput->setParameters(keyValuePair); + } + if (status == NO_ERROR && reconfig) { + delete mAudioMixer; + readOutputParameters(); + mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); + for (size_t i = 0; i < mTracks.size() ; i++) { + int name = getTrackName_l(); + if (name < 0) break; + mTracks[i]->mName = name; + // limit track sample rate to 2 x new output sample rate + if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { + mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); + } + } + sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); + } + } + + mNewParameters.removeAt(0); + + mParamStatus = status; + mParamCond.signal(); + mWaitWorkCV.wait(mLock); + } + return reconfig; +} + +status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<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; +} + +uint32_t AudioFlinger::MixerThread::activeSleepTimeUs() +{ + return (uint32_t)(mOutput->latency() * 1000) / 2; +} + +uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() +{ + return (uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000; +} + +// ---------------------------------------------------------------------------- +AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) + : PlaybackThread(audioFlinger, output, id, device) +{ + mType = PlaybackThread::DIRECT; +} + +AudioFlinger::DirectOutputThread::~DirectOutputThread() +{ +} + + +static inline int16_t clamp16(int32_t sample) +{ + if ((sample>>15) ^ (sample>>31)) + sample = 0x7FFF ^ (sample>>31); + return sample; +} + +static inline +int32_t mul(int16_t in, int16_t v) +{ +#if defined(__arm__) && !defined(__thumb__) + int32_t out; + asm( "smulbb %[out], %[in], %[v] \n" + : [out]"=r"(out) + : [in]"%r"(in), [v]"r"(v) + : ); + return out; +#else + return in * int32_t(v); +#endif +} + +void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) +{ + // Do not apply volume on compressed audio + if (!AudioSystem::isLinearPCM(mFormat)) { + return; + } + + // convert to signed 16 bit before volume calculation + if (mFormat == AudioSystem::PCM_8_BIT) { + size_t count = mFrameCount * mChannelCount; + uint8_t *src = (uint8_t *)mMixBuffer + count-1; + int16_t *dst = mMixBuffer + count-1; + while(count--) { + *dst-- = (int16_t)(*src--^0x80) << 8; + } + } + + size_t frameCount = mFrameCount; + int16_t *out = mMixBuffer; + if (ramp) { + if (mChannelCount == 1) { + int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; + int32_t vlInc = d / (int32_t)frameCount; + int32_t vl = ((int32_t)mLeftVolShort << 16); + do { + out[0] = clamp16(mul(out[0], vl >> 16) >> 12); + out++; + vl += vlInc; + } while (--frameCount); + + } else { + int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; + int32_t vlInc = d / (int32_t)frameCount; + d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; + int32_t vrInc = d / (int32_t)frameCount; + int32_t vl = ((int32_t)mLeftVolShort << 16); + int32_t vr = ((int32_t)mRightVolShort << 16); + do { + out[0] = clamp16(mul(out[0], vl >> 16) >> 12); + out[1] = clamp16(mul(out[1], vr >> 16) >> 12); + out += 2; + vl += vlInc; + vr += vrInc; + } while (--frameCount); + } + } else { + if (mChannelCount == 1) { + do { + out[0] = clamp16(mul(out[0], leftVol) >> 12); + out++; + } while (--frameCount); + } else { + do { + out[0] = clamp16(mul(out[0], leftVol) >> 12); + out[1] = clamp16(mul(out[1], rightVol) >> 12); + out += 2; + } while (--frameCount); + } + } + + // convert back to unsigned 8 bit after volume calculation + if (mFormat == AudioSystem::PCM_8_BIT) { + size_t count = mFrameCount * mChannelCount; + int16_t *src = mMixBuffer; + uint8_t *dst = (uint8_t *)mMixBuffer; + while(count--) { + *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; + } + } + + mLeftVolShort = leftVol; + mRightVolShort = rightVol; +} + +bool AudioFlinger::DirectOutputThread::threadLoop() +{ + uint32_t mixerStatus = MIXER_IDLE; + sp<Track> trackToRemove; + sp<Track> activeTrack; + nsecs_t standbyTime = systemTime(); + int8_t *curBuf; + size_t mixBufferSize = mFrameCount*mFrameSize; + uint32_t activeSleepTime = activeSleepTimeUs(); + uint32_t idleSleepTime = idleSleepTimeUs(); + uint32_t sleepTime = idleSleepTime; + // use shorter standby delay as on normal output to release + // hardware resources as soon as possible + nsecs_t standbyDelay = microseconds(activeSleepTime*2); + + + while (!exitPending()) + { + bool rampVolume; + uint16_t leftVol; + uint16_t rightVol; + Vector< sp<EffectChain> > effectChains; + + processConfigEvents(); + + mixerStatus = MIXER_IDLE; + + { // scope for the mLock + + Mutex::Autolock _l(mLock); + + if (checkForNewParameters_l()) { + mixBufferSize = mFrameCount*mFrameSize; + activeSleepTime = activeSleepTimeUs(); + idleSleepTime = idleSleepTimeUs(); + standbyDelay = microseconds(activeSleepTime*2); + } + + // put audio hardware into standby after short delay + if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || + mSuspended) { + // wait until we have something to do... + if (!mStandby) { + LOGV("Audio hardware entering standby, mixer %p\n", this); + mOutput->standby(); + mStandby = true; + mBytesWritten = 0; + } + + if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { + // we're about to wait, flush the binder command buffer + IPCThreadState::self()->flushCommands(); + + if (exitPending()) break; + + LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); + mWaitWorkCV.wait(mLock); + LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); + + if (mMasterMute == false) { + char value[PROPERTY_VALUE_MAX]; + property_get("ro.audio.silent", value, "0"); + if (atoi(value)) { + LOGD("Silence is golden"); + setMasterMute(true); + } + } + + standbyTime = systemTime() + standbyDelay; + sleepTime = idleSleepTime; + continue; + } + } + + effectChains = mEffectChains; + + // find out which tracks need to be processed + if (mActiveTracks.size() != 0) { + sp<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() && !track->isTerminated()) + { + //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); + + if (track->mFillingUpStatus == Track::FS_FILLED) { + track->mFillingUpStatus = Track::FS_ACTIVE; + mLeftVolFloat = mRightVolFloat = 0; + mLeftVolShort = mRightVolShort = 0; + if (track->mState == TrackBase::RESUMING) { + track->mState = TrackBase::ACTIVE; + rampVolume = true; + } + } else if (cblk->server != 0) { + // If the track is stopped before the first frame was mixed, + // do not apply ramp + rampVolume = true; + } + // compute volume for this track + float left, right; + if (track->isMuted() || mMasterMute || track->isPausing() || + mStreamTypes[track->type()].mute) { + left = right = 0; + if (track->isPausing()) { + track->setPaused(); + } + } else { + float typeVolume = mStreamTypes[track->type()].volume; + float v = mMasterVolume * typeVolume; + float v_clamped = v * cblk->volume[0]; + if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; + left = v_clamped/MAX_GAIN; + v_clamped = v * cblk->volume[1]; + if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; + right = v_clamped/MAX_GAIN; + } + + if (left != mLeftVolFloat || right != mRightVolFloat) { + mLeftVolFloat = left; + mRightVolFloat = right; + + // If audio HAL implements volume control, + // force software volume to nominal value + if (mOutput->setVolume(left, right) == NO_ERROR) { + left = 1.0f; + right = 1.0f; + } + + // Convert volumes from float to 8.24 + uint32_t vl = (uint32_t)(left * (1 << 24)); + uint32_t vr = (uint32_t)(right * (1 << 24)); + + // Delegate volume control to effect in track effect chain if needed + // only one effect chain can be present on DirectOutputThread, so if + // there is one, the track is connected to it + if (!effectChains.isEmpty()) { + // Do not ramp volume is volume is controlled by effect + if(effectChains[0]->setVolume(&vl, &vr)) { + rampVolume = false; + } + } + + // Convert volumes from 8.24 to 4.12 format + uint32_t v_clamped = (vl + (1 << 11)) >> 12; + if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; + leftVol = (uint16_t)v_clamped; + v_clamped = (vr + (1 << 11)) >> 12; + if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; + rightVol = (uint16_t)v_clamped; + } else { + leftVol = mLeftVolShort; + rightVol = mRightVolShort; + rampVolume = false; + } + + // reset retry count + track->mRetryCount = kMaxTrackRetriesDirect; + activeTrack = t; + mixerStatus = MIXER_TRACKS_READY; + } else { + //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); + if (track->isStopped()) { + track->reset(); + } + if (track->isTerminated() || track->isStopped() || track->isPaused()) { + // We have consumed all the buffers of this track. + // Remove it from the list of active tracks. + trackToRemove = track; + } else { + // No buffers for this track. Give it a few chances to + // fill a buffer, then remove it from active list. + if (--(track->mRetryCount) <= 0) { + LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); + trackToRemove = track; + } else { + mixerStatus = MIXER_TRACKS_ENABLED; + } + } + } + } + + // remove all the tracks that need to be... + if (UNLIKELY(trackToRemove != 0)) { + mActiveTracks.remove(trackToRemove); + if (!effectChains.isEmpty()) { + LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), trackToRemove->sessionId()); + effectChains[0]->stopTrack(); + } + if (trackToRemove->isTerminated()) { + mTracks.remove(trackToRemove); + deleteTrackName_l(trackToRemove->mName); + } + } + + lockEffectChains_l(); + } + + if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { + AudioBufferProvider::Buffer buffer; + size_t frameCount = mFrameCount; + curBuf = (int8_t *)mMixBuffer; + // output audio to hardware + while (frameCount) { + buffer.frameCount = frameCount; + activeTrack->getNextBuffer(&buffer); + if (UNLIKELY(buffer.raw == 0)) { + memset(curBuf, 0, frameCount * mFrameSize); + break; + } + memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); + frameCount -= buffer.frameCount; + curBuf += buffer.frameCount * mFrameSize; + activeTrack->releaseBuffer(&buffer); + } + sleepTime = 0; + standbyTime = systemTime() + standbyDelay; + } else { + if (sleepTime == 0) { + if (mixerStatus == MIXER_TRACKS_ENABLED) { + sleepTime = activeSleepTime; + } else { + sleepTime = idleSleepTime; + } + } else if (mBytesWritten != 0 && AudioSystem::isLinearPCM(mFormat)) { + memset (mMixBuffer, 0, mFrameCount * mFrameSize); + sleepTime = 0; + } + } + + if (mSuspended) { + sleepTime = idleSleepTime; + } + // sleepTime == 0 means we must write to audio hardware + if (sleepTime == 0) { + if (mixerStatus == MIXER_TRACKS_READY) { + applyVolume(leftVol, rightVol, rampVolume); + } + for (size_t i = 0; i < effectChains.size(); i ++) { + effectChains[i]->process_l(); + } + unlockEffectChains(); + + mLastWriteTime = systemTime(); + mInWrite = true; + mBytesWritten += mixBufferSize; + int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); + if (bytesWritten < 0) mBytesWritten -= mixBufferSize; + mNumWrites++; + mInWrite = false; + mStandby = false; + } else { + unlockEffectChains(); + usleep(sleepTime); + } + + // finally let go of removed track, without the lock held + // since we can't guarantee the destructors won't acquire that + // same lock. + trackToRemove.clear(); + activeTrack.clear(); + + // Effect chains will be actually deleted here if they were removed from + // mEffectChains list during mixing or effects processing + effectChains.clear(); + } + + if (!mStandby) { + mOutput->standby(); + } + + LOGV("DirectOutputThread %p exiting", this); + return false; +} + +// getTrackName_l() must be called with ThreadBase::mLock held +int AudioFlinger::DirectOutputThread::getTrackName_l() +{ + return 0; +} + +// deleteTrackName_l() must be called with ThreadBase::mLock held +void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) +{ +} + +// checkForNewParameters_l() must be called with ThreadBase::mLock held +bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() +{ + bool reconfig = false; + + while (!mNewParameters.isEmpty()) { + status_t status = NO_ERROR; + String8 keyValuePair = mNewParameters[0]; + AudioParameter param = AudioParameter(keyValuePair); + int value; + + if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { + // do not accept frame count changes if tracks are open as the track buffer + // size depends on frame count and correct behavior would not be garantied + // if frame count is changed after track creation + if (!mTracks.isEmpty()) { + status = INVALID_OPERATION; + } else { + reconfig = true; + } + } + if (status == NO_ERROR) { + status = mOutput->setParameters(keyValuePair); + if (!mStandby && status == INVALID_OPERATION) { + mOutput->standby(); + mStandby = true; + mBytesWritten = 0; + status = mOutput->setParameters(keyValuePair); + } + if (status == NO_ERROR && reconfig) { + readOutputParameters(); + sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); + } + } + + mNewParameters.removeAt(0); + + mParamStatus = status; + mParamCond.signal(); + mWaitWorkCV.wait(mLock); + } + return reconfig; +} + +uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() +{ + uint32_t time; + if (AudioSystem::isLinearPCM(mFormat)) { + time = (uint32_t)(mOutput->latency() * 1000) / 2; + } else { + time = 10000; + } + return time; +} + +uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() +{ + uint32_t time; + if (AudioSystem::isLinearPCM(mFormat)) { + time = (uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000; + } else { + time = 10000; + } + return time; +} + +// ---------------------------------------------------------------------------- + +AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) + : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) +{ + mType = PlaybackThread::DUPLICATING; + addOutputTrack(mainThread); +} + +AudioFlinger::DuplicatingThread::~DuplicatingThread() +{ + for (size_t i = 0; i < mOutputTracks.size(); i++) { + mOutputTracks[i]->destroy(); + } + mOutputTracks.clear(); +} + +bool AudioFlinger::DuplicatingThread::threadLoop() +{ + Vector< sp<Track> > tracksToRemove; + uint32_t mixerStatus = MIXER_IDLE; + nsecs_t standbyTime = systemTime(); + size_t mixBufferSize = mFrameCount*mFrameSize; + SortedVector< sp<OutputTrack> > outputTracks; + uint32_t writeFrames = 0; + uint32_t activeSleepTime = activeSleepTimeUs(); + uint32_t idleSleepTime = idleSleepTimeUs(); + uint32_t sleepTime = idleSleepTime; + Vector< sp<EffectChain> > effectChains; + + while (!exitPending()) + { + processConfigEvents(); + + mixerStatus = MIXER_IDLE; + { // scope for the mLock + + Mutex::Autolock _l(mLock); + + if (checkForNewParameters_l()) { + mixBufferSize = mFrameCount*mFrameSize; + updateWaitTime(); + activeSleepTime = activeSleepTimeUs(); + idleSleepTime = idleSleepTimeUs(); + } + + const SortedVector< wp<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++) { + outputTracks[i]->stop(); + } + mStandby = true; + mBytesWritten = 0; + } + + if (!activeTracks.size() && mConfigEvents.isEmpty()) { + // we're about to wait, flush the binder command buffer + IPCThreadState::self()->flushCommands(); + outputTracks.clear(); + + if (exitPending()) break; + + LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); + mWaitWorkCV.wait(mLock); + LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); + if (mMasterMute == false) { + char value[PROPERTY_VALUE_MAX]; + property_get("ro.audio.silent", value, "0"); + if (atoi(value)) { + LOGD("Silence is golden"); + setMasterMute(true); + } + } + + standbyTime = systemTime() + kStandbyTimeInNsecs; + sleepTime = idleSleepTime; + continue; + } + } + + mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); + + // prevent any changes in effect chain list and in each effect chain + // during mixing and effect process as the audio buffers could be deleted + // or modified if an effect is created or deleted + effectChains = mEffectChains; + lockEffectChains_l(); + } + + if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { + // mix buffers... + if (outputsReady(outputTracks)) { + mAudioMixer->process(); + } else { + memset(mMixBuffer, 0, mixBufferSize); + } + sleepTime = 0; + writeFrames = mFrameCount; + } else { + if (sleepTime == 0) { + if (mixerStatus == MIXER_TRACKS_ENABLED) { + sleepTime = activeSleepTime; + } else { + sleepTime = idleSleepTime; + } + } else if (mBytesWritten != 0) { + // flush remaining overflow buffers in output tracks + for (size_t i = 0; i < outputTracks.size(); i++) { + if (outputTracks[i]->isActive()) { + sleepTime = 0; + writeFrames = 0; + memset(mMixBuffer, 0, mixBufferSize); + break; + } + } + } + } + + if (mSuspended) { + sleepTime = idleSleepTime; + } + // sleepTime == 0 means we must write to audio hardware + if (sleepTime == 0) { + for (size_t i = 0; i < effectChains.size(); i ++) { + effectChains[i]->process_l(); + } + // enable changes in effect chain + unlockEffectChains(); + + standbyTime = systemTime() + kStandbyTimeInNsecs; + for (size_t i = 0; i < outputTracks.size(); i++) { + outputTracks[i]->write(mMixBuffer, writeFrames); + } + mStandby = false; + mBytesWritten += mixBufferSize; + } else { + // enable changes in effect chain + unlockEffectChains(); + usleep(sleepTime); + } + + // finally let go of all our tracks, without the lock held + // since we can't guarantee the destructors won't acquire that + // same lock. + tracksToRemove.clear(); + outputTracks.clear(); + + // Effect chains will be actually deleted here if they were removed from + // mEffectChains list during mixing or effects processing + effectChains.clear(); + } + + return false; +} + +void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) +{ + int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); + OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, + this, + mSampleRate, + mFormat, + mChannelCount, + frameCount); + if (outputTrack->cblk() != NULL) { + thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f); + mOutputTracks.add(outputTrack); + LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); + updateWaitTime(); + } +} + +void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) +{ + Mutex::Autolock _l(mLock); + for (size_t i = 0; i < mOutputTracks.size(); i++) { + if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { + mOutputTracks[i]->destroy(); + mOutputTracks.removeAt(i); + updateWaitTime(); + return; + } + } + LOGV("removeOutputTrack(): unkonwn thread: %p", thread); +} + +void AudioFlinger::DuplicatingThread::updateWaitTime() +{ + mWaitTimeMs = UINT_MAX; + for (size_t i = 0; i < mOutputTracks.size(); i++) { + sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); + if (strong != NULL) { + uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); + if (waitTimeMs < mWaitTimeMs) { + mWaitTimeMs = waitTimeMs; + } + } + } +} + + +bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) +{ + for (size_t i = 0; i < outputTracks.size(); i++) { + sp <ThreadBase> thread = outputTracks[i]->thread().promote(); + if (thread == 0) { + LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); + return false; + } + PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); + if (playbackThread->standby() && !playbackThread->isSuspended()) { + LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); + return false; + } + } + return true; +} + +uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() +{ + return (mWaitTimeMs * 1000) / 2; +} + +// ---------------------------------------------------------------------------- + +// TrackBase constructor must be called with AudioFlinger::mLock held +AudioFlinger::ThreadBase::TrackBase::TrackBase( + const wp<ThreadBase>& thread, + const sp<Client>& client, + uint32_t sampleRate, + int format, + int channelCount, + int frameCount, + uint32_t flags, + const sp<IMemory>& sharedBuffer, + int sessionId) + : RefBase(), + mThread(thread), + mClient(client), + mCblk(0), + mFrameCount(0), + mState(IDLE), + mClientTid(-1), + mFormat(format), + mFlags(flags & ~SYSTEM_FLAGS_MASK), + mSessionId(sessionId) +{ + LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); + + // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); + size_t size = sizeof(audio_track_cblk_t); + size_t bufferSize = 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->channelCount = (uint8_t)channelCount; + if (sharedBuffer == 0) { + mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); + memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); + // Force underrun condition to avoid false underrun callback until first data is + // written to buffer + mCblk->flags = CBLK_UNDERRUN_ON; + } else { + mBuffer = sharedBuffer->pointer(); + } + mBufferEnd = (uint8_t *)mBuffer + bufferSize; + } + } else { + LOGE("not enough memory for AudioTrack size=%u", size); + client->heap()->dump("AudioTrack"); + return; + } + } else { + mCblk = (audio_track_cblk_t *)(new uint8_t[size]); + if (mCblk) { // construct the shared structure in-place. + new(mCblk) audio_track_cblk_t(); + // clear all buffers + mCblk->frameCount = frameCount; + mCblk->sampleRate = sampleRate; + mCblk->channelCount = (uint8_t)channelCount; + mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); + memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); + // Force underrun condition to avoid false underrun callback until first data is + // written to buffer + mCblk->flags = CBLK_UNDERRUN_ON; + mBufferEnd = (uint8_t *)mBuffer + bufferSize; + } + } +} + +AudioFlinger::ThreadBase::TrackBase::~TrackBase() +{ + if (mCblk) { + mCblk->~audio_track_cblk_t(); // destroy our shared-structure. + if (mClient == NULL) { + delete mCblk; + } + } + mCblkMemory.clear(); // and free the shared memory + if (mClient != NULL) { + Mutex::Autolock _l(mClient->audioFlinger()->mLock); + mClient.clear(); + } +} + +void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) +{ + buffer->raw = 0; + mFrameCount = buffer->frameCount; + step(); + buffer->frameCount = 0; +} + +bool AudioFlinger::ThreadBase::TrackBase::step() { + bool result; + audio_track_cblk_t* cblk = this->cblk(); + + result = cblk->stepServer(mFrameCount); + if (!result) { + LOGV("stepServer failed acquiring cblk mutex"); + mFlags |= STEPSERVER_FAILED; + } + return result; +} + +void AudioFlinger::ThreadBase::TrackBase::reset() { + audio_track_cblk_t* cblk = this->cblk(); + + cblk->user = 0; + cblk->server = 0; + cblk->userBase = 0; + cblk->serverBase = 0; + mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); + LOGV("TrackBase::reset"); +} + +sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const +{ + return mCblkMemory; +} + +int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { + return (int)mCblk->sampleRate; +} + +int AudioFlinger::ThreadBase::TrackBase::channelCount() const { + return (int)mCblk->channelCount; +} + +void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { + audio_track_cblk_t* cblk = this->cblk(); + int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; + int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; + + // Check validity of returned pointer in case the track control block would have been corrupted. + if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || + ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { + LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ + server %d, serverBase %d, user %d, userBase %d, channelCount %d", + bufferStart, bufferEnd, mBuffer, mBufferEnd, + cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount); + return 0; + } + + return bufferStart; +} + +// ---------------------------------------------------------------------------- + +// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held +AudioFlinger::PlaybackThread::Track::Track( + const wp<ThreadBase>& thread, + const sp<Client>& client, + int streamType, + uint32_t sampleRate, + int format, + int channelCount, + int frameCount, + const sp<IMemory>& sharedBuffer, + int sessionId) + : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId), + mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), mAuxEffectId(0) +{ + if (mCblk != NULL) { + sp<ThreadBase> baseThread = thread.promote(); + if (baseThread != 0) { + PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); + mName = playbackThread->getTrackName_l(); + mMainBuffer = playbackThread->mixBuffer(); + } + 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) { + if (!isOutputTrack()) { + if (mState == ACTIVE || mState == RESUMING) { + AudioSystem::stopOutput(thread->id(), (AudioSystem::stream_type)mStreamType); + } + AudioSystem::releaseOutput(thread->id()); + } + Mutex::Autolock _l(thread->mLock); + PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); + playbackThread->destroyTrack_l(this); + } + } +} + +void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) +{ + snprintf(buffer, size, " %05d %05d %03u %03u %03u %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n", + mName - AudioMixer::TRACK0, + (mClient == NULL) ? getpid() : mClient->pid(), + mStreamType, + mFormat, + mCblk->channelCount, + mSessionId, + mFrameCount, + mState, + mMute, + mFillingUpStatus, + mCblk->sampleRate, + mCblk->volume[0], + mCblk->volume[1], + mCblk->server, + mCblk->user, + (int)mMainBuffer, + (int)mAuxBuffer); +} + +status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) +{ + audio_track_cblk_t* cblk = this->cblk(); + uint32_t framesReady; + uint32_t framesReq = buffer->frameCount; + + // Check if last stepServer failed, try to step now + if (mFlags & TrackBase::STEPSERVER_FAILED) { + if (!step()) goto getNextBuffer_exit; + LOGV("stepServer recovered"); + mFlags &= ~TrackBase::STEPSERVER_FAILED; + } + + framesReady = cblk->framesReady(); + + if (LIKELY(framesReady)) { + uint32_t s = cblk->server; + uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; + + bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; + if (framesReq > framesReady) { + framesReq = framesReady; + } + if (s + framesReq > bufferEnd) { + framesReq = bufferEnd - s; + } + + buffer->raw = getBuffer(s, framesReq); + if (buffer->raw == 0) goto getNextBuffer_exit; + + buffer->frameCount = framesReq; + return NO_ERROR; + } + +getNextBuffer_exit: + buffer->raw = 0; + buffer->frameCount = 0; + LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); + return NOT_ENOUGH_DATA; +} + +bool AudioFlinger::PlaybackThread::Track::isReady() const { + if (mFillingUpStatus != FS_FILLING) return true; + + if (mCblk->framesReady() >= mCblk->frameCount || + (mCblk->flags & CBLK_FORCEREADY_MSK)) { + mFillingUpStatus = FS_FILLED; + mCblk->flags &= ~CBLK_FORCEREADY_MSK; + return true; + } + return false; +} + +status_t AudioFlinger::PlaybackThread::Track::start() +{ + status_t status = NO_ERROR; + LOGV("start(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + Mutex::Autolock _l(thread->mLock); + int state = mState; + // here the track could be either new, or restarted + // in both cases "unstop" the track + if (mState == PAUSED) { + mState = TrackBase::RESUMING; + LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); + } else { + mState = TrackBase::ACTIVE; + LOGV("? => ACTIVE (%d) on thread %p", mName, this); + } + + if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { + thread->mLock.unlock(); + status = AudioSystem::startOutput(thread->id(), (AudioSystem::stream_type)mStreamType); + thread->mLock.lock(); + } + if (status == NO_ERROR) { + PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); + playbackThread->addTrack_l(this); + } else { + mState = state; + } + } else { + status = BAD_VALUE; + } + return status; +} + +void AudioFlinger::PlaybackThread::Track::stop() +{ + LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + Mutex::Autolock _l(thread->mLock); + int state = mState; + if (mState > STOPPED) { + mState = STOPPED; + // If the track is not active (PAUSED and buffers full), flush buffers + PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); + if (playbackThread->mActiveTracks.indexOf(this) < 0) { + reset(); + } + LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); + } + if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { + thread->mLock.unlock(); + AudioSystem::stopOutput(thread->id(), (AudioSystem::stream_type)mStreamType); + thread->mLock.lock(); + } + } +} + +void AudioFlinger::PlaybackThread::Track::pause() +{ + LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + Mutex::Autolock _l(thread->mLock); + if (mState == ACTIVE || mState == RESUMING) { + mState = PAUSING; + LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); + if (!isOutputTrack()) { + thread->mLock.unlock(); + AudioSystem::stopOutput(thread->id(), (AudioSystem::stream_type)mStreamType); + thread->mLock.lock(); + } + } + } +} + +void AudioFlinger::PlaybackThread::Track::flush() +{ + LOGV("flush(%d)", mName); + sp<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->flags |= CBLK_UNDERRUN_ON; + mCblk->flags &= ~CBLK_FORCEREADY_MSK; + mFillingUpStatus = FS_FILLING; + mResetDone = true; + } +} + +void AudioFlinger::PlaybackThread::Track::mute(bool muted) +{ + mMute = muted; +} + +void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) +{ + mVolume[0] = left; + mVolume[1] = right; +} + +status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) +{ + status_t status = DEAD_OBJECT; + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); + status = playbackThread->attachAuxEffect(this, EffectId); + } + return status; +} + +void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) +{ + mAuxEffectId = EffectId; + mAuxBuffer = buffer; +} + +// ---------------------------------------------------------------------------- + +// RecordTrack constructor must be called with AudioFlinger::mLock held +AudioFlinger::RecordThread::RecordTrack::RecordTrack( + const wp<ThreadBase>& thread, + const sp<Client>& client, + uint32_t sampleRate, + int format, + int channelCount, + int frameCount, + uint32_t flags, + int sessionId) + : TrackBase(thread, client, sampleRate, format, + channelCount, frameCount, flags, 0, sessionId), + mOverflow(false) +{ + if (mCblk != NULL) { + LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); + if (format == AudioSystem::PCM_16_BIT) { + mCblk->frameSize = channelCount * sizeof(int16_t); + } else if (format == AudioSystem::PCM_8_BIT) { + mCblk->frameSize = channelCount * sizeof(int8_t); + } else { + mCblk->frameSize = sizeof(int8_t); + } + } +} + +AudioFlinger::RecordThread::RecordTrack::~RecordTrack() +{ + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + AudioSystem::releaseInput(thread->id()); + } +} + +status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) +{ + audio_track_cblk_t* cblk = this->cblk(); + uint32_t framesAvail; + uint32_t framesReq = buffer->frameCount; + + // Check if last stepServer failed, try to step now + if (mFlags & TrackBase::STEPSERVER_FAILED) { + if (!step()) goto getNextBuffer_exit; + LOGV("stepServer recovered"); + mFlags &= ~TrackBase::STEPSERVER_FAILED; + } + + framesAvail = cblk->framesAvailable_l(); + + if (LIKELY(framesAvail)) { + uint32_t s = cblk->server; + uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; + + if (framesReq > framesAvail) { + framesReq = framesAvail; + } + if (s + framesReq > bufferEnd) { + framesReq = bufferEnd - s; + } + + buffer->raw = getBuffer(s, framesReq); + if (buffer->raw == 0) goto getNextBuffer_exit; + + buffer->frameCount = framesReq; + return NO_ERROR; + } + +getNextBuffer_exit: + buffer->raw = 0; + buffer->frameCount = 0; + return NOT_ENOUGH_DATA; +} + +status_t AudioFlinger::RecordThread::RecordTrack::start() +{ + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + RecordThread *recordThread = (RecordThread *)thread.get(); + return recordThread->start(this); + } else { + return BAD_VALUE; + } +} + +void AudioFlinger::RecordThread::RecordTrack::stop() +{ + sp<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->flags |= CBLK_UNDERRUN_ON; + } +} + +void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) +{ + snprintf(buffer, size, " %05d %03u %03u %05d %04u %01d %05u %08x %08x\n", + (mClient == NULL) ? getpid() : mClient->pid(), + mFormat, + mCblk->channelCount, + mSessionId, + mFrameCount, + mState, + mCblk->sampleRate, + mCblk->server, + mCblk->user); +} + + +// ---------------------------------------------------------------------------- + +AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( + const wp<ThreadBase>& thread, + DuplicatingThread *sourceThread, + uint32_t sampleRate, + int format, + int channelCount, + int frameCount) + : Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL, 0), + mActive(false), mSourceThread(sourceThread) +{ + + PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); + if (mCblk != NULL) { + mCblk->flags |= CBLK_DIRECTION_OUT; + mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); + mCblk->volume[0] = mCblk->volume[1] = 0x1000; + mOutBuffer.frameCount = 0; + playbackThread->mTracks.add(this); + LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p", + mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd); + } else { + LOGW("Error creating output track on thread %p", playbackThread); + } +} + +AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() +{ + clearBufferQueue(); +} + +status_t AudioFlinger::PlaybackThread::OutputTrack::start() +{ + status_t status = Track::start(); + if (status != NO_ERROR) { + return status; + } + + mActive = true; + mRetryCount = 127; + return status; +} + +void AudioFlinger::PlaybackThread::OutputTrack::stop() +{ + Track::stop(); + clearBufferQueue(); + mOutBuffer.frameCount = 0; + mActive = false; +} + +bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) +{ + Buffer *pInBuffer; + Buffer inBuffer; + uint32_t channelCount = mCblk->channelCount; + bool outputBufferFull = false; + inBuffer.frameCount = frames; + inBuffer.i16 = data; + + uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); + + if (!mActive && frames != 0) { + start(); + sp<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 * channelCount]; + pInBuffer->frameCount = startFrames; + pInBuffer->i16 = pInBuffer->mBuffer; + memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); + mBufferQueue.add(pInBuffer); + } else { + LOGW ("OutputTrack::write() %p no more buffers in queue", this); + } + } + } + } + + while (waitTimeLeftMs) { + // First write pending buffers, then new data + if (mBufferQueue.size()) { + pInBuffer = mBufferQueue.itemAt(0); + } else { + pInBuffer = &inBuffer; + } + + if (pInBuffer->frameCount == 0) { + break; + } + + if (mOutBuffer.frameCount == 0) { + mOutBuffer.frameCount = pInBuffer->frameCount; + nsecs_t startTime = systemTime(); + if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { + LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); + outputBufferFull = true; + break; + } + uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); + if (waitTimeLeftMs >= waitTimeMs) { + waitTimeLeftMs -= waitTimeMs; + } else { + waitTimeLeftMs = 0; + } + } + + uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; + memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); + mCblk->stepUser(outFrames); + pInBuffer->frameCount -= outFrames; + pInBuffer->i16 += outFrames * channelCount; + mOutBuffer.frameCount -= outFrames; + mOutBuffer.i16 += outFrames * channelCount; + + if (pInBuffer->frameCount == 0) { + if (mBufferQueue.size()) { + mBufferQueue.removeAt(0); + delete [] pInBuffer->mBuffer; + delete pInBuffer; + LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); + } else { + break; + } + } + } + + // If we could not write all frames, allocate a buffer and queue it for next time. + if (inBuffer.frameCount) { + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0 && !thread->standby()) { + if (mBufferQueue.size() < kMaxOverFlowBuffers) { + pInBuffer = new Buffer; + pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; + pInBuffer->frameCount = inBuffer.frameCount; + pInBuffer->i16 = pInBuffer->mBuffer; + memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); + mBufferQueue.add(pInBuffer); + LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); + } else { + LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); + } + } + } + + // Calling write() with a 0 length buffer, means that no more data will be written: + // If no more buffers are pending, fill output track buffer to make sure it is started + // by output mixer. + if (frames == 0 && mBufferQueue.size() == 0) { + if (mCblk->user < mCblk->frameCount) { + frames = mCblk->frameCount - mCblk->user; + pInBuffer = new Buffer; + pInBuffer->mBuffer = new int16_t[frames * channelCount]; + pInBuffer->frameCount = frames; + pInBuffer->i16 = pInBuffer->mBuffer; + memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); + mBufferQueue.add(pInBuffer); + } else if (mActive) { + stop(); + } + } + + return outputBufferFull; +} + +status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) +{ + int active; + status_t result; + audio_track_cblk_t* cblk = mCblk; + uint32_t framesReq = buffer->frameCount; + +// LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); + buffer->frameCount = 0; + + uint32_t framesAvail = cblk->framesAvailable(); + + + if (framesAvail == 0) { + Mutex::Autolock _l(cblk->lock); + goto start_loop_here; + while (framesAvail == 0) { + active = mActive; + if (UNLIKELY(!active)) { + LOGV("Not active and NO_MORE_BUFFERS"); + return AudioTrack::NO_MORE_BUFFERS; + } + result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); + if (result != NO_ERROR) { + return AudioTrack::NO_MORE_BUFFERS; + } + // read the server count again + start_loop_here: + framesAvail = cblk->framesAvailable_l(); + } + } + +// if (framesAvail < framesReq) { +// return AudioTrack::NO_MORE_BUFFERS; +// } + + if (framesReq > framesAvail) { + framesReq = framesAvail; + } + + uint32_t u = cblk->user; + uint32_t bufferEnd = cblk->userBase + cblk->frameCount; + + if (u + framesReq > bufferEnd) { + framesReq = bufferEnd - u; + } + + buffer->frameCount = framesReq; + buffer->raw = (void *)cblk->buffer(u); + return NO_ERROR; +} + + +void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() +{ + size_t size = mBufferQueue.size(); + Buffer *pBuffer; + + for (size_t i = 0; i < size; i++) { + pBuffer = mBufferQueue.itemAt(i); + delete [] pBuffer->mBuffer; + delete pBuffer; + } + mBufferQueue.clear(); +} + +// ---------------------------------------------------------------------------- + +AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) + : RefBase(), + mAudioFlinger(audioFlinger), + mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), + mPid(pid) +{ + // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer +} + +// Client destructor must be called with AudioFlinger::mLock held +AudioFlinger::Client::~Client() +{ + mAudioFlinger->removeClient_l(mPid); +} + +const sp<MemoryDealer>& AudioFlinger::Client::heap() const +{ + return mMemoryDealer; +} + +// ---------------------------------------------------------------------------- + +AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, + const sp<IAudioFlingerClient>& client, + pid_t pid) + : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) +{ +} + +AudioFlinger::NotificationClient::~NotificationClient() +{ + mClient.clear(); +} + +void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) +{ + sp<NotificationClient> keep(this); + { + mAudioFlinger->removeNotificationClient(mPid); + } +} + +// ---------------------------------------------------------------------------- + +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::attachAuxEffect(int EffectId) +{ + return mTrack->attachAuxEffect(EffectId); +} + +status_t AudioFlinger::TrackHandle::onTransact( + uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) +{ + return BnAudioTrack::onTransact(code, data, reply, flags); +} + +// ---------------------------------------------------------------------------- + +sp<IAudioRecord> AudioFlinger::openRecord( + pid_t pid, + int input, + uint32_t sampleRate, + int format, + int channelCount, + int frameCount, + uint32_t flags, + int *sessionId, + status_t *status) +{ + sp<RecordThread::RecordTrack> recordTrack; + sp<RecordHandle> recordHandle; + sp<Client> client; + wp<Client> wclient; + status_t lStatus; + RecordThread *thread; + size_t inFrameCount; + int lSessionId; + + // 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); + } + + // If no audio session id is provided, create one here + if (sessionId != NULL && *sessionId != 0) { + lSessionId = *sessionId; + } else { + lSessionId = nextUniqueId(); + if (sessionId != NULL) { + *sessionId = lSessionId; + } + } + // create new record track. The record track uses one track in mHardwareMixerThread by convention. + recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate, + format, channelCount, frameCount, flags, lSessionId); + } + if (recordTrack->getCblk() == NULL) { + // remove local strong reference to Client before deleting the RecordTrack so that the Client + // destructor is called by the TrackBase destructor with mLock held + client.clear(); + recordTrack.clear(); + lStatus = NO_MEMORY; + goto Exit; + } + + // return to handle to client + recordHandle = new RecordHandle(recordTrack); + lStatus = NO_ERROR; + +Exit: + if (status) { + *status = lStatus; + } + return recordHandle; +} + +// ---------------------------------------------------------------------------- + +AudioFlinger::RecordHandle::RecordHandle(const sp<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, int id) : + ThreadBase(audioFlinger, id), + mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) +{ + mReqChannelCount = AudioSystem::popCount(channels); + mReqSampleRate = sampleRate; + readInputParameters(); +} + + +AudioFlinger::RecordThread::~RecordThread() +{ + delete[] mRsmpInBuffer; + if (mResampler != 0) { + delete mResampler; + delete[] mRsmpOutBuffer; + } +} + +void AudioFlinger::RecordThread::onFirstRef() +{ + const size_t SIZE = 256; + char buffer[SIZE]; + + snprintf(buffer, SIZE, "Record Thread %p", this); + + run(buffer, PRIORITY_URGENT_AUDIO); +} + +bool AudioFlinger::RecordThread::threadLoop() +{ + AudioBufferProvider::Buffer buffer; + sp<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) { + if (!mStandby) { + mInput->standby(); + mStandby = true; + } + mActiveTrack.clear(); + mStartStopCond.broadcast(); + } else if (mActiveTrack->mState == TrackBase::RESUMING) { + if (mReqChannelCount != mActiveTrack->channelCount()) { + mActiveTrack.clear(); + mStartStopCond.broadcast(); + } else if (mBytesRead != 0) { + // record start succeeds only if first read from audio input + // succeeds + if (mBytesRead > 0) { + mActiveTrack->mState = TrackBase::ACTIVE; + } else { + mActiveTrack.clear(); + } + mStartStopCond.broadcast(); + } + mStandby = false; + } + } + } + + if (mActiveTrack != 0) { + if (mActiveTrack->mState != TrackBase::ACTIVE && + mActiveTrack->mState != TrackBase::RESUMING) { + usleep(5000); + continue; + } + buffer.frameCount = mFrameCount; + if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { + size_t framesOut = buffer.frameCount; + if (mResampler == 0) { + // no resampling + while (framesOut) { + size_t framesIn = mFrameCount - mRsmpInIndex; + if (framesIn) { + int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; + int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; + if (framesIn > framesOut) + framesIn = framesOut; + mRsmpInIndex += framesIn; + framesOut -= framesIn; + if ((int)mChannelCount == mReqChannelCount || + mFormat != AudioSystem::PCM_16_BIT) { + memcpy(dst, src, framesIn * mFrameSize); + } else { + int16_t *src16 = (int16_t *)src; + int16_t *dst16 = (int16_t *)dst; + if (mChannelCount == 1) { + while (framesIn--) { + *dst16++ = *src16; + *dst16++ = *src16++; + } + } else { + while (framesIn--) { + *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); + src16 += 2; + } + } + } + } + if (framesOut && mFrameCount == mRsmpInIndex) { + if (framesOut == mFrameCount && + ((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) { + mBytesRead = mInput->read(buffer.raw, mInputBytes); + framesOut = 0; + } else { + mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); + mRsmpInIndex = 0; + } + if (mBytesRead < 0) { + LOGE("Error reading audio input"); + if (mActiveTrack->mState == TrackBase::ACTIVE) { + // Force input into standby so that it tries to + // recover at next read attempt + mInput->standby(); + usleep(5000); + } + mRsmpInIndex = mFrameCount; + framesOut = 0; + buffer.frameCount = 0; + } + } + } + } else { + // resampling + + memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); + // alter output frame count as if we were expecting stereo samples + if (mChannelCount == 1 && mReqChannelCount == 1) { + framesOut >>= 1; + } + mResampler->resample(mRsmpOutBuffer, framesOut, this); + // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() + // are 32 bit aligned which should be always true. + if (mChannelCount == 2 && mReqChannelCount == 1) { + AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); + // the resampler always outputs stereo samples: do post stereo to mono conversion + int16_t *src = (int16_t *)mRsmpOutBuffer; + int16_t *dst = buffer.i16; + while (framesOut--) { + *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); + src += 2; + } + } else { + AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); + } + + } + mActiveTrack->releaseBuffer(&buffer); + mActiveTrack->overflow(); + } + // client isn't retrieving buffers fast enough + else { + if (!mActiveTrack->setOverflow()) + LOGW("RecordThread: buffer overflow"); + // Release the processor for a while before asking for a new buffer. + // This will give the application more chance to read from the buffer and + // clear the overflow. + usleep(5000); + } + } + } + + if (!mStandby) { + mInput->standby(); + } + mActiveTrack.clear(); + + mStartStopCond.broadcast(); + + LOGV("RecordThread %p exiting", this); + return false; +} + +status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) +{ + LOGV("RecordThread::start"); + sp <ThreadBase> strongMe = this; + status_t status = NO_ERROR; + { + AutoMutex lock(&mLock); + if (mActiveTrack != 0) { + if (recordTrack != mActiveTrack.get()) { + status = -EBUSY; + } else if (mActiveTrack->mState == TrackBase::PAUSING) { + mActiveTrack->mState = TrackBase::ACTIVE; + } + return status; + } + + recordTrack->mState = TrackBase::IDLE; + mActiveTrack = recordTrack; + mLock.unlock(); + status_t status = AudioSystem::startInput(mId); + mLock.lock(); + if (status != NO_ERROR) { + mActiveTrack.clear(); + return status; + } + mActiveTrack->mState = TrackBase::RESUMING; + mRsmpInIndex = mFrameCount; + mBytesRead = 0; + // signal thread to start + LOGV("Signal record thread"); + mWaitWorkCV.signal(); + // do not wait for mStartStopCond if exiting + if (mExiting) { + mActiveTrack.clear(); + status = INVALID_OPERATION; + goto startError; + } + mStartStopCond.wait(mLock); + if (mActiveTrack == 0) { + LOGV("Record failed to start"); + status = BAD_VALUE; + goto startError; + } + LOGV("Record started OK"); + return status; + } +startError: + AudioSystem::stopInput(mId); + return status; +} + +void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { + LOGV("RecordThread::stop"); + sp <ThreadBase> strongMe = this; + { + AutoMutex lock(&mLock); + if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { + mActiveTrack->mState = TrackBase::PAUSING; + // do not wait for mStartStopCond if exiting + if (mExiting) { + return; + } + mStartStopCond.wait(mLock); + // if we have been restarted, recordTrack == mActiveTrack.get() here + if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { + mLock.unlock(); + AudioSystem::stopInput(mId); + mLock.lock(); + LOGV("Record stopped OK"); + } + } + } +} + +status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + pid_t pid = 0; + + snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); + result.append(buffer); + + if (mActiveTrack != 0) { + result.append("Active Track:\n"); + result.append(" Clien Fmt Chn Session Buf S SRate Serv User\n"); + mActiveTrack->dump(buffer, SIZE); + result.append(buffer); + + snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); + result.append(buffer); + snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); + result.append(buffer); + snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); + result.append(buffer); + snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); + result.append(buffer); + snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); + result.append(buffer); + + + } else { + result.append("No record client\n"); + } + write(fd, result.string(), result.size()); + + dumpBase(fd, args); + + return NO_ERROR; +} + +status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) +{ + size_t framesReq = buffer->frameCount; + size_t framesReady = mFrameCount - mRsmpInIndex; + int channelCount; + + if (framesReady == 0) { + mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes); + if (mBytesRead < 0) { + LOGE("RecordThread::getNextBuffer() Error reading audio input"); + if (mActiveTrack->mState == TrackBase::ACTIVE) { + // Force input into standby so that it tries to + // recover at next read attempt + mInput->standby(); + usleep(5000); + } + buffer->raw = 0; + buffer->frameCount = 0; + return NOT_ENOUGH_DATA; + } + mRsmpInIndex = 0; + framesReady = mFrameCount; + } + + if (framesReq > framesReady) { + framesReq = framesReady; + } + + if (mChannelCount == 1 && mReqChannelCount == 2) { + channelCount = 1; + } else { + channelCount = 2; + } + buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; + buffer->frameCount = framesReq; + return NO_ERROR; +} + +void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) +{ + mRsmpInIndex += buffer->frameCount; + buffer->frameCount = 0; +} + +bool AudioFlinger::RecordThread::checkForNewParameters_l() +{ + bool reconfig = false; + + while (!mNewParameters.isEmpty()) { + status_t status = NO_ERROR; + String8 keyValuePair = mNewParameters[0]; + AudioParameter param = AudioParameter(keyValuePair); + int value; + int reqFormat = mFormat; + int reqSamplingRate = mReqSampleRate; + int reqChannelCount = mReqChannelCount; + + if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { + reqSamplingRate = value; + reconfig = true; + } + if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { + reqFormat = value; + reconfig = true; + } + if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { + reqChannelCount = AudioSystem::popCount(value); + reconfig = true; + } + if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { + // do not accept frame count changes if tracks are open as the track buffer + // size depends on frame count and correct behavior would not be garantied + // if frame count is changed after track creation + if (mActiveTrack != 0) { + status = INVALID_OPERATION; + } else { + reconfig = true; + } + } + if (status == NO_ERROR) { + status = mInput->setParameters(keyValuePair); + if (status == INVALID_OPERATION) { + mInput->standby(); + status = mInput->setParameters(keyValuePair); + } + if (reconfig) { + if (status == BAD_VALUE && + reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT && + ((int)mInput->sampleRate() <= 2 * reqSamplingRate) && + (AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) { + status = NO_ERROR; + } + if (status == NO_ERROR) { + readInputParameters(); + sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); + } + } + } + + mNewParameters.removeAt(0); + + mParamStatus = status; + mParamCond.signal(); + mWaitWorkCV.wait(mLock); + } + return reconfig; +} + +String8 AudioFlinger::RecordThread::getParameters(const String8& keys) +{ + return mInput->getParameters(keys); +} + +void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { + AudioSystem::OutputDescriptor desc; + void *param2 = 0; + + switch (event) { + case AudioSystem::INPUT_OPENED: + case AudioSystem::INPUT_CONFIG_CHANGED: + desc.channels = mChannels; + desc.samplingRate = mSampleRate; + desc.format = mFormat; + desc.frameCount = mFrameCount; + desc.latency = 0; + param2 = &desc; + break; + + case AudioSystem::INPUT_CLOSED: + default: + break; + } + mAudioFlinger->audioConfigChanged_l(event, mId, param2); +} + +void AudioFlinger::RecordThread::readInputParameters() +{ + if (mRsmpInBuffer) delete mRsmpInBuffer; + if (mRsmpOutBuffer) delete mRsmpOutBuffer; + if (mResampler) delete mResampler; + mResampler = 0; + + mSampleRate = mInput->sampleRate(); + mChannels = mInput->channels(); + mChannelCount = (uint16_t)AudioSystem::popCount(mChannels); + mFormat = mInput->format(); + mFrameSize = (uint16_t)mInput->frameSize(); + mInputBytes = mInput->bufferSize(); + mFrameCount = mInputBytes / mFrameSize; + mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; + + if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) + { + int channelCount; + // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid + // stereo to mono post process as the resampler always outputs stereo. + if (mChannelCount == 1 && mReqChannelCount == 2) { + channelCount = 1; + } else { + channelCount = 2; + } + mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); + mResampler->setSampleRate(mSampleRate); + mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); + mRsmpOutBuffer = new int32_t[mFrameCount * 2]; + + // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples + if (mChannelCount == 1 && mReqChannelCount == 1) { + mFrameCount >>= 1; + } + + } + mRsmpInIndex = mFrameCount; +} + +unsigned int AudioFlinger::RecordThread::getInputFramesLost() +{ + return mInput->getInputFramesLost(); +} + +// ---------------------------------------------------------------------------- + +int AudioFlinger::openOutput(uint32_t *pDevices, + uint32_t *pSamplingRate, + uint32_t *pFormat, + uint32_t *pChannels, + uint32_t *pLatencyMs, + uint32_t flags) +{ + status_t status; + PlaybackThread *thread = NULL; + mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; + uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; + uint32_t format = pFormat ? *pFormat : 0; + uint32_t channels = pChannels ? *pChannels : 0; + uint32_t latency = pLatencyMs ? *pLatencyMs : 0; + + LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", + pDevices ? *pDevices : 0, + samplingRate, + format, + channels, + flags); + + if (pDevices == NULL || *pDevices == 0) { + return 0; + } + Mutex::Autolock _l(mLock); + + AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices, + (int *)&format, + &channels, + &samplingRate, + &status); + LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", + output, + samplingRate, + format, + channels, + status); + + mHardwareStatus = AUDIO_HW_IDLE; + if (output != 0) { + int id = nextUniqueId(); + if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) || + (format != AudioSystem::PCM_16_BIT) || + (channels != AudioSystem::CHANNEL_OUT_STEREO)) { + thread = new DirectOutputThread(this, output, id, *pDevices); + LOGV("openOutput() created direct output: ID %d thread %p", id, thread); + } else { + thread = new MixerThread(this, output, id, *pDevices); + LOGV("openOutput() created mixer output: ID %d thread %p", id, thread); + +#ifdef LVMX + unsigned bitsPerSample = + (format == AudioSystem::PCM_16_BIT) ? 16 : + ((format == AudioSystem::PCM_8_BIT) ? 8 : 0); + unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1; + int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id()); + + LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount); + LifeVibes::setDevice(audioOutputType, *pDevices); +#endif + + } + mPlaybackThreads.add(id, thread); + + if (pSamplingRate) *pSamplingRate = samplingRate; + if (pFormat) *pFormat = format; + if (pChannels) *pChannels = channels; + if (pLatencyMs) *pLatencyMs = thread->latency(); + + // notify client processes of the new output creation + thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); + return id; + } + + return 0; +} + +int AudioFlinger::openDuplicateOutput(int output1, int output2) +{ + Mutex::Autolock _l(mLock); + MixerThread *thread1 = checkMixerThread_l(output1); + MixerThread *thread2 = checkMixerThread_l(output2); + + if (thread1 == NULL || thread2 == NULL) { + LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); + return 0; + } + + int id = nextUniqueId(); + DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); + thread->addOutputTrack(thread2); + mPlaybackThreads.add(id, thread); + // notify client processes of the new output creation + thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); + return id; +} + +status_t AudioFlinger::closeOutput(int output) +{ + // keep strong reference on the playback thread so that + // it is not destroyed while exit() is executed + sp <PlaybackThread> thread; + { + Mutex::Autolock _l(mLock); + thread = checkPlaybackThread_l(output); + if (thread == NULL) { + return BAD_VALUE; + } + + LOGV("closeOutput() %d", output); + + if (thread->type() == PlaybackThread::MIXER) { + for (size_t i = 0; i < mPlaybackThreads.size(); i++) { + if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) { + DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); + dupThread->removeOutputTrack((MixerThread *)thread.get()); + } + } + } + void *param2 = 0; + audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); + mPlaybackThreads.removeItem(output); + } + thread->exit(); + + if (thread->type() != PlaybackThread::DUPLICATING) { + mAudioHardware->closeOutputStream(thread->getOutput()); + } + return NO_ERROR; +} + +status_t AudioFlinger::suspendOutput(int output) +{ + Mutex::Autolock _l(mLock); + PlaybackThread *thread = checkPlaybackThread_l(output); + + if (thread == NULL) { + return BAD_VALUE; + } + + LOGV("suspendOutput() %d", output); + thread->suspend(); + + return NO_ERROR; +} + +status_t AudioFlinger::restoreOutput(int output) +{ + Mutex::Autolock _l(mLock); + PlaybackThread *thread = checkPlaybackThread_l(output); + + if (thread == NULL) { + return BAD_VALUE; + } + + LOGV("restoreOutput() %d", output); + + thread->restore(); + + return NO_ERROR; +} + +int AudioFlinger::openInput(uint32_t *pDevices, + uint32_t *pSamplingRate, + uint32_t *pFormat, + uint32_t *pChannels, + uint32_t acoustics) +{ + status_t status; + RecordThread *thread = NULL; + uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; + uint32_t format = pFormat ? *pFormat : 0; + uint32_t channels = pChannels ? *pChannels : 0; + uint32_t reqSamplingRate = samplingRate; + uint32_t reqFormat = format; + uint32_t reqChannels = channels; + + if (pDevices == NULL || *pDevices == 0) { + return 0; + } + Mutex::Autolock _l(mLock); + + AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices, + (int *)&format, + &channels, + &samplingRate, + &status, + (AudioSystem::audio_in_acoustics)acoustics); + LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", + input, + samplingRate, + format, + channels, + acoustics, + status); + + // If the input could not be opened with the requested parameters and we can handle the conversion internally, + // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo + // or stereo to mono conversions on 16 bit PCM inputs. + if (input == 0 && status == BAD_VALUE && + reqFormat == format && format == AudioSystem::PCM_16_BIT && + (samplingRate <= 2 * reqSamplingRate) && + (AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) { + LOGV("openInput() reopening with proposed sampling rate and channels"); + input = mAudioHardware->openInputStream(*pDevices, + (int *)&format, + &channels, + &samplingRate, + &status, + (AudioSystem::audio_in_acoustics)acoustics); + } + + if (input != 0) { + int id = nextUniqueId(); + // Start record thread + thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id); + mRecordThreads.add(id, thread); + LOGV("openInput() created record thread: ID %d thread %p", id, thread); + if (pSamplingRate) *pSamplingRate = reqSamplingRate; + if (pFormat) *pFormat = format; + if (pChannels) *pChannels = reqChannels; + + input->standby(); + + // notify client processes of the new input creation + thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); + return id; + } + + return 0; +} + +status_t AudioFlinger::closeInput(int input) +{ + // keep strong reference on the record thread so that + // it is not destroyed while exit() is executed + sp <RecordThread> thread; + { + Mutex::Autolock _l(mLock); + thread = checkRecordThread_l(input); + if (thread == NULL) { + return BAD_VALUE; + } + + LOGV("closeInput() %d", input); + void *param2 = 0; + audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); + mRecordThreads.removeItem(input); + } + thread->exit(); + + mAudioHardware->closeInputStream(thread->getInput()); + + return NO_ERROR; +} + +status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) +{ + Mutex::Autolock _l(mLock); + MixerThread *dstThread = checkMixerThread_l(output); + if (dstThread == NULL) { + LOGW("setStreamOutput() bad output id %d", output); + return BAD_VALUE; + } + + LOGV("setStreamOutput() stream %d to output %d", stream, output); + audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); + + for (size_t i = 0; i < mPlaybackThreads.size(); i++) { + PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); + if (thread != dstThread && + thread->type() != PlaybackThread::DIRECT) { + MixerThread *srcThread = (MixerThread *)thread; + srcThread->invalidateTracks(stream); + } + } + + return NO_ERROR; +} + + +int AudioFlinger::newAudioSessionId() +{ + return nextUniqueId(); +} + +// checkPlaybackThread_l() must be called with AudioFlinger::mLock held +AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const +{ + PlaybackThread *thread = NULL; + if (mPlaybackThreads.indexOfKey(output) >= 0) { + thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); + } + return thread; +} + +// checkMixerThread_l() must be called with AudioFlinger::mLock held +AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const +{ + PlaybackThread *thread = checkPlaybackThread_l(output); + if (thread != NULL) { + if (thread->type() == PlaybackThread::DIRECT) { + thread = NULL; + } + } + return (MixerThread *)thread; +} + +// checkRecordThread_l() must be called with AudioFlinger::mLock held +AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const +{ + RecordThread *thread = NULL; + if (mRecordThreads.indexOfKey(input) >= 0) { + thread = (RecordThread *)mRecordThreads.valueFor(input).get(); + } + return thread; +} + +int AudioFlinger::nextUniqueId() +{ + return android_atomic_inc(&mNextUniqueId); +} + +// ---------------------------------------------------------------------------- +// Effect management +// ---------------------------------------------------------------------------- + + +status_t AudioFlinger::loadEffectLibrary(const char *libPath, int *handle) +{ + Mutex::Autolock _l(mLock); + return EffectLoadLibrary(libPath, handle); +} + +status_t AudioFlinger::unloadEffectLibrary(int handle) +{ + Mutex::Autolock _l(mLock); + return EffectUnloadLibrary(handle); +} + +status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) +{ + Mutex::Autolock _l(mLock); + return EffectQueryNumberEffects(numEffects); +} + +status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) +{ + Mutex::Autolock _l(mLock); + return EffectQueryEffect(index, descriptor); +} + +status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) +{ + Mutex::Autolock _l(mLock); + return EffectGetDescriptor(pUuid, descriptor); +} + + +// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp +static const effect_uuid_t VISUALIZATION_UUID_ = + {0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; + +sp<IEffect> AudioFlinger::createEffect(pid_t pid, + effect_descriptor_t *pDesc, + const sp<IEffectClient>& effectClient, + int32_t priority, + int output, + int sessionId, + status_t *status, + int *id, + int *enabled) +{ + status_t lStatus = NO_ERROR; + sp<EffectHandle> handle; + effect_interface_t itfe; + effect_descriptor_t desc; + sp<Client> client; + wp<Client> wclient; + + LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, output %d", pid, effectClient.get(), priority, sessionId, output); + + if (pDesc == NULL) { + lStatus = BAD_VALUE; + goto Exit; + } + + { + Mutex::Autolock _l(mLock); + + // check recording permission for visualizer + if (memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 || + memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) { + if (!recordingAllowed()) { + lStatus = PERMISSION_DENIED; + goto Exit; + } + } + + if (!EffectIsNullUuid(&pDesc->uuid)) { + // if uuid is specified, request effect descriptor + lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); + if (lStatus < 0) { + LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); + goto Exit; + } + } else { + // if uuid is not specified, look for an available implementation + // of the required type in effect factory + if (EffectIsNullUuid(&pDesc->type)) { + LOGW("createEffect() no effect type"); + lStatus = BAD_VALUE; + goto Exit; + } + uint32_t numEffects = 0; + effect_descriptor_t d; + bool found = false; + + lStatus = EffectQueryNumberEffects(&numEffects); + if (lStatus < 0) { + LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); + goto Exit; + } + for (uint32_t i = 0; i < numEffects; i++) { + lStatus = EffectQueryEffect(i, &desc); + if (lStatus < 0) { + LOGW("createEffect() error %d from EffectQueryEffect", lStatus); + continue; + } + if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { + // If matching type found save effect descriptor. If the session is + // 0 and the effect is not auxiliary, continue enumeration in case + // an auxiliary version of this effect type is available + found = true; + memcpy(&d, &desc, sizeof(effect_descriptor_t)); + if (sessionId != 0 || + (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { + break; + } + } + } + if (!found) { + lStatus = BAD_VALUE; + LOGW("createEffect() effect not found"); + goto Exit; + } + // For same effect type, chose auxiliary version over insert version if + // connect to output mix (Compliance to OpenSL ES) + if (sessionId == 0 && + (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { + memcpy(&desc, &d, sizeof(effect_descriptor_t)); + } + } + + // Do not allow auxiliary effects on a session different from 0 (output mix) + if (sessionId != 0 && + (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { + lStatus = INVALID_OPERATION; + goto Exit; + } + + // Session -1 is reserved for output stage effects that can only be created + // by audio policy manager (running in same process) + if (sessionId == -1 && getpid() != IPCThreadState::self()->getCallingPid()) { + lStatus = INVALID_OPERATION; + goto Exit; + } + + // return effect descriptor + memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); + + // If output is not specified try to find a matching audio session ID in one of the + // output threads. + // TODO: allow attachment of effect to inputs + if (output == 0) { + if (sessionId <= 0) { + // default to first output + // TODO: define criteria to choose output when not specified. Or + // receive output from audio policy manager + if (mPlaybackThreads.size() != 0) { + output = mPlaybackThreads.keyAt(0); + } + } else { + // look for the thread where the specified audio session is present + for (size_t i = 0; i < mPlaybackThreads.size(); i++) { + if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId)) { + output = mPlaybackThreads.keyAt(i); + break; + } + } + } + } + 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); + } + + // create effect on selected output trhead + handle = thread->createEffect_l(client, effectClient, priority, sessionId, &desc, enabled, &lStatus); + if (handle != 0 && id != NULL) { + *id = handle->id(); + } + } + +Exit: + if(status) { + *status = lStatus; + } + return handle; +} + +status_t AudioFlinger::registerEffectResource_l(effect_descriptor_t *desc) { + if (mTotalEffectsCpuLoad + desc->cpuLoad > MAX_EFFECTS_CPU_LOAD) { + LOGW("registerEffectResource() CPU Load limit exceeded for Fx %s, CPU %f MIPS", + desc->name, (float)desc->cpuLoad/10); + return INVALID_OPERATION; + } + if (mTotalEffectsMemory + desc->memoryUsage > MAX_EFFECTS_MEMORY) { + LOGW("registerEffectResource() memory limit exceeded for Fx %s, Memory %d KB", + desc->name, desc->memoryUsage); + return INVALID_OPERATION; + } + mTotalEffectsCpuLoad += desc->cpuLoad; + mTotalEffectsMemory += desc->memoryUsage; + LOGV("registerEffectResource_l() effect %s, CPU %d, memory %d", + desc->name, desc->cpuLoad, desc->memoryUsage); + LOGV(" total CPU %d, total memory %d", mTotalEffectsCpuLoad, mTotalEffectsMemory); + return NO_ERROR; +} + +void AudioFlinger::unregisterEffectResource_l(effect_descriptor_t *desc) { + mTotalEffectsCpuLoad -= desc->cpuLoad; + mTotalEffectsMemory -= desc->memoryUsage; + LOGV("unregisterEffectResource_l() effect %s, CPU %d, memory %d", + desc->name, desc->cpuLoad, desc->memoryUsage); + LOGV(" total CPU %d, total memory %d", mTotalEffectsCpuLoad, mTotalEffectsMemory); +} + +// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held +sp<AudioFlinger::EffectHandle> AudioFlinger::PlaybackThread::createEffect_l( + const sp<AudioFlinger::Client>& client, + const sp<IEffectClient>& effectClient, + int32_t priority, + int sessionId, + effect_descriptor_t *desc, + int *enabled, + status_t *status + ) +{ + sp<EffectModule> effect; + sp<EffectHandle> handle; + status_t lStatus; + sp<Track> track; + sp<EffectChain> chain; + bool effectCreated = false; + bool effectRegistered = false; + + if (mOutput == 0) { + LOGW("createEffect_l() Audio driver not initialized."); + lStatus = NO_INIT; + goto Exit; + } + + // Do not allow auxiliary effect on session other than 0 + if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY && + sessionId != 0) { + LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", desc->name, sessionId); + lStatus = BAD_VALUE; + goto Exit; + } + + // Do not allow effects with session ID 0 on direct output or duplicating threads + // TODO: add rule for hw accelerated effects on direct outputs with non PCM format + if (sessionId == 0 && mType != MIXER) { + LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", desc->name, sessionId); + lStatus = BAD_VALUE; + goto Exit; + } + + LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); + + { // scope for mLock + Mutex::Autolock _l(mLock); + + // check for existing effect chain with the requested audio session + chain = getEffectChain_l(sessionId); + if (chain == 0) { + // create a new chain for this session + LOGV("createEffect_l() new effect chain for session %d", sessionId); + chain = new EffectChain(this, sessionId); + addEffectChain_l(chain); + } else { + effect = chain->getEffectFromDesc(desc); + } + + LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); + + if (effect == 0) { + // Check CPU and memory usage + lStatus = mAudioFlinger->registerEffectResource_l(desc); + if (lStatus != NO_ERROR) { + goto Exit; + } + effectRegistered = true; + // create a new effect module if none present in the chain + effect = new EffectModule(this, chain, desc, mAudioFlinger->nextUniqueId(), sessionId); + lStatus = effect->status(); + if (lStatus != NO_ERROR) { + goto Exit; + } + lStatus = chain->addEffect(effect); + if (lStatus != NO_ERROR) { + goto Exit; + } + effectCreated = true; + + effect->setDevice(mDevice); + effect->setMode(mAudioFlinger->getMode()); + } + // create effect handle and connect it to effect module + handle = new EffectHandle(effect, client, effectClient, priority); + lStatus = effect->addHandle(handle); + if (enabled) { + *enabled = (int)effect->isEnabled(); + } + } + +Exit: + if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { + if (effectCreated) { + if (chain->removeEffect(effect) == 0) { + removeEffectChain_l(chain); + } + } + if (effectRegistered) { + mAudioFlinger->unregisterEffectResource_l(desc); + } + handle.clear(); + } + + if(status) { + *status = lStatus; + } + return handle; +} + +void AudioFlinger::PlaybackThread::disconnectEffect(const sp< EffectModule>& effect, + const wp<EffectHandle>& handle) { + effect_descriptor_t desc = effect->desc(); + Mutex::Autolock _l(mLock); + // delete the effect module if removing last handle on it + if (effect->removeHandle(handle) == 0) { + if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { + detachAuxEffect_l(effect->id()); + } + sp<EffectChain> chain = effect->chain().promote(); + if (chain != 0) { + // remove effect chain if remove last effect + if (chain->removeEffect(effect) == 0) { + removeEffectChain_l(chain); + } + } + mLock.unlock(); + mAudioFlinger->mLock.lock(); + mAudioFlinger->unregisterEffectResource_l(&desc); + mAudioFlinger->mLock.unlock(); + } +} + +status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) +{ + int session = chain->sessionId(); + int16_t *buffer = mMixBuffer; + bool ownsBuffer = false; + + LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); + if (session > 0) { + // Only one effect chain can be present in direct output thread and it uses + // the mix buffer as input + if (mType != DIRECT) { + size_t numSamples = mFrameCount * mChannelCount; + buffer = new int16_t[numSamples]; + memset(buffer, 0, numSamples * sizeof(int16_t)); + LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); + ownsBuffer = true; + } + + // Attach all tracks with same session ID to this chain. + for (size_t i = 0; i < mTracks.size(); ++i) { + sp<Track> track = mTracks[i]; + if (session == track->sessionId()) { + LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); + track->setMainBuffer(buffer); + } + } + + // indicate all active tracks in the chain + for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { + sp<Track> track = mActiveTracks[i].promote(); + if (track == 0) continue; + if (session == track->sessionId()) { + LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); + chain->startTrack(); + } + } + } + + chain->setInBuffer(buffer, ownsBuffer); + chain->setOutBuffer(mMixBuffer); + // Effect chain for session -1 is inserted at end of effect chains list + // in order to be processed last as it contains output stage effects + // Effect chain for session 0 is inserted before session -1 to be processed + // after track specific effects and before output stage + // Effect chain for session other than 0 is inserted at beginning of effect + // chains list to be processed before output mix effects. Relative order between + // sessions other than 0 is not important + size_t size = mEffectChains.size(); + size_t i = 0; + for (i = 0; i < size; i++) { + if (mEffectChains[i]->sessionId() < session) break; + } + mEffectChains.insertAt(chain, i); + + return NO_ERROR; +} + +size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) +{ + int session = chain->sessionId(); + + LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); + + for (size_t i = 0; i < mEffectChains.size(); i++) { + if (chain == mEffectChains[i]) { + mEffectChains.removeAt(i); + // detach all tracks with same session ID from this chain + for (size_t i = 0; i < mTracks.size(); ++i) { + sp<Track> track = mTracks[i]; + if (session == track->sessionId()) { + track->setMainBuffer(mMixBuffer); + } + } + } + } + return mEffectChains.size(); +} + +void AudioFlinger::PlaybackThread::lockEffectChains_l() +{ + for (size_t i = 0; i < mEffectChains.size(); i++) { + mEffectChains[i]->lock(); + } +} + +void AudioFlinger::PlaybackThread::unlockEffectChains() +{ + Mutex::Autolock _l(mLock); + for (size_t i = 0; i < mEffectChains.size(); i++) { + mEffectChains[i]->unlock(); + } +} + +sp<AudioFlinger::EffectModule> AudioFlinger::PlaybackThread::getEffect_l(int sessionId, int effectId) +{ + sp<EffectModule> effect; + + sp<EffectChain> chain = getEffectChain_l(sessionId); + if (chain != 0) { + effect = chain->getEffectFromId(effectId); + } + return effect; +} + +status_t AudioFlinger::PlaybackThread::attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) +{ + Mutex::Autolock _l(mLock); + return attachAuxEffect_l(track, EffectId); +} + +status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) +{ + status_t status = NO_ERROR; + + if (EffectId == 0) { + track->setAuxBuffer(0, NULL); + } else { + // Auxiliary effects are always in audio session 0 + sp<EffectModule> effect = getEffect_l(0, EffectId); + if (effect != 0) { + if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { + track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); + } else { + status = INVALID_OPERATION; + } + } else { + status = BAD_VALUE; + } + } + return status; +} + +void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) +{ + for (size_t i = 0; i < mTracks.size(); ++i) { + sp<Track> track = mTracks[i]; + if (track->auxEffectId() == effectId) { + attachAuxEffect_l(track, 0); + } + } +} + +// ---------------------------------------------------------------------------- +// EffectModule implementation +// ---------------------------------------------------------------------------- + +#undef LOG_TAG +#define LOG_TAG "AudioFlinger::EffectModule" + +AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, + const wp<AudioFlinger::EffectChain>& chain, + effect_descriptor_t *desc, + int id, + int sessionId) + : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), + mStatus(NO_INIT), mState(IDLE) +{ + LOGV("Constructor %p", this); + int lStatus; + sp<ThreadBase> thread = mThread.promote(); + if (thread == 0) { + return; + } + PlaybackThread *p = (PlaybackThread *)thread.get(); + + memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); + + // create effect engine from effect factory + mStatus = EffectCreate(&desc->uuid, sessionId, p->id(), &mEffectInterface); + + if (mStatus != NO_ERROR) { + return; + } + lStatus = init(); + if (lStatus < 0) { + mStatus = lStatus; + goto Error; + } + + LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); + return; +Error: + EffectRelease(mEffectInterface); + mEffectInterface = NULL; + LOGV("Constructor Error %d", mStatus); +} + +AudioFlinger::EffectModule::~EffectModule() +{ + LOGV("Destructor %p", this); + if (mEffectInterface != NULL) { + // release effect engine + EffectRelease(mEffectInterface); + } +} + +status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) +{ + status_t status; + + Mutex::Autolock _l(mLock); + // First handle in mHandles has highest priority and controls the effect module + int priority = handle->priority(); + size_t size = mHandles.size(); + sp<EffectHandle> h; + size_t i; + for (i = 0; i < size; i++) { + h = mHandles[i].promote(); + if (h == 0) continue; + if (h->priority() <= priority) break; + } + // if inserted in first place, move effect control from previous owner to this handle + if (i == 0) { + if (h != 0) { + h->setControl(false, true); + } + handle->setControl(true, false); + status = NO_ERROR; + } else { + status = ALREADY_EXISTS; + } + mHandles.insertAt(handle, i); + return status; +} + +size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) +{ + Mutex::Autolock _l(mLock); + size_t size = mHandles.size(); + size_t i; + for (i = 0; i < size; i++) { + if (mHandles[i] == handle) break; + } + if (i == size) { + return size; + } + mHandles.removeAt(i); + size = mHandles.size(); + // if removed from first place, move effect control from this handle to next in line + if (i == 0 && size != 0) { + sp<EffectHandle> h = mHandles[0].promote(); + if (h != 0) { + h->setControl(true, true); + } + } + + return size; +} + +void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle) +{ + // keep a strong reference on this EffectModule to avoid calling the + // destructor before we exit + sp<EffectModule> keep(this); + { + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); + playbackThread->disconnectEffect(keep, handle); + } + } +} + +void AudioFlinger::EffectModule::updateState() { + Mutex::Autolock _l(mLock); + + switch (mState) { + case RESTART: + reset_l(); + // FALL THROUGH + + case STARTING: + // clear auxiliary effect input buffer for next accumulation + if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { + memset(mConfig.inputCfg.buffer.raw, + 0, + mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); + } + start_l(); + mState = ACTIVE; + break; + case STOPPING: + stop_l(); + mDisableWaitCnt = mMaxDisableWaitCnt; + mState = STOPPED; + break; + case STOPPED: + // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the + // turn off sequence. + if (--mDisableWaitCnt == 0) { + reset_l(); + mState = IDLE; + } + break; + default: //IDLE , ACTIVE + break; + } +} + +void AudioFlinger::EffectModule::process() +{ + Mutex::Autolock _l(mLock); + + if (mEffectInterface == NULL || + mConfig.inputCfg.buffer.raw == NULL || + mConfig.outputCfg.buffer.raw == NULL) { + return; + } + + if (mState == ACTIVE || mState == STOPPING || mState == STOPPED) { + // do 32 bit to 16 bit conversion for auxiliary effect input buffer + if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { + AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32, + mConfig.inputCfg.buffer.s32, + mConfig.inputCfg.buffer.frameCount); + } + + // do the actual processing in the effect engine + int ret = (*mEffectInterface)->process(mEffectInterface, + &mConfig.inputCfg.buffer, + &mConfig.outputCfg.buffer); + + // force transition to IDLE state when engine is ready + if (mState == STOPPED && ret == -ENODATA) { + mDisableWaitCnt = 1; + } + + // clear auxiliary effect input buffer for next accumulation + if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { + memset(mConfig.inputCfg.buffer.raw, 0, mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); + } + } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && + mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw){ + // If an insert effect is idle and input buffer is different from output buffer, copy input to + // output + sp<EffectChain> chain = mChain.promote(); + if (chain != 0 && chain->activeTracks() != 0) { + size_t size = mConfig.inputCfg.buffer.frameCount * sizeof(int16_t); + if (mConfig.inputCfg.channels == CHANNEL_STEREO) { + size *= 2; + } + memcpy(mConfig.outputCfg.buffer.raw, mConfig.inputCfg.buffer.raw, size); + } + } +} + +void AudioFlinger::EffectModule::reset_l() +{ + if (mEffectInterface == NULL) { + return; + } + (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); +} + +status_t AudioFlinger::EffectModule::configure() +{ + uint32_t channels; + if (mEffectInterface == NULL) { + return NO_INIT; + } + + sp<ThreadBase> thread = mThread.promote(); + if (thread == 0) { + return DEAD_OBJECT; + } + + // TODO: handle configuration of effects replacing track process + if (thread->channelCount() == 1) { + channels = CHANNEL_MONO; + } else { + channels = CHANNEL_STEREO; + } + + if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { + mConfig.inputCfg.channels = CHANNEL_MONO; + } else { + mConfig.inputCfg.channels = channels; + } + mConfig.outputCfg.channels = channels; + mConfig.inputCfg.format = SAMPLE_FORMAT_PCM_S15; + mConfig.outputCfg.format = SAMPLE_FORMAT_PCM_S15; + mConfig.inputCfg.samplingRate = thread->sampleRate(); + mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; + mConfig.inputCfg.bufferProvider.cookie = NULL; + mConfig.inputCfg.bufferProvider.getBuffer = NULL; + mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; + mConfig.outputCfg.bufferProvider.cookie = NULL; + mConfig.outputCfg.bufferProvider.getBuffer = NULL; + mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; + mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; + // Insert effect: + // - in session 0 or -1, always overwrites output buffer: input buffer == output buffer + // - in other sessions: + // last effect in the chain accumulates in output buffer: input buffer != output buffer + // other effect: overwrites output buffer: input buffer == output buffer + // Auxiliary effect: + // accumulates in output buffer: input buffer != output buffer + // Therefore: accumulate <=> input buffer != output buffer + if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { + mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; + } else { + mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; + } + mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; + mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; + mConfig.inputCfg.buffer.frameCount = thread->frameCount(); + mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; + + status_t cmdStatus; + int size = sizeof(int); + status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_CONFIGURE, sizeof(effect_config_t), &mConfig, &size, &cmdStatus); + if (status == 0) { + status = cmdStatus; + } + + mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / + (1000 * mConfig.outputCfg.buffer.frameCount); + + return status; +} + +status_t AudioFlinger::EffectModule::init() +{ + Mutex::Autolock _l(mLock); + if (mEffectInterface == NULL) { + return NO_INIT; + } + status_t cmdStatus; + int size = sizeof(status_t); + status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_INIT, 0, NULL, &size, &cmdStatus); + if (status == 0) { + status = cmdStatus; + } + return status; +} + +status_t AudioFlinger::EffectModule::start_l() +{ + if (mEffectInterface == NULL) { + return NO_INIT; + } + status_t cmdStatus; + int size = sizeof(status_t); + status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_ENABLE, 0, NULL, &size, &cmdStatus); + if (status == 0) { + status = cmdStatus; + } + return status; +} + +status_t AudioFlinger::EffectModule::stop_l() +{ + if (mEffectInterface == NULL) { + return NO_INIT; + } + status_t cmdStatus; + int size = sizeof(status_t); + status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_DISABLE, 0, NULL, &size, &cmdStatus); + if (status == 0) { + status = cmdStatus; + } + return status; +} + +status_t AudioFlinger::EffectModule::command(int cmdCode, int cmdSize, void *pCmdData, int *replySize, void *pReplyData) +{ + Mutex::Autolock _l(mLock); +// LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); + + if (mEffectInterface == NULL) { + return NO_INIT; + } + status_t status = (*mEffectInterface)->command(mEffectInterface, cmdCode, cmdSize, pCmdData, replySize, pReplyData); + if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { + int size = (replySize == NULL) ? 0 : *replySize; + for (size_t i = 1; i < mHandles.size(); i++) { + sp<EffectHandle> h = mHandles[i].promote(); + if (h != 0) { + h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); + } + } + } + return status; +} + +status_t AudioFlinger::EffectModule::setEnabled(bool enabled) +{ + Mutex::Autolock _l(mLock); + LOGV("setEnabled %p enabled %d", this, enabled); + + if (enabled != isEnabled()) { + switch (mState) { + // going from disabled to enabled + case IDLE: + mState = STARTING; + break; + case STOPPED: + mState = RESTART; + break; + case STOPPING: + mState = ACTIVE; + break; + + // going from enabled to disabled + case RESTART: + case STARTING: + mState = IDLE; + break; + case ACTIVE: + mState = STOPPING; + break; + } + for (size_t i = 1; i < mHandles.size(); i++) { + sp<EffectHandle> h = mHandles[i].promote(); + if (h != 0) { + h->setEnabled(enabled); + } + } + } + return NO_ERROR; +} + +bool AudioFlinger::EffectModule::isEnabled() +{ + switch (mState) { + case RESTART: + case STARTING: + case ACTIVE: + return true; + case IDLE: + case STOPPING: + case STOPPED: + default: + return false; + } +} + +status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) +{ + Mutex::Autolock _l(mLock); + status_t status = NO_ERROR; + + // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume + // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) + if ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) & (EFFECT_FLAG_VOLUME_CTRL|EFFECT_FLAG_VOLUME_IND)) { + status_t cmdStatus; + uint32_t volume[2]; + uint32_t *pVolume = NULL; + int size = sizeof(volume); + volume[0] = *left; + volume[1] = *right; + if (controller) { + pVolume = volume; + } + status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_VOLUME, size, volume, &size, pVolume); + if (controller && status == NO_ERROR && size == sizeof(volume)) { + *left = volume[0]; + *right = volume[1]; + } + } + return status; +} + +status_t AudioFlinger::EffectModule::setDevice(uint32_t device) +{ + Mutex::Autolock _l(mLock); + status_t status = NO_ERROR; + if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { + // convert device bit field from AudioSystem to EffectApi format. + device = deviceAudioSystemToEffectApi(device); + if (device == 0) { + return BAD_VALUE; + } + status_t cmdStatus; + int size = sizeof(status_t); + status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_DEVICE, sizeof(uint32_t), &device, &size, &cmdStatus); + if (status == NO_ERROR) { + status = cmdStatus; + } + } + return status; +} + +status_t AudioFlinger::EffectModule::setMode(uint32_t mode) +{ + Mutex::Autolock _l(mLock); + status_t status = NO_ERROR; + if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { + // convert audio mode from AudioSystem to EffectApi format. + int effectMode = modeAudioSystemToEffectApi(mode); + if (effectMode < 0) { + return BAD_VALUE; + } + status_t cmdStatus; + int size = sizeof(status_t); + status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_AUDIO_MODE, sizeof(int), &effectMode, &size, &cmdStatus); + if (status == NO_ERROR) { + status = cmdStatus; + } + } + return status; +} + +// update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified +const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = { + DEVICE_EARPIECE, // AudioSystem::DEVICE_OUT_EARPIECE + DEVICE_SPEAKER, // AudioSystem::DEVICE_OUT_SPEAKER + DEVICE_WIRED_HEADSET, // case AudioSystem::DEVICE_OUT_WIRED_HEADSET + DEVICE_WIRED_HEADPHONE, // AudioSystem::DEVICE_OUT_WIRED_HEADPHONE + DEVICE_BLUETOOTH_SCO, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO + DEVICE_BLUETOOTH_SCO_HEADSET, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET + DEVICE_BLUETOOTH_SCO_CARKIT, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT + DEVICE_BLUETOOTH_A2DP, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP + DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES + DEVICE_BLUETOOTH_A2DP_SPEAKER, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER + DEVICE_AUX_DIGITAL // AudioSystem::DEVICE_OUT_AUX_DIGITAL +}; + +uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device) +{ + uint32_t deviceOut = 0; + while (device) { + const uint32_t i = 31 - __builtin_clz(device); + device &= ~(1 << i); + if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) { + LOGE("device convertion error for AudioSystem device 0x%08x", device); + return 0; + } + deviceOut |= (uint32_t)sDeviceConvTable[i]; + } + return deviceOut; +} + +// update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified +const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = { + AUDIO_MODE_NORMAL, // AudioSystem::MODE_NORMAL + AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE + AUDIO_MODE_IN_CALL // AudioSystem::MODE_IN_CALL +}; + +int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode) +{ + int modeOut = -1; + if (mode < sizeof(sModeConvTable) / sizeof(uint32_t)) { + modeOut = (int)sModeConvTable[mode]; + } + return modeOut; +} + +status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); + result.append(buffer); + + bool locked = tryLock(mLock); + // failed to lock - AudioFlinger is probably deadlocked + if (!locked) { + result.append("\t\tCould not lock Fx mutex:\n"); + } + + result.append("\t\tSession Status State Engine:\n"); + snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", + mSessionId, mStatus, mState, (uint32_t)mEffectInterface); + result.append(buffer); + + result.append("\t\tDescriptor:\n"); + snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", + mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, + mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], + mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); + result.append(buffer); + snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", + mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, + mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], + mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); + result.append(buffer); + snprintf(buffer, SIZE, "\t\t- apiVersion: %04X\n\t\t- flags: %08X\n", + mDescriptor.apiVersion, + mDescriptor.flags); + result.append(buffer); + snprintf(buffer, SIZE, "\t\t- name: %s\n", + mDescriptor.name); + result.append(buffer); + snprintf(buffer, SIZE, "\t\t- implementor: %s\n", + mDescriptor.implementor); + result.append(buffer); + + result.append("\t\t- Input configuration:\n"); + result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); + snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", + (uint32_t)mConfig.inputCfg.buffer.raw, + mConfig.inputCfg.buffer.frameCount, + mConfig.inputCfg.samplingRate, + mConfig.inputCfg.channels, + mConfig.inputCfg.format); + result.append(buffer); + + result.append("\t\t- Output configuration:\n"); + result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); + snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", + (uint32_t)mConfig.outputCfg.buffer.raw, + mConfig.outputCfg.buffer.frameCount, + mConfig.outputCfg.samplingRate, + mConfig.outputCfg.channels, + mConfig.outputCfg.format); + result.append(buffer); + + snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); + result.append(buffer); + result.append("\t\t\tPid Priority Ctrl Locked client server\n"); + for (size_t i = 0; i < mHandles.size(); ++i) { + sp<EffectHandle> handle = mHandles[i].promote(); + if (handle != 0) { + handle->dump(buffer, SIZE); + result.append(buffer); + } + } + + result.append("\n"); + + write(fd, result.string(), result.length()); + + if (locked) { + mLock.unlock(); + } + + return NO_ERROR; +} + +// ---------------------------------------------------------------------------- +// EffectHandle implementation +// ---------------------------------------------------------------------------- + +#undef LOG_TAG +#define LOG_TAG "AudioFlinger::EffectHandle" + +AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, + const sp<AudioFlinger::Client>& client, + const sp<IEffectClient>& effectClient, + int32_t priority) + : BnEffect(), + mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false) +{ + LOGV("constructor %p", this); + + int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); + mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); + if (mCblkMemory != 0) { + mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); + + if (mCblk) { + new(mCblk) effect_param_cblk_t(); + mBuffer = (uint8_t *)mCblk + bufOffset; + } + } else { + LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); + return; + } +} + +AudioFlinger::EffectHandle::~EffectHandle() +{ + LOGV("Destructor %p", this); + disconnect(); +} + +status_t AudioFlinger::EffectHandle::enable() +{ + if (!mHasControl) return INVALID_OPERATION; + if (mEffect == 0) return DEAD_OBJECT; + + return mEffect->setEnabled(true); +} + +status_t AudioFlinger::EffectHandle::disable() +{ + if (!mHasControl) return INVALID_OPERATION; + if (mEffect == NULL) return DEAD_OBJECT; + + return mEffect->setEnabled(false); +} + +void AudioFlinger::EffectHandle::disconnect() +{ + if (mEffect == 0) { + return; + } + mEffect->disconnect(this); + // release sp on module => module destructor can be called now + mEffect.clear(); + if (mCblk) { + mCblk->~effect_param_cblk_t(); // destroy our shared-structure. + } + mCblkMemory.clear(); // and free the shared memory + if (mClient != 0) { + Mutex::Autolock _l(mClient->audioFlinger()->mLock); + mClient.clear(); + } +} + +status_t AudioFlinger::EffectHandle::command(int cmdCode, int cmdSize, void *pCmdData, int *replySize, void *pReplyData) +{ +// LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); + + // only get parameter command is permitted for applications not controlling the effect + if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { + return INVALID_OPERATION; + } + if (mEffect == 0) return DEAD_OBJECT; + + // handle commands that are not forwarded transparently to effect engine + if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { + // No need to trylock() here as this function is executed in the binder thread serving a particular client process: + // no risk to block the whole media server process or mixer threads is we are stuck here + Mutex::Autolock _l(mCblk->lock); + if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || + mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { + mCblk->serverIndex = 0; + mCblk->clientIndex = 0; + return BAD_VALUE; + } + status_t status = NO_ERROR; + while (mCblk->serverIndex < mCblk->clientIndex) { + int reply; + int rsize = sizeof(int); + int *p = (int *)(mBuffer + mCblk->serverIndex); + int size = *p++; + if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { + LOGW("command(): invalid parameter block size"); + break; + } + effect_param_t *param = (effect_param_t *)p; + if (param->psize == 0 || param->vsize == 0) { + LOGW("command(): null parameter or value size"); + mCblk->serverIndex += size; + continue; + } + int psize = sizeof(effect_param_t) + ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + param->vsize; + status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, psize, p, &rsize, &reply); + if (ret == NO_ERROR) { + if (reply != NO_ERROR) { + status = reply; + } + } else { + status = ret; + } + mCblk->serverIndex += size; + } + mCblk->serverIndex = 0; + mCblk->clientIndex = 0; + return status; + } else if (cmdCode == EFFECT_CMD_ENABLE) { + return enable(); + } else if (cmdCode == EFFECT_CMD_DISABLE) { + return disable(); + } + + return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); +} + +sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { + return mCblkMemory; +} + +void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal) +{ + LOGV("setControl %p control %d", this, hasControl); + + mHasControl = hasControl; + if (signal && mEffectClient != 0) { + mEffectClient->controlStatusChanged(hasControl); + } +} + +void AudioFlinger::EffectHandle::commandExecuted(int cmdCode, int cmdSize, void *pCmdData, int replySize, void *pReplyData) +{ + if (mEffectClient != 0) { + mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); + } +} + + + +void AudioFlinger::EffectHandle::setEnabled(bool enabled) +{ + if (mEffectClient != 0) { + mEffectClient->enableStatusChanged(enabled); + } +} + +status_t AudioFlinger::EffectHandle::onTransact( + uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) +{ + return BnEffect::onTransact(code, data, reply, flags); +} + + +void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) +{ + bool locked = tryLock(mCblk->lock); + + snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", + (mClient == NULL) ? getpid() : mClient->pid(), + mPriority, + mHasControl, + !locked, + mCblk->clientIndex, + mCblk->serverIndex + ); + + if (locked) { + mCblk->lock.unlock(); + } +} + +#undef LOG_TAG +#define LOG_TAG "AudioFlinger::EffectChain" + +AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, + int sessionId) + : mThread(wThread), mSessionId(sessionId), mVolumeCtrlIdx(-1), mActiveTrackCnt(0), mOwnInBuffer(false) +{ + +} + +AudioFlinger::EffectChain::~EffectChain() +{ + if (mOwnInBuffer) { + delete mInBuffer; + } + +} + +sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc(effect_descriptor_t *descriptor) +{ + sp<EffectModule> effect; + size_t size = mEffects.size(); + + for (size_t i = 0; i < size; i++) { + if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { + effect = mEffects[i]; + break; + } + } + return effect; +} + +sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId(int id) +{ + sp<EffectModule> effect; + size_t size = mEffects.size(); + + for (size_t i = 0; i < size; i++) { + if (mEffects[i]->id() == id) { + effect = mEffects[i]; + break; + } + } + return effect; +} + +// Must be called with EffectChain::mLock locked +void AudioFlinger::EffectChain::process_l() +{ + size_t size = mEffects.size(); + for (size_t i = 0; i < size; i++) { + mEffects[i]->process(); + } + for (size_t i = 0; i < size; i++) { + mEffects[i]->updateState(); + } + // if no track is active, input buffer must be cleared here as the mixer process + // will not do it + if (mSessionId > 0 && activeTracks() == 0) { + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + size_t numSamples = thread->frameCount() * thread->channelCount(); + memset(mInBuffer, 0, numSamples * sizeof(int16_t)); + } + } +} + +status_t AudioFlinger::EffectChain::addEffect(sp<EffectModule>& effect) +{ + effect_descriptor_t desc = effect->desc(); + uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; + + Mutex::Autolock _l(mLock); + + if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { + // Auxiliary effects are inserted at the beginning of mEffects vector as + // they are processed first and accumulated in chain input buffer + mEffects.insertAt(effect, 0); + sp<ThreadBase> thread = mThread.promote(); + if (thread == 0) { + return NO_INIT; + } + // the input buffer for auxiliary effect contains mono samples in + // 32 bit format. This is to avoid saturation in AudoMixer + // accumulation stage. Saturation is done in EffectModule::process() before + // calling the process in effect engine + size_t numSamples = thread->frameCount(); + int32_t *buffer = new int32_t[numSamples]; + memset(buffer, 0, numSamples * sizeof(int32_t)); + effect->setInBuffer((int16_t *)buffer); + // auxiliary effects output samples to chain input buffer for further processing + // by insert effects + effect->setOutBuffer(mInBuffer); + } else { + // Insert effects are inserted at the end of mEffects vector as they are processed + // after track and auxiliary effects. + // Insert effect order as a function of indicated preference: + // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if + // another effect is present + // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the + // last effect claiming first position + // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the + // first effect claiming last position + // else if EFFECT_FLAG_INSERT_ANY insert after first or before last + // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is + // already present + + int size = (int)mEffects.size(); + int idx_insert = size; + int idx_insert_first = -1; + int idx_insert_last = -1; + + for (int i = 0; i < size; i++) { + effect_descriptor_t d = mEffects[i]->desc(); + uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; + uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; + if (iMode == EFFECT_FLAG_TYPE_INSERT) { + // check invalid effect chaining combinations + if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || + iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { + LOGW("addEffect() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); + return INVALID_OPERATION; + } + // remember position of first insert effect and by default + // select this as insert position for new effect + if (idx_insert == size) { + idx_insert = i; + } + // remember position of last insert effect claiming + // first position + if (iPref == EFFECT_FLAG_INSERT_FIRST) { + idx_insert_first = i; + } + // remember position of first insert effect claiming + // last position + if (iPref == EFFECT_FLAG_INSERT_LAST && + idx_insert_last == -1) { + idx_insert_last = i; + } + } + } + + // modify idx_insert from first position if needed + if (insertPref == EFFECT_FLAG_INSERT_LAST) { + if (idx_insert_last != -1) { + idx_insert = idx_insert_last; + } else { + idx_insert = size; + } + } else { + if (idx_insert_first != -1) { + idx_insert = idx_insert_first + 1; + } + } + + // always read samples from chain input buffer + effect->setInBuffer(mInBuffer); + + // if last effect in the chain, output samples to chain + // output buffer, otherwise to chain input buffer + if (idx_insert == size) { + if (idx_insert != 0) { + mEffects[idx_insert-1]->setOutBuffer(mInBuffer); + mEffects[idx_insert-1]->configure(); + } + effect->setOutBuffer(mOutBuffer); + } else { + effect->setOutBuffer(mInBuffer); + } + mEffects.insertAt(effect, idx_insert); + // Always give volume control to last effect in chain with volume control capability + if (((desc.flags & EFFECT_FLAG_VOLUME_MASK) & EFFECT_FLAG_VOLUME_CTRL) && + mVolumeCtrlIdx < idx_insert) { + mVolumeCtrlIdx = idx_insert; + } + + LOGV("addEffect() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); + } + effect->configure(); + return NO_ERROR; +} + +size_t AudioFlinger::EffectChain::removeEffect(const sp<EffectModule>& effect) +{ + Mutex::Autolock _l(mLock); + + int size = (int)mEffects.size(); + int i; + uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; + + for (i = 0; i < size; i++) { + if (effect == mEffects[i]) { + if (type == EFFECT_FLAG_TYPE_AUXILIARY) { + delete[] effect->inBuffer(); + } else { + if (i == size - 1 && i != 0) { + mEffects[i - 1]->setOutBuffer(mOutBuffer); + mEffects[i - 1]->configure(); + } + } + mEffects.removeAt(i); + LOGV("removeEffect() effect %p, removed from chain %p at rank %d", effect.get(), this, i); + break; + } + } + // Return volume control to last effect in chain with volume control capability + if (mVolumeCtrlIdx == i) { + size = (int)mEffects.size(); + for (i = size; i > 0; i--) { + if ((mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) & EFFECT_FLAG_VOLUME_CTRL) { + break; + } + } + // mVolumeCtrlIdx reset to -1 if no effect found with volume control flag set + mVolumeCtrlIdx = i - 1; + } + + return mEffects.size(); +} + +void AudioFlinger::EffectChain::setDevice(uint32_t device) +{ + size_t size = mEffects.size(); + for (size_t i = 0; i < size; i++) { + mEffects[i]->setDevice(device); + } +} + +void AudioFlinger::EffectChain::setMode(uint32_t mode) +{ + size_t size = mEffects.size(); + for (size_t i = 0; i < size; i++) { + mEffects[i]->setMode(mode); + } +} + +bool AudioFlinger::EffectChain::setVolume(uint32_t *left, uint32_t *right) +{ + uint32_t newLeft = *left; + uint32_t newRight = *right; + bool hasControl = false; + + // first get volume update from volume controller + if (mVolumeCtrlIdx >= 0) { + mEffects[mVolumeCtrlIdx]->setVolume(&newLeft, &newRight, true); + hasControl = true; + } + // then indicate volume to all other effects in chain. + // Pass altered volume to effects before volume controller + // and requested volume to effects after controller + uint32_t lVol = newLeft; + uint32_t rVol = newRight; + size_t size = mEffects.size(); + for (size_t i = 0; i < size; i++) { + if ((int)i == mVolumeCtrlIdx) continue; + // this also works for mVolumeCtrlIdx == -1 when there is no volume controller + if ((int)i > mVolumeCtrlIdx) { + lVol = *left; + rVol = *right; + } + mEffects[i]->setVolume(&lVol, &rVol, false); + } + *left = newLeft; + *right = newRight; + + return hasControl; +} + +sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getVolumeController() +{ + sp<EffectModule> effect; + if (mVolumeCtrlIdx >= 0) { + effect = mEffects[mVolumeCtrlIdx]; + } + return effect; +} + + +status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); + result.append(buffer); + + bool locked = tryLock(mLock); + // failed to lock - AudioFlinger is probably deadlocked + if (!locked) { + result.append("\tCould not lock mutex:\n"); + } + + result.append("\tNum fx In buffer Out buffer Vol ctrl Active tracks:\n"); + snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %02d %d\n", + mEffects.size(), + (uint32_t)mInBuffer, + (uint32_t)mOutBuffer, + (mVolumeCtrlIdx == -1) ? 0 : mEffects[mVolumeCtrlIdx]->id(), + mActiveTrackCnt); + result.append(buffer); + write(fd, result.string(), result.size()); + + for (size_t i = 0; i < mEffects.size(); ++i) { + sp<EffectModule> effect = mEffects[i]; + if (effect != 0) { + effect->dump(fd, args); + } + } + + if (locked) { + mLock.unlock(); + } + + return NO_ERROR; +} + +#undef LOG_TAG +#define LOG_TAG "AudioFlinger" + +// ---------------------------------------------------------------------------- + +status_t AudioFlinger::onTransact( + uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) +{ + return BnAudioFlinger::onTransact(code, data, reply, flags); +} + +// ---------------------------------------------------------------------------- + +void AudioFlinger::instantiate() { + defaultServiceManager()->addService( + String16("media.audio_flinger"), new AudioFlinger()); +} + +}; // namespace android |