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Diffstat (limited to 'services/audioflinger/AudioFlinger.cpp')
| -rw-r--r-- | services/audioflinger/AudioFlinger.cpp | 4055 |
1 files changed, 4055 insertions, 0 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp new file mode 100644 index 0000000..2414e8d --- /dev/null +++ b/services/audioflinger/AudioFlinger.cpp @@ -0,0 +1,4055 @@ +/* //device/include/server/AudioFlinger/AudioFlinger.cpp +** +** Copyright 2007, The Android Open Source Project +** +** Licensed under the Apache License, Version 2.0 (the "License"); +** you may not use this file except in compliance with the License. +** You may obtain a copy of the License at +** +** http://www.apache.org/licenses/LICENSE-2.0 +** +** Unless required by applicable law or agreed to in writing, software +** distributed under the License is distributed on an "AS IS" BASIS, +** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +** See the License for the specific language governing permissions and +** limitations under the License. +*/ + + +#define LOG_TAG "AudioFlinger" +//#define LOG_NDEBUG 0 + +#include <math.h> +#include <signal.h> +#include <sys/time.h> +#include <sys/resource.h> + +#include <binder/IServiceManager.h> +#include <utils/Log.h> +#include <binder/Parcel.h> +#include <binder/IPCThreadState.h> +#include <utils/String16.h> +#include <utils/threads.h> + +#include <cutils/properties.h> + +#include <media/AudioTrack.h> +#include <media/AudioRecord.h> + +#include <private/media/AudioTrackShared.h> + +#include <hardware_legacy/AudioHardwareInterface.h> + +#include "AudioMixer.h" +#include "AudioFlinger.h" + +#ifdef WITH_A2DP +#include "A2dpAudioInterface.h" +#endif + +#ifdef LVMX +#include "lifevibes.h" +#endif + +// ---------------------------------------------------------------------------- +// the sim build doesn't have gettid + +#ifndef HAVE_GETTID +# define gettid getpid +#endif + +// ---------------------------------------------------------------------------- + +namespace android { + +static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n"; +static const char* kHardwareLockedString = "Hardware lock is taken\n"; + +//static const nsecs_t kStandbyTimeInNsecs = seconds(3); +static const float MAX_GAIN = 4096.0f; + +// retry counts for buffer fill timeout +// 50 * ~20msecs = 1 second +static const int8_t kMaxTrackRetries = 50; +static const int8_t kMaxTrackStartupRetries = 50; +// allow less retry attempts on direct output thread. +// direct outputs can be a scarce resource in audio hardware and should +// be released as quickly as possible. +static const int8_t kMaxTrackRetriesDirect = 2; + +static const int kDumpLockRetries = 50; +static const int kDumpLockSleep = 20000; + +static const nsecs_t kWarningThrottle = seconds(5); + + +#define AUDIOFLINGER_SECURITY_ENABLED 1 + +// ---------------------------------------------------------------------------- + +static bool recordingAllowed() { +#ifndef HAVE_ANDROID_OS + return true; +#endif +#if AUDIOFLINGER_SECURITY_ENABLED + if (getpid() == IPCThreadState::self()->getCallingPid()) return true; + bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); + if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO"); + return ok; +#else + if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO"))) + LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest"); + return true; +#endif +} + +static bool settingsAllowed() { +#ifndef HAVE_ANDROID_OS + return true; +#endif +#if AUDIOFLINGER_SECURITY_ENABLED + if (getpid() == IPCThreadState::self()->getCallingPid()) return true; + bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); + if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); + return ok; +#else + if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"))) + LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest"); + return true; +#endif +} + +// ---------------------------------------------------------------------------- + +AudioFlinger::AudioFlinger() + : BnAudioFlinger(), + mAudioHardware(0), mMasterVolume(1.0f), mMasterMute(false), mNextThreadId(0) +{ + mHardwareStatus = AUDIO_HW_IDLE; + + mAudioHardware = AudioHardwareInterface::create(); + + mHardwareStatus = AUDIO_HW_INIT; + if (mAudioHardware->initCheck() == NO_ERROR) { + // open 16-bit output stream for s/w mixer + + setMode(AudioSystem::MODE_NORMAL); + + setMasterVolume(1.0f); + setMasterMute(false); + } else { + LOGE("Couldn't even initialize the stubbed audio hardware!"); + } +#ifdef LVMX + LifeVibes::init(); +#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, + status_t *status) +{ + sp<PlaybackThread::Track> track; + sp<TrackHandle> trackHandle; + sp<Client> client; + wp<Client> wclient; + status_t lStatus; + + if (streamType >= AudioSystem::NUM_STREAM_TYPES) { + LOGE("invalid stream type"); + lStatus = BAD_VALUE; + goto Exit; + } + + { + Mutex::Autolock _l(mLock); + PlaybackThread *thread = checkPlaybackThread_l(output); + if (thread == NULL) { + LOGE("unknown output thread"); + lStatus = BAD_VALUE; + goto Exit; + } + + wclient = mClients.valueFor(pid); + + if (wclient != NULL) { + client = wclient.promote(); + } else { + client = new Client(this, pid); + mClients.add(pid, client); + } + track = thread->createTrack_l(client, streamType, sampleRate, format, + channelCount, frameCount, sharedBuffer, &lStatus); + } + if (lStatus == NO_ERROR) { + trackHandle = new TrackHandle(track); + } else { + // remove local strong reference to Client before deleting the Track so that the Client + // destructor is called by the TrackBase destructor with mLock held + client.clear(); + track.clear(); + } + +Exit: + if(status) { + *status = lStatus; + } + return trackHandle; +} + +uint32_t AudioFlinger::sampleRate(int output) const +{ + Mutex::Autolock _l(mLock); + PlaybackThread *thread = checkPlaybackThread_l(output); + if (thread == NULL) { + LOGW("sampleRate() unknown thread %d", output); + return 0; + } + return thread->sampleRate(); +} + +int AudioFlinger::channelCount(int output) const +{ + Mutex::Autolock _l(mLock); + PlaybackThread *thread = checkPlaybackThread_l(output); + if (thread == NULL) { + LOGW("channelCount() unknown thread %d", output); + return 0; + } + return thread->channelCount(); +} + +int AudioFlinger::format(int output) const +{ + Mutex::Autolock _l(mLock); + PlaybackThread *thread = checkPlaybackThread_l(output); + if (thread == NULL) { + LOGW("format() unknown thread %d", output); + return 0; + } + return thread->format(); +} + +size_t AudioFlinger::frameCount(int output) const +{ + Mutex::Autolock _l(mLock); + PlaybackThread *thread = checkPlaybackThread_l(output); + if (thread == NULL) { + LOGW("frameCount() unknown thread %d", output); + return 0; + } + return thread->frameCount(); +} + +uint32_t AudioFlinger::latency(int output) const +{ + Mutex::Autolock _l(mLock); + PlaybackThread *thread = checkPlaybackThread_l(output); + if (thread == NULL) { + LOGW("latency() unknown thread %d", output); + return 0; + } + return thread->latency(); +} + +status_t AudioFlinger::setMasterVolume(float value) +{ + // check calling permissions + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + + // when hw supports master volume, don't scale in sw mixer + AutoMutex lock(mHardwareLock); + mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; + if (mAudioHardware->setMasterVolume(value) == NO_ERROR) { + value = 1.0f; + } + mHardwareStatus = AUDIO_HW_IDLE; + + mMasterVolume = value; + for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) + mPlaybackThreads.valueAt(i)->setMasterVolume(value); + + return NO_ERROR; +} + +status_t AudioFlinger::setMode(int mode) +{ + // check calling permissions + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) { + LOGW("Illegal value: setMode(%d)", mode); + return BAD_VALUE; + } + + AutoMutex lock(mHardwareLock); + mHardwareStatus = AUDIO_HW_SET_MODE; + status_t ret = mAudioHardware->setMode(mode); +#ifdef LVMX + if (NO_ERROR == ret) { + LifeVibes::setMode(mode); + } +#endif + mHardwareStatus = AUDIO_HW_IDLE; + return ret; +} + +status_t AudioFlinger::setMicMute(bool state) +{ + // check calling permissions + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + + AutoMutex lock(mHardwareLock); + mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; + status_t ret = mAudioHardware->setMicMute(state); + mHardwareStatus = AUDIO_HW_IDLE; + return ret; +} + +bool AudioFlinger::getMicMute() const +{ + bool state = AudioSystem::MODE_INVALID; + mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; + mAudioHardware->getMicMute(&state); + mHardwareStatus = AUDIO_HW_IDLE; + return state; +} + +status_t AudioFlinger::setMasterMute(bool muted) +{ + // check calling permissions + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + + mMasterMute = muted; + for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) + mPlaybackThreads.valueAt(i)->setMasterMute(muted); + + return NO_ERROR; +} + +float AudioFlinger::masterVolume() const +{ + return mMasterVolume; +} + +bool AudioFlinger::masterMute() const +{ + return mMasterMute; +} + +status_t AudioFlinger::setStreamVolume(int stream, float value, int output) +{ + // check calling permissions + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + + if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { + return BAD_VALUE; + } + + AutoMutex lock(mLock); + PlaybackThread *thread = NULL; + if (output) { + thread = checkPlaybackThread_l(output); + if (thread == NULL) { + return BAD_VALUE; + } + } + + mStreamTypes[stream].volume = value; + + if (thread == NULL) { + for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { + mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); + } + } else { + thread->setStreamVolume(stream, value); + } + + return NO_ERROR; +} + +status_t AudioFlinger::setStreamMute(int stream, bool muted) +{ + // check calling permissions + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + + if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES || + uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) { + return BAD_VALUE; + } + + mStreamTypes[stream].mute = muted; + for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) + mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); + + return NO_ERROR; +} + +float AudioFlinger::streamVolume(int stream, int output) const +{ + if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) { + return 0.0f; + } + + AutoMutex lock(mLock); + float volume; + if (output) { + PlaybackThread *thread = checkPlaybackThread_l(output); + if (thread == NULL) { + return 0.0f; + } + volume = thread->streamVolume(stream); + } else { + volume = mStreamTypes[stream].volume; + } + + return volume; +} + +bool AudioFlinger::streamMute(int stream) const +{ + if (stream < 0 || stream >= (int)AudioSystem::NUM_STREAM_TYPES) { + return true; + } + + return mStreamTypes[stream].mute; +} + +bool AudioFlinger::isStreamActive(int stream) const +{ + Mutex::Autolock _l(mLock); + for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { + if (mPlaybackThreads.valueAt(i)->isStreamActive(stream)) { + return true; + } + } + return false; +} + +status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) +{ + status_t result; + + LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", + ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); + // check calling permissions + if (!settingsAllowed()) { + return PERMISSION_DENIED; + } + +#ifdef LVMX + AudioParameter param = AudioParameter(keyValuePairs); + LifeVibes::setParameters(ioHandle,keyValuePairs); + String8 key = String8(AudioParameter::keyRouting); + int device; + if (NO_ERROR != param.getInt(key, device)) { + device = -1; + } + + key = String8(LifevibesTag); + String8 value; + int musicEnabled = -1; + if (NO_ERROR == param.get(key, value)) { + if (value == LifevibesEnable) { + musicEnabled = 1; + } else if (value == LifevibesDisable) { + 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 ((NO_ERROR == result) && (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) +{ + + LOGV("registerClient() %p, tid %d, calling tid %d", client.get(), gettid(), IPCThreadState::self()->getCallingPid()); + Mutex::Autolock _l(mLock); + + sp<IBinder> binder = client->asBinder(); + if (mNotificationClients.indexOf(binder) < 0) { + LOGV("Adding notification client %p", binder.get()); + binder->linkToDeath(this); + mNotificationClients.add(binder); + } + + // the config change is always sent from playback or record threads to avoid deadlock + // with AudioSystem::gLock + for (size_t i = 0; i < mPlaybackThreads.size(); i++) { + mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); + } + + for (size_t i = 0; i < mRecordThreads.size(); i++) { + mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); + } +} + +void AudioFlinger::binderDied(const wp<IBinder>& who) { + + LOGV("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(), IPCThreadState::self()->getCallingPid()); + Mutex::Autolock _l(mLock); + + IBinder *binder = who.unsafe_get(); + + if (binder != NULL) { + int index = mNotificationClients.indexOf(binder); + if (index >= 0) { + LOGV("Removing notification client %p", binder); + mNotificationClients.removeAt(index); + } + } +} + +// 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++) { + sp<IBinder> binder = mNotificationClients.itemAt(i); + LOGV("audioConfigChanged_l() Notifying change to client %p", binder.get()); + sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient> (binder); + 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), + mFormat(0), mFrameSize(1), 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 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 because audioConfigChanged() will lock AudioFlinger mLock + // before calling Audioflinger::audioConfigChanged_l() thus creating + // potential cross deadlock between AudioFlinger::mLock and mLock + mLock.unlock(); + audioConfigChanged(configEvent->mEvent, configEvent->mParam); + 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) + : ThreadBase(audioFlinger, id), + mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output), + mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) +{ + readOutputParameters(); + + mMasterVolume = mAudioFlinger->masterVolume(); + mMasterMute = mAudioFlinger->masterMute(); + + for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) { + mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); + mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); + } + // notify client processes that a new input has been opened + sendConfigEvent(AudioSystem::OUTPUT_OPENED); +} + +AudioFlinger::PlaybackThread::~PlaybackThread() +{ + delete [] mMixBuffer; +} + +status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) +{ + dumpInternals(fd, args); + dumpTracks(fd, args); + return NO_ERROR; +} + +status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "Output thread %p tracks\n", this); + result.append(buffer); + result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n"); + for (size_t i = 0; i < mTracks.size(); ++i) { + sp<Track> track = mTracks[i]; + if (track != 0) { + track->dump(buffer, SIZE); + result.append(buffer); + } + } + + snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); + result.append(buffer); + result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n"); + for (size_t i = 0; i < mActiveTracks.size(); ++i) { + wp<Track> wTrack = mActiveTracks[i]; + if (wTrack != 0) { + sp<Track> track = wTrack.promote(); + if (track != 0) { + track->dump(buffer, SIZE); + result.append(buffer); + } + } + } + write(fd, result.string(), result.size()); + return NO_ERROR; +} + +status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); + result.append(buffer); + snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); + result.append(buffer); + snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); + result.append(buffer); + snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); + result.append(buffer); + snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); + result.append(buffer); + snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); + result.append(buffer); + write(fd, result.string(), result.size()); + + dumpBase(fd, args); + + return NO_ERROR; +} + +// Thread virtuals +status_t AudioFlinger::PlaybackThread::readyToRun() +{ + if (mSampleRate == 0) { + LOGE("No working audio driver found."); + return NO_INIT; + } + LOGI("AudioFlinger's thread %p ready to run", this); + return NO_ERROR; +} + +void AudioFlinger::PlaybackThread::onFirstRef() +{ + const size_t SIZE = 256; + char buffer[SIZE]; + + snprintf(buffer, SIZE, "Playback Thread %p", this); + + run(buffer, ANDROID_PRIORITY_URGENT_AUDIO); +} + +// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held +sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( + const sp<AudioFlinger::Client>& client, + int streamType, + uint32_t sampleRate, + int format, + int channelCount, + int frameCount, + const sp<IMemory>& sharedBuffer, + status_t *status) +{ + sp<Track> track; + status_t lStatus; + + if (mType == DIRECT) { + if (sampleRate != mSampleRate || format != mFormat || channelCount != mChannelCount) { + LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p", + sampleRate, format, channelCount, mOutput); + lStatus = BAD_VALUE; + goto Exit; + } + } else { + // Resampler implementation limits input sampling rate to 2 x output sampling rate. + if (sampleRate > mSampleRate*2) { + LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); + lStatus = BAD_VALUE; + goto Exit; + } + } + + if (mOutput == 0) { + LOGE("Audio driver not initialized."); + lStatus = NO_INIT; + goto Exit; + } + + { // scope for mLock + Mutex::Autolock _l(mLock); + track = new Track(this, client, streamType, sampleRate, format, + channelCount, frameCount, sharedBuffer); + if (track->getCblk() == NULL || track->name() < 0) { + lStatus = NO_MEMORY; + goto Exit; + } + mTracks.add(track); + } + lStatus = NO_ERROR; + +Exit: + if(status) { + *status = lStatus; + } + return track; +} + +uint32_t AudioFlinger::PlaybackThread::latency() const +{ + if (mOutput) { + return mOutput->latency(); + } + else { + return 0; + } +} + +status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) +{ +#ifdef LVMX + int audioOutputType = LifeVibes::getMixerType(mId, mType); + if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { + LifeVibes::setMasterVolume(audioOutputType, value); + } +#endif + mMasterVolume = value; + return NO_ERROR; +} + +status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) +{ +#ifdef LVMX + int audioOutputType = LifeVibes::getMixerType(mId, mType); + if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { + LifeVibes::setMasterMute(audioOutputType, muted); + } +#endif + mMasterMute = muted; + return NO_ERROR; +} + +float AudioFlinger::PlaybackThread::masterVolume() const +{ + return mMasterVolume; +} + +bool AudioFlinger::PlaybackThread::masterMute() const +{ + return mMasterMute; +} + +status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) +{ +#ifdef LVMX + int audioOutputType = LifeVibes::getMixerType(mId, mType); + if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { + LifeVibes::setStreamVolume(audioOutputType, stream, value); + } +#endif + mStreamTypes[stream].volume = value; + return NO_ERROR; +} + +status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) +{ +#ifdef LVMX + int audioOutputType = LifeVibes::getMixerType(mId, mType); + if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { + LifeVibes::setStreamMute(audioOutputType, stream, muted); + } +#endif + mStreamTypes[stream].mute = muted; + return NO_ERROR; +} + +float AudioFlinger::PlaybackThread::streamVolume(int stream) const +{ + return mStreamTypes[stream].volume; +} + +bool AudioFlinger::PlaybackThread::streamMute(int stream) const +{ + return mStreamTypes[stream].mute; +} + +bool AudioFlinger::PlaybackThread::isStreamActive(int stream) const +{ + Mutex::Autolock _l(mLock); + size_t count = mActiveTracks.size(); + for (size_t i = 0 ; i < count ; ++i) { + sp<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); + 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); +} + +void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { + AudioSystem::OutputDescriptor desc; + void *param2 = 0; + + LOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, param); + + switch (event) { + case AudioSystem::OUTPUT_OPENED: + case AudioSystem::OUTPUT_CONFIG_CHANGED: + desc.channels = mChannelCount; + desc.samplingRate = mSampleRate; + desc.format = mFormat; + desc.frameCount = mFrameCount; + desc.latency = latency(); + param2 = &desc; + break; + + case AudioSystem::STREAM_CONFIG_CHANGED: + param2 = ¶m; + case AudioSystem::OUTPUT_CLOSED: + default: + break; + } + Mutex::Autolock _l(mAudioFlinger->mLock); + mAudioFlinger->audioConfigChanged_l(event, mId, param2); +} + +void AudioFlinger::PlaybackThread::readOutputParameters() +{ + mSampleRate = mOutput->sampleRate(); + mChannelCount = AudioSystem::popCount(mOutput->channels()); + + mFormat = mOutput->format(); + mFrameSize = mOutput->frameSize(); + mFrameCount = mOutput->bufferSize() / mFrameSize; + + // FIXME - Current mixer implementation only supports stereo output: Always + // Allocate a stereo buffer even if HW output is mono. + if (mMixBuffer != NULL) delete mMixBuffer; + mMixBuffer = new int16_t[mFrameCount * 2]; + memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); +} + +status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) +{ + if (halFrames == 0 || dspFrames == 0) { + return BAD_VALUE; + } + if (mOutput == 0) { + return INVALID_OPERATION; + } + *halFrames = mBytesWritten/mOutput->frameSize(); + + return mOutput->getRenderPosition(dspFrames); +} + +// ---------------------------------------------------------------------------- + +AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id) + : PlaybackThread(audioFlinger, output, id), + mAudioMixer(0) +{ + mType = PlaybackThread::MIXER; + mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); + + // FIXME - Current mixer implementation only supports stereo output + if (mChannelCount == 1) { + LOGE("Invalid audio hardware channel count"); + } +} + +AudioFlinger::MixerThread::~MixerThread() +{ + delete mAudioMixer; +} + +bool AudioFlinger::MixerThread::threadLoop() +{ + int16_t* curBuf = mMixBuffer; + Vector< sp<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; + + 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); + } + + if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { + // mix buffers... + mAudioMixer->process(curBuf); + sleepTime = 0; + standbyTime = systemTime() + kStandbyTimeInNsecs; + } else { + // If no tracks are ready, sleep once for the duration of an output + // buffer size, then write 0s to the output + if (sleepTime == 0) { + if (mixerStatus == MIXER_TRACKS_ENABLED) { + sleepTime = activeSleepTime; + } else { + sleepTime = idleSleepTime; + } + } else if (mBytesWritten != 0 || + (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { + memset (curBuf, 0, mixBufferSize); + sleepTime = 0; + LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); + } + } + + if (mSuspended) { + sleepTime = idleSleepTime; + } + // sleepTime == 0 means we must write to audio hardware + if (sleepTime == 0) { + mLastWriteTime = systemTime(); + mInWrite = true; + mBytesWritten += mixBufferSize; +#ifdef LVMX + int audioOutputType = LifeVibes::getMixerType(mId, mType); + if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) { + LifeVibes::process(audioOutputType, curBuf, mixBufferSize); + } +#endif + int bytesWritten = (int)mOutput->write(curBuf, mixBufferSize); + if (bytesWritten < 0) mBytesWritten -= mixBufferSize; + mNumWrites++; + mInWrite = false; + nsecs_t now = systemTime(); + nsecs_t delta = now - mLastWriteTime; + if (delta > maxPeriod) { + mNumDelayedWrites++; + if ((now - lastWarning) > kWarningThrottle) { + LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", + ns2ms(delta), mNumDelayedWrites, this); + lastWarning = now; + } + if (mStandby) { + longStandbyExit = true; + } + } + mStandby = false; + } else { + usleep(sleepTime); + } + + // finally let go of all our tracks, without the lock held + // since we can't guarantee the destructors won't acquire that + // same lock. + tracksToRemove.clear(); + } + + if (!mStandby) { + mOutput->standby(); + } + + LOGV("MixerThread %p exiting", this); + return false; +} + +// prepareTracks_l() must be called with ThreadBase::mLock held +uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) +{ + + uint32_t mixerStatus = MIXER_IDLE; + // find out which tracks need to be processed + size_t count = activeTracks.size(); + + float masterVolume = mMasterVolume; + bool masterMute = mMasterMute; + +#ifdef LVMX + bool tracksConnectedChanged = false; + bool stateChanged = false; + + int audioOutputType = LifeVibes::getMixerType(mId, mType); + if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) + { + int activeTypes = 0; + for (size_t i=0 ; i<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 + + 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); + + // compute volume for this track + int16_t left, right; + if (track->isMuted() || masterMute || track->isPausing() || + mStreamTypes[track->type()].mute) { + left = right = 0; + if (track->isPausing()) { + track->setPaused(); + } + } else { + // read original volumes with volume control + float typeVolume = mStreamTypes[track->type()].volume; +#ifdef LVMX + bool streamMute=false; + // read the volume from the LivesVibes audio engine. + if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) + { + LifeVibes::getStreamVolumes(audioOutputType, track->type(), &typeVolume, &streamMute); + if (streamMute) { + typeVolume = 0; + } + } +#endif + float v = masterVolume * typeVolume; + float v_clamped = v * cblk->volume[0]; + if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; + left = int16_t(v_clamped); + v_clamped = v * cblk->volume[1]; + if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; + right = int16_t(v_clamped); + } + + // XXX: these things DON'T need to be done each time + mAudioMixer->setBufferProvider(track); + mAudioMixer->enable(AudioMixer::MIXING); + + int param = AudioMixer::VOLUME; + if (track->mFillingUpStatus == Track::FS_FILLED) { + // no ramp for the first volume setting + track->mFillingUpStatus = Track::FS_ACTIVE; + if (track->mState == TrackBase::RESUMING) { + track->mState = TrackBase::ACTIVE; + param = AudioMixer::RAMP_VOLUME; + } + } else if (cblk->server != 0) { + // If the track is stopped before the first frame was mixed, + // do not apply ramp + param = AudioMixer::RAMP_VOLUME; + } +#ifdef LVMX + if ( tracksConnectedChanged || stateChanged ) + { + // only do the ramp when the volume is changed by the user / application + param = AudioMixer::VOLUME; + } +#endif + mAudioMixer->setParameter(param, AudioMixer::VOLUME0, left); + mAudioMixer->setParameter(param, AudioMixer::VOLUME1, right); + mAudioMixer->setParameter( + AudioMixer::TRACK, + AudioMixer::FORMAT, track->format()); + mAudioMixer->setParameter( + AudioMixer::TRACK, + AudioMixer::CHANNEL_COUNT, track->channelCount()); + mAudioMixer->setParameter( + AudioMixer::RESAMPLE, + AudioMixer::SAMPLE_RATE, + int(cblk->sampleRate)); + + // reset retry count + track->mRetryCount = kMaxTrackRetries; + mixerStatus = MIXER_TRACKS_READY; + } else { + //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); + if (track->isStopped()) { + track->reset(); + } + if (track->isTerminated() || track->isStopped() || track->isPaused()) { + // We have consumed all the buffers of this track. + // Remove it from the list of active tracks. + tracksToRemove->add(track); + mAudioMixer->disable(AudioMixer::MIXING); + } else { + // No buffers for this track. Give it a few chances to + // fill a buffer, then remove it from active list. + if (--(track->mRetryCount) <= 0) { + LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); + tracksToRemove->add(track); + } else if (mixerStatus != MIXER_TRACKS_READY) { + mixerStatus = MIXER_TRACKS_ENABLED; + } + + mAudioMixer->disable(AudioMixer::MIXING); + } + } + } + + // remove all the tracks that need to be... + count = tracksToRemove->size(); + if (UNLIKELY(count)) { + for (size_t i=0 ; i<count ; i++) { + const sp<Track>& track = tracksToRemove->itemAt(i); + mActiveTracks.remove(track); + if (track->isTerminated()) { + mTracks.remove(track); + deleteTrackName_l(track->mName); + } + } + } + + return mixerStatus; +} + +void AudioFlinger::MixerThread::getTracks( + SortedVector < sp<Track> >& tracks, + SortedVector < wp<Track> >& activeTracks, + int streamType) +{ + LOGV ("MixerThread::getTracks() mixer %p, mTracks.size %d, mActiveTracks.size %d", this, mTracks.size(), mActiveTracks.size()); + Mutex::Autolock _l(mLock); + size_t size = mTracks.size(); + for (size_t i = 0; i < size; i++) { + sp<Track> t = mTracks[i]; + if (t->type() == streamType) { + tracks.add(t); + int j = mActiveTracks.indexOf(t); + if (j >= 0) { + t = mActiveTracks[j].promote(); + if (t != NULL) { + activeTracks.add(t); + } + } + } + } + + size = activeTracks.size(); + for (size_t i = 0; i < size; i++) { + mActiveTracks.remove(activeTracks[i]); + } + + size = tracks.size(); + for (size_t i = 0; i < size; i++) { + sp<Track> t = tracks[i]; + mTracks.remove(t); + deleteTrackName_l(t->name()); + } +} + +void AudioFlinger::MixerThread::putTracks( + SortedVector < sp<Track> >& tracks, + SortedVector < wp<Track> >& activeTracks) +{ + LOGV ("MixerThread::putTracks() mixer %p, tracks.size %d, activeTracks.size %d", this, tracks.size(), activeTracks.size()); + Mutex::Autolock _l(mLock); + size_t size = tracks.size(); + for (size_t i = 0; i < size ; i++) { + sp<Track> t = tracks[i]; + int name = getTrackName_l(); + + if (name < 0) return; + + t->mName = name; + t->mThread = this; + mTracks.add(t); + + int j = activeTracks.indexOf(t); + if (j >= 0) { + mActiveTracks.add(t); + // force buffer refilling and no ramp volume when the track is mixed for the first time + t->mFillingUpStatus = Track::FS_FILLING; + } + } +} + +// getTrackName_l() must be called with ThreadBase::mLock held +int AudioFlinger::MixerThread::getTrackName_l() +{ + return mAudioMixer->getTrackName(); +} + +// deleteTrackName_l() must be called with ThreadBase::mLock held +void AudioFlinger::MixerThread::deleteTrackName_l(int name) +{ + LOGV("remove track (%d) and delete from mixer", name); + mAudioMixer->deleteTrackName(name); +} + +// checkForNewParameters_l() must be called with ThreadBase::mLock held +bool AudioFlinger::MixerThread::checkForNewParameters_l() +{ + bool reconfig = false; + + while (!mNewParameters.isEmpty()) { + status_t status = NO_ERROR; + String8 keyValuePair = mNewParameters[0]; + AudioParameter param = AudioParameter(keyValuePair); + int value; + + if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { + reconfig = true; + } + if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { + if (value != AudioSystem::PCM_16_BIT) { + status = BAD_VALUE; + } else { + reconfig = true; + } + } + if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { + if (value != AudioSystem::CHANNEL_OUT_STEREO) { + status = BAD_VALUE; + } else { + reconfig = true; + } + } + if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { + // do not accept frame count changes if tracks are open as the track buffer + // size depends on frame count and correct behavior would not be garantied + // if frame count is changed after track creation + if (!mTracks.isEmpty()) { + status = INVALID_OPERATION; + } else { + reconfig = true; + } + } + if (status == NO_ERROR) { + status = mOutput->setParameters(keyValuePair); + if (!mStandby && status == INVALID_OPERATION) { + mOutput->standby(); + mStandby = true; + mBytesWritten = 0; + status = mOutput->setParameters(keyValuePair); + } + if (status == NO_ERROR && reconfig) { + delete mAudioMixer; + readOutputParameters(); + mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); + for (size_t i = 0; i < mTracks.size() ; i++) { + int name = getTrackName_l(); + if (name < 0) break; + mTracks[i]->mName = name; + // limit track sample rate to 2 x new output sample rate + if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { + mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); + } + } + sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); + } + } + + mNewParameters.removeAt(0); + + mParamStatus = status; + mParamCond.signal(); + mWaitWorkCV.wait(mLock); + } + return reconfig; +} + +status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<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) + : PlaybackThread(audioFlinger, output, id), + mLeftVolume (1.0), mRightVolume(1.0) +{ + mType = PlaybackThread::DIRECT; +} + +AudioFlinger::DirectOutputThread::~DirectOutputThread() +{ +} + + +bool AudioFlinger::DirectOutputThread::threadLoop() +{ + uint32_t mixerStatus = MIXER_IDLE; + sp<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()) + { + processConfigEvents(); + + mixerStatus = MIXER_IDLE; + + { // scope for the mLock + + Mutex::Autolock _l(mLock); + + if (checkForNewParameters_l()) { + mixBufferSize = mFrameCount*mFrameSize; + activeSleepTime = activeSleepTimeUs(); + idleSleepTime = idleSleepTimeUs(); + standbyDelay = microseconds(activeSleepTime*2); + } + + // put audio hardware into standby after short delay + if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || + mSuspended) { + // wait until we have something to do... + if (!mStandby) { + LOGV("Audio hardware entering standby, mixer %p\n", this); + mOutput->standby(); + mStandby = true; + mBytesWritten = 0; + } + + if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { + // we're about to wait, flush the binder command buffer + IPCThreadState::self()->flushCommands(); + + if (exitPending()) break; + + LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); + mWaitWorkCV.wait(mLock); + LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); + + if (mMasterMute == false) { + char value[PROPERTY_VALUE_MAX]; + property_get("ro.audio.silent", value, "0"); + if (atoi(value)) { + LOGD("Silence is golden"); + setMasterMute(true); + } + } + + standbyTime = systemTime() + standbyDelay; + sleepTime = idleSleepTime; + continue; + } + } + + // find out which tracks need to be processed + if (mActiveTracks.size() != 0) { + sp<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); + + // compute volume for this track + float left, right; + if (track->isMuted() || mMasterMute || track->isPausing() || + mStreamTypes[track->type()].mute) { + left = right = 0; + if (track->isPausing()) { + track->setPaused(); + } + } else { + float typeVolume = mStreamTypes[track->type()].volume; + float v = mMasterVolume * typeVolume; + float v_clamped = v * cblk->volume[0]; + if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; + left = v_clamped/MAX_GAIN; + v_clamped = v * cblk->volume[1]; + if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; + right = v_clamped/MAX_GAIN; + } + + if (left != mLeftVolume || right != mRightVolume) { + mOutput->setVolume(left, right); + left = mLeftVolume; + right = mRightVolume; + } + + if (track->mFillingUpStatus == Track::FS_FILLED) { + track->mFillingUpStatus = Track::FS_ACTIVE; + if (track->mState == TrackBase::RESUMING) { + track->mState = TrackBase::ACTIVE; + } + } + + // reset retry count + track->mRetryCount = kMaxTrackRetriesDirect; + activeTrack = t; + mixerStatus = MIXER_TRACKS_READY; + } else { + //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); + if (track->isStopped()) { + track->reset(); + } + if (track->isTerminated() || track->isStopped() || track->isPaused()) { + // We have consumed all the buffers of this track. + // Remove it from the list of active tracks. + trackToRemove = track; + } else { + // No buffers for this track. Give it a few chances to + // fill a buffer, then remove it from active list. + if (--(track->mRetryCount) <= 0) { + LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); + trackToRemove = track; + } else { + mixerStatus = MIXER_TRACKS_ENABLED; + } + } + } + } + + // remove all the tracks that need to be... + if (UNLIKELY(trackToRemove != 0)) { + mActiveTracks.remove(trackToRemove); + if (trackToRemove->isTerminated()) { + mTracks.remove(trackToRemove); + deleteTrackName_l(trackToRemove->mName); + } + } + } + + if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { + AudioBufferProvider::Buffer buffer; + size_t frameCount = mFrameCount; + curBuf = (int8_t *)mMixBuffer; + // output audio to hardware + while(frameCount) { + buffer.frameCount = frameCount; + activeTrack->getNextBuffer(&buffer); + if (UNLIKELY(buffer.raw == 0)) { + memset(curBuf, 0, frameCount * mFrameSize); + break; + } + memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); + frameCount -= buffer.frameCount; + curBuf += buffer.frameCount * mFrameSize; + activeTrack->releaseBuffer(&buffer); + } + sleepTime = 0; + standbyTime = systemTime() + standbyDelay; + } else { + if (sleepTime == 0) { + if (mixerStatus == MIXER_TRACKS_ENABLED) { + sleepTime = activeSleepTime; + } else { + sleepTime = idleSleepTime; + } + } else if (mBytesWritten != 0 && AudioSystem::isLinearPCM(mFormat)) { + memset (mMixBuffer, 0, mFrameCount * mFrameSize); + sleepTime = 0; + } + } + + if (mSuspended) { + sleepTime = idleSleepTime; + } + // sleepTime == 0 means we must write to audio hardware + if (sleepTime == 0) { + mLastWriteTime = systemTime(); + mInWrite = true; + mBytesWritten += mixBufferSize; + int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize); + if (bytesWritten < 0) mBytesWritten -= mixBufferSize; + mNumWrites++; + mInWrite = false; + mStandby = false; + } else { + usleep(sleepTime); + } + + // finally let go of removed track, without the lock held + // since we can't guarantee the destructors won't acquire that + // same lock. + trackToRemove.clear(); + activeTrack.clear(); + } + + if (!mStandby) { + mOutput->standby(); + } + + LOGV("DirectOutputThread %p exiting", this); + return false; +} + +// getTrackName_l() must be called with ThreadBase::mLock held +int AudioFlinger::DirectOutputThread::getTrackName_l() +{ + return 0; +} + +// deleteTrackName_l() must be called with ThreadBase::mLock held +void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) +{ +} + +// checkForNewParameters_l() must be called with ThreadBase::mLock held +bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() +{ + bool reconfig = false; + + while (!mNewParameters.isEmpty()) { + status_t status = NO_ERROR; + String8 keyValuePair = mNewParameters[0]; + AudioParameter param = AudioParameter(keyValuePair); + int value; + + if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { + // do not accept frame count changes if tracks are open as the track buffer + // size depends on frame count and correct behavior would not be garantied + // if frame count is changed after track creation + if (!mTracks.isEmpty()) { + status = INVALID_OPERATION; + } else { + reconfig = true; + } + } + if (status == NO_ERROR) { + status = mOutput->setParameters(keyValuePair); + if (!mStandby && status == INVALID_OPERATION) { + mOutput->standby(); + mStandby = true; + mBytesWritten = 0; + status = mOutput->setParameters(keyValuePair); + } + if (status == NO_ERROR && reconfig) { + readOutputParameters(); + sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); + } + } + + mNewParameters.removeAt(0); + + mParamStatus = status; + mParamCond.signal(); + mWaitWorkCV.wait(mLock); + } + return reconfig; +} + +uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() +{ + uint32_t time; + if (AudioSystem::isLinearPCM(mFormat)) { + time = (uint32_t)(mOutput->latency() * 1000) / 2; + } else { + time = 10000; + } + return time; +} + +uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() +{ + uint32_t time; + if (AudioSystem::isLinearPCM(mFormat)) { + time = (uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000; + } else { + time = 10000; + } + return time; +} + +// ---------------------------------------------------------------------------- + +AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) + : MixerThread(audioFlinger, mainThread->getOutput(), id), mWaitTimeMs(UINT_MAX) +{ + mType = PlaybackThread::DUPLICATING; + addOutputTrack(mainThread); +} + +AudioFlinger::DuplicatingThread::~DuplicatingThread() +{ + for (size_t i = 0; i < mOutputTracks.size(); i++) { + mOutputTracks[i]->destroy(); + } + mOutputTracks.clear(); +} + +bool AudioFlinger::DuplicatingThread::threadLoop() +{ + int16_t* curBuf = mMixBuffer; + Vector< sp<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; + + 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); + } + + if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { + // mix buffers... + if (outputsReady(outputTracks)) { + mAudioMixer->process(curBuf); + } else { + memset(curBuf, 0, mixBufferSize); + } + sleepTime = 0; + writeFrames = mFrameCount; + } else { + if (sleepTime == 0) { + if (mixerStatus == MIXER_TRACKS_ENABLED) { + sleepTime = activeSleepTime; + } else { + sleepTime = idleSleepTime; + } + } else if (mBytesWritten != 0) { + // flush remaining overflow buffers in output tracks + for (size_t i = 0; i < outputTracks.size(); i++) { + if (outputTracks[i]->isActive()) { + sleepTime = 0; + writeFrames = 0; + break; + } + } + } + } + + if (mSuspended) { + sleepTime = idleSleepTime; + } + // sleepTime == 0 means we must write to audio hardware + if (sleepTime == 0) { + standbyTime = systemTime() + kStandbyTimeInNsecs; + for (size_t i = 0; i < outputTracks.size(); i++) { + outputTracks[i]->write(curBuf, writeFrames); + } + mStandby = false; + mBytesWritten += mixBufferSize; + } else { + usleep(sleepTime); + } + + // finally let go of all our tracks, without the lock held + // since we can't guarantee the destructors won't acquire that + // same lock. + tracksToRemove.clear(); + outputTracks.clear(); + } + + return false; +} + +void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) +{ + int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); + OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, + this, + mSampleRate, + mFormat, + mChannelCount, + frameCount); + if (outputTrack->cblk() != NULL) { + thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f); + mOutputTracks.add(outputTrack); + LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); + updateWaitTime(); + } +} + +void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) +{ + Mutex::Autolock _l(mLock); + for (size_t i = 0; i < mOutputTracks.size(); i++) { + if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { + mOutputTracks[i]->destroy(); + mOutputTracks.removeAt(i); + updateWaitTime(); + return; + } + } + LOGV("removeOutputTrack(): unkonwn thread: %p", thread); +} + +void AudioFlinger::DuplicatingThread::updateWaitTime() +{ + mWaitTimeMs = UINT_MAX; + for (size_t i = 0; i < mOutputTracks.size(); i++) { + sp<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) + : RefBase(), + mThread(thread), + mClient(client), + mCblk(0), + mFrameCount(0), + mState(IDLE), + mClientTid(-1), + mFormat(format), + mFlags(flags & ~SYSTEM_FLAGS_MASK) +{ + LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); + + // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); + size_t size = sizeof(audio_track_cblk_t); + size_t bufferSize = frameCount*channelCount*sizeof(int16_t); + if (sharedBuffer == 0) { + size += bufferSize; + } + + if (client != NULL) { + mCblkMemory = client->heap()->allocate(size); + if (mCblkMemory != 0) { + mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); + if (mCblk) { // construct the shared structure in-place. + new(mCblk) audio_track_cblk_t(); + // clear all buffers + mCblk->frameCount = frameCount; + mCblk->sampleRate = sampleRate; + mCblk->channels = (uint8_t)channelCount; + if (sharedBuffer == 0) { + mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); + memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); + // Force underrun condition to avoid false underrun callback until first data is + // written to buffer + mCblk->flowControlFlag = 1; + } else { + mBuffer = sharedBuffer->pointer(); + } + mBufferEnd = (uint8_t *)mBuffer + bufferSize; + } + } else { + LOGE("not enough memory for AudioTrack size=%u", size); + client->heap()->dump("AudioTrack"); + return; + } + } else { + mCblk = (audio_track_cblk_t *)(new uint8_t[size]); + if (mCblk) { // construct the shared structure in-place. + new(mCblk) audio_track_cblk_t(); + // clear all buffers + mCblk->frameCount = frameCount; + mCblk->sampleRate = sampleRate; + mCblk->channels = (uint8_t)channelCount; + mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); + memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); + // Force underrun condition to avoid false underrun callback until first data is + // written to buffer + mCblk->flowControlFlag = 1; + mBufferEnd = (uint8_t *)mBuffer + bufferSize; + } + } +} + +AudioFlinger::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->channels; +} + +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, channels %d", + bufferStart, bufferEnd, mBuffer, mBufferEnd, + cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channels); + return 0; + } + + return bufferStart; +} + +// ---------------------------------------------------------------------------- + +// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held +AudioFlinger::PlaybackThread::Track::Track( + const wp<ThreadBase>& thread, + const sp<Client>& client, + int streamType, + uint32_t sampleRate, + int format, + int channelCount, + int frameCount, + const sp<IMemory>& sharedBuffer) + : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer), + mMute(false), mSharedBuffer(sharedBuffer), mName(-1) +{ + if (mCblk != NULL) { + sp<ThreadBase> baseThread = thread.promote(); + if (baseThread != 0) { + PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); + mName = playbackThread->getTrackName_l(); + } + LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); + if (mName < 0) { + LOGE("no more track names available"); + } + mVolume[0] = 1.0f; + mVolume[1] = 1.0f; + mStreamType = streamType; + // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of + // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack + mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t); + } +} + +AudioFlinger::PlaybackThread::Track::~Track() +{ + LOGV("PlaybackThread::Track destructor"); + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + Mutex::Autolock _l(thread->mLock); + mState = TERMINATED; + } +} + +void AudioFlinger::PlaybackThread::Track::destroy() +{ + // NOTE: destroyTrack_l() can remove a strong reference to this Track + // by removing it from mTracks vector, so there is a risk that this Tracks's + // desctructor is called. As the destructor needs to lock mLock, + // we must acquire a strong reference on this Track before locking mLock + // here so that the destructor is called only when exiting this function. + // On the other hand, as long as Track::destroy() is only called by + // TrackHandle destructor, the TrackHandle still holds a strong ref on + // this Track with its member mTrack. + sp<Track> keep(this); + { // scope for mLock + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + if (!isOutputTrack()) { + if (mState == ACTIVE || mState == RESUMING) { + AudioSystem::stopOutput(thread->id(), (AudioSystem::stream_type)mStreamType); + } + AudioSystem::releaseOutput(thread->id()); + } + Mutex::Autolock _l(thread->mLock); + PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); + playbackThread->destroyTrack_l(this); + } + } +} + +void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) +{ + snprintf(buffer, size, " %5d %5d %3u %3u %3u %04u %1d %1d %1d %5u %5u %5u %08x %08x\n", + mName - AudioMixer::TRACK0, + (mClient == NULL) ? getpid() : mClient->pid(), + mStreamType, + mFormat, + mCblk->channels, + mFrameCount, + mState, + mMute, + mFillingUpStatus, + mCblk->sampleRate, + mCblk->volume[0], + mCblk->volume[1], + mCblk->server, + mCblk->user); +} + +status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) +{ + audio_track_cblk_t* cblk = this->cblk(); + uint32_t framesReady; + uint32_t framesReq = buffer->frameCount; + + // Check if last stepServer failed, try to step now + if (mFlags & TrackBase::STEPSERVER_FAILED) { + if (!step()) goto getNextBuffer_exit; + LOGV("stepServer recovered"); + mFlags &= ~TrackBase::STEPSERVER_FAILED; + } + + framesReady = cblk->framesReady(); + + if (LIKELY(framesReady)) { + uint32_t s = cblk->server; + uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; + + bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; + if (framesReq > framesReady) { + framesReq = framesReady; + } + if (s + framesReq > bufferEnd) { + framesReq = bufferEnd - s; + } + + buffer->raw = getBuffer(s, framesReq); + if (buffer->raw == 0) goto getNextBuffer_exit; + + buffer->frameCount = framesReq; + return NO_ERROR; + } + +getNextBuffer_exit: + buffer->raw = 0; + buffer->frameCount = 0; + LOGV("getNextBuffer() no more data 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->forceReady) { + mFillingUpStatus = FS_FILLED; + mCblk->forceReady = 0; + 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->flowControlFlag = 1; + mCblk->forceReady = 0; + mFillingUpStatus = FS_FILLING; + mResetDone = true; + } +} + +void AudioFlinger::PlaybackThread::Track::mute(bool muted) +{ + mMute = muted; +} + +void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) +{ + mVolume[0] = left; + mVolume[1] = right; +} + +// ---------------------------------------------------------------------------- + +// RecordTrack constructor must be called with AudioFlinger::mLock held +AudioFlinger::RecordThread::RecordTrack::RecordTrack( + const wp<ThreadBase>& thread, + const sp<Client>& client, + uint32_t sampleRate, + int format, + int channelCount, + int frameCount, + uint32_t flags) + : TrackBase(thread, client, sampleRate, format, + channelCount, frameCount, flags, 0), + mOverflow(false) +{ + 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->flowControlFlag = 1; + } +} + +void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) +{ + snprintf(buffer, size, " %05d %03u %03u %04u %01d %05u %08x %08x\n", + (mClient == NULL) ? getpid() : mClient->pid(), + mFormat, + mCblk->channels, + 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), + mActive(false), mSourceThread(sourceThread) +{ + + PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); + if (mCblk != NULL) { + mCblk->out = 1; + 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->channels %d mBufferEnd %p", + mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channels, 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 channels = mCblk->channels; + 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 * channels]; + pInBuffer->frameCount = startFrames; + pInBuffer->i16 = pInBuffer->mBuffer; + memset(pInBuffer->raw, 0, startFrames * channels * sizeof(int16_t)); + mBufferQueue.add(pInBuffer); + } else { + LOGW ("OutputTrack::write() %p no more buffers in queue", this); + } + } + } + } + + while (waitTimeLeftMs) { + // First write pending buffers, then new data + if (mBufferQueue.size()) { + pInBuffer = mBufferQueue.itemAt(0); + } else { + pInBuffer = &inBuffer; + } + + if (pInBuffer->frameCount == 0) { + break; + } + + if (mOutBuffer.frameCount == 0) { + mOutBuffer.frameCount = pInBuffer->frameCount; + nsecs_t startTime = systemTime(); + if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { + LOGV ("OutputTrack::write() %p 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 * channels * sizeof(int16_t)); + mCblk->stepUser(outFrames); + pInBuffer->frameCount -= outFrames; + pInBuffer->i16 += outFrames * channels; + mOutBuffer.frameCount -= outFrames; + mOutBuffer.i16 += outFrames * channels; + + if (pInBuffer->frameCount == 0) { + if (mBufferQueue.size()) { + mBufferQueue.removeAt(0); + delete [] pInBuffer->mBuffer; + delete pInBuffer; + LOGV("OutputTrack::write() %p 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 * channels]; + pInBuffer->frameCount = inBuffer.frameCount; + pInBuffer->i16 = pInBuffer->mBuffer; + memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channels * sizeof(int16_t)); + mBufferQueue.add(pInBuffer); + LOGV("OutputTrack::write() %p 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 * channels]; + pInBuffer->frameCount = frames; + pInBuffer->i16 = pInBuffer->mBuffer; + memset(pInBuffer->raw, 0, frames * channels * 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::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) + : BnAudioTrack(), + mTrack(track) +{ +} + +AudioFlinger::TrackHandle::~TrackHandle() { + // just stop the track on deletion, associated resources + // will be freed from the main thread once all pending buffers have + // been played. Unless it's not in the active track list, in which + // case we free everything now... + mTrack->destroy(); +} + +status_t AudioFlinger::TrackHandle::start() { + return mTrack->start(); +} + +void AudioFlinger::TrackHandle::stop() { + mTrack->stop(); +} + +void AudioFlinger::TrackHandle::flush() { + mTrack->flush(); +} + +void AudioFlinger::TrackHandle::mute(bool e) { + mTrack->mute(e); +} + +void AudioFlinger::TrackHandle::pause() { + mTrack->pause(); +} + +void AudioFlinger::TrackHandle::setVolume(float left, float right) { + mTrack->setVolume(left, right); +} + +sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { + return mTrack->getCblk(); +} + +status_t AudioFlinger::TrackHandle::onTransact( + uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) +{ + return BnAudioTrack::onTransact(code, data, reply, flags); +} + +// ---------------------------------------------------------------------------- + +sp<IAudioRecord> AudioFlinger::openRecord( + pid_t pid, + int input, + uint32_t sampleRate, + int format, + int channelCount, + int frameCount, + uint32_t flags, + status_t *status) +{ + sp<RecordThread::RecordTrack> recordTrack; + sp<RecordHandle> recordHandle; + sp<Client> client; + wp<Client> wclient; + status_t lStatus; + RecordThread *thread; + size_t inFrameCount; + + // check calling permissions + if (!recordingAllowed()) { + lStatus = PERMISSION_DENIED; + goto Exit; + } + + // add client to list + { // scope for mLock + Mutex::Autolock _l(mLock); + thread = checkRecordThread_l(input); + if (thread == NULL) { + lStatus = BAD_VALUE; + goto Exit; + } + + wclient = mClients.valueFor(pid); + if (wclient != NULL) { + client = wclient.promote(); + } else { + client = new Client(this, pid); + mClients.add(pid, client); + } + + // create new record track. The record track uses one track in mHardwareMixerThread by convention. + recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate, + format, channelCount, frameCount, flags); + } + if (recordTrack->getCblk() == NULL) { + // 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(); + sendConfigEvent(AudioSystem::INPUT_OPENED); +} + + +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 (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 && + (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 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(int event, int param) { + AudioSystem::OutputDescriptor desc; + void *param2 = 0; + + switch (event) { + case AudioSystem::INPUT_OPENED: + case AudioSystem::INPUT_CONFIG_CHANGED: + desc.channels = mChannelCount; + desc.samplingRate = mSampleRate; + desc.format = mFormat; + desc.frameCount = mFrameCount; + desc.latency = 0; + param2 = &desc; + break; + + case AudioSystem::INPUT_CLOSED: + default: + break; + } + Mutex::Autolock _l(mAudioFlinger->mLock); + 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(); + mChannelCount = AudioSystem::popCount(mInput->channels()); + mFormat = mInput->format(); + mFrameSize = mInput->frameSize(); + mInputBytes = mInput->bufferSize(); + mFrameCount = mInputBytes / mFrameSize; + mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; + + if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) + { + int channelCount; + // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid + // stereo to mono post process as the resampler always outputs stereo. + if (mChannelCount == 1 && mReqChannelCount == 2) { + channelCount = 1; + } else { + channelCount = 2; + } + mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); + mResampler->setSampleRate(mSampleRate); + mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); + mRsmpOutBuffer = new int32_t[mFrameCount * 2]; + + // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples + if (mChannelCount == 1 && mReqChannelCount == 1) { + mFrameCount >>= 1; + } + + } + mRsmpInIndex = mFrameCount; +} + +unsigned int AudioFlinger::RecordThread::getInputFramesLost() +{ + return mInput->getInputFramesLost(); +} + +// ---------------------------------------------------------------------------- + +int AudioFlinger::openOutput(uint32_t *pDevices, + uint32_t *pSamplingRate, + uint32_t *pFormat, + uint32_t *pChannels, + uint32_t *pLatencyMs, + uint32_t flags) +{ + status_t status; + PlaybackThread *thread = NULL; + mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; + uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; + uint32_t format = pFormat ? *pFormat : 0; + uint32_t channels = pChannels ? *pChannels : 0; + uint32_t latency = pLatencyMs ? *pLatencyMs : 0; + + LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", + pDevices ? *pDevices : 0, + samplingRate, + format, + channels, + flags); + + if (pDevices == NULL || *pDevices == 0) { + return 0; + } + Mutex::Autolock _l(mLock); + + AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices, + (int *)&format, + &channels, + &samplingRate, + &status); + LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", + output, + samplingRate, + format, + channels, + status); + + mHardwareStatus = AUDIO_HW_IDLE; + if (output != 0) { + if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) || + (format != AudioSystem::PCM_16_BIT) || + (channels != AudioSystem::CHANNEL_OUT_STEREO)) { + thread = new DirectOutputThread(this, output, ++mNextThreadId); + LOGV("openOutput() created direct output: ID %d thread %p", mNextThreadId, thread); + } else { + thread = new MixerThread(this, output, ++mNextThreadId); + LOGV("openOutput() created mixer output: ID %d thread %p", mNextThreadId, thread); + +#ifdef LVMX + unsigned bitsPerSample = + (format == AudioSystem::PCM_16_BIT) ? 16 : + ((format == AudioSystem::PCM_8_BIT) ? 8 : 0); + unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1; + int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id()); + + LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount); + LifeVibes::setDevice(audioOutputType, *pDevices); +#endif + + } + mPlaybackThreads.add(mNextThreadId, thread); + + if (pSamplingRate) *pSamplingRate = samplingRate; + if (pFormat) *pFormat = format; + if (pChannels) *pChannels = channels; + if (pLatencyMs) *pLatencyMs = thread->latency(); + + return mNextThreadId; + } + + return 0; +} + +int AudioFlinger::openDuplicateOutput(int output1, int output2) +{ + Mutex::Autolock _l(mLock); + MixerThread *thread1 = checkMixerThread_l(output1); + MixerThread *thread2 = checkMixerThread_l(output2); + + if (thread1 == NULL || thread2 == NULL) { + LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); + return 0; + } + + + DuplicatingThread *thread = new DuplicatingThread(this, thread1, ++mNextThreadId); + thread->addOutputTrack(thread2); + mPlaybackThreads.add(mNextThreadId, thread); + return mNextThreadId; +} + +status_t AudioFlinger::closeOutput(int output) +{ + // keep strong reference on the playback thread so that + // it is not destroyed while exit() is executed + sp <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) { + // Start record thread + thread = new RecordThread(this, input, reqSamplingRate, reqChannels, ++mNextThreadId); + mRecordThreads.add(mNextThreadId, thread); + LOGV("openInput() created record thread: ID %d thread %p", mNextThreadId, thread); + if (pSamplingRate) *pSamplingRate = reqSamplingRate; + if (pFormat) *pFormat = format; + if (pChannels) *pChannels = reqChannels; + + input->standby(); + + return mNextThreadId; + } + + return 0; +} + +status_t AudioFlinger::closeInput(int input) +{ + // keep strong reference on the record thread so that + // it is not destroyed while exit() is executed + sp <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); + + 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; + SortedVector < sp<MixerThread::Track> > tracks; + SortedVector < wp<MixerThread::Track> > activeTracks; + srcThread->getTracks(tracks, activeTracks, stream); + if (tracks.size()) { + dstThread->putTracks(tracks, activeTracks); + } + } + } + + dstThread->sendConfigEvent(AudioSystem::STREAM_CONFIG_CHANGED, stream); + + return NO_ERROR; +} + +// checkPlaybackThread_l() must be called with AudioFlinger::mLock held +AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const +{ + PlaybackThread *thread = NULL; + if (mPlaybackThreads.indexOfKey(output) >= 0) { + thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); + } + return thread; +} + +// checkMixerThread_l() must be called with AudioFlinger::mLock held +AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const +{ + PlaybackThread *thread = checkPlaybackThread_l(output); + if (thread != NULL) { + if (thread->type() == PlaybackThread::DIRECT) { + thread = NULL; + } + } + return (MixerThread *)thread; +} + +// checkRecordThread_l() must be called with AudioFlinger::mLock held +AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const +{ + RecordThread *thread = NULL; + if (mRecordThreads.indexOfKey(input) >= 0) { + thread = (RecordThread *)mRecordThreads.valueFor(input).get(); + } + return thread; +} + +// ---------------------------------------------------------------------------- + +status_t AudioFlinger::onTransact( + uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) +{ + return BnAudioFlinger::onTransact(code, data, reply, flags); +} + +// ---------------------------------------------------------------------------- + +void AudioFlinger::instantiate() { + defaultServiceManager()->addService( + String16("media.audio_flinger"), new AudioFlinger()); +} + +}; // namespace android |