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Diffstat (limited to 'libs/audioflinger/AudioFlinger.cpp')
| -rw-r--r-- | libs/audioflinger/AudioFlinger.cpp | 4055 |
1 files changed, 0 insertions, 4055 deletions
diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp deleted file mode 100644 index 2414e8d..0000000 --- a/libs/audioflinger/AudioFlinger.cpp +++ /dev/null @@ -1,4055 +0,0 @@ -/* //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 |