diff options
| -rw-r--r-- | services/audioflinger/Android.mk | 3 | ||||
| -rw-r--r-- | services/audioflinger/AudioFlinger.cpp | 7843 | ||||
| -rw-r--r-- | services/audioflinger/AudioFlinger.h | 1592 | ||||
| -rw-r--r-- | services/audioflinger/Effects.cpp | 1684 | ||||
| -rw-r--r-- | services/audioflinger/Effects.h | 359 | ||||
| -rw-r--r-- | services/audioflinger/PlaybackTracks.h | 285 | ||||
| -rw-r--r-- | services/audioflinger/RecordTracks.h | 62 | ||||
| -rw-r--r-- | services/audioflinger/Threads.cpp | 4426 | ||||
| -rw-r--r-- | services/audioflinger/Threads.h | 801 | ||||
| -rw-r--r-- | services/audioflinger/TrackBase.h | 139 | ||||
| -rw-r--r-- | services/audioflinger/Tracks.cpp | 1789 |
11 files changed, 9647 insertions, 9336 deletions
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk index 4416b52..c4050b8 100644 --- a/services/audioflinger/Android.mk +++ b/services/audioflinger/Android.mk @@ -15,6 +15,9 @@ include $(CLEAR_VARS) LOCAL_SRC_FILES:= \ AudioFlinger.cpp \ + Threads.cpp \ + Tracks.cpp \ + Effects.cpp \ AudioMixer.cpp.arm \ AudioResampler.cpp.arm \ AudioPolicyService.cpp \ diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 0c1ab3c..514fcb1 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -29,7 +29,6 @@ #include <utils/Log.h> #include <utils/Trace.h> #include <binder/Parcel.h> -#include <binder/IPCThreadState.h> #include <utils/String16.h> #include <utils/threads.h> #include <utils/Atomic.h> @@ -38,15 +37,8 @@ #include <cutils/properties.h> #include <cutils/compiler.h> -#undef ADD_BATTERY_DATA - -#ifdef ADD_BATTERY_DATA -#include <media/IMediaPlayerService.h> -#include <media/IMediaDeathNotifier.h> -#endif - -#include <private/media/AudioTrackShared.h> -#include <private/media/AudioEffectShared.h> +//#include <private/media/AudioTrackShared.h> +//#include <private/media/AudioEffectShared.h> #include <system/audio.h> #include <hardware/audio.h> @@ -64,26 +56,8 @@ #include <powermanager/PowerManager.h> -// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds -#ifdef DEBUG_CPU_USAGE -#include <cpustats/CentralTendencyStatistics.h> -#include <cpustats/ThreadCpuUsage.h> -#endif - #include <common_time/cc_helper.h> -#include <common_time/local_clock.h> - -#include "FastMixer.h" - -// NBAIO implementations -#include <media/nbaio/AudioStreamOutSink.h> -#include <media/nbaio/MonoPipe.h> -#include <media/nbaio/MonoPipeReader.h> -#include <media/nbaio/Pipe.h> -#include <media/nbaio/PipeReader.h> -#include <media/nbaio/SourceAudioBufferProvider.h> - -#include "SchedulingPolicyService.h" +//#include <common_time/local_clock.h> // ---------------------------------------------------------------------------- @@ -105,90 +79,13 @@ namespace android { static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; static const char kHardwareLockedString[] = "Hardware lock is taken\n"; -static const float MAX_GAIN = 4096.0f; -static const uint32_t MAX_GAIN_INT = 0x1000; - -// retry counts for buffer fill timeout -// 50 * ~20msecs = 1 second -static const int8_t kMaxTrackRetries = 50; -static const int8_t kMaxTrackStartupRetries = 50; -// allow less retry attempts on direct output thread. -// direct outputs can be a scarce resource in audio hardware and should -// be released as quickly as possible. -static const int8_t kMaxTrackRetriesDirect = 2; - -static const int kDumpLockRetries = 50; -static const int kDumpLockSleepUs = 20000; - -// don't warn about blocked writes or record buffer overflows more often than this -static const nsecs_t kWarningThrottleNs = seconds(5); - -// RecordThread loop sleep time upon application overrun or audio HAL read error -static const int kRecordThreadSleepUs = 5000; - -// maximum time to wait for setParameters to complete -static const nsecs_t kSetParametersTimeoutNs = seconds(2); - -// minimum sleep time for the mixer thread loop when tracks are active but in underrun -static const uint32_t kMinThreadSleepTimeUs = 5000; -// maximum divider applied to the active sleep time in the mixer thread loop -static const uint32_t kMaxThreadSleepTimeShift = 2; - -// minimum normal mix buffer size, expressed in milliseconds rather than frames -static const uint32_t kMinNormalMixBufferSizeMs = 20; -// maximum normal mix buffer size -static const uint32_t kMaxNormalMixBufferSizeMs = 24; nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; -// Whether to use fast mixer -static const enum { - FastMixer_Never, // never initialize or use: for debugging only - FastMixer_Always, // always initialize and use, even if not needed: for debugging only - // normal mixer multiplier is 1 - FastMixer_Static, // initialize if needed, then use all the time if initialized, - // multiplier is calculated based on min & max normal mixer buffer size - FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, - // multiplier is calculated based on min & max normal mixer buffer size - // FIXME for FastMixer_Dynamic: - // Supporting this option will require fixing HALs that can't handle large writes. - // For example, one HAL implementation returns an error from a large write, - // and another HAL implementation corrupts memory, possibly in the sample rate converter. - // We could either fix the HAL implementations, or provide a wrapper that breaks - // up large writes into smaller ones, and the wrapper would need to deal with scheduler. -} kUseFastMixer = FastMixer_Static; - -static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off" - // AudioFlinger::setParameters() updates, other threads read w/o lock - -// Priorities for requestPriority -static const int kPriorityAudioApp = 2; -static const int kPriorityFastMixer = 3; - -// IAudioFlinger::createTrack() reports back to client the total size of shared memory area -// for the track. The client then sub-divides this into smaller buffers for its use. -// Currently the client uses double-buffering by default, but doesn't tell us about that. -// So for now we just assume that client is double-buffered. -// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or -// N-buffering, so AudioFlinger could allocate the right amount of memory. -// See the client's minBufCount and mNotificationFramesAct calculations for details. -static const int kFastTrackMultiplier = 2; +uint32_t AudioFlinger::mScreenState; // ---------------------------------------------------------------------------- -#ifdef ADD_BATTERY_DATA -// To collect the amplifier usage -static void addBatteryData(uint32_t params) { - sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); - if (service == NULL) { - // it already logged - return; - } - - service->addBatteryData(params); -} -#endif - static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) { const hw_module_t *mod; @@ -364,7 +261,7 @@ void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) write(fd, result.string(), result.size()); } -static bool tryLock(Mutex& mutex) +bool AudioFlinger::dumpTryLock(Mutex& mutex) { bool locked = false; for (int i = 0; i < kDumpLockRetries; ++i) { @@ -383,7 +280,7 @@ status_t AudioFlinger::dump(int fd, const Vector<String16>& args) dumpPermissionDenial(fd, args); } else { // get state of hardware lock - bool hardwareLocked = tryLock(mHardwareLock); + bool hardwareLocked = dumpTryLock(mHardwareLock); if (!hardwareLocked) { String8 result(kHardwareLockedString); write(fd, result.string(), result.size()); @@ -391,7 +288,7 @@ status_t AudioFlinger::dump(int fd, const Vector<String16>& args) mHardwareLock.unlock(); } - bool locked = tryLock(mLock); + bool locked = dumpTryLock(mLock); // failed to lock - AudioFlinger is probably deadlocked if (!locked) { @@ -874,6 +771,7 @@ status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& { ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); + // check calling permissions if (!settingsAllowed()) { return PERMISSION_DENIED; @@ -922,8 +820,8 @@ status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& String8 screenState; if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { bool isOff = screenState == "off"; - if (isOff != (gScreenState & 1)) { - gScreenState = ((gScreenState & ~1) + 2) | isOff; + if (isOff != (AudioFlinger::mScreenState & 1)) { + AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; } } return final_result; @@ -1148,4640 +1046,7 @@ sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, return thread; } -// ---------------------------------------------------------------------------- - -AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, - audio_devices_t outDevice, audio_devices_t inDevice, type_t type) - : Thread(false /*canCallJava*/), - mType(type), - mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), - // mChannelMask - mChannelCount(0), - mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), - mParamStatus(NO_ERROR), - mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), - mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), - // mName will be set by concrete (non-virtual) subclass - mDeathRecipient(new PMDeathRecipient(this)) -{ -} - -AudioFlinger::ThreadBase::~ThreadBase() -{ - mParamCond.broadcast(); - // do not lock the mutex in destructor - releaseWakeLock_l(); - if (mPowerManager != 0) { - sp<IBinder> binder = mPowerManager->asBinder(); - binder->unlinkToDeath(mDeathRecipient); - } -} - -void AudioFlinger::ThreadBase::exit() -{ - ALOGV("ThreadBase::exit"); - // do any cleanup required for exit to succeed - preExit(); - { - // This lock prevents the following race in thread (uniprocessor for illustration): - // if (!exitPending()) { - // // context switch from here to exit() - // // exit() calls requestExit(), what exitPending() observes - // // exit() calls signal(), which is dropped since no waiters - // // context switch back from exit() to here - // mWaitWorkCV.wait(...); - // // now thread is hung - // } - AutoMutex lock(mLock); - requestExit(); - mWaitWorkCV.broadcast(); - } - // When Thread::requestExitAndWait is made virtual and this method is renamed to - // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" - requestExitAndWait(); -} - -status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) -{ - status_t status; - - ALOGV("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, kSetParametersTimeoutNs) == NO_ERROR) { - status = mParamStatus; - mWaitWorkCV.signal(); - } else { - status = TIMED_OUT; - } - return status; -} - -void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) -{ - Mutex::Autolock _l(mLock); - sendIoConfigEvent_l(event, param); -} - -// sendIoConfigEvent_l() must be called with ThreadBase::mLock held -void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) -{ - IoConfigEvent *ioEvent = new IoConfigEvent(event, param); - mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); - ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, - param); - mWaitWorkCV.signal(); -} - -// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held -void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) -{ - PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); - mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); - ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", - mConfigEvents.size(), pid, tid, prio); - mWaitWorkCV.signal(); -} - -void AudioFlinger::ThreadBase::processConfigEvents() -{ - mLock.lock(); - while (!mConfigEvents.isEmpty()) { - ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); - ConfigEvent *event = mConfigEvents[0]; - mConfigEvents.removeAt(0); - // release mLock before locking AudioFlinger mLock: lock order is always - // AudioFlinger then ThreadBase to avoid cross deadlock - mLock.unlock(); - switch(event->type()) { - case CFG_EVENT_PRIO: { - PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); - int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); - if (err != 0) { - ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " - "error %d", - prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); - } - } break; - case CFG_EVENT_IO: { - IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); - mAudioFlinger->mLock.lock(); - audioConfigChanged_l(ioEvent->event(), ioEvent->param()); - mAudioFlinger->mLock.unlock(); - } break; - default: - ALOGE("processConfigEvents() unknown event type %d", event->type()); - break; - } - delete event; - mLock.lock(); - } - mLock.unlock(); -} - -void 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, "io handle: %d\n", mId); - result.append(buffer); - snprintf(buffer, SIZE, "TID: %d\n", getTid()); - result.append(buffer); - snprintf(buffer, SIZE, "standby: %d\n", mStandby); - result.append(buffer); - snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); - result.append(buffer); - snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); - result.append(buffer); - snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); - result.append(buffer); - snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); - result.append(buffer); - snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); - result.append(buffer); - snprintf(buffer, SIZE, "Format: %d\n", mFormat); - result.append(buffer); - snprintf(buffer, SIZE, "Frame size: %u\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); - for (size_t i = 0; i < mConfigEvents.size(); i++) { - mConfigEvents[i]->dump(buffer, SIZE); - result.append(buffer); - } - result.append("\n"); - - write(fd, result.string(), result.size()); - - if (locked) { - mLock.unlock(); - } -} - -void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); - write(fd, buffer, strlen(buffer)); - - for (size_t i = 0; i < mEffectChains.size(); ++i) { - sp<EffectChain> chain = mEffectChains[i]; - if (chain != 0) { - chain->dump(fd, args); - } - } -} - -void AudioFlinger::ThreadBase::acquireWakeLock() -{ - Mutex::Autolock _l(mLock); - acquireWakeLock_l(); -} - -void AudioFlinger::ThreadBase::acquireWakeLock_l() -{ - if (mPowerManager == 0) { - // use checkService() to avoid blocking if power service is not up yet - sp<IBinder> binder = - defaultServiceManager()->checkService(String16("power")); - if (binder == 0) { - ALOGW("Thread %s cannot connect to the power manager service", mName); - } else { - mPowerManager = interface_cast<IPowerManager>(binder); - binder->linkToDeath(mDeathRecipient); - } - } - if (mPowerManager != 0) { - sp<IBinder> binder = new BBinder(); - status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, - binder, - String16(mName)); - if (status == NO_ERROR) { - mWakeLockToken = binder; - } - ALOGV("acquireWakeLock_l() %s status %d", mName, status); - } -} - -void AudioFlinger::ThreadBase::releaseWakeLock() -{ - Mutex::Autolock _l(mLock); - releaseWakeLock_l(); -} - -void AudioFlinger::ThreadBase::releaseWakeLock_l() -{ - if (mWakeLockToken != 0) { - ALOGV("releaseWakeLock_l() %s", mName); - if (mPowerManager != 0) { - mPowerManager->releaseWakeLock(mWakeLockToken, 0); - } - mWakeLockToken.clear(); - } -} - -void AudioFlinger::ThreadBase::clearPowerManager() -{ - Mutex::Autolock _l(mLock); - releaseWakeLock_l(); - mPowerManager.clear(); -} - -void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) -{ - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - thread->clearPowerManager(); - } - ALOGW("power manager service died !!!"); -} - -void AudioFlinger::ThreadBase::setEffectSuspended( - const effect_uuid_t *type, bool suspend, int sessionId) -{ - Mutex::Autolock _l(mLock); - setEffectSuspended_l(type, suspend, sessionId); -} - -void AudioFlinger::ThreadBase::setEffectSuspended_l( - const effect_uuid_t *type, bool suspend, int sessionId) -{ - sp<EffectChain> chain = getEffectChain_l(sessionId); - if (chain != 0) { - if (type != NULL) { - chain->setEffectSuspended_l(type, suspend); - } else { - chain->setEffectSuspendedAll_l(suspend); - } - } - - updateSuspendedSessions_l(type, suspend, sessionId); -} - -void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) -{ - ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); - if (index < 0) { - return; - } - - const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = - mSuspendedSessions.valueAt(index); - - for (size_t i = 0; i < sessionEffects.size(); i++) { - sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); - for (int j = 0; j < desc->mRefCount; j++) { - if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { - chain->setEffectSuspendedAll_l(true); - } else { - ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", - desc->mType.timeLow); - chain->setEffectSuspended_l(&desc->mType, true); - } - } - } -} - -void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, - bool suspend, - int sessionId) -{ - ssize_t index = mSuspendedSessions.indexOfKey(sessionId); - - KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; - - if (suspend) { - if (index >= 0) { - sessionEffects = mSuspendedSessions.valueAt(index); - } else { - mSuspendedSessions.add(sessionId, sessionEffects); - } - } else { - if (index < 0) { - return; - } - sessionEffects = mSuspendedSessions.valueAt(index); - } - - - int key = EffectChain::kKeyForSuspendAll; - if (type != NULL) { - key = type->timeLow; - } - index = sessionEffects.indexOfKey(key); - - sp<SuspendedSessionDesc> desc; - if (suspend) { - if (index >= 0) { - desc = sessionEffects.valueAt(index); - } else { - desc = new SuspendedSessionDesc(); - if (type != NULL) { - desc->mType = *type; - } - sessionEffects.add(key, desc); - ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); - } - desc->mRefCount++; - } else { - if (index < 0) { - return; - } - desc = sessionEffects.valueAt(index); - if (--desc->mRefCount == 0) { - ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); - sessionEffects.removeItemsAt(index); - if (sessionEffects.isEmpty()) { - ALOGV("updateSuspendedSessions_l() restore removing session %d", - sessionId); - mSuspendedSessions.removeItem(sessionId); - } - } - } - if (!sessionEffects.isEmpty()) { - mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); - } -} - -void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, - bool enabled, - int sessionId) -{ - Mutex::Autolock _l(mLock); - checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); -} - -void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, - bool enabled, - int sessionId) -{ - if (mType != RECORD) { - // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on - // another session. This gives the priority to well behaved effect control panels - // and applications not using global effects. - // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect - // global effects - if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { - setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); - } - } - sp<EffectChain> chain = getEffectChain_l(sessionId); - if (chain != 0) { - chain->checkSuspendOnEffectEnabled(effect, enabled); - } -} - -// ---------------------------------------------------------------------------- - -AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, - AudioStreamOut* output, - audio_io_handle_t id, - audio_devices_t device, - type_t type) - : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), - mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), - // mStreamTypes[] initialized in constructor body - mOutput(output), - mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), - mMixerStatus(MIXER_IDLE), - mMixerStatusIgnoringFastTracks(MIXER_IDLE), - standbyDelay(AudioFlinger::mStandbyTimeInNsecs), - mScreenState(gScreenState), - // index 0 is reserved for normal mixer's submix - mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) -{ - snprintf(mName, kNameLength, "AudioOut_%X", id); - - // Assumes constructor is called by AudioFlinger with it's mLock held, but - // it would be safer to explicitly pass initial masterVolume/masterMute as - // parameter. - // - // If the HAL we are using has support for master volume or master mute, - // then do not attenuate or mute during mixing (just leave the volume at 1.0 - // and the mute set to false). - mMasterVolume = audioFlinger->masterVolume_l(); - mMasterMute = audioFlinger->masterMute_l(); - if (mOutput && mOutput->audioHwDev) { - if (mOutput->audioHwDev->canSetMasterVolume()) { - mMasterVolume = 1.0; - } - - if (mOutput->audioHwDev->canSetMasterMute()) { - mMasterMute = false; - } - } - - readOutputParameters(); - - // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor - // There is no AUDIO_STREAM_MIN, and ++ operator does not compile - for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; - stream = (audio_stream_type_t) (stream + 1)) { - mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); - mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); - } - // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, - // because mAudioFlinger doesn't have one to copy from -} - -AudioFlinger::PlaybackThread::~PlaybackThread() -{ - delete [] mMixBuffer; -} - -void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) -{ - dumpInternals(fd, args); - dumpTracks(fd, args); - dumpEffectChains(fd, args); -} - -void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - result.appendFormat("Output thread %p stream volumes in dB:\n ", this); - for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { - const stream_type_t *st = &mStreamTypes[i]; - if (i > 0) { - result.appendFormat(", "); - } - result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); - if (st->mute) { - result.append("M"); - } - } - result.append("\n"); - write(fd, result.string(), result.length()); - result.clear(); - - snprintf(buffer, SIZE, "Output thread %p tracks\n", this); - result.append(buffer); - Track::appendDumpHeader(result); - 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); - Track::appendDumpHeader(result); - for (size_t i = 0; i < mActiveTracks.size(); ++i) { - sp<Track> track = mActiveTracks[i].promote(); - if (track != 0) { - track->dump(buffer, SIZE); - result.append(buffer); - } - } - write(fd, result.string(), result.size()); - - // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. - FastTrackUnderruns underruns = getFastTrackUnderruns(0); - fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", - underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); -} - -void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); - result.append(buffer); - snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", - ns2ms(systemTime() - mLastWriteTime)); - result.append(buffer); - snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); - result.append(buffer); - snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); - result.append(buffer); - snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); - result.append(buffer); - snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); - result.append(buffer); - snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); - result.append(buffer); - write(fd, result.string(), result.size()); - fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); - - dumpBase(fd, args); -} - -// Thread virtuals -status_t AudioFlinger::PlaybackThread::readyToRun() -{ - status_t status = initCheck(); - if (status == NO_ERROR) { - ALOGI("AudioFlinger's thread %p ready to run", this); - } else { - ALOGE("No working audio driver found."); - } - return status; -} - -void AudioFlinger::PlaybackThread::onFirstRef() -{ - run(mName, ANDROID_PRIORITY_URGENT_AUDIO); -} - -// ThreadBase virtuals -void AudioFlinger::PlaybackThread::preExit() -{ - ALOGV(" preExit()"); - // FIXME this is using hard-coded strings but in the future, this functionality will be - // converted to use audio HAL extensions required to support tunneling - mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); -} - -// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held -sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( - const sp<AudioFlinger::Client>& client, - audio_stream_type_t streamType, - uint32_t sampleRate, - audio_format_t format, - audio_channel_mask_t channelMask, - size_t frameCount, - const sp<IMemory>& sharedBuffer, - int sessionId, - IAudioFlinger::track_flags_t *flags, - pid_t tid, - status_t *status) -{ - sp<Track> track; - status_t lStatus; - - bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; - - // client expresses a preference for FAST, but we get the final say - if (*flags & IAudioFlinger::TRACK_FAST) { - if ( - // not timed - (!isTimed) && - // either of these use cases: - ( - // use case 1: shared buffer with any frame count - ( - (sharedBuffer != 0) - ) || - // use case 2: callback handler and frame count is default or at least as large as HAL - ( - (tid != -1) && - ((frameCount == 0) || - (frameCount >= (mFrameCount * kFastTrackMultiplier))) - ) - ) && - // PCM data - audio_is_linear_pcm(format) && - // mono or stereo - ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || - (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && -#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE - // hardware sample rate - (sampleRate == mSampleRate) && -#endif - // normal mixer has an associated fast mixer - hasFastMixer() && - // there are sufficient fast track slots available - (mFastTrackAvailMask != 0) - // FIXME test that MixerThread for this fast track has a capable output HAL - // FIXME add a permission test also? - ) { - // if frameCount not specified, then it defaults to fast mixer (HAL) frame count - if (frameCount == 0) { - frameCount = mFrameCount * kFastTrackMultiplier; - } - ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", - frameCount, mFrameCount); - } else { - ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " - "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " - "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", - isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, - audio_is_linear_pcm(format), - channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); - *flags &= ~IAudioFlinger::TRACK_FAST; - // For compatibility with AudioTrack calculation, buffer depth is forced - // to be at least 2 x the normal mixer frame count and cover audio hardware latency. - // This is probably too conservative, but legacy application code may depend on it. - // If you change this calculation, also review the start threshold which is related. - uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); - uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); - if (minBufCount < 2) { - minBufCount = 2; - } - size_t minFrameCount = mNormalFrameCount * minBufCount; - if (frameCount < minFrameCount) { - frameCount = minFrameCount; - } - } - } - - if (mType == DIRECT) { - if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { - if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { - ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " - "for output %p with format %d", - sampleRate, format, channelMask, mOutput, mFormat); - lStatus = BAD_VALUE; - goto Exit; - } - } - } else { - // Resampler implementation limits input sampling rate to 2 x output sampling rate. - if (sampleRate > mSampleRate*2) { - ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); - lStatus = BAD_VALUE; - goto Exit; - } - } - - lStatus = initCheck(); - if (lStatus != NO_ERROR) { - ALOGE("Audio driver not initialized."); - goto Exit; - } - - { // scope for mLock - Mutex::Autolock _l(mLock); - - // all tracks in same audio session must share the same routing strategy otherwise - // conflicts will happen when tracks are moved from one output to another by audio policy - // manager - uint32_t strategy = AudioSystem::getStrategyForStream(streamType); - for (size_t i = 0; i < mTracks.size(); ++i) { - sp<Track> t = mTracks[i]; - if (t != 0 && !t->isOutputTrack()) { - uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); - if (sessionId == t->sessionId() && strategy != actual) { - ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", - strategy, actual); - lStatus = BAD_VALUE; - goto Exit; - } - } - } - - if (!isTimed) { - track = new Track(this, client, streamType, sampleRate, format, - channelMask, frameCount, sharedBuffer, sessionId, *flags); - } else { - track = TimedTrack::create(this, client, streamType, sampleRate, format, - channelMask, frameCount, sharedBuffer, sessionId); - } - if (track == 0 || track->getCblk() == NULL || track->name() < 0) { - lStatus = NO_MEMORY; - goto Exit; - } - mTracks.add(track); - - sp<EffectChain> chain = getEffectChain_l(sessionId); - if (chain != 0) { - ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); - track->setMainBuffer(chain->inBuffer()); - chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); - chain->incTrackCnt(); - } - - if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { - pid_t callingPid = IPCThreadState::self()->getCallingPid(); - // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, - // so ask activity manager to do this on our behalf - sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); - } - } - - lStatus = NO_ERROR; - -Exit: - if (status) { - *status = lStatus; - } - return track; -} - -uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const -{ - if (mFastMixer != NULL) { - MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); - latency += (pipe->getAvgFrames() * 1000) / mSampleRate; - } - return latency; -} - -uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const -{ - return latency; -} - -uint32_t AudioFlinger::PlaybackThread::latency() const -{ - Mutex::Autolock _l(mLock); - return latency_l(); -} -uint32_t AudioFlinger::PlaybackThread::latency_l() const -{ - if (initCheck() == NO_ERROR) { - return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); - } else { - return 0; - } -} - -void AudioFlinger::PlaybackThread::setMasterVolume(float value) -{ - Mutex::Autolock _l(mLock); - // Don't apply master volume in SW if our HAL can do it for us. - if (mOutput && mOutput->audioHwDev && - mOutput->audioHwDev->canSetMasterVolume()) { - mMasterVolume = 1.0; - } else { - mMasterVolume = value; - } -} - -void AudioFlinger::PlaybackThread::setMasterMute(bool muted) -{ - Mutex::Autolock _l(mLock); - // Don't apply master mute in SW if our HAL can do it for us. - if (mOutput && mOutput->audioHwDev && - mOutput->audioHwDev->canSetMasterMute()) { - mMasterMute = false; - } else { - mMasterMute = muted; - } -} - -void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) -{ - Mutex::Autolock _l(mLock); - mStreamTypes[stream].volume = value; -} - -void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) -{ - Mutex::Autolock _l(mLock); - mStreamTypes[stream].mute = muted; -} - -float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const -{ - Mutex::Autolock _l(mLock); - return mStreamTypes[stream].volume; -} - -// 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; - track->mPresentationCompleteFrames = 0; - mActiveTracks.add(track); - if (track->mainBuffer() != mMixBuffer) { - sp<EffectChain> chain = getEffectChain_l(track->sessionId()); - if (chain != 0) { - ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), - track->sessionId()); - chain->incActiveTrackCnt(); - } - } - - status = NO_ERROR; - } - - ALOGV("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; - // active tracks are removed by threadLoop() - if (mActiveTracks.indexOf(track) < 0) { - removeTrack_l(track); - } -} - -void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) -{ - track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); - mTracks.remove(track); - deleteTrackName_l(track->name()); - // redundant as track is about to be destroyed, for dumpsys only - track->mName = -1; - if (track->isFastTrack()) { - int index = track->mFastIndex; - ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); - ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); - mFastTrackAvailMask |= 1 << index; - // redundant as track is about to be destroyed, for dumpsys only - track->mFastIndex = -1; - } - sp<EffectChain> chain = getEffectChain_l(track->sessionId()); - if (chain != 0) { - chain->decTrackCnt(); - } -} - -String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) -{ - String8 out_s8 = String8(""); - char *s; - - Mutex::Autolock _l(mLock); - if (initCheck() != NO_ERROR) { - return out_s8; - } - - s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); - out_s8 = String8(s); - free(s); - return out_s8; -} - -// audioConfigChanged_l() must be called with AudioFlinger::mLock held -void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { - AudioSystem::OutputDescriptor desc; - void *param2 = NULL; - - ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, - param); - - switch (event) { - case AudioSystem::OUTPUT_OPENED: - case AudioSystem::OUTPUT_CONFIG_CHANGED: - desc.channels = mChannelMask; - desc.samplingRate = mSampleRate; - desc.format = mFormat; - desc.frameCount = mNormalFrameCount; // FIXME see - // AudioFlinger::frameCount(audio_io_handle_t) - desc.latency = latency(); - param2 = &desc; - break; - - case AudioSystem::STREAM_CONFIG_CHANGED: - param2 = ¶m; - case AudioSystem::OUTPUT_CLOSED: - default: - break; - } - mAudioFlinger->audioConfigChanged_l(event, mId, param2); -} - -void AudioFlinger::PlaybackThread::readOutputParameters() -{ - mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); - mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); - mChannelCount = (uint16_t)popcount(mChannelMask); - mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); - mFrameSize = audio_stream_frame_size(&mOutput->stream->common); - mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; - if (mFrameCount & 15) { - ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", - mFrameCount); - } - - // Calculate size of normal mix buffer relative to the HAL output buffer size - double multiplier = 1.0; - if (mType == MIXER && (kUseFastMixer == FastMixer_Static || - kUseFastMixer == FastMixer_Dynamic)) { - size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; - size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; - // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer - minNormalFrameCount = (minNormalFrameCount + 15) & ~15; - maxNormalFrameCount = maxNormalFrameCount & ~15; - if (maxNormalFrameCount < minNormalFrameCount) { - maxNormalFrameCount = minNormalFrameCount; - } - multiplier = (double) minNormalFrameCount / (double) mFrameCount; - if (multiplier <= 1.0) { - multiplier = 1.0; - } else if (multiplier <= 2.0) { - if (2 * mFrameCount <= maxNormalFrameCount) { - multiplier = 2.0; - } else { - multiplier = (double) maxNormalFrameCount / (double) mFrameCount; - } - } else { - // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL - // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast - // track, but we sometimes have to do this to satisfy the maximum frame count - // constraint) - // FIXME this rounding up should not be done if no HAL SRC - uint32_t truncMult = (uint32_t) multiplier; - if ((truncMult & 1)) { - if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { - ++truncMult; - } - } - multiplier = (double) truncMult; - } - } - mNormalFrameCount = multiplier * mFrameCount; - // round up to nearest 16 frames to satisfy AudioMixer - mNormalFrameCount = (mNormalFrameCount + 15) & ~15; - ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, - mNormalFrameCount); - - delete[] mMixBuffer; - mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; - memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); - - // force reconfiguration of effect chains and engines to take new buffer size and audio - // parameters into account - // Note that mLock is not held when readOutputParameters() is called from the constructor - // but in this case nothing is done below as no audio sessions have effect yet so it doesn't - // matter. - // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains - Vector< sp<EffectChain> > effectChains = mEffectChains; - for (size_t i = 0; i < effectChains.size(); i ++) { - mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); - } -} - - -status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) -{ - if (halFrames == NULL || dspFrames == NULL) { - return BAD_VALUE; - } - Mutex::Autolock _l(mLock); - if (initCheck() != NO_ERROR) { - return INVALID_OPERATION; - } - size_t framesWritten = mBytesWritten / mFrameSize; - *halFrames = framesWritten; - - if (isSuspended()) { - // return an estimation of rendered frames when the output is suspended - size_t latencyFrames = (latency_l() * mSampleRate) / 1000; - *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; - return NO_ERROR; - } else { - return mOutput->stream->get_render_position(mOutput->stream, dspFrames); - } -} - -uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const -{ - Mutex::Autolock _l(mLock); - uint32_t result = 0; - if (getEffectChain_l(sessionId) != 0) { - result = EFFECT_SESSION; - } - - for (size_t i = 0; i < mTracks.size(); ++i) { - sp<Track> track = mTracks[i]; - if (sessionId == track->sessionId() && - !(track->mCblk->flags & CBLK_INVALID)) { - result |= TRACK_SESSION; - break; - } - } - - return result; -} - -uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) -{ - // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that - // it is moved to correct output by audio policy manager when A2DP is connected or disconnected - if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { - return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); - } - for (size_t i = 0; i < mTracks.size(); i++) { - sp<Track> track = mTracks[i]; - if (sessionId == track->sessionId() && - !(track->mCblk->flags & CBLK_INVALID)) { - return AudioSystem::getStrategyForStream(track->streamType()); - } - } - return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); -} - - -AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const -{ - Mutex::Autolock _l(mLock); - return mOutput; -} - -AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() -{ - Mutex::Autolock _l(mLock); - AudioStreamOut *output = mOutput; - mOutput = NULL; - // FIXME FastMixer might also have a raw ptr to mOutputSink; - // must push a NULL and wait for ack - mOutputSink.clear(); - mPipeSink.clear(); - mNormalSink.clear(); - return output; -} - -// this method must always be called either with ThreadBase mLock held or inside the thread loop -audio_stream_t* AudioFlinger::PlaybackThread::stream() const -{ - if (mOutput == NULL) { - return NULL; - } - return &mOutput->stream->common; -} - -uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const -{ - return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); -} - -status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) -{ - if (!isValidSyncEvent(event)) { - return BAD_VALUE; - } - - Mutex::Autolock _l(mLock); - - for (size_t i = 0; i < mTracks.size(); ++i) { - sp<Track> track = mTracks[i]; - if (event->triggerSession() == track->sessionId()) { - (void) track->setSyncEvent(event); - return NO_ERROR; - } - } - - return NAME_NOT_FOUND; -} - -bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const -{ - return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; -} - -void AudioFlinger::PlaybackThread::threadLoop_removeTracks( - const Vector< sp<Track> >& tracksToRemove) -{ - size_t count = tracksToRemove.size(); - if (CC_UNLIKELY(count)) { - for (size_t i = 0 ; i < count ; i++) { - const sp<Track>& track = tracksToRemove.itemAt(i); - if ((track->sharedBuffer() != 0) && - (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { - AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); - } - } - } - -} - -// ---------------------------------------------------------------------------- - -AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, - audio_io_handle_t id, audio_devices_t device, type_t type) - : PlaybackThread(audioFlinger, output, id, device, type), - // mAudioMixer below - // mFastMixer below - mFastMixerFutex(0) - // mOutputSink below - // mPipeSink below - // mNormalSink below -{ - ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); - ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " - "mFrameCount=%d, mNormalFrameCount=%d", - mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, - mNormalFrameCount); - mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); - - // FIXME - Current mixer implementation only supports stereo output - if (mChannelCount != FCC_2) { - ALOGE("Invalid audio hardware channel count %d", mChannelCount); - } - - // create an NBAIO sink for the HAL output stream, and negotiate - mOutputSink = new AudioStreamOutSink(output->stream); - size_t numCounterOffers = 0; - const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; - ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); - ALOG_ASSERT(index == 0); - - // initialize fast mixer depending on configuration - bool initFastMixer; - switch (kUseFastMixer) { - case FastMixer_Never: - initFastMixer = false; - break; - case FastMixer_Always: - initFastMixer = true; - break; - case FastMixer_Static: - case FastMixer_Dynamic: - initFastMixer = mFrameCount < mNormalFrameCount; - break; - } - if (initFastMixer) { - - // create a MonoPipe to connect our submix to FastMixer - NBAIO_Format format = mOutputSink->format(); - // This pipe depth compensates for scheduling latency of the normal mixer thread. - // When it wakes up after a maximum latency, it runs a few cycles quickly before - // finally blocking. Note the pipe implementation rounds up the request to a power of 2. - MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); - const NBAIO_Format offers[1] = {format}; - size_t numCounterOffers = 0; - ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); - ALOG_ASSERT(index == 0); - monoPipe->setAvgFrames((mScreenState & 1) ? - (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); - mPipeSink = monoPipe; - -#ifdef TEE_SINK_FRAMES - // create a Pipe to archive a copy of FastMixer's output for dumpsys - Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); - numCounterOffers = 0; - index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); - ALOG_ASSERT(index == 0); - mTeeSink = teeSink; - PipeReader *teeSource = new PipeReader(*teeSink); - numCounterOffers = 0; - index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); - ALOG_ASSERT(index == 0); - mTeeSource = teeSource; -#endif - - // create fast mixer and configure it initially with just one fast track for our submix - mFastMixer = new FastMixer(); - FastMixerStateQueue *sq = mFastMixer->sq(); -#ifdef STATE_QUEUE_DUMP - sq->setObserverDump(&mStateQueueObserverDump); - sq->setMutatorDump(&mStateQueueMutatorDump); -#endif - FastMixerState *state = sq->begin(); - FastTrack *fastTrack = &state->mFastTracks[0]; - // wrap the source side of the MonoPipe to make it an AudioBufferProvider - fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); - fastTrack->mVolumeProvider = NULL; - fastTrack->mGeneration++; - state->mFastTracksGen++; - state->mTrackMask = 1; - // fast mixer will use the HAL output sink - state->mOutputSink = mOutputSink.get(); - state->mOutputSinkGen++; - state->mFrameCount = mFrameCount; - state->mCommand = FastMixerState::COLD_IDLE; - // already done in constructor initialization list - //mFastMixerFutex = 0; - state->mColdFutexAddr = &mFastMixerFutex; - state->mColdGen++; - state->mDumpState = &mFastMixerDumpState; - state->mTeeSink = mTeeSink.get(); - sq->end(); - sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); - - // start the fast mixer - mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); - pid_t tid = mFastMixer->getTid(); - int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); - if (err != 0) { - ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", - kPriorityFastMixer, getpid_cached, tid, err); - } - -#ifdef AUDIO_WATCHDOG - // create and start the watchdog - mAudioWatchdog = new AudioWatchdog(); - mAudioWatchdog->setDump(&mAudioWatchdogDump); - mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); - tid = mAudioWatchdog->getTid(); - err = requestPriority(getpid_cached, tid, kPriorityFastMixer); - if (err != 0) { - ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", - kPriorityFastMixer, getpid_cached, tid, err); - } -#endif - - } else { - mFastMixer = NULL; - } - - switch (kUseFastMixer) { - case FastMixer_Never: - case FastMixer_Dynamic: - mNormalSink = mOutputSink; - break; - case FastMixer_Always: - mNormalSink = mPipeSink; - break; - case FastMixer_Static: - mNormalSink = initFastMixer ? mPipeSink : mOutputSink; - break; - } -} - -AudioFlinger::MixerThread::~MixerThread() -{ - if (mFastMixer != NULL) { - FastMixerStateQueue *sq = mFastMixer->sq(); - FastMixerState *state = sq->begin(); - if (state->mCommand == FastMixerState::COLD_IDLE) { - int32_t old = android_atomic_inc(&mFastMixerFutex); - if (old == -1) { - __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); - } - } - state->mCommand = FastMixerState::EXIT; - sq->end(); - sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); - mFastMixer->join(); - // Though the fast mixer thread has exited, it's state queue is still valid. - // We'll use that extract the final state which contains one remaining fast track - // corresponding to our sub-mix. - state = sq->begin(); - ALOG_ASSERT(state->mTrackMask == 1); - FastTrack *fastTrack = &state->mFastTracks[0]; - ALOG_ASSERT(fastTrack->mBufferProvider != NULL); - delete fastTrack->mBufferProvider; - sq->end(false /*didModify*/); - delete mFastMixer; -#ifdef AUDIO_WATCHDOG - if (mAudioWatchdog != 0) { - mAudioWatchdog->requestExit(); - mAudioWatchdog->requestExitAndWait(); - mAudioWatchdog.clear(); - } -#endif - } - delete mAudioMixer; -} - -class CpuStats { -public: - CpuStats(); - void sample(const String8 &title); -#ifdef DEBUG_CPU_USAGE -private: - ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns - CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns - - CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles - - int mCpuNum; // thread's current CPU number - int mCpukHz; // frequency of thread's current CPU in kHz -#endif -}; - -CpuStats::CpuStats() -#ifdef DEBUG_CPU_USAGE - : mCpuNum(-1), mCpukHz(-1) -#endif -{ -} - -void CpuStats::sample(const String8 &title) { -#ifdef DEBUG_CPU_USAGE - // get current thread's delta CPU time in wall clock ns - double wcNs; - bool valid = mCpuUsage.sampleAndEnable(wcNs); - - // record sample for wall clock statistics - if (valid) { - mWcStats.sample(wcNs); - } - - // get the current CPU number - int cpuNum = sched_getcpu(); - - // get the current CPU frequency in kHz - int cpukHz = mCpuUsage.getCpukHz(cpuNum); - - // check if either CPU number or frequency changed - if (cpuNum != mCpuNum || cpukHz != mCpukHz) { - mCpuNum = cpuNum; - mCpukHz = cpukHz; - // ignore sample for purposes of cycles - valid = false; - } - - // if no change in CPU number or frequency, then record sample for cycle statistics - if (valid && mCpukHz > 0) { - double cycles = wcNs * cpukHz * 0.000001; - mHzStats.sample(cycles); - } - - unsigned n = mWcStats.n(); - // mCpuUsage.elapsed() is expensive, so don't call it every loop - if ((n & 127) == 1) { - long long elapsed = mCpuUsage.elapsed(); - if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { - double perLoop = elapsed / (double) n; - double perLoop100 = perLoop * 0.01; - double perLoop1k = perLoop * 0.001; - double mean = mWcStats.mean(); - double stddev = mWcStats.stddev(); - double minimum = mWcStats.minimum(); - double maximum = mWcStats.maximum(); - double meanCycles = mHzStats.mean(); - double stddevCycles = mHzStats.stddev(); - double minCycles = mHzStats.minimum(); - double maxCycles = mHzStats.maximum(); - mCpuUsage.resetElapsed(); - mWcStats.reset(); - mHzStats.reset(); - ALOGD("CPU usage for %s over past %.1f secs\n" - " (%u mixer loops at %.1f mean ms per loop):\n" - " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" - " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" - " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", - title.string(), - elapsed * .000000001, n, perLoop * .000001, - mean * .001, - stddev * .001, - minimum * .001, - maximum * .001, - mean / perLoop100, - stddev / perLoop100, - minimum / perLoop100, - maximum / perLoop100, - meanCycles / perLoop1k, - stddevCycles / perLoop1k, - minCycles / perLoop1k, - maxCycles / perLoop1k); - - } - } -#endif -}; - -void AudioFlinger::PlaybackThread::checkSilentMode_l() -{ - if (!mMasterMute) { - char value[PROPERTY_VALUE_MAX]; - if (property_get("ro.audio.silent", value, "0") > 0) { - char *endptr; - unsigned long ul = strtoul(value, &endptr, 0); - if (*endptr == '\0' && ul != 0) { - ALOGD("Silence is golden"); - // The setprop command will not allow a property to be changed after - // the first time it is set, so we don't have to worry about un-muting. - setMasterMute_l(true); - } - } - } -} - -bool AudioFlinger::PlaybackThread::threadLoop() -{ - Vector< sp<Track> > tracksToRemove; - - standbyTime = systemTime(); - - // MIXER - nsecs_t lastWarning = 0; - - // DUPLICATING - // FIXME could this be made local to while loop? - writeFrames = 0; - - cacheParameters_l(); - sleepTime = idleSleepTime; - - if (mType == MIXER) { - sleepTimeShift = 0; - } - - CpuStats cpuStats; - const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); - - acquireWakeLock(); - - while (!exitPending()) - { - cpuStats.sample(myName); - - Vector< sp<EffectChain> > effectChains; - - processConfigEvents(); - - { // scope for mLock - - Mutex::Autolock _l(mLock); - - if (checkForNewParameters_l()) { - cacheParameters_l(); - } - - saveOutputTracks(); - - // put audio hardware into standby after short delay - if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || - isSuspended())) { - if (!mStandby) { - - threadLoop_standby(); - - mStandby = true; - } - - if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { - // we're about to wait, flush the binder command buffer - IPCThreadState::self()->flushCommands(); - - clearOutputTracks(); - - if (exitPending()) { - break; - } - - releaseWakeLock_l(); - // wait until we have something to do... - ALOGV("%s going to sleep", myName.string()); - mWaitWorkCV.wait(mLock); - ALOGV("%s waking up", myName.string()); - acquireWakeLock_l(); - - mMixerStatus = MIXER_IDLE; - mMixerStatusIgnoringFastTracks = MIXER_IDLE; - mBytesWritten = 0; - - checkSilentMode_l(); - - standbyTime = systemTime() + standbyDelay; - sleepTime = idleSleepTime; - if (mType == MIXER) { - sleepTimeShift = 0; - } - - continue; - } - } - - // mMixerStatusIgnoringFastTracks is also updated internally - mMixerStatus = prepareTracks_l(&tracksToRemove); - - // prevent any changes in effect chain list and in each effect chain - // during mixing and effect process as the audio buffers could be deleted - // or modified if an effect is created or deleted - lockEffectChains_l(effectChains); - } - - if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { - threadLoop_mix(); - } else { - threadLoop_sleepTime(); - } - - if (isSuspended()) { - sleepTime = suspendSleepTimeUs(); - mBytesWritten += mixBufferSize; - } - - // only process effects if we're going to write - if (sleepTime == 0) { - for (size_t i = 0; i < effectChains.size(); i ++) { - effectChains[i]->process_l(); - } - } - - // enable changes in effect chain - unlockEffectChains(effectChains); - - // sleepTime == 0 means we must write to audio hardware - if (sleepTime == 0) { - - threadLoop_write(); - -if (mType == MIXER) { - // write blocked detection - nsecs_t now = systemTime(); - nsecs_t delta = now - mLastWriteTime; - if (!mStandby && delta > maxPeriod) { - mNumDelayedWrites++; - if ((now - lastWarning) > kWarningThrottleNs) { -#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) - ScopedTrace st(ATRACE_TAG, "underrun"); -#endif - ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", - ns2ms(delta), mNumDelayedWrites, this); - lastWarning = now; - } - } -} - - mStandby = false; - } else { - usleep(sleepTime); - } - - // Finally let go of removed track(s), without the lock held - // since we can't guarantee the destructors won't acquire that - // same lock. This will also mutate and push a new fast mixer state. - threadLoop_removeTracks(tracksToRemove); - tracksToRemove.clear(); - - // FIXME I don't understand the need for this here; - // it was in the original code but maybe the - // assignment in saveOutputTracks() makes this unnecessary? - clearOutputTracks(); - - // Effect chains will be actually deleted here if they were removed from - // mEffectChains list during mixing or effects processing - effectChains.clear(); - - // FIXME Note that the above .clear() is no longer necessary since effectChains - // is now local to this block, but will keep it for now (at least until merge done). - } - - // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... - if (mType == MIXER || mType == DIRECT) { - // put output stream into standby mode - if (!mStandby) { - mOutput->stream->common.standby(&mOutput->stream->common); - } - } - - releaseWakeLock(); - - ALOGV("Thread %p type %d exiting", this, mType); - return false; -} - -void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) -{ - PlaybackThread::threadLoop_removeTracks(tracksToRemove); -} - -void AudioFlinger::MixerThread::threadLoop_write() -{ - // FIXME we should only do one push per cycle; confirm this is true - // Start the fast mixer if it's not already running - if (mFastMixer != NULL) { - FastMixerStateQueue *sq = mFastMixer->sq(); - FastMixerState *state = sq->begin(); - if (state->mCommand != FastMixerState::MIX_WRITE && - (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { - if (state->mCommand == FastMixerState::COLD_IDLE) { - int32_t old = android_atomic_inc(&mFastMixerFutex); - if (old == -1) { - __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); - } -#ifdef AUDIO_WATCHDOG - if (mAudioWatchdog != 0) { - mAudioWatchdog->resume(); - } -#endif - } - state->mCommand = FastMixerState::MIX_WRITE; - sq->end(); - sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); - if (kUseFastMixer == FastMixer_Dynamic) { - mNormalSink = mPipeSink; - } - } else { - sq->end(false /*didModify*/); - } - } - PlaybackThread::threadLoop_write(); -} - -// shared by MIXER and DIRECT, overridden by DUPLICATING -void AudioFlinger::PlaybackThread::threadLoop_write() -{ - // FIXME rewrite to reduce number of system calls - mLastWriteTime = systemTime(); - mInWrite = true; - int bytesWritten; - - // If an NBAIO sink is present, use it to write the normal mixer's submix - if (mNormalSink != 0) { -#define mBitShift 2 // FIXME - size_t count = mixBufferSize >> mBitShift; -#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) - Tracer::traceBegin(ATRACE_TAG, "write"); -#endif - // update the setpoint when gScreenState changes - uint32_t screenState = gScreenState; - if (screenState != mScreenState) { - mScreenState = screenState; - MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); - if (pipe != NULL) { - pipe->setAvgFrames((mScreenState & 1) ? - (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); - } - } - ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); -#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) - Tracer::traceEnd(ATRACE_TAG); -#endif - if (framesWritten > 0) { - bytesWritten = framesWritten << mBitShift; - } else { - bytesWritten = framesWritten; - } - // otherwise use the HAL / AudioStreamOut directly - } else { - // Direct output thread. - bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); - } - - if (bytesWritten > 0) { - mBytesWritten += mixBufferSize; - } - mNumWrites++; - mInWrite = false; -} - -void AudioFlinger::MixerThread::threadLoop_standby() -{ - // Idle the fast mixer if it's currently running - if (mFastMixer != NULL) { - FastMixerStateQueue *sq = mFastMixer->sq(); - FastMixerState *state = sq->begin(); - if (!(state->mCommand & FastMixerState::IDLE)) { - state->mCommand = FastMixerState::COLD_IDLE; - state->mColdFutexAddr = &mFastMixerFutex; - state->mColdGen++; - mFastMixerFutex = 0; - sq->end(); - // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now - sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); - if (kUseFastMixer == FastMixer_Dynamic) { - mNormalSink = mOutputSink; - } -#ifdef AUDIO_WATCHDOG - if (mAudioWatchdog != 0) { - mAudioWatchdog->pause(); - } -#endif - } else { - sq->end(false /*didModify*/); - } - } - PlaybackThread::threadLoop_standby(); -} - -// shared by MIXER and DIRECT, overridden by DUPLICATING -void AudioFlinger::PlaybackThread::threadLoop_standby() -{ - ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); - mOutput->stream->common.standby(&mOutput->stream->common); -} - -void AudioFlinger::MixerThread::threadLoop_mix() -{ - // obtain the presentation timestamp of the next output buffer - int64_t pts; - status_t status = INVALID_OPERATION; - - if (mNormalSink != 0) { - status = mNormalSink->getNextWriteTimestamp(&pts); - } else { - status = mOutputSink->getNextWriteTimestamp(&pts); - } - - if (status != NO_ERROR) { - pts = AudioBufferProvider::kInvalidPTS; - } - - // mix buffers... - mAudioMixer->process(pts); - // increase sleep time progressively when application underrun condition clears. - // Only increase sleep time if the mixer is ready for two consecutive times to avoid - // that a steady state of alternating ready/not ready conditions keeps the sleep time - // such that we would underrun the audio HAL. - if ((sleepTime == 0) && (sleepTimeShift > 0)) { - sleepTimeShift--; - } - sleepTime = 0; - standbyTime = systemTime() + standbyDelay; - //TODO: delay standby when effects have a tail -} - -void AudioFlinger::MixerThread::threadLoop_sleepTime() -{ - // 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 (mMixerStatus == MIXER_TRACKS_ENABLED) { - sleepTime = activeSleepTime >> sleepTimeShift; - if (sleepTime < kMinThreadSleepTimeUs) { - sleepTime = kMinThreadSleepTimeUs; - } - // reduce sleep time in case of consecutive application underruns to avoid - // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer - // duration we would end up writing less data than needed by the audio HAL if - // the condition persists. - if (sleepTimeShift < kMaxThreadSleepTimeShift) { - sleepTimeShift++; - } - } else { - sleepTime = idleSleepTime; - } - } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { - memset (mMixBuffer, 0, mixBufferSize); - sleepTime = 0; - ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), - "anticipated start"); - } - // TODO add standby time extension fct of effect tail -} - -// prepareTracks_l() must be called with ThreadBase::mLock held -AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( - Vector< sp<Track> > *tracksToRemove) -{ - - mixer_state mixerStatus = MIXER_IDLE; - // find out which tracks need to be processed - size_t count = mActiveTracks.size(); - size_t mixedTracks = 0; - size_t tracksWithEffect = 0; - // counts only _active_ fast tracks - size_t fastTracks = 0; - uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset - - float masterVolume = mMasterVolume; - bool masterMute = mMasterMute; - - if (masterMute) { - masterVolume = 0; - } - // Delegate master volume control to effect in output mix effect chain if needed - sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); - if (chain != 0) { - uint32_t v = (uint32_t)(masterVolume * (1 << 24)); - chain->setVolume_l(&v, &v); - masterVolume = (float)((v + (1 << 23)) >> 24); - chain.clear(); - } - - // prepare a new state to push - FastMixerStateQueue *sq = NULL; - FastMixerState *state = NULL; - bool didModify = false; - FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; - if (mFastMixer != NULL) { - sq = mFastMixer->sq(); - state = sq->begin(); - } - - for (size_t i=0 ; i<count ; i++) { - sp<Track> t = mActiveTracks[i].promote(); - if (t == 0) { - continue; - } - - // this const just means the local variable doesn't change - Track* const track = t.get(); - - // process fast tracks - if (track->isFastTrack()) { - - // It's theoretically possible (though unlikely) for a fast track to be created - // and then removed within the same normal mix cycle. This is not a problem, as - // the track never becomes active so it's fast mixer slot is never touched. - // The converse, of removing an (active) track and then creating a new track - // at the identical fast mixer slot within the same normal mix cycle, - // is impossible because the slot isn't marked available until the end of each cycle. - int j = track->mFastIndex; - ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); - ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); - FastTrack *fastTrack = &state->mFastTracks[j]; - - // Determine whether the track is currently in underrun condition, - // and whether it had a recent underrun. - FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; - FastTrackUnderruns underruns = ftDump->mUnderruns; - uint32_t recentFull = (underruns.mBitFields.mFull - - track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; - uint32_t recentPartial = (underruns.mBitFields.mPartial - - track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; - uint32_t recentEmpty = (underruns.mBitFields.mEmpty - - track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; - uint32_t recentUnderruns = recentPartial + recentEmpty; - track->mObservedUnderruns = underruns; - // don't count underruns that occur while stopping or pausing - // or stopped which can occur when flush() is called while active - if (!(track->isStopping() || track->isPausing() || track->isStopped())) { - track->mUnderrunCount += recentUnderruns; - } - - // This is similar to the state machine for normal tracks, - // with a few modifications for fast tracks. - bool isActive = true; - switch (track->mState) { - case TrackBase::STOPPING_1: - // track stays active in STOPPING_1 state until first underrun - if (recentUnderruns > 0) { - track->mState = TrackBase::STOPPING_2; - } - break; - case TrackBase::PAUSING: - // ramp down is not yet implemented - track->setPaused(); - break; - case TrackBase::RESUMING: - // ramp up is not yet implemented - track->mState = TrackBase::ACTIVE; - break; - case TrackBase::ACTIVE: - if (recentFull > 0 || recentPartial > 0) { - // track has provided at least some frames recently: reset retry count - track->mRetryCount = kMaxTrackRetries; - } - if (recentUnderruns == 0) { - // no recent underruns: stay active - break; - } - // there has recently been an underrun of some kind - if (track->sharedBuffer() == 0) { - // were any of the recent underruns "empty" (no frames available)? - if (recentEmpty == 0) { - // no, then ignore the partial underruns as they are allowed indefinitely - break; - } - // there has recently been an "empty" underrun: decrement the retry counter - if (--(track->mRetryCount) > 0) { - break; - } - // indicate to client process that the track was disabled because of underrun; - // it will then automatically call start() when data is available - android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); - // remove from active list, but state remains ACTIVE [confusing but true] - isActive = false; - break; - } - // fall through - case TrackBase::STOPPING_2: - case TrackBase::PAUSED: - case TrackBase::TERMINATED: - case TrackBase::STOPPED: - case TrackBase::FLUSHED: // flush() while active - // Check for presentation complete if track is inactive - // We have consumed all the buffers of this track. - // This would be incomplete if we auto-paused on underrun - { - size_t audioHALFrames = - (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; - size_t framesWritten = mBytesWritten / mFrameSize; - if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { - // track stays in active list until presentation is complete - break; - } - } - if (track->isStopping_2()) { - track->mState = TrackBase::STOPPED; - } - if (track->isStopped()) { - // Can't reset directly, as fast mixer is still polling this track - // track->reset(); - // So instead mark this track as needing to be reset after push with ack - resetMask |= 1 << i; - } - isActive = false; - break; - case TrackBase::IDLE: - default: - LOG_FATAL("unexpected track state %d", track->mState); - } - - if (isActive) { - // was it previously inactive? - if (!(state->mTrackMask & (1 << j))) { - ExtendedAudioBufferProvider *eabp = track; - VolumeProvider *vp = track; - fastTrack->mBufferProvider = eabp; - fastTrack->mVolumeProvider = vp; - fastTrack->mSampleRate = track->mSampleRate; - fastTrack->mChannelMask = track->mChannelMask; - fastTrack->mGeneration++; - state->mTrackMask |= 1 << j; - didModify = true; - // no acknowledgement required for newly active tracks - } - // cache the combined master volume and stream type volume for fast mixer; this - // lacks any synchronization or barrier so VolumeProvider may read a stale value - track->mCachedVolume = track->isMuted() ? - 0 : masterVolume * mStreamTypes[track->streamType()].volume; - ++fastTracks; - } else { - // was it previously active? - if (state->mTrackMask & (1 << j)) { - fastTrack->mBufferProvider = NULL; - fastTrack->mGeneration++; - state->mTrackMask &= ~(1 << j); - didModify = true; - // If any fast tracks were removed, we must wait for acknowledgement - // because we're about to decrement the last sp<> on those tracks. - block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; - } else { - LOG_FATAL("fast track %d should have been active", j); - } - tracksToRemove->add(track); - // Avoids a misleading display in dumpsys - track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; - } - continue; - } - - { // local variable scope to avoid goto warning - - 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 - int name = track->name(); - // make sure that we have enough frames to mix one full buffer. - // enforce this condition only once to enable draining the buffer in case the client - // app does not call stop() and relies on underrun to stop: - // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed - // during last round - uint32_t minFrames = 1; - if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && - (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { - if (t->sampleRate() == mSampleRate) { - minFrames = mNormalFrameCount; - } else { - // +1 for rounding and +1 for additional sample needed for interpolation - minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; - // add frames already consumed but not yet released by the resampler - // because cblk->framesReady() will include these frames - minFrames += mAudioMixer->getUnreleasedFrames(track->name()); - // the minimum track buffer size is normally twice the number of frames necessary - // to fill one buffer and the resampler should not leave more than one buffer worth - // of unreleased frames after each pass, but just in case... - ALOG_ASSERT(minFrames <= cblk->frameCount); - } - } - if ((track->framesReady() >= minFrames) && track->isReady() && - !track->isPaused() && !track->isTerminated()) - { - ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, - this); - - mixedTracks++; - - // track->mainBuffer() != mMixBuffer means there is an effect chain - // connected to the track - chain.clear(); - if (track->mainBuffer() != mMixBuffer) { - chain = getEffectChain_l(track->sessionId()); - // Delegate volume control to effect in track effect chain if needed - if (chain != 0) { - tracksWithEffect++; - } else { - ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " - "session %d", - name, track->sessionId()); - } - } - - - int param = AudioMixer::VOLUME; - if (track->mFillingUpStatus == Track::FS_FILLED) { - // no ramp for the first volume setting - track->mFillingUpStatus = Track::FS_ACTIVE; - if (track->mState == TrackBase::RESUMING) { - track->mState = TrackBase::ACTIVE; - param = AudioMixer::RAMP_VOLUME; - } - mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); - } else if (cblk->server != 0) { - // If the track is stopped before the first frame was mixed, - // do not apply ramp - param = AudioMixer::RAMP_VOLUME; - } - - // compute volume for this track - uint32_t vl, vr, va; - if (track->isMuted() || track->isPausing() || - mStreamTypes[track->streamType()].mute) { - vl = vr = va = 0; - if (track->isPausing()) { - track->setPaused(); - } - } else { - - // read original volumes with volume control - float typeVolume = mStreamTypes[track->streamType()].volume; - float v = masterVolume * typeVolume; - uint32_t vlr = cblk->getVolumeLR(); - vl = vlr & 0xFFFF; - vr = vlr >> 16; - // track volumes come from shared memory, so can't be trusted and must be clamped - if (vl > MAX_GAIN_INT) { - ALOGV("Track left volume out of range: %04X", vl); - vl = MAX_GAIN_INT; - } - if (vr > MAX_GAIN_INT) { - ALOGV("Track right volume out of range: %04X", vr); - vr = MAX_GAIN_INT; - } - // now apply the master volume and stream type volume - vl = (uint32_t)(v * vl) << 12; - vr = (uint32_t)(v * vr) << 12; - // assuming master volume and stream type volume each go up to 1.0, - // vl and vr are now in 8.24 format - - uint16_t sendLevel = cblk->getSendLevel_U4_12(); - // send level comes from shared memory and so may be corrupt - if (sendLevel > MAX_GAIN_INT) { - ALOGV("Track send level out of range: %04X", sendLevel); - sendLevel = MAX_GAIN_INT; - } - va = (uint32_t)(v * sendLevel); - } - // Delegate volume control to effect in track effect chain if needed - if (chain != 0 && chain->setVolume_l(&vl, &vr)) { - // Do not ramp volume if volume is controlled by effect - param = AudioMixer::VOLUME; - track->mHasVolumeController = true; - } else { - // force no volume ramp when volume controller was just disabled or removed - // from effect chain to avoid volume spike - if (track->mHasVolumeController) { - param = AudioMixer::VOLUME; - } - track->mHasVolumeController = false; - } - - // Convert volumes from 8.24 to 4.12 format - // This additional clamping is needed in case chain->setVolume_l() overshot - vl = (vl + (1 << 11)) >> 12; - if (vl > MAX_GAIN_INT) { - vl = MAX_GAIN_INT; - } - vr = (vr + (1 << 11)) >> 12; - if (vr > MAX_GAIN_INT) { - vr = MAX_GAIN_INT; - } - - if (va > MAX_GAIN_INT) { - va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - - } - - // XXX: these things DON'T need to be done each time - mAudioMixer->setBufferProvider(name, track); - mAudioMixer->enable(name); - - mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); - mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); - mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); - mAudioMixer->setParameter( - name, - AudioMixer::TRACK, - AudioMixer::FORMAT, (void *)track->format()); - mAudioMixer->setParameter( - name, - AudioMixer::TRACK, - AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); - mAudioMixer->setParameter( - name, - AudioMixer::RESAMPLE, - AudioMixer::SAMPLE_RATE, - (void *)(cblk->sampleRate)); - mAudioMixer->setParameter( - name, - AudioMixer::TRACK, - AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); - mAudioMixer->setParameter( - name, - AudioMixer::TRACK, - AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); - - // reset retry count - track->mRetryCount = kMaxTrackRetries; - - // If one track is ready, set the mixer ready if: - // - the mixer was not ready during previous round OR - // - no other track is not ready - if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || - mixerStatus != MIXER_TRACKS_ENABLED) { - mixerStatus = MIXER_TRACKS_READY; - } - } else { - // clear effect chain input buffer if an active track underruns to avoid sending - // previous audio buffer again to effects - chain = getEffectChain_l(track->sessionId()); - if (chain != 0) { - chain->clearInputBuffer(); - } - - ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, - cblk->server, this); - if ((track->sharedBuffer() != 0) || track->isTerminated() || - track->isStopped() || track->isPaused()) { - // We have consumed all the buffers of this track. - // Remove it from the list of active tracks. - // TODO: use actual buffer filling status instead of latency when available from - // audio HAL - size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; - size_t framesWritten = mBytesWritten / mFrameSize; - if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { - if (track->isStopped()) { - track->reset(); - } - tracksToRemove->add(track); - } - } else { - track->mUnderrunCount++; - // No buffers for this track. Give it a few chances to - // fill a buffer, then remove it from active list. - if (--(track->mRetryCount) <= 0) { - ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); - tracksToRemove->add(track); - // indicate to client process that the track was disabled because of underrun; - // it will then automatically call start() when data is available - android_atomic_or(CBLK_DISABLED, &cblk->flags); - // If one track is not ready, mark the mixer also not ready if: - // - the mixer was ready during previous round OR - // - no other track is ready - } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || - mixerStatus != MIXER_TRACKS_READY) { - mixerStatus = MIXER_TRACKS_ENABLED; - } - } - mAudioMixer->disable(name); - } - - } // local variable scope to avoid goto warning -track_is_ready: ; - - } - - // Push the new FastMixer state if necessary - bool pauseAudioWatchdog = false; - if (didModify) { - state->mFastTracksGen++; - // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle - if (kUseFastMixer == FastMixer_Dynamic && - state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { - state->mCommand = FastMixerState::COLD_IDLE; - state->mColdFutexAddr = &mFastMixerFutex; - state->mColdGen++; - mFastMixerFutex = 0; - if (kUseFastMixer == FastMixer_Dynamic) { - mNormalSink = mOutputSink; - } - // If we go into cold idle, need to wait for acknowledgement - // so that fast mixer stops doing I/O. - block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; - pauseAudioWatchdog = true; - } - sq->end(); - } - if (sq != NULL) { - sq->end(didModify); - sq->push(block); - } -#ifdef AUDIO_WATCHDOG - if (pauseAudioWatchdog && mAudioWatchdog != 0) { - mAudioWatchdog->pause(); - } -#endif - - // Now perform the deferred reset on fast tracks that have stopped - while (resetMask != 0) { - size_t i = __builtin_ctz(resetMask); - ALOG_ASSERT(i < count); - resetMask &= ~(1 << i); - sp<Track> t = mActiveTracks[i].promote(); - if (t == 0) { - continue; - } - Track* track = t.get(); - ALOG_ASSERT(track->isFastTrack() && track->isStopped()); - track->reset(); - } - - // remove all the tracks that need to be... - count = tracksToRemove->size(); - if (CC_UNLIKELY(count)) { - for (size_t i=0 ; i<count ; i++) { - const sp<Track>& track = tracksToRemove->itemAt(i); - mActiveTracks.remove(track); - if (track->mainBuffer() != mMixBuffer) { - chain = getEffectChain_l(track->sessionId()); - if (chain != 0) { - ALOGV("stopping track on chain %p for session Id: %d", chain.get(), - track->sessionId()); - chain->decActiveTrackCnt(); - } - } - if (track->isTerminated()) { - removeTrack_l(track); - } - } - } - - // mix buffer must be cleared if all tracks are connected to an - // effect chain as in this case the mixer will not write to - // mix buffer and track effects will accumulate into it - if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || - (mixedTracks == 0 && fastTracks > 0)) { - // FIXME as a performance optimization, should remember previous zero status - memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); - } - - // if any fast tracks, then status is ready - mMixerStatusIgnoringFastTracks = mixerStatus; - if (fastTracks > 0) { - mixerStatus = MIXER_TRACKS_READY; - } - return mixerStatus; -} - -/* -The derived values that are cached: - - mixBufferSize from frame count * frame size - - activeSleepTime from activeSleepTimeUs() - - idleSleepTime from idleSleepTimeUs() - - standbyDelay from mActiveSleepTimeUs (DIRECT only) - - maxPeriod from frame count and sample rate (MIXER only) - -The parameters that affect these derived values are: - - frame count - - frame size - - sample rate - - device type: A2DP or not - - device latency - - format: PCM or not - - active sleep time - - idle sleep time -*/ - -void AudioFlinger::PlaybackThread::cacheParameters_l() -{ - mixBufferSize = mNormalFrameCount * mFrameSize; - activeSleepTime = activeSleepTimeUs(); - idleSleepTime = idleSleepTimeUs(); -} - -void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) -{ - ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", - this, streamType, mTracks.size()); - Mutex::Autolock _l(mLock); - - size_t size = mTracks.size(); - for (size_t i = 0; i < size; i++) { - sp<Track> t = mTracks[i]; - if (t->streamType() == streamType) { - android_atomic_or(CBLK_INVALID, &t->mCblk->flags); - t->mCblk->cv.signal(); - } - } -} - -// getTrackName_l() must be called with ThreadBase::mLock held -int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) -{ - return mAudioMixer->getTrackName(channelMask, sessionId); -} - -// deleteTrackName_l() must be called with ThreadBase::mLock held -void AudioFlinger::MixerThread::deleteTrackName_l(int name) -{ - ALOGV("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() -{ - // if !&IDLE, holds the FastMixer state to restore after new parameters processed - FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; - bool reconfig = false; - - while (!mNewParameters.isEmpty()) { - - if (mFastMixer != NULL) { - FastMixerStateQueue *sq = mFastMixer->sq(); - FastMixerState *state = sq->begin(); - if (!(state->mCommand & FastMixerState::IDLE)) { - previousCommand = state->mCommand; - state->mCommand = FastMixerState::HOT_IDLE; - sq->end(); - sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); - } else { - sq->end(false /*didModify*/); - } - } - - 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 ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { - status = BAD_VALUE; - } else { - reconfig = true; - } - } - if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { - if (value != AUDIO_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 guaranteed - // if frame count is changed after track creation - if (!mTracks.isEmpty()) { - status = INVALID_OPERATION; - } else { - reconfig = true; - } - } - if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { -#ifdef ADD_BATTERY_DATA - // when changing the audio output device, call addBatteryData to notify - // the change - if (mOutDevice != value) { - uint32_t params = 0; - // check whether speaker is on - if (value & AUDIO_DEVICE_OUT_SPEAKER) { - params |= IMediaPlayerService::kBatteryDataSpeakerOn; - } - - audio_devices_t deviceWithoutSpeaker - = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; - // check if any other device (except speaker) is on - if (value & deviceWithoutSpeaker ) { - params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; - } - - if (params != 0) { - addBatteryData(params); - } - } -#endif - - // forward device change to effects that have requested to be - // aware of attached audio device. - mOutDevice = value; - for (size_t i = 0; i < mEffectChains.size(); i++) { - mEffectChains[i]->setDevice_l(mOutDevice); - } - } - - if (status == NO_ERROR) { - status = mOutput->stream->common.set_parameters(&mOutput->stream->common, - keyValuePair.string()); - if (!mStandby && status == INVALID_OPERATION) { - mOutput->stream->common.standby(&mOutput->stream->common); - mStandby = true; - mBytesWritten = 0; - status = mOutput->stream->common.set_parameters(&mOutput->stream->common, - keyValuePair.string()); - } - if (status == NO_ERROR && reconfig) { - delete mAudioMixer; - // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) - mAudioMixer = NULL; - readOutputParameters(); - mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); - for (size_t i = 0; i < mTracks.size() ; i++) { - int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); - 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(); - } - } - sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); - } - } - - mNewParameters.removeAt(0); - - mParamStatus = status; - mParamCond.signal(); - // wait for condition with time out in case the thread calling ThreadBase::setParameters() - // already timed out waiting for the status and will never signal the condition. - mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); - } - - if (!(previousCommand & FastMixerState::IDLE)) { - ALOG_ASSERT(mFastMixer != NULL); - FastMixerStateQueue *sq = mFastMixer->sq(); - FastMixerState *state = sq->begin(); - ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); - state->mCommand = previousCommand; - sq->end(); - sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); - } - - return reconfig; -} - -void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) -{ - NBAIO_Source *teeSource = source.get(); - if (teeSource != NULL) { - char teeTime[16]; - struct timeval tv; - gettimeofday(&tv, NULL); - struct tm tm; - localtime_r(&tv.tv_sec, &tm); - strftime(teeTime, sizeof(teeTime), "%T", &tm); - char teePath[64]; - sprintf(teePath, "/data/misc/media/%s_%d.wav", teeTime, id); - int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); - if (teeFd >= 0) { - char wavHeader[44]; - memcpy(wavHeader, - "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0", - sizeof(wavHeader)); - NBAIO_Format format = teeSource->format(); - unsigned channelCount = Format_channelCount(format); - ALOG_ASSERT(channelCount <= FCC_2); - uint32_t sampleRate = Format_sampleRate(format); - wavHeader[22] = channelCount; // number of channels - wavHeader[24] = sampleRate; // sample rate - wavHeader[25] = sampleRate >> 8; - wavHeader[32] = channelCount * 2; // block alignment - write(teeFd, wavHeader, sizeof(wavHeader)); - size_t total = 0; - bool firstRead = true; - for (;;) { -#define TEE_SINK_READ 1024 - short buffer[TEE_SINK_READ * FCC_2]; - size_t count = TEE_SINK_READ; - ssize_t actual = teeSource->read(buffer, count, - AudioBufferProvider::kInvalidPTS); - bool wasFirstRead = firstRead; - firstRead = false; - if (actual <= 0) { - if (actual == (ssize_t) OVERRUN && wasFirstRead) { - continue; - } - break; - } - ALOG_ASSERT(actual <= (ssize_t)count); - write(teeFd, buffer, actual * channelCount * sizeof(short)); - total += actual; - } - lseek(teeFd, (off_t) 4, SEEK_SET); - uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; - write(teeFd, &temp, sizeof(temp)); - lseek(teeFd, (off_t) 40, SEEK_SET); - temp = total * channelCount * sizeof(short); - write(teeFd, &temp, sizeof(temp)); - close(teeFd); - fdprintf(fd, "FastMixer tee copied to %s\n", teePath); - } else { - fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); - } - } -} - -void 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()); - - // Make a non-atomic copy of fast mixer dump state so it won't change underneath us - FastMixerDumpState copy = mFastMixerDumpState; - copy.dump(fd); - -#ifdef STATE_QUEUE_DUMP - // Similar for state queue - StateQueueObserverDump observerCopy = mStateQueueObserverDump; - observerCopy.dump(fd); - StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; - mutatorCopy.dump(fd); -#endif - - // Write the tee output to a .wav file - dumpTee(fd, mTeeSource, mId); - -#ifdef AUDIO_WATCHDOG - if (mAudioWatchdog != 0) { - // Make a non-atomic copy of audio watchdog dump so it won't change underneath us - AudioWatchdogDump wdCopy = mAudioWatchdogDump; - wdCopy.dump(fd); - } -#endif -} - -uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const -{ - return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; -} - -uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const -{ - return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); -} - -void AudioFlinger::MixerThread::cacheParameters_l() -{ - PlaybackThread::cacheParameters_l(); - - // 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 - // increase threshold again due to low power audio mode. The way this warning - // threshold is calculated and its usefulness should be reconsidered anyway. - maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; -} - -// ---------------------------------------------------------------------------- -AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, - AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) - : PlaybackThread(audioFlinger, output, id, device, DIRECT) - // mLeftVolFloat, mRightVolFloat -{ -} - -AudioFlinger::DirectOutputThread::~DirectOutputThread() -{ -} - -AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( - Vector< sp<Track> > *tracksToRemove -) -{ - sp<Track> trackToRemove; - - mixer_state mixerStatus = MIXER_IDLE; - - // find out which tracks need to be processed - if (mActiveTracks.size() != 0) { - sp<Track> t = mActiveTracks[0].promote(); - // The track died recently - if (t == 0) { - return MIXER_IDLE; - } - - 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 - uint32_t minFrames; - if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { - minFrames = mNormalFrameCount; - } else { - minFrames = 1; - } - if ((track->framesReady() >= minFrames) && track->isReady() && - !track->isPaused() && !track->isTerminated()) - { - ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); - - if (track->mFillingUpStatus == Track::FS_FILLED) { - track->mFillingUpStatus = Track::FS_ACTIVE; - mLeftVolFloat = mRightVolFloat = 0; - if (track->mState == TrackBase::RESUMING) { - track->mState = TrackBase::ACTIVE; - } - } - - // compute volume for this track - float left, right; - if (track->isMuted() || mMasterMute || track->isPausing() || - mStreamTypes[track->streamType()].mute) { - left = right = 0; - if (track->isPausing()) { - track->setPaused(); - } - } else { - float typeVolume = mStreamTypes[track->streamType()].volume; - float v = mMasterVolume * typeVolume; - uint32_t vlr = cblk->getVolumeLR(); - float v_clamped = v * (vlr & 0xFFFF); - if (v_clamped > MAX_GAIN) { - v_clamped = MAX_GAIN; - } - left = v_clamped/MAX_GAIN; - v_clamped = v * (vlr >> 16); - if (v_clamped > MAX_GAIN) { - v_clamped = MAX_GAIN; - } - right = v_clamped/MAX_GAIN; - } - - if (left != mLeftVolFloat || right != mRightVolFloat) { - mLeftVolFloat = left; - mRightVolFloat = right; - - // Convert volumes from float to 8.24 - uint32_t vl = (uint32_t)(left * (1 << 24)); - uint32_t vr = (uint32_t)(right * (1 << 24)); - - // Delegate volume control to effect in track effect chain if needed - // only one effect chain can be present on DirectOutputThread, so if - // there is one, the track is connected to it - if (!mEffectChains.isEmpty()) { - // Do not ramp volume if volume is controlled by effect - mEffectChains[0]->setVolume_l(&vl, &vr); - left = (float)vl / (1 << 24); - right = (float)vr / (1 << 24); - } - mOutput->stream->set_volume(mOutput->stream, left, right); - } - - // reset retry count - track->mRetryCount = kMaxTrackRetriesDirect; - mActiveTrack = t; - mixerStatus = MIXER_TRACKS_READY; - } else { - // clear effect chain input buffer if an active track underruns to avoid sending - // previous audio buffer again to effects - if (!mEffectChains.isEmpty()) { - mEffectChains[0]->clearInputBuffer(); - } - - ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); - if ((track->sharedBuffer() != 0) || track->isTerminated() || - track->isStopped() || track->isPaused()) { - // We have consumed all the buffers of this track. - // Remove it from the list of active tracks. - // TODO: implement behavior for compressed audio - size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; - size_t framesWritten = mBytesWritten / mFrameSize; - if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { - if (track->isStopped()) { - track->reset(); - } - 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) { - ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); - trackToRemove = track; - } else { - mixerStatus = MIXER_TRACKS_ENABLED; - } - } - } - } - - // FIXME merge this with similar code for removing multiple tracks - // remove all the tracks that need to be... - if (CC_UNLIKELY(trackToRemove != 0)) { - tracksToRemove->add(trackToRemove); - mActiveTracks.remove(trackToRemove); - if (!mEffectChains.isEmpty()) { - ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), - trackToRemove->sessionId()); - mEffectChains[0]->decActiveTrackCnt(); - } - if (trackToRemove->isTerminated()) { - removeTrack_l(trackToRemove); - } - } - - return mixerStatus; -} - -void AudioFlinger::DirectOutputThread::threadLoop_mix() -{ - AudioBufferProvider::Buffer buffer; - size_t frameCount = mFrameCount; - int8_t *curBuf = (int8_t *)mMixBuffer; - // output audio to hardware - while (frameCount) { - buffer.frameCount = frameCount; - mActiveTrack->getNextBuffer(&buffer); - if (CC_UNLIKELY(buffer.raw == NULL)) { - memset(curBuf, 0, frameCount * mFrameSize); - break; - } - memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); - frameCount -= buffer.frameCount; - curBuf += buffer.frameCount * mFrameSize; - mActiveTrack->releaseBuffer(&buffer); - } - sleepTime = 0; - standbyTime = systemTime() + standbyDelay; - mActiveTrack.clear(); - -} - -void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() -{ - if (sleepTime == 0) { - if (mMixerStatus == MIXER_TRACKS_ENABLED) { - sleepTime = activeSleepTime; - } else { - sleepTime = idleSleepTime; - } - } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { - memset(mMixBuffer, 0, mFrameCount * mFrameSize); - sleepTime = 0; - } -} - -// getTrackName_l() must be called with ThreadBase::mLock held -int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, - int sessionId) -{ - 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->stream->common.set_parameters(&mOutput->stream->common, - keyValuePair.string()); - if (!mStandby && status == INVALID_OPERATION) { - mOutput->stream->common.standby(&mOutput->stream->common); - mStandby = true; - mBytesWritten = 0; - status = mOutput->stream->common.set_parameters(&mOutput->stream->common, - keyValuePair.string()); - } - if (status == NO_ERROR && reconfig) { - readOutputParameters(); - sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); - } - } - - mNewParameters.removeAt(0); - - mParamStatus = status; - mParamCond.signal(); - // wait for condition with time out in case the thread calling ThreadBase::setParameters() - // already timed out waiting for the status and will never signal the condition. - mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); - } - return reconfig; -} - -uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const -{ - uint32_t time; - if (audio_is_linear_pcm(mFormat)) { - time = PlaybackThread::activeSleepTimeUs(); - } else { - time = 10000; - } - return time; -} - -uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const -{ - uint32_t time; - if (audio_is_linear_pcm(mFormat)) { - time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; - } else { - time = 10000; - } - return time; -} - -uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const -{ - uint32_t time; - if (audio_is_linear_pcm(mFormat)) { - time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); - } else { - time = 10000; - } - return time; -} - -void AudioFlinger::DirectOutputThread::cacheParameters_l() -{ - PlaybackThread::cacheParameters_l(); - - // use shorter standby delay as on normal output to release - // hardware resources as soon as possible - standbyDelay = microseconds(activeSleepTime*2); -} - -// ---------------------------------------------------------------------------- - -AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, - AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) - : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), - DUPLICATING), - mWaitTimeMs(UINT_MAX) -{ - addOutputTrack(mainThread); -} - -AudioFlinger::DuplicatingThread::~DuplicatingThread() -{ - for (size_t i = 0; i < mOutputTracks.size(); i++) { - mOutputTracks[i]->destroy(); - } -} - -void AudioFlinger::DuplicatingThread::threadLoop_mix() -{ - // mix buffers... - if (outputsReady(outputTracks)) { - mAudioMixer->process(AudioBufferProvider::kInvalidPTS); - } else { - memset(mMixBuffer, 0, mixBufferSize); - } - sleepTime = 0; - writeFrames = mNormalFrameCount; - standbyTime = systemTime() + standbyDelay; -} - -void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() -{ - if (sleepTime == 0) { - if (mMixerStatus == MIXER_TRACKS_ENABLED) { - sleepTime = activeSleepTime; - } else { - sleepTime = idleSleepTime; - } - } else if (mBytesWritten != 0) { - if (mMixerStatus == MIXER_TRACKS_ENABLED) { - writeFrames = mNormalFrameCount; - memset(mMixBuffer, 0, mixBufferSize); - } else { - // flush remaining overflow buffers in output tracks - writeFrames = 0; - } - sleepTime = 0; - } -} - -void AudioFlinger::DuplicatingThread::threadLoop_write() -{ - for (size_t i = 0; i < outputTracks.size(); i++) { - outputTracks[i]->write(mMixBuffer, writeFrames); - } - mBytesWritten += mixBufferSize; -} - -void AudioFlinger::DuplicatingThread::threadLoop_standby() -{ - // DuplicatingThread implements standby by stopping all tracks - for (size_t i = 0; i < outputTracks.size(); i++) { - outputTracks[i]->stop(); - } -} - -void AudioFlinger::DuplicatingThread::saveOutputTracks() -{ - outputTracks = mOutputTracks; -} - -void AudioFlinger::DuplicatingThread::clearOutputTracks() -{ - outputTracks.clear(); -} - -void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) -{ - Mutex::Autolock _l(mLock); - // FIXME explain this formula - size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); - OutputTrack *outputTrack = new OutputTrack(thread, - this, - mSampleRate, - mFormat, - mChannelMask, - frameCount); - if (outputTrack->cblk() != NULL) { - thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); - mOutputTracks.add(outputTrack); - ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); - updateWaitTime_l(); - } -} - -void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) -{ - Mutex::Autolock _l(mLock); - for (size_t i = 0; i < mOutputTracks.size(); i++) { - if (mOutputTracks[i]->thread() == thread) { - mOutputTracks[i]->destroy(); - mOutputTracks.removeAt(i); - updateWaitTime_l(); - return; - } - } - ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); -} - -// caller must hold mLock -void AudioFlinger::DuplicatingThread::updateWaitTime_l() -{ - mWaitTimeMs = UINT_MAX; - for (size_t i = 0; i < mOutputTracks.size(); i++) { - sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); - if (strong != 0) { - uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); - if (waitTimeMs < mWaitTimeMs) { - mWaitTimeMs = waitTimeMs; - } - } - } -} - - -bool AudioFlinger::DuplicatingThread::outputsReady( - const SortedVector< sp<OutputTrack> > &outputTracks) -{ - for (size_t i = 0; i < outputTracks.size(); i++) { - sp<ThreadBase> thread = outputTracks[i]->thread().promote(); - if (thread == 0) { - ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", - outputTracks[i].get()); - return false; - } - PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); - // see note at standby() declaration - if (playbackThread->standby() && !playbackThread->isSuspended()) { - ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), - thread.get()); - return false; - } - } - return true; -} - -uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const -{ - return (mWaitTimeMs * 1000) / 2; -} - -void AudioFlinger::DuplicatingThread::cacheParameters_l() -{ - // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first - updateWaitTime_l(); - - MixerThread::cacheParameters_l(); -} - -// ---------------------------------------------------------------------------- - -// TrackBase constructor must be called with AudioFlinger::mLock held -AudioFlinger::ThreadBase::TrackBase::TrackBase( - ThreadBase *thread, - const sp<Client>& client, - uint32_t sampleRate, - audio_format_t format, - audio_channel_mask_t channelMask, - size_t frameCount, - const sp<IMemory>& sharedBuffer, - int sessionId) - : RefBase(), - mThread(thread), - mClient(client), - mCblk(NULL), - // mBuffer - // mBufferEnd - mStepCount(0), - mState(IDLE), - mSampleRate(sampleRate), - mFormat(format), - mChannelMask(channelMask), - mChannelCount(popcount(channelMask)), - mFrameSize(audio_is_linear_pcm(format) ? - mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), - mFrameCount(frameCount), - mStepServerFailed(false), - mSessionId(sessionId) -{ - // client == 0 implies sharedBuffer == 0 - ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); - - ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), - sharedBuffer->size()); - - // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); - size_t size = sizeof(audio_track_cblk_t); - size_t bufferSize = frameCount * mFrameSize; - if (sharedBuffer == 0) { - size += bufferSize; - } - - if (client != 0) { - mCblkMemory = client->heap()->allocate(size); - if (mCblkMemory != 0) { - mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); - // can't assume mCblk != NULL - } else { - ALOGE("not enough memory for AudioTrack size=%u", size); - client->heap()->dump("AudioTrack"); - return; - } - } else { - mCblk = (audio_track_cblk_t *)(new uint8_t[size]); - // assume mCblk != NULL - } - - // construct the shared structure in-place. - if (mCblk != NULL) { - new(mCblk) audio_track_cblk_t(); - // clear all buffers - mCblk->frameCount_ = frameCount; - mCblk->sampleRate = sampleRate; -// uncomment the following lines to quickly test 32-bit wraparound -// mCblk->user = 0xffff0000; -// mCblk->server = 0xffff0000; -// mCblk->userBase = 0xffff0000; -// mCblk->serverBase = 0xffff0000; - if (sharedBuffer == 0) { - mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); - memset(mBuffer, 0, bufferSize); - // Force underrun condition to avoid false underrun callback until first data is - // written to buffer (other flags are cleared) - mCblk->flags = CBLK_UNDERRUN; - } else { - mBuffer = sharedBuffer->pointer(); - } - mBufferEnd = (uint8_t *)mBuffer + bufferSize; - } -} - -AudioFlinger::ThreadBase::TrackBase::~TrackBase() -{ - if (mCblk != NULL) { - if (mClient == 0) { - delete mCblk; - } else { - mCblk->~audio_track_cblk_t(); // destroy our shared-structure. - } - } - mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to - if (mClient != 0) { - // Client destructor must run with AudioFlinger mutex locked - Mutex::Autolock _l(mClient->audioFlinger()->mLock); - // If the client's reference count drops to zero, the associated destructor - // must run with AudioFlinger lock held. Thus the explicit clear() rather than - // relying on the automatic clear() at end of scope. - mClient.clear(); - } -} - -// AudioBufferProvider interface -// getNextBuffer() = 0; -// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack -void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) -{ - buffer->raw = NULL; - mStepCount = buffer->frameCount; - // FIXME See note at getNextBuffer() - (void) step(); // ignore return value of step() - buffer->frameCount = 0; -} - -bool AudioFlinger::ThreadBase::TrackBase::step() { - bool result; - audio_track_cblk_t* cblk = this->cblk(); - - result = cblk->stepServer(mStepCount, mFrameCount, isOut()); - if (!result) { - ALOGV("stepServer failed acquiring cblk mutex"); - mStepServerFailed = true; - } - 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; - mStepServerFailed = false; - ALOGV("TrackBase::reset"); -} - -uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const { - return mCblk->sampleRate; -} - -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) * mFrameSize; - int8_t *bufferEnd = bufferStart + frames * mFrameSize; - - // Check validity of returned pointer in case the track control block would have been corrupted. - ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), - "TrackBase::getBuffer buffer out of range:\n" - " start: %p, end %p , mBuffer %p mBufferEnd %p\n" - " server %u, serverBase %u, user %u, userBase %u, frameSize %u", - bufferStart, bufferEnd, mBuffer, mBufferEnd, - cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize); - - return bufferStart; -} - -status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) -{ - mSyncEvents.add(event); - return NO_ERROR; -} - -// ---------------------------------------------------------------------------- - -// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held -AudioFlinger::PlaybackThread::Track::Track( - PlaybackThread *thread, - const sp<Client>& client, - audio_stream_type_t streamType, - uint32_t sampleRate, - audio_format_t format, - audio_channel_mask_t channelMask, - size_t frameCount, - const sp<IMemory>& sharedBuffer, - int sessionId, - IAudioFlinger::track_flags_t flags) - : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, - sessionId), - mMute(false), - mFillingUpStatus(FS_INVALID), - // mRetryCount initialized later when needed - mSharedBuffer(sharedBuffer), - mStreamType(streamType), - mName(-1), // see note below - mMainBuffer(thread->mixBuffer()), - mAuxBuffer(NULL), - mAuxEffectId(0), mHasVolumeController(false), - mPresentationCompleteFrames(0), - mFlags(flags), - mFastIndex(-1), - mUnderrunCount(0), - mCachedVolume(1.0) -{ - if (mCblk != NULL) { - // to avoid leaking a track name, do not allocate one unless there is an mCblk - mName = thread->getTrackName_l(channelMask, sessionId); - mCblk->mName = mName; - if (mName < 0) { - ALOGE("no more track names available"); - return; - } - // only allocate a fast track index if we were able to allocate a normal track name - if (flags & IAudioFlinger::TRACK_FAST) { - ALOG_ASSERT(thread->mFastTrackAvailMask != 0); - int i = __builtin_ctz(thread->mFastTrackAvailMask); - ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); - // FIXME This is too eager. We allocate a fast track index before the - // fast track becomes active. Since fast tracks are a scarce resource, - // this means we are potentially denying other more important fast tracks from - // being created. It would be better to allocate the index dynamically. - mFastIndex = i; - mCblk->mName = i; - // Read the initial underruns because this field is never cleared by the fast mixer - mObservedUnderruns = thread->getFastTrackUnderruns(i); - thread->mFastTrackAvailMask &= ~(1 << i); - } - } - ALOGV("Track constructor name %d, calling pid %d", mName, - IPCThreadState::self()->getCallingPid()); -} - -AudioFlinger::PlaybackThread::Track::~Track() -{ - ALOGV("PlaybackThread::Track destructor"); -} - -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 - // destructor 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(), mStreamType, mSessionId); - -#ifdef ADD_BATTERY_DATA - // to track the speaker usage - addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); -#endif - } - AudioSystem::releaseOutput(thread->id()); - } - Mutex::Autolock _l(thread->mLock); - PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); - playbackThread->destroyTrack_l(this); - } - } -} - -/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) -{ - result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S M F SRate " - "L dB R dB Server User Main buf Aux Buf Flags Underruns\n"); -} - -void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) -{ - uint32_t vlr = mCblk->getVolumeLR(); - if (isFastTrack()) { - sprintf(buffer, " F %2d", mFastIndex); - } else { - sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); - } - track_state state = mState; - char stateChar; - switch (state) { - case IDLE: - stateChar = 'I'; - break; - case TERMINATED: - stateChar = 'T'; - break; - case STOPPING_1: - stateChar = 's'; - break; - case STOPPING_2: - stateChar = '5'; - break; - case STOPPED: - stateChar = 'S'; - break; - case RESUMING: - stateChar = 'R'; - break; - case ACTIVE: - stateChar = 'A'; - break; - case PAUSING: - stateChar = 'p'; - break; - case PAUSED: - stateChar = 'P'; - break; - case FLUSHED: - stateChar = 'F'; - break; - default: - stateChar = '?'; - break; - } - char nowInUnderrun; - switch (mObservedUnderruns.mBitFields.mMostRecent) { - case UNDERRUN_FULL: - nowInUnderrun = ' '; - break; - case UNDERRUN_PARTIAL: - nowInUnderrun = '<'; - break; - case UNDERRUN_EMPTY: - nowInUnderrun = '*'; - break; - default: - nowInUnderrun = '?'; - break; - } - snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " - "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", - (mClient == 0) ? getpid_cached : mClient->pid(), - mStreamType, - mFormat, - mChannelMask, - mSessionId, - mStepCount, - mFrameCount, - stateChar, - mMute, - mFillingUpStatus, - mCblk->sampleRate, - 20.0 * log10((vlr & 0xFFFF) / 4096.0), - 20.0 * log10((vlr >> 16) / 4096.0), - mCblk->server, - mCblk->user, - (int)mMainBuffer, - (int)mAuxBuffer, - mCblk->flags, - mUnderrunCount, - nowInUnderrun); -} - -// AudioBufferProvider interface -status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( - AudioBufferProvider::Buffer* buffer, int64_t pts) -{ - 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 (mStepServerFailed) { - // FIXME When called by fast mixer, this takes a mutex with tryLock(). - // Since the fast mixer is higher priority than client callback thread, - // it does not result in priority inversion for client. - // But a non-blocking solution would be preferable to avoid - // fast mixer being unable to tryLock(), and - // to avoid the extra context switches if the client wakes up, - // discovers the mutex is locked, then has to wait for fast mixer to unlock. - if (!step()) goto getNextBuffer_exit; - ALOGV("stepServer recovered"); - mStepServerFailed = false; - } - - // FIXME Same as above - framesReady = cblk->framesReadyOut(); - - if (CC_LIKELY(framesReady)) { - uint32_t s = cblk->server; - uint32_t bufferEnd = cblk->serverBase + mFrameCount; - - bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; - if (framesReq > framesReady) { - framesReq = framesReady; - } - if (framesReq > bufferEnd - s) { - framesReq = bufferEnd - s; - } - - buffer->raw = getBuffer(s, framesReq); - buffer->frameCount = framesReq; - return NO_ERROR; - } - -getNextBuffer_exit: - buffer->raw = NULL; - buffer->frameCount = 0; - ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); - return NOT_ENOUGH_DATA; -} - -// Note that framesReady() takes a mutex on the control block using tryLock(). -// This could result in priority inversion if framesReady() is called by the normal mixer, -// as the normal mixer thread runs at lower -// priority than the client's callback thread: there is a short window within framesReady() -// during which the normal mixer could be preempted, and the client callback would block. -// Another problem can occur if framesReady() is called by the fast mixer: -// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. -// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. -size_t AudioFlinger::PlaybackThread::Track::framesReady() const { - return mCblk->framesReadyOut(); -} - -// Don't call for fast tracks; the framesReady() could result in priority inversion -bool AudioFlinger::PlaybackThread::Track::isReady() const { - if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { - return true; - } - - if (framesReady() >= mFrameCount || - (mCblk->flags & CBLK_FORCEREADY)) { - mFillingUpStatus = FS_FILLED; - android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); - return true; - } - return false; -} - -status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, - int triggerSession) -{ - status_t status = NO_ERROR; - ALOGV("start(%d), calling pid %d session %d", - mName, IPCThreadState::self()->getCallingPid(), mSessionId); - - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - Mutex::Autolock _l(thread->mLock); - track_state state = mState; - // here the track could be either new, or restarted - // in both cases "unstop" the track - if (mState == PAUSED) { - mState = TrackBase::RESUMING; - ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); - } else { - mState = TrackBase::ACTIVE; - ALOGV("? => ACTIVE (%d) on thread %p", mName, this); - } - - if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { - thread->mLock.unlock(); - status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); - thread->mLock.lock(); - -#ifdef ADD_BATTERY_DATA - // to track the speaker usage - if (status == NO_ERROR) { - addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); - } -#endif - } - if (status == NO_ERROR) { - PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); - playbackThread->addTrack_l(this); - } else { - mState = state; - triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); - } - } else { - status = BAD_VALUE; - } - return status; -} - -void AudioFlinger::PlaybackThread::Track::stop() -{ - ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - Mutex::Autolock _l(thread->mLock); - track_state state = mState; - if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { - // If the track is not active (PAUSED and buffers full), flush buffers - PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); - if (playbackThread->mActiveTracks.indexOf(this) < 0) { - reset(); - mState = STOPPED; - } else if (!isFastTrack()) { - mState = STOPPED; - } else { - // prepareTracks_l() will set state to STOPPING_2 after next underrun, - // and then to STOPPED and reset() when presentation is complete - mState = STOPPING_1; - } - ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, - playbackThread); - } - if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { - thread->mLock.unlock(); - AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); - thread->mLock.lock(); - -#ifdef ADD_BATTERY_DATA - // to track the speaker usage - addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); -#endif - } - } -} - -void AudioFlinger::PlaybackThread::Track::pause() -{ - ALOGV("pause(%d), calling pid %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; - ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); - if (!isOutputTrack()) { - thread->mLock.unlock(); - AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); - thread->mLock.lock(); - -#ifdef ADD_BATTERY_DATA - // to track the speaker usage - addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); -#endif - } - } - } -} - -void AudioFlinger::PlaybackThread::Track::flush() -{ - ALOGV("flush(%d)", mName); - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - Mutex::Autolock _l(thread->mLock); - if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && - mState != PAUSING && mState != IDLE && mState != FLUSHED) { - return; - } - // No point remaining in PAUSED state after a flush => go to - // FLUSHED state - mState = FLUSHED; - // do not reset the track if it is still in the process of being stopped or paused. - // this will be done by prepareTracks_l() when the track is stopped. - // prepareTracks_l() will see mState == FLUSHED, then - // remove from active track list, reset(), and trigger presentation complete - PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); - if (playbackThread->mActiveTracks.indexOf(this) < 0) { - reset(); - } - } -} - -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 - android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); - android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); - mFillingUpStatus = FS_FILLING; - mResetDone = true; - if (mState == FLUSHED) { - mState = IDLE; - } - } -} - -void AudioFlinger::PlaybackThread::Track::mute(bool muted) -{ - mMute = muted; -} - -status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) -{ - status_t status = DEAD_OBJECT; - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); - sp<AudioFlinger> af = mClient->audioFlinger(); - - Mutex::Autolock _l(af->mLock); - - sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); - - if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { - Mutex::Autolock _dl(playbackThread->mLock); - Mutex::Autolock _sl(srcThread->mLock); - sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); - if (chain == 0) { - return INVALID_OPERATION; - } - - sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); - if (effect == 0) { - return INVALID_OPERATION; - } - srcThread->removeEffect_l(effect); - playbackThread->addEffect_l(effect); - // removeEffect_l() has stopped the effect if it was active so it must be restarted - if (effect->state() == EffectModule::ACTIVE || - effect->state() == EffectModule::STOPPING) { - effect->start(); - } - - sp<EffectChain> dstChain = effect->chain().promote(); - if (dstChain == 0) { - srcThread->addEffect_l(effect); - return INVALID_OPERATION; - } - AudioSystem::unregisterEffect(effect->id()); - AudioSystem::registerEffect(&effect->desc(), - srcThread->id(), - dstChain->strategy(), - AUDIO_SESSION_OUTPUT_MIX, - effect->id()); - } - status = playbackThread->attachAuxEffect(this, EffectId); - } - return status; -} - -void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) -{ - mAuxEffectId = EffectId; - mAuxBuffer = buffer; -} - -bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, - size_t audioHalFrames) -{ - // a track is considered presented when the total number of frames written to audio HAL - // corresponds to the number of frames written when presentationComplete() is called for the - // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. - if (mPresentationCompleteFrames == 0) { - mPresentationCompleteFrames = framesWritten + audioHalFrames; - ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", - mPresentationCompleteFrames, audioHalFrames); - } - if (framesWritten >= mPresentationCompleteFrames) { - ALOGV("presentationComplete() session %d complete: framesWritten %d", - mSessionId, framesWritten); - triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); - return true; - } - return false; -} - -void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) -{ - for (int i = 0; i < (int)mSyncEvents.size(); i++) { - if (mSyncEvents[i]->type() == type) { - mSyncEvents[i]->trigger(); - mSyncEvents.removeAt(i); - i--; - } - } -} - -// implement VolumeBufferProvider interface - -uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() -{ - // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs - ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); - uint32_t vlr = mCblk->getVolumeLR(); - uint32_t vl = vlr & 0xFFFF; - uint32_t vr = vlr >> 16; - // track volumes come from shared memory, so can't be trusted and must be clamped - if (vl > MAX_GAIN_INT) { - vl = MAX_GAIN_INT; - } - if (vr > MAX_GAIN_INT) { - vr = MAX_GAIN_INT; - } - // now apply the cached master volume and stream type volume; - // this is trusted but lacks any synchronization or barrier so may be stale - float v = mCachedVolume; - vl *= v; - vr *= v; - // re-combine into U4.16 - vlr = (vr << 16) | (vl & 0xFFFF); - // FIXME look at mute, pause, and stop flags - return vlr; -} - -status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) -{ - if (mState == TERMINATED || mState == PAUSED || - ((framesReady() == 0) && ((mSharedBuffer != 0) || - (mState == STOPPED)))) { - ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", - mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); - event->cancel(); - return INVALID_OPERATION; - } - (void) TrackBase::setSyncEvent(event); - return NO_ERROR; -} - -bool AudioFlinger::PlaybackThread::Track::isOut() const -{ - return true; -} - -// timed audio tracks - -sp<AudioFlinger::PlaybackThread::TimedTrack> -AudioFlinger::PlaybackThread::TimedTrack::create( - PlaybackThread *thread, - const sp<Client>& client, - audio_stream_type_t streamType, - uint32_t sampleRate, - audio_format_t format, - audio_channel_mask_t channelMask, - size_t frameCount, - const sp<IMemory>& sharedBuffer, - int sessionId) { - if (!client->reserveTimedTrack()) - return 0; - - return new TimedTrack( - thread, client, streamType, sampleRate, format, channelMask, frameCount, - sharedBuffer, sessionId); -} - -AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( - PlaybackThread *thread, - const sp<Client>& client, - audio_stream_type_t streamType, - uint32_t sampleRate, - audio_format_t format, - audio_channel_mask_t channelMask, - size_t frameCount, - const sp<IMemory>& sharedBuffer, - int sessionId) - : Track(thread, client, streamType, sampleRate, format, channelMask, - frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), - mQueueHeadInFlight(false), - mTrimQueueHeadOnRelease(false), - mFramesPendingInQueue(0), - mTimedSilenceBuffer(NULL), - mTimedSilenceBufferSize(0), - mTimedAudioOutputOnTime(false), - mMediaTimeTransformValid(false) -{ - LocalClock lc; - mLocalTimeFreq = lc.getLocalFreq(); - - mLocalTimeToSampleTransform.a_zero = 0; - mLocalTimeToSampleTransform.b_zero = 0; - mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; - mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; - LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, - &mLocalTimeToSampleTransform.a_to_b_denom); - - mMediaTimeToSampleTransform.a_zero = 0; - mMediaTimeToSampleTransform.b_zero = 0; - mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; - mMediaTimeToSampleTransform.a_to_b_denom = 1000000; - LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, - &mMediaTimeToSampleTransform.a_to_b_denom); -} - -AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { - mClient->releaseTimedTrack(); - delete [] mTimedSilenceBuffer; -} - -status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( - size_t size, sp<IMemory>* buffer) { - - Mutex::Autolock _l(mTimedBufferQueueLock); - - trimTimedBufferQueue_l(); - - // lazily initialize the shared memory heap for timed buffers - if (mTimedMemoryDealer == NULL) { - const int kTimedBufferHeapSize = 512 << 10; - - mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, - "AudioFlingerTimed"); - if (mTimedMemoryDealer == NULL) - return NO_MEMORY; - } - - sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); - if (newBuffer == NULL) { - newBuffer = mTimedMemoryDealer->allocate(size); - if (newBuffer == NULL) - return NO_MEMORY; - } - - *buffer = newBuffer; - return NO_ERROR; -} - -// caller must hold mTimedBufferQueueLock -void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { - int64_t mediaTimeNow; - { - Mutex::Autolock mttLock(mMediaTimeTransformLock); - if (!mMediaTimeTransformValid) - return; - - int64_t targetTimeNow; - status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) - ? mCCHelper.getCommonTime(&targetTimeNow) - : mCCHelper.getLocalTime(&targetTimeNow); - - if (OK != res) - return; - - if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, - &mediaTimeNow)) { - return; - } - } - - size_t trimEnd; - for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { - int64_t bufEnd; - - if ((trimEnd + 1) < mTimedBufferQueue.size()) { - // We have a next buffer. Just use its PTS as the PTS of the frame - // following the last frame in this buffer. If the stream is sparse - // (ie, there are deliberate gaps left in the stream which should be - // filled with silence by the TimedAudioTrack), then this can result - // in one extra buffer being left un-trimmed when it could have - // been. In general, this is not typical, and we would rather - // optimized away the TS calculation below for the more common case - // where PTSes are contiguous. - bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); - } else { - // We have no next buffer. Compute the PTS of the frame following - // the last frame in this buffer by computing the duration of of - // this frame in media time units and adding it to the PTS of the - // buffer. - int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() - / mFrameSize; - - if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, - &bufEnd)) { - ALOGE("Failed to convert frame count of %lld to media time" - " duration" " (scale factor %d/%u) in %s", - frameCount, - mMediaTimeToSampleTransform.a_to_b_numer, - mMediaTimeToSampleTransform.a_to_b_denom, - __PRETTY_FUNCTION__); - break; - } - bufEnd += mTimedBufferQueue[trimEnd].pts(); - } - - if (bufEnd > mediaTimeNow) - break; - - // Is the buffer we want to use in the middle of a mix operation right - // now? If so, don't actually trim it. Just wait for the releaseBuffer - // from the mixer which should be coming back shortly. - if (!trimEnd && mQueueHeadInFlight) { - mTrimQueueHeadOnRelease = true; - } - } - - size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; - if (trimStart < trimEnd) { - // Update the bookkeeping for framesReady() - for (size_t i = trimStart; i < trimEnd; ++i) { - updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); - } - - // Now actually remove the buffers from the queue. - mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); - } -} - -void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( - const char* logTag) { - ALOG_ASSERT(mTimedBufferQueue.size() > 0, - "%s called (reason \"%s\"), but timed buffer queue has no" - " elements to trim.", __FUNCTION__, logTag); - - updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); - mTimedBufferQueue.removeAt(0); -} - -void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( - const TimedBuffer& buf, - const char* logTag) { - uint32_t bufBytes = buf.buffer()->size(); - uint32_t consumedAlready = buf.position(); - - ALOG_ASSERT(consumedAlready <= bufBytes, - "Bad bookkeeping while updating frames pending. Timed buffer is" - " only %u bytes long, but claims to have consumed %u" - " bytes. (update reason: \"%s\")", - bufBytes, consumedAlready, logTag); - - uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; - ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, - "Bad bookkeeping while updating frames pending. Should have at" - " least %u queued frames, but we think we have only %u. (update" - " reason: \"%s\")", - bufFrames, mFramesPendingInQueue, logTag); - - mFramesPendingInQueue -= bufFrames; -} - -status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( - const sp<IMemory>& buffer, int64_t pts) { - - { - Mutex::Autolock mttLock(mMediaTimeTransformLock); - if (!mMediaTimeTransformValid) - return INVALID_OPERATION; - } - - Mutex::Autolock _l(mTimedBufferQueueLock); - - uint32_t bufFrames = buffer->size() / mFrameSize; - mFramesPendingInQueue += bufFrames; - mTimedBufferQueue.add(TimedBuffer(buffer, pts)); - - return NO_ERROR; -} - -status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( - const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { - - ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", - xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, - target); - - if (!(target == TimedAudioTrack::LOCAL_TIME || - target == TimedAudioTrack::COMMON_TIME)) { - return BAD_VALUE; - } - - Mutex::Autolock lock(mMediaTimeTransformLock); - mMediaTimeTransform = xform; - mMediaTimeTransformTarget = target; - mMediaTimeTransformValid = true; - - return NO_ERROR; -} - -#define min(a, b) ((a) < (b) ? (a) : (b)) - -// implementation of getNextBuffer for tracks whose buffers have timestamps -status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( - AudioBufferProvider::Buffer* buffer, int64_t pts) -{ - if (pts == AudioBufferProvider::kInvalidPTS) { - buffer->raw = NULL; - buffer->frameCount = 0; - mTimedAudioOutputOnTime = false; - return INVALID_OPERATION; - } - - Mutex::Autolock _l(mTimedBufferQueueLock); - - ALOG_ASSERT(!mQueueHeadInFlight, - "getNextBuffer called without releaseBuffer!"); - - while (true) { - - // if we have no timed buffers, then fail - if (mTimedBufferQueue.isEmpty()) { - buffer->raw = NULL; - buffer->frameCount = 0; - return NOT_ENOUGH_DATA; - } - - TimedBuffer& head = mTimedBufferQueue.editItemAt(0); - - // calculate the PTS of the head of the timed buffer queue expressed in - // local time - int64_t headLocalPTS; - { - Mutex::Autolock mttLock(mMediaTimeTransformLock); - - ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); - - if (mMediaTimeTransform.a_to_b_denom == 0) { - // the transform represents a pause, so yield silence - timedYieldSilence_l(buffer->frameCount, buffer); - return NO_ERROR; - } - - int64_t transformedPTS; - if (!mMediaTimeTransform.doForwardTransform(head.pts(), - &transformedPTS)) { - // the transform failed. this shouldn't happen, but if it does - // then just drop this buffer - ALOGW("timedGetNextBuffer transform failed"); - buffer->raw = NULL; - buffer->frameCount = 0; - trimTimedBufferQueueHead_l("getNextBuffer; no transform"); - return NO_ERROR; - } - - if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { - if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, - &headLocalPTS)) { - buffer->raw = NULL; - buffer->frameCount = 0; - return INVALID_OPERATION; - } - } else { - headLocalPTS = transformedPTS; - } - } - - // adjust the head buffer's PTS to reflect the portion of the head buffer - // that has already been consumed - int64_t effectivePTS = headLocalPTS + - ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate()); - - // Calculate the delta in samples between the head of the input buffer - // queue and the start of the next output buffer that will be written. - // If the transformation fails because of over or underflow, it means - // that the sample's position in the output stream is so far out of - // whack that it should just be dropped. - int64_t sampleDelta; - if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { - ALOGV("*** head buffer is too far from PTS: dropped buffer"); - trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" - " mix"); - continue; - } - if (!mLocalTimeToSampleTransform.doForwardTransform( - (effectivePTS - pts) << 32, &sampleDelta)) { - ALOGV("*** too late during sample rate transform: dropped buffer"); - trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); - continue; - } - - ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" - " sampleDelta=[%d.%08x]", - head.pts(), head.position(), pts, - static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) - + (sampleDelta >> 32)), - static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); - - // if the delta between the ideal placement for the next input sample and - // the current output position is within this threshold, then we will - // concatenate the next input samples to the previous output - const int64_t kSampleContinuityThreshold = - (static_cast<int64_t>(sampleRate()) << 32) / 250; - - // if this is the first buffer of audio that we're emitting from this track - // then it should be almost exactly on time. - const int64_t kSampleStartupThreshold = 1LL << 32; - - if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || - (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { - // the next input is close enough to being on time, so concatenate it - // with the last output - timedYieldSamples_l(buffer); - - ALOGVV("*** on time: head.pos=%d frameCount=%u", - head.position(), buffer->frameCount); - return NO_ERROR; - } - - // Looks like our output is not on time. Reset our on timed status. - // Next time we mix samples from our input queue, then should be within - // the StartupThreshold. - mTimedAudioOutputOnTime = false; - if (sampleDelta > 0) { - // the gap between the current output position and the proper start of - // the next input sample is too big, so fill it with silence - uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; - - timedYieldSilence_l(framesUntilNextInput, buffer); - ALOGV("*** silence: frameCount=%u", buffer->frameCount); - return NO_ERROR; - } else { - // the next input sample is late - uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); - size_t onTimeSamplePosition = - head.position() + lateFrames * mFrameSize; - - if (onTimeSamplePosition > head.buffer()->size()) { - // all the remaining samples in the head are too late, so - // drop it and move on - ALOGV("*** too late: dropped buffer"); - trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); - continue; - } else { - // skip over the late samples - head.setPosition(onTimeSamplePosition); - - // yield the available samples - timedYieldSamples_l(buffer); - - ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); - return NO_ERROR; - } - } - } -} - -// Yield samples from the timed buffer queue head up to the given output -// buffer's capacity. -// -// Caller must hold mTimedBufferQueueLock -void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( - AudioBufferProvider::Buffer* buffer) { - - const TimedBuffer& head = mTimedBufferQueue[0]; - - buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + - head.position()); - - uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / - mFrameSize); - size_t framesRequested = buffer->frameCount; - buffer->frameCount = min(framesLeftInHead, framesRequested); - - mQueueHeadInFlight = true; - mTimedAudioOutputOnTime = true; -} - -// Yield samples of silence up to the given output buffer's capacity -// -// Caller must hold mTimedBufferQueueLock -void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( - uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { - - // lazily allocate a buffer filled with silence - if (mTimedSilenceBufferSize < numFrames * mFrameSize) { - delete [] mTimedSilenceBuffer; - mTimedSilenceBufferSize = numFrames * mFrameSize; - mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; - memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); - } - - buffer->raw = mTimedSilenceBuffer; - size_t framesRequested = buffer->frameCount; - buffer->frameCount = min(numFrames, framesRequested); - - mTimedAudioOutputOnTime = false; -} - -// AudioBufferProvider interface -void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( - AudioBufferProvider::Buffer* buffer) { - - Mutex::Autolock _l(mTimedBufferQueueLock); - - // If the buffer which was just released is part of the buffer at the head - // of the queue, be sure to update the amt of the buffer which has been - // consumed. If the buffer being returned is not part of the head of the - // queue, its either because the buffer is part of the silence buffer, or - // because the head of the timed queue was trimmed after the mixer called - // getNextBuffer but before the mixer called releaseBuffer. - if (buffer->raw == mTimedSilenceBuffer) { - ALOG_ASSERT(!mQueueHeadInFlight, - "Queue head in flight during release of silence buffer!"); - goto done; - } - - ALOG_ASSERT(mQueueHeadInFlight, - "TimedTrack::releaseBuffer of non-silence buffer, but no queue" - " head in flight."); - - if (mTimedBufferQueue.size()) { - TimedBuffer& head = mTimedBufferQueue.editItemAt(0); - - void* start = head.buffer()->pointer(); - void* end = reinterpret_cast<void*>( - reinterpret_cast<uint8_t*>(head.buffer()->pointer()) - + head.buffer()->size()); - - ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), - "released buffer not within the head of the timed buffer" - " queue; qHead = [%p, %p], released buffer = %p", - start, end, buffer->raw); - - head.setPosition(head.position() + - (buffer->frameCount * mFrameSize)); - mQueueHeadInFlight = false; - - ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, - "Bad bookkeeping during releaseBuffer! Should have at" - " least %u queued frames, but we think we have only %u", - buffer->frameCount, mFramesPendingInQueue); - - mFramesPendingInQueue -= buffer->frameCount; - - if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) - || mTrimQueueHeadOnRelease) { - trimTimedBufferQueueHead_l("releaseBuffer"); - mTrimQueueHeadOnRelease = false; - } - } else { - LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" - " buffers in the timed buffer queue"); - } - -done: - buffer->raw = 0; - buffer->frameCount = 0; -} - -size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { - Mutex::Autolock _l(mTimedBufferQueueLock); - return mFramesPendingInQueue; -} - -AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() - : mPTS(0), mPosition(0) {} - -AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( - const sp<IMemory>& buffer, int64_t pts) - : mBuffer(buffer), mPTS(pts), mPosition(0) {} - -// ---------------------------------------------------------------------------- - -// RecordTrack constructor must be called with AudioFlinger::mLock held -AudioFlinger::RecordThread::RecordTrack::RecordTrack( - RecordThread *thread, - const sp<Client>& client, - uint32_t sampleRate, - audio_format_t format, - audio_channel_mask_t channelMask, - size_t frameCount, - int sessionId) - : TrackBase(thread, client, sampleRate, format, - channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), - mOverflow(false) -{ - ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); -} - -AudioFlinger::RecordThread::RecordTrack::~RecordTrack() -{ - ALOGV("%s", __func__); -} - -// AudioBufferProvider interface -status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, - int64_t pts) -{ - 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 (mStepServerFailed) { - if (!step()) { - goto getNextBuffer_exit; - } - ALOGV("stepServer recovered"); - mStepServerFailed = false; - } - - // FIXME lock is not actually held, so overrun is possible - framesAvail = cblk->framesAvailableIn_l(mFrameCount); - - if (CC_LIKELY(framesAvail)) { - uint32_t s = cblk->server; - uint32_t bufferEnd = cblk->serverBase + mFrameCount; - - if (framesReq > framesAvail) { - framesReq = framesAvail; - } - if (framesReq > bufferEnd - s) { - framesReq = bufferEnd - s; - } - - buffer->raw = getBuffer(s, framesReq); - buffer->frameCount = framesReq; - return NO_ERROR; - } - -getNextBuffer_exit: - buffer->raw = NULL; - buffer->frameCount = 0; - return NOT_ENOUGH_DATA; -} - -status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, - int triggerSession) -{ - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - RecordThread *recordThread = (RecordThread *)thread.get(); - return recordThread->start(this, event, triggerSession); - } else { - return BAD_VALUE; - } -} - -void AudioFlinger::RecordThread::RecordTrack::stop() -{ - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - RecordThread *recordThread = (RecordThread *)thread.get(); - recordThread->mLock.lock(); - bool doStop = recordThread->stop_l(this); - if (doStop) { - TrackBase::reset(); - // Force overrun condition to avoid false overrun callback until first data is - // read from buffer - android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); - } - recordThread->mLock.unlock(); - if (doStop) { - AudioSystem::stopInput(recordThread->id()); - } - } -} - -/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) -{ - result.append(" Clien Fmt Chn mask Session Step S SRate Serv User FrameCount\n"); -} - -void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) -{ - snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x %05d\n", - (mClient == 0) ? getpid_cached : mClient->pid(), - mFormat, - mChannelMask, - mSessionId, - mStepCount, - mState, - mCblk->sampleRate, - mCblk->server, - mCblk->user, - mFrameCount); -} - -bool AudioFlinger::RecordThread::RecordTrack::isOut() const -{ - return false; -} - -// ---------------------------------------------------------------------------- - -AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( - PlaybackThread *playbackThread, - DuplicatingThread *sourceThread, - uint32_t sampleRate, - audio_format_t format, - audio_channel_mask_t channelMask, - size_t frameCount) - : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, - NULL, 0, IAudioFlinger::TRACK_DEFAULT), - mActive(false), mSourceThread(sourceThread), mBuffers(NULL) -{ - - if (mCblk != NULL) { - mBuffers = (char*)mCblk + sizeof(audio_track_cblk_t); - mOutBuffer.frameCount = 0; - playbackThread->mTracks.add(this); - ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mBuffers %p, " \ - "mCblk->frameCount %d, mCblk->sampleRate %u, mChannelMask 0x%08x mBufferEnd %p", - mCblk, mBuffer, mBuffers, - mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); - } else { - ALOGW("Error creating output track on thread %p", playbackThread); - } -} - -AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() -{ - clearBufferQueue(); -} - -status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, - int triggerSession) -{ - status_t status = Track::start(event, triggerSession); - if (status != NO_ERROR) { - return status; - } - - mActive = true; - mRetryCount = 127; - return status; -} - -void AudioFlinger::PlaybackThread::OutputTrack::stop() -{ - Track::stop(); - clearBufferQueue(); - mOutBuffer.frameCount = 0; - mActive = false; -} - -bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) -{ - Buffer *pInBuffer; - Buffer inBuffer; - uint32_t channelCount = mChannelCount; - 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 (mFrameCount > frames){ - if (mBufferQueue.size() < kMaxOverFlowBuffers) { - uint32_t startFrames = (mFrameCount - frames); - pInBuffer = new Buffer; - pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; - pInBuffer->frameCount = startFrames; - pInBuffer->i16 = pInBuffer->mBuffer; - memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); - mBufferQueue.add(pInBuffer); - } else { - ALOGW ("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)NO_MORE_BUFFERS) { - ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, - mThread.unsafe_get()); - outputBufferFull = true; - break; - } - uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); - if (waitTimeLeftMs >= waitTimeMs) { - waitTimeLeftMs -= waitTimeMs; - } else { - waitTimeLeftMs = 0; - } - } - - uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : - pInBuffer->frameCount; - memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); - mCblk->stepUserOut(outFrames, mFrameCount); - pInBuffer->frameCount -= outFrames; - pInBuffer->i16 += outFrames * channelCount; - mOutBuffer.frameCount -= outFrames; - mOutBuffer.i16 += outFrames * channelCount; - - if (pInBuffer->frameCount == 0) { - if (mBufferQueue.size()) { - mBufferQueue.removeAt(0); - delete [] pInBuffer->mBuffer; - delete pInBuffer; - ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, - mThread.unsafe_get(), mBufferQueue.size()); - } else { - break; - } - } - } - - // If we could not write all frames, allocate a buffer and queue it for next time. - if (inBuffer.frameCount) { - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0 && !thread->standby()) { - if (mBufferQueue.size() < kMaxOverFlowBuffers) { - pInBuffer = new Buffer; - pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; - pInBuffer->frameCount = inBuffer.frameCount; - pInBuffer->i16 = pInBuffer->mBuffer; - memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * - sizeof(int16_t)); - mBufferQueue.add(pInBuffer); - ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, - mThread.unsafe_get(), mBufferQueue.size()); - } else { - ALOGW("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 < mFrameCount) { - frames = mFrameCount - mCblk->user; - pInBuffer = new Buffer; - pInBuffer->mBuffer = new int16_t[frames * channelCount]; - pInBuffer->frameCount = frames; - pInBuffer->i16 = pInBuffer->mBuffer; - memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); - mBufferQueue.add(pInBuffer); - } else if (mActive) { - stop(); - } - } - - return outputBufferFull; -} - -status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( - AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) -{ - int active; - status_t result; - audio_track_cblk_t* cblk = mCblk; - uint32_t framesReq = buffer->frameCount; - - ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); - buffer->frameCount = 0; - - uint32_t framesAvail = cblk->framesAvailableOut(mFrameCount); - - - if (framesAvail == 0) { - Mutex::Autolock _l(cblk->lock); - goto start_loop_here; - while (framesAvail == 0) { - active = mActive; - if (CC_UNLIKELY(!active)) { - ALOGV("Not active and NO_MORE_BUFFERS"); - return NO_MORE_BUFFERS; - } - result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); - if (result != NO_ERROR) { - return NO_MORE_BUFFERS; - } - // read the server count again - start_loop_here: - framesAvail = cblk->framesAvailableOut_l(mFrameCount); - } - } - -// if (framesAvail < framesReq) { -// return NO_MORE_BUFFERS; -// } - - if (framesReq > framesAvail) { - framesReq = framesAvail; - } - - uint32_t u = cblk->user; - uint32_t bufferEnd = cblk->userBase + mFrameCount; - - if (framesReq > bufferEnd - u) { - framesReq = bufferEnd - u; - } - - buffer->frameCount = framesReq; - buffer->raw = cblk->buffer(mBuffers, mFrameSize, u); - return NO_ERROR; -} - - -void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() -{ - size_t size = mBufferQueue.size(); - - for (size_t i = 0; i < size; i++) { - Buffer *pBuffer = mBufferQueue.itemAt(i); - delete [] pBuffer->mBuffer; - delete pBuffer; - } - mBufferQueue.clear(); -} // ---------------------------------------------------------------------------- @@ -5851,88 +1116,6 @@ void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) mAudioFlinger->removeNotificationClient(mPid); } -// ---------------------------------------------------------------------------- - -AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) - : BnAudioTrack(), - mTrack(track) -{ -} - -AudioFlinger::TrackHandle::~TrackHandle() { - // just stop the track on deletion, associated resources - // will be freed from the main thread once all pending buffers have - // been played. Unless it's not in the active track list, in which - // case we free everything now... - mTrack->destroy(); -} - -sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { - return mTrack->getCblk(); -} - -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(); -} - -status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) -{ - return mTrack->attachAuxEffect(EffectId); -} - -status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, - sp<IMemory>* buffer) { - if (!mTrack->isTimedTrack()) - return INVALID_OPERATION; - - PlaybackThread::TimedTrack* tt = - reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); - return tt->allocateTimedBuffer(size, buffer); -} - -status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, - int64_t pts) { - if (!mTrack->isTimedTrack()) - return INVALID_OPERATION; - - PlaybackThread::TimedTrack* tt = - reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); - return tt->queueTimedBuffer(buffer, pts); -} - -status_t AudioFlinger::TrackHandle::setMediaTimeTransform( - const LinearTransform& xform, int target) { - - if (!mTrack->isTimedTrack()) - return INVALID_OPERATION; - - PlaybackThread::TimedTrack* tt = - reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); - return tt->setMediaTimeTransform( - xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); -} - -status_t AudioFlinger::TrackHandle::onTransact( - uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) -{ - return BnAudioTrack::onTransact(code, data, reply, flags); -} // ---------------------------------------------------------------------------- @@ -6006,912 +1189,6 @@ Exit: return recordHandle; } -// ---------------------------------------------------------------------------- - -AudioFlinger::RecordHandle::RecordHandle( - const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) - : BnAudioRecord(), - mRecordTrack(recordTrack) -{ -} - -AudioFlinger::RecordHandle::~RecordHandle() { - stop_nonvirtual(); - mRecordTrack->destroy(); -} - -sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { - return mRecordTrack->getCblk(); -} - -status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, - int triggerSession) { - ALOGV("RecordHandle::start()"); - return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); -} - -void AudioFlinger::RecordHandle::stop() { - stop_nonvirtual(); -} - -void AudioFlinger::RecordHandle::stop_nonvirtual() { - ALOGV("RecordHandle::stop()"); - mRecordTrack->stop(); -} - -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, - audio_channel_mask_t channelMask, - audio_io_handle_t id, - audio_devices_t device, - const sp<NBAIO_Sink>& teeSink) : - ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), - mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), - // mRsmpInIndex and mInputBytes set by readInputParameters() - mReqChannelCount(popcount(channelMask)), - mReqSampleRate(sampleRate), - // mBytesRead is only meaningful while active, and so is cleared in start() - // (but might be better to also clear here for dump?) - mTeeSink(teeSink) -{ - snprintf(mName, kNameLength, "AudioIn_%X", id); - - readInputParameters(); - -} - - -AudioFlinger::RecordThread::~RecordThread() -{ - delete[] mRsmpInBuffer; - delete mResampler; - delete[] mRsmpOutBuffer; -} - -void AudioFlinger::RecordThread::onFirstRef() -{ - run(mName, PRIORITY_URGENT_AUDIO); -} - -status_t AudioFlinger::RecordThread::readyToRun() -{ - status_t status = initCheck(); - ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); - return status; -} - -bool AudioFlinger::RecordThread::threadLoop() -{ - AudioBufferProvider::Buffer buffer; - sp<RecordTrack> activeTrack; - Vector< sp<EffectChain> > effectChains; - - nsecs_t lastWarning = 0; - - inputStandBy(); - acquireWakeLock(); - - // used to verify we've read at least once before evaluating how many bytes were read - bool readOnce = false; - - // start recording - while (!exitPending()) { - - processConfigEvents(); - - { // scope for mLock - Mutex::Autolock _l(mLock); - checkForNewParameters_l(); - if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { - standby(); - - if (exitPending()) { - break; - } - - releaseWakeLock_l(); - ALOGV("RecordThread: loop stopping"); - // go to sleep - mWaitWorkCV.wait(mLock); - ALOGV("RecordThread: loop starting"); - acquireWakeLock_l(); - continue; - } - if (mActiveTrack != 0) { - if (mActiveTrack->mState == TrackBase::PAUSING) { - standby(); - mActiveTrack.clear(); - mStartStopCond.broadcast(); - } else if (mActiveTrack->mState == TrackBase::RESUMING) { - if (mReqChannelCount != mActiveTrack->channelCount()) { - mActiveTrack.clear(); - mStartStopCond.broadcast(); - } else if (readOnce) { - // 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; - } else if (mActiveTrack->mState == TrackBase::TERMINATED) { - removeTrack_l(mActiveTrack); - mActiveTrack.clear(); - } - } - lockEffectChains_l(effectChains); - } - - if (mActiveTrack != 0) { - if (mActiveTrack->mState != TrackBase::ACTIVE && - mActiveTrack->mState != TrackBase::RESUMING) { - unlockEffectChains(effectChains); - usleep(kRecordThreadSleepUs); - continue; - } - for (size_t i = 0; i < effectChains.size(); i ++) { - effectChains[i]->process_l(); - } - - buffer.frameCount = mFrameCount; - if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { - readOnce = true; - size_t framesOut = buffer.frameCount; - if (mResampler == NULL) { - // 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->mFrameSize; - if (framesIn > framesOut) - framesIn = framesOut; - mRsmpInIndex += framesIn; - framesOut -= framesIn; - if (mChannelCount == mReqChannelCount || - mFormat != AUDIO_FORMAT_PCM_16_BIT) { - memcpy(dst, src, framesIn * mFrameSize); - } else { - if (mChannelCount == 1) { - upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, - (int16_t *)src, framesIn); - } else { - downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, - (int16_t *)src, framesIn); - } - } - } - if (framesOut && mFrameCount == mRsmpInIndex) { - void *readInto; - if (framesOut == mFrameCount && - (mChannelCount == mReqChannelCount || - mFormat != AUDIO_FORMAT_PCM_16_BIT)) { - readInto = buffer.raw; - framesOut = 0; - } else { - readInto = mRsmpInBuffer; - mRsmpInIndex = 0; - } - mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); - if (mBytesRead <= 0) { - if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) - { - ALOGE("Error reading audio input"); - // Force input into standby so that it tries to - // recover at next read attempt - inputStandBy(); - usleep(kRecordThreadSleepUs); - } - mRsmpInIndex = mFrameCount; - framesOut = 0; - buffer.frameCount = 0; - } else if (mTeeSink != 0) { - (void) mTeeSink->write(readInto, - mBytesRead >> Format_frameBitShift(mTeeSink->format())); - } - } - } - } 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 /* AudioBufferProvider* */); - // 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) { - ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); - // the resampler always outputs stereo samples: - // do post stereo to mono conversion - downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, - framesOut); - } else { - ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); - } - - } - if (mFramestoDrop == 0) { - mActiveTrack->releaseBuffer(&buffer); - } else { - if (mFramestoDrop > 0) { - mFramestoDrop -= buffer.frameCount; - if (mFramestoDrop <= 0) { - clearSyncStartEvent(); - } - } else { - mFramestoDrop += buffer.frameCount; - if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || - mSyncStartEvent->isCancelled()) { - ALOGW("Synced record %s, session %d, trigger session %d", - (mFramestoDrop >= 0) ? "timed out" : "cancelled", - mActiveTrack->sessionId(), - (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); - clearSyncStartEvent(); - } - } - } - mActiveTrack->clearOverflow(); - } - // client isn't retrieving buffers fast enough - else { - if (!mActiveTrack->setOverflow()) { - nsecs_t now = systemTime(); - if ((now - lastWarning) > kWarningThrottleNs) { - ALOGW("RecordThread: buffer overflow"); - lastWarning = now; - } - } - // 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(kRecordThreadSleepUs); - } - } - // enable changes in effect chain - unlockEffectChains(effectChains); - effectChains.clear(); - } - - standby(); - - { - Mutex::Autolock _l(mLock); - mActiveTrack.clear(); - mStartStopCond.broadcast(); - } - - releaseWakeLock(); - - ALOGV("RecordThread %p exiting", this); - return false; -} - -void AudioFlinger::RecordThread::standby() -{ - if (!mStandby) { - inputStandBy(); - mStandby = true; - } -} - -void AudioFlinger::RecordThread::inputStandBy() -{ - mInput->stream->common.standby(&mInput->stream->common); -} - -sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( - const sp<AudioFlinger::Client>& client, - uint32_t sampleRate, - audio_format_t format, - audio_channel_mask_t channelMask, - size_t frameCount, - int sessionId, - IAudioFlinger::track_flags_t flags, - pid_t tid, - status_t *status) -{ - sp<RecordTrack> track; - status_t lStatus; - - lStatus = initCheck(); - if (lStatus != NO_ERROR) { - ALOGE("Audio driver not initialized."); - goto Exit; - } - - // FIXME use flags and tid similar to createTrack_l() - - { // scope for mLock - Mutex::Autolock _l(mLock); - - track = new RecordTrack(this, client, sampleRate, - format, channelMask, frameCount, sessionId); - - if (track->getCblk() == 0) { - lStatus = NO_MEMORY; - goto Exit; - } - mTracks.add(track); - - // disable AEC and NS if the device is a BT SCO headset supporting those pre processings - bool suspend = audio_is_bluetooth_sco_device(mInDevice) && - mAudioFlinger->btNrecIsOff(); - setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); - setEffectSuspended_l(FX_IID_NS, suspend, sessionId); - } - lStatus = NO_ERROR; - -Exit: - if (status) { - *status = lStatus; - } - return track; -} - -status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, - AudioSystem::sync_event_t event, - int triggerSession) -{ - ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); - sp<ThreadBase> strongMe = this; - status_t status = NO_ERROR; - - if (event == AudioSystem::SYNC_EVENT_NONE) { - clearSyncStartEvent(); - } else if (event != AudioSystem::SYNC_EVENT_SAME) { - mSyncStartEvent = mAudioFlinger->createSyncEvent(event, - triggerSession, - recordTrack->sessionId(), - syncStartEventCallback, - this); - // Sync event can be cancelled by the trigger session if the track is not in a - // compatible state in which case we start record immediately - if (mSyncStartEvent->isCancelled()) { - clearSyncStartEvent(); - } else { - // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs - mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); - } - } - - { - 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(); - clearSyncStartEvent(); - return status; - } - mRsmpInIndex = mFrameCount; - mBytesRead = 0; - if (mResampler != NULL) { - mResampler->reset(); - } - mActiveTrack->mState = TrackBase::RESUMING; - // signal thread to start - ALOGV("Signal record thread"); - mWaitWorkCV.broadcast(); - // do not wait for mStartStopCond if exiting - if (exitPending()) { - mActiveTrack.clear(); - status = INVALID_OPERATION; - goto startError; - } - mStartStopCond.wait(mLock); - if (mActiveTrack == 0) { - ALOGV("Record failed to start"); - status = BAD_VALUE; - goto startError; - } - ALOGV("Record started OK"); - return status; - } -startError: - AudioSystem::stopInput(mId); - clearSyncStartEvent(); - return status; -} - -void AudioFlinger::RecordThread::clearSyncStartEvent() -{ - if (mSyncStartEvent != 0) { - mSyncStartEvent->cancel(); - } - mSyncStartEvent.clear(); - mFramestoDrop = 0; -} - -void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) -{ - sp<SyncEvent> strongEvent = event.promote(); - - if (strongEvent != 0) { - RecordThread *me = (RecordThread *)strongEvent->cookie(); - me->handleSyncStartEvent(strongEvent); - } -} - -void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) -{ - if (event == mSyncStartEvent) { - // TODO: use actual buffer filling status instead of 2 buffers when info is available - // from audio HAL - mFramestoDrop = mFrameCount * 2; - } -} - -bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { - ALOGV("RecordThread::stop"); - if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { - return false; - } - recordTrack->mState = TrackBase::PAUSING; - // do not wait for mStartStopCond if exiting - if (exitPending()) { - return true; - } - mStartStopCond.wait(mLock); - // if we have been restarted, recordTrack == mActiveTrack.get() here - if (exitPending() || recordTrack != mActiveTrack.get()) { - ALOGV("Record stopped OK"); - return true; - } - return false; -} - -bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const -{ - return false; -} - -status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) -{ -#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future - if (!isValidSyncEvent(event)) { - return BAD_VALUE; - } - - int eventSession = event->triggerSession(); - status_t ret = NAME_NOT_FOUND; - - Mutex::Autolock _l(mLock); - - for (size_t i = 0; i < mTracks.size(); i++) { - sp<RecordTrack> track = mTracks[i]; - if (eventSession == track->sessionId()) { - (void) track->setSyncEvent(event); - ret = NO_ERROR; - } - } - return ret; -#else - return BAD_VALUE; -#endif -} - -void AudioFlinger::RecordThread::RecordTrack::destroy() -{ - // see comments at AudioFlinger::PlaybackThread::Track::destroy() - sp<RecordTrack> keep(this); - { - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - if (mState == ACTIVE || mState == RESUMING) { - AudioSystem::stopInput(thread->id()); - } - AudioSystem::releaseInput(thread->id()); - Mutex::Autolock _l(thread->mLock); - RecordThread *recordThread = (RecordThread *) thread.get(); - recordThread->destroyTrack_l(this); - } - } -} - -// destroyTrack_l() must be called with ThreadBase::mLock held -void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) -{ - track->mState = TrackBase::TERMINATED; - // active tracks are removed by threadLoop() - if (mActiveTrack != track) { - removeTrack_l(track); - } -} - -void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) -{ - mTracks.remove(track); - // need anything related to effects here? -} - -void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) -{ - dumpInternals(fd, args); - dumpTracks(fd, args); - dumpEffectChains(fd, args); -} - -void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); - result.append(buffer); - - if (mActiveTrack != 0) { - 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 != NULL)); - result.append(buffer); - snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); - result.append(buffer); - snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); - result.append(buffer); - } else { - result.append("No active record client\n"); - } - - write(fd, result.string(), result.size()); - - dumpBase(fd, args); -} - -void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "Input thread %p tracks\n", this); - result.append(buffer); - RecordTrack::appendDumpHeader(result); - for (size_t i = 0; i < mTracks.size(); ++i) { - sp<RecordTrack> track = mTracks[i]; - if (track != 0) { - track->dump(buffer, SIZE); - result.append(buffer); - } - } - - if (mActiveTrack != 0) { - snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); - result.append(buffer); - RecordTrack::appendDumpHeader(result); - mActiveTrack->dump(buffer, SIZE); - result.append(buffer); - - } - write(fd, result.string(), result.size()); -} - -// AudioBufferProvider interface -status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) -{ - size_t framesReq = buffer->frameCount; - size_t framesReady = mFrameCount - mRsmpInIndex; - int channelCount; - - if (framesReady == 0) { - mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); - if (mBytesRead <= 0) { - if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { - ALOGE("RecordThread::getNextBuffer() Error reading audio input"); - // Force input into standby so that it tries to - // recover at next read attempt - inputStandBy(); - usleep(kRecordThreadSleepUs); - } - buffer->raw = NULL; - 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; -} - -// AudioBufferProvider interface -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; - audio_format_t reqFormat = mFormat; - uint32_t reqSamplingRate = mReqSampleRate; - uint32_t 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 = (audio_format_t) value; - reconfig = true; - } - if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { - reqChannelCount = 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 guaranteed - // if frame count is changed after track creation - if (mActiveTrack != 0) { - status = INVALID_OPERATION; - } else { - reconfig = true; - } - } - if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { - // forward device change to effects that have requested to be - // aware of attached audio device. - for (size_t i = 0; i < mEffectChains.size(); i++) { - mEffectChains[i]->setDevice_l(value); - } - - // store input device and output device but do not forward output device to audio HAL. - // Note that status is ignored by the caller for output device - // (see AudioFlinger::setParameters() - if (audio_is_output_devices(value)) { - mOutDevice = value; - status = BAD_VALUE; - } else { - mInDevice = value; - // disable AEC and NS if the device is a BT SCO headset supporting those - // pre processings - if (mTracks.size() > 0) { - bool suspend = audio_is_bluetooth_sco_device(mInDevice) && - mAudioFlinger->btNrecIsOff(); - for (size_t i = 0; i < mTracks.size(); i++) { - sp<RecordTrack> track = mTracks[i]; - setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); - setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); - } - } - } - } - if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && - mAudioSource != (audio_source_t)value) { - // forward device change to effects that have requested to be - // aware of attached audio device. - for (size_t i = 0; i < mEffectChains.size(); i++) { - mEffectChains[i]->setAudioSource_l((audio_source_t)value); - } - mAudioSource = (audio_source_t)value; - } - if (status == NO_ERROR) { - status = mInput->stream->common.set_parameters(&mInput->stream->common, - keyValuePair.string()); - if (status == INVALID_OPERATION) { - inputStandBy(); - status = mInput->stream->common.set_parameters(&mInput->stream->common, - keyValuePair.string()); - } - if (reconfig) { - if (status == BAD_VALUE && - reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && - reqFormat == AUDIO_FORMAT_PCM_16_BIT && - ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) - <= (2 * reqSamplingRate)) && - popcount(mInput->stream->common.get_channels(&mInput->stream->common)) - <= FCC_2 && - (reqChannelCount <= FCC_2)) { - status = NO_ERROR; - } - if (status == NO_ERROR) { - readInputParameters(); - sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); - } - } - } - - mNewParameters.removeAt(0); - - mParamStatus = status; - mParamCond.signal(); - // wait for condition with time out in case the thread calling ThreadBase::setParameters() - // already timed out waiting for the status and will never signal the condition. - mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); - } - return reconfig; -} - -String8 AudioFlinger::RecordThread::getParameters(const String8& keys) -{ - char *s; - String8 out_s8 = String8(); - - Mutex::Autolock _l(mLock); - if (initCheck() != NO_ERROR) { - return out_s8; - } - - s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); - out_s8 = String8(s); - free(s); - return out_s8; -} - -void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { - AudioSystem::OutputDescriptor desc; - void *param2 = NULL; - - switch (event) { - case AudioSystem::INPUT_OPENED: - case AudioSystem::INPUT_CONFIG_CHANGED: - desc.channels = mChannelMask; - desc.samplingRate = mSampleRate; - desc.format = mFormat; - desc.frameCount = mFrameCount; - desc.latency = 0; - param2 = &desc; - break; - - case AudioSystem::INPUT_CLOSED: - default: - break; - } - mAudioFlinger->audioConfigChanged_l(event, mId, param2); -} - -void AudioFlinger::RecordThread::readInputParameters() -{ - delete mRsmpInBuffer; - // mRsmpInBuffer is always assigned a new[] below - delete mRsmpOutBuffer; - mRsmpOutBuffer = NULL; - delete mResampler; - mResampler = NULL; - - mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); - mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); - mChannelCount = (uint16_t)popcount(mChannelMask); - mFormat = mInput->stream->common.get_format(&mInput->stream->common); - mFrameSize = audio_stream_frame_size(&mInput->stream->common); - mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); - mFrameCount = mInputBytes / mFrameSize; - mNormalFrameCount = mFrameCount; // not used by record, but used by input effects - mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; - - if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) - { - int channelCount; - // optimization: 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() -{ - Mutex::Autolock _l(mLock); - if (initCheck() != NO_ERROR) { - return 0; - } - - return mInput->stream->get_input_frames_lost(mInput->stream); -} - -uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const -{ - Mutex::Autolock _l(mLock); - uint32_t result = 0; - if (getEffectChain_l(sessionId) != 0) { - result = EFFECT_SESSION; - } - - for (size_t i = 0; i < mTracks.size(); ++i) { - if (sessionId == mTracks[i]->sessionId()) { - result |= TRACK_SESSION; - break; - } - } - - return result; -} - -KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const -{ - KeyedVector<int, bool> ids; - Mutex::Autolock _l(mLock); - for (size_t j = 0; j < mTracks.size(); ++j) { - sp<RecordThread::RecordTrack> track = mTracks[j]; - int sessionId = track->sessionId(); - if (ids.indexOfKey(sessionId) < 0) { - ids.add(sessionId, true); - } - } - return ids; -} - -AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() -{ - Mutex::Autolock _l(mLock); - AudioStreamIn *input = mInput; - mInput = NULL; - return input; -} - -// this method must always be called either with ThreadBase mLock held or inside the thread loop -audio_stream_t* AudioFlinger::RecordThread::stream() const -{ - if (mInput == NULL) { - return NULL; - } - return &mInput->stream->common; -} // ---------------------------------------------------------------------------- @@ -7871,2063 +2148,67 @@ status_t AudioFlinger::moveEffectChain_l(int sessionId, return NO_ERROR; } - -// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held -sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( - const sp<AudioFlinger::Client>& client, - const sp<IEffectClient>& effectClient, - int32_t priority, - int sessionId, - effect_descriptor_t *desc, - int *enabled, - status_t *status - ) -{ - sp<EffectModule> effect; - sp<EffectHandle> handle; - status_t lStatus; - sp<EffectChain> chain; - bool chainCreated = false; - bool effectCreated = false; - bool effectRegistered = false; - - lStatus = initCheck(); - if (lStatus != NO_ERROR) { - ALOGW("createEffect_l() Audio driver not initialized."); - goto Exit; - } - - // Do not allow effects with session ID 0 on direct output or duplicating threads - // TODO: add rule for hw accelerated effects on direct outputs with non PCM format - if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { - ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", - desc->name, sessionId); - lStatus = BAD_VALUE; - goto Exit; - } - // Only Pre processor effects are allowed on input threads and only on input threads - if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { - ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", - desc->name, desc->flags, mType); - lStatus = BAD_VALUE; - goto Exit; - } - - ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); - - { // scope for mLock - Mutex::Autolock _l(mLock); - - // check for existing effect chain with the requested audio session - chain = getEffectChain_l(sessionId); - if (chain == 0) { - // create a new chain for this session - ALOGV("createEffect_l() new effect chain for session %d", sessionId); - chain = new EffectChain(this, sessionId); - addEffectChain_l(chain); - chain->setStrategy(getStrategyForSession_l(sessionId)); - chainCreated = true; - } else { - effect = chain->getEffectFromDesc_l(desc); - } - - ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); - - if (effect == 0) { - int id = mAudioFlinger->nextUniqueId(); - // Check CPU and memory usage - lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); - if (lStatus != NO_ERROR) { - goto Exit; - } - effectRegistered = true; - // create a new effect module if none present in the chain - effect = new EffectModule(this, chain, desc, id, sessionId); - lStatus = effect->status(); - if (lStatus != NO_ERROR) { - goto Exit; - } - lStatus = chain->addEffect_l(effect); - if (lStatus != NO_ERROR) { - goto Exit; - } - effectCreated = true; - - effect->setDevice(mOutDevice); - effect->setDevice(mInDevice); - effect->setMode(mAudioFlinger->getMode()); - effect->setAudioSource(mAudioSource); - } - // create effect handle and connect it to effect module - handle = new EffectHandle(effect, client, effectClient, priority); - lStatus = effect->addHandle(handle.get()); - if (enabled != NULL) { - *enabled = (int)effect->isEnabled(); - } - } - -Exit: - if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { - Mutex::Autolock _l(mLock); - if (effectCreated) { - chain->removeEffect_l(effect); - } - if (effectRegistered) { - AudioSystem::unregisterEffect(effect->id()); - } - if (chainCreated) { - removeEffectChain_l(chain); - } - handle.clear(); - } - - if (status != NULL) { - *status = lStatus; - } - return handle; -} - -sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) -{ - Mutex::Autolock _l(mLock); - return getEffect_l(sessionId, effectId); -} - -sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) -{ - sp<EffectChain> chain = getEffectChain_l(sessionId); - return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; -} - -// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and -// PlaybackThread::mLock held -status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) -{ - // check for existing effect chain with the requested audio session - int sessionId = effect->sessionId(); - sp<EffectChain> chain = getEffectChain_l(sessionId); - bool chainCreated = false; - - if (chain == 0) { - // create a new chain for this session - ALOGV("addEffect_l() new effect chain for session %d", sessionId); - chain = new EffectChain(this, sessionId); - addEffectChain_l(chain); - chain->setStrategy(getStrategyForSession_l(sessionId)); - chainCreated = true; - } - ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); - - if (chain->getEffectFromId_l(effect->id()) != 0) { - ALOGW("addEffect_l() %p effect %s already present in chain %p", - this, effect->desc().name, chain.get()); - return BAD_VALUE; - } - - status_t status = chain->addEffect_l(effect); - if (status != NO_ERROR) { - if (chainCreated) { - removeEffectChain_l(chain); - } - return status; - } - - effect->setDevice(mOutDevice); - effect->setDevice(mInDevice); - effect->setMode(mAudioFlinger->getMode()); - effect->setAudioSource(mAudioSource); - return NO_ERROR; -} - -void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { - - ALOGV("removeEffect_l() %p effect %p", this, effect.get()); - effect_descriptor_t desc = effect->desc(); - if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { - detachAuxEffect_l(effect->id()); - } - - sp<EffectChain> chain = effect->chain().promote(); - if (chain != 0) { - // remove effect chain if removing last effect - if (chain->removeEffect_l(effect) == 0) { - removeEffectChain_l(chain); - } - } else { - ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); - } -} - -void AudioFlinger::ThreadBase::lockEffectChains_l( - Vector< sp<AudioFlinger::EffectChain> >& effectChains) -{ - effectChains = mEffectChains; - for (size_t i = 0; i < mEffectChains.size(); i++) { - mEffectChains[i]->lock(); - } -} - -void AudioFlinger::ThreadBase::unlockEffectChains( - const Vector< sp<AudioFlinger::EffectChain> >& effectChains) -{ - for (size_t i = 0; i < effectChains.size(); i++) { - effectChains[i]->unlock(); - } -} - -sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) -{ - Mutex::Autolock _l(mLock); - return getEffectChain_l(sessionId); -} - -sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const -{ - size_t size = mEffectChains.size(); - for (size_t i = 0; i < size; i++) { - if (mEffectChains[i]->sessionId() == sessionId) { - return mEffectChains[i]; - } - } - return 0; -} - -void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) -{ - Mutex::Autolock _l(mLock); - size_t size = mEffectChains.size(); - for (size_t i = 0; i < size; i++) { - mEffectChains[i]->setMode_l(mode); - } -} - -void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, - EffectHandle *handle, - bool unpinIfLast) { - - Mutex::Autolock _l(mLock); - ALOGV("disconnectEffect() %p effect %p", this, effect.get()); - // delete the effect module if removing last handle on it - if (effect->removeHandle(handle) == 0) { - if (!effect->isPinned() || unpinIfLast) { - removeEffect_l(effect); - AudioSystem::unregisterEffect(effect->id()); - } - } -} - -status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) -{ - int session = chain->sessionId(); - int16_t *buffer = mMixBuffer; - bool ownsBuffer = false; - - ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); - if (session > 0) { - // Only one effect chain can be present in direct output thread and it uses - // the mix buffer as input - if (mType != DIRECT) { - size_t numSamples = mNormalFrameCount * mChannelCount; - buffer = new int16_t[numSamples]; - memset(buffer, 0, numSamples * sizeof(int16_t)); - ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); - ownsBuffer = true; - } - - // Attach all tracks with same session ID to this chain. - for (size_t i = 0; i < mTracks.size(); ++i) { - sp<Track> track = mTracks[i]; - if (session == track->sessionId()) { - ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), - buffer); - track->setMainBuffer(buffer); - chain->incTrackCnt(); - } - } - - // indicate all active tracks in the chain - for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { - sp<Track> track = mActiveTracks[i].promote(); - if (track == 0) { - continue; - } - if (session == track->sessionId()) { - ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); - chain->incActiveTrackCnt(); - } - } - } - - chain->setInBuffer(buffer, ownsBuffer); - chain->setOutBuffer(mMixBuffer); - // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect - // chains list in order to be processed last as it contains output stage effects - // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before - // session AUDIO_SESSION_OUTPUT_STAGE to be processed - // after track specific effects and before output stage - // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and - // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX - // Effect chain for other sessions are inserted at beginning of effect - // chains list to be processed before output mix effects. Relative order between other - // sessions is not important - size_t size = mEffectChains.size(); - size_t i = 0; - for (i = 0; i < size; i++) { - if (mEffectChains[i]->sessionId() < session) { - break; - } - } - mEffectChains.insertAt(chain, i); - checkSuspendOnAddEffectChain_l(chain); - - return NO_ERROR; -} - -size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) -{ - int session = chain->sessionId(); - - ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); - - for (size_t i = 0; i < mEffectChains.size(); i++) { - if (chain == mEffectChains[i]) { - mEffectChains.removeAt(i); - // detach all active tracks from the chain - for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { - sp<Track> track = mActiveTracks[i].promote(); - if (track == 0) { - continue; - } - if (session == track->sessionId()) { - ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", - chain.get(), session); - chain->decActiveTrackCnt(); - } - } - - // detach all tracks with same session ID from this chain - for (size_t i = 0; i < mTracks.size(); ++i) { - sp<Track> track = mTracks[i]; - if (session == track->sessionId()) { - track->setMainBuffer(mMixBuffer); - chain->decTrackCnt(); - } - } - break; - } - } - return mEffectChains.size(); -} - -status_t AudioFlinger::PlaybackThread::attachAuxEffect( - const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) -{ - Mutex::Autolock _l(mLock); - return attachAuxEffect_l(track, EffectId); -} - -status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( - const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) -{ - status_t status = NO_ERROR; - - if (EffectId == 0) { - track->setAuxBuffer(0, NULL); - } else { - // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX - sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); - if (effect != 0) { - if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { - track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); - } else { - status = INVALID_OPERATION; - } - } else { - status = BAD_VALUE; - } - } - return status; -} - -void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) -{ - for (size_t i = 0; i < mTracks.size(); ++i) { - sp<Track> track = mTracks[i]; - if (track->auxEffectId() == effectId) { - attachAuxEffect_l(track, 0); - } - } -} - -status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) -{ - // only one chain per input thread - if (mEffectChains.size() != 0) { - return INVALID_OPERATION; - } - ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); - - chain->setInBuffer(NULL); - chain->setOutBuffer(NULL); - - checkSuspendOnAddEffectChain_l(chain); - - mEffectChains.add(chain); - - return NO_ERROR; -} - -size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) -{ - ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); - ALOGW_IF(mEffectChains.size() != 1, - "removeEffectChain_l() %p invalid chain size %d on thread %p", - chain.get(), mEffectChains.size(), this); - if (mEffectChains.size() == 1) { - mEffectChains.removeAt(0); - } - return 0; -} - -// ---------------------------------------------------------------------------- -// EffectModule implementation -// ---------------------------------------------------------------------------- - -#undef LOG_TAG -#define LOG_TAG "AudioFlinger::EffectModule" - -AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, - const wp<AudioFlinger::EffectChain>& chain, - effect_descriptor_t *desc, - int id, - int sessionId) - : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), - mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), - mDescriptor(*desc), - // mConfig is set by configure() and not used before then - mEffectInterface(NULL), - mStatus(NO_INIT), mState(IDLE), - // mMaxDisableWaitCnt is set by configure() and not used before then - // mDisableWaitCnt is set by process() and updateState() and not used before then - mSuspended(false) -{ - ALOGV("Constructor %p", this); - int lStatus; - - // create effect engine from effect factory - mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); - - if (mStatus != NO_ERROR) { - return; - } - lStatus = init(); - if (lStatus < 0) { - mStatus = lStatus; - goto Error; - } - - ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); - return; -Error: - EffectRelease(mEffectInterface); - mEffectInterface = NULL; - ALOGV("Constructor Error %d", mStatus); -} - -AudioFlinger::EffectModule::~EffectModule() -{ - ALOGV("Destructor %p", this); - if (mEffectInterface != NULL) { - if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || - (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - audio_stream_t *stream = thread->stream(); - if (stream != NULL) { - stream->remove_audio_effect(stream, mEffectInterface); - } - } - } - // release effect engine - EffectRelease(mEffectInterface); - } -} - -status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) -{ - status_t status; - - Mutex::Autolock _l(mLock); - int priority = handle->priority(); - size_t size = mHandles.size(); - EffectHandle *controlHandle = NULL; - size_t i; - for (i = 0; i < size; i++) { - EffectHandle *h = mHandles[i]; - if (h == NULL || h->destroyed_l()) { - continue; - } - // first non destroyed handle is considered in control - if (controlHandle == NULL) - controlHandle = h; - if (h->priority() <= priority) { - break; - } - } - // if inserted in first place, move effect control from previous owner to this handle - if (i == 0) { - bool enabled = false; - if (controlHandle != NULL) { - enabled = controlHandle->enabled(); - controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); - } - handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); - status = NO_ERROR; - } else { - status = ALREADY_EXISTS; - } - ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); - mHandles.insertAt(handle, i); - return status; -} - -size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) -{ - Mutex::Autolock _l(mLock); - size_t size = mHandles.size(); - size_t i; - for (i = 0; i < size; i++) { - if (mHandles[i] == handle) { - break; - } - } - if (i == size) { - return size; - } - ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); - - mHandles.removeAt(i); - // if removed from first place, move effect control from this handle to next in line - if (i == 0) { - EffectHandle *h = controlHandle_l(); - if (h != NULL) { - h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); - } - } - - // Prevent calls to process() and other functions on effect interface from now on. - // The effect engine will be released by the destructor when the last strong reference on - // this object is released which can happen after next process is called. - if (mHandles.size() == 0 && !mPinned) { - mState = DESTROYED; - } - - return mHandles.size(); -} - -// must be called with EffectModule::mLock held -AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() -{ - // the first valid handle in the list has control over the module - for (size_t i = 0; i < mHandles.size(); i++) { - EffectHandle *h = mHandles[i]; - if (h != NULL && !h->destroyed_l()) { - return h; - } - } - - return NULL; -} - -size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) -{ - ALOGV("disconnect() %p handle %p", this, handle); - // keep a strong reference on this EffectModule to avoid calling the - // destructor before we exit - sp<EffectModule> keep(this); - { - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - thread->disconnectEffect(keep, handle, unpinIfLast); - } - } - return mHandles.size(); -} - -void AudioFlinger::EffectModule::updateState() { - Mutex::Autolock _l(mLock); - - switch (mState) { - case RESTART: - reset_l(); - // FALL THROUGH - - case STARTING: - // clear auxiliary effect input buffer for next accumulation - if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { - memset(mConfig.inputCfg.buffer.raw, - 0, - mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); - } - start_l(); - mState = ACTIVE; - break; - case STOPPING: - stop_l(); - mDisableWaitCnt = mMaxDisableWaitCnt; - mState = STOPPED; - break; - case STOPPED: - // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the - // turn off sequence. - if (--mDisableWaitCnt == 0) { - reset_l(); - mState = IDLE; - } - break; - default: //IDLE , ACTIVE, DESTROYED - break; - } -} - -void AudioFlinger::EffectModule::process() -{ - Mutex::Autolock _l(mLock); - - if (mState == DESTROYED || mEffectInterface == NULL || - mConfig.inputCfg.buffer.raw == NULL || - mConfig.outputCfg.buffer.raw == NULL) { - return; - } - - if (isProcessEnabled()) { - // do 32 bit to 16 bit conversion for auxiliary effect input buffer - if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { - ditherAndClamp(mConfig.inputCfg.buffer.s32, - mConfig.inputCfg.buffer.s32, - mConfig.inputCfg.buffer.frameCount/2); - } - - // do the actual processing in the effect engine - int ret = (*mEffectInterface)->process(mEffectInterface, - &mConfig.inputCfg.buffer, - &mConfig.outputCfg.buffer); - - // force transition to IDLE state when engine is ready - if (mState == STOPPED && ret == -ENODATA) { - mDisableWaitCnt = 1; - } - - // clear auxiliary effect input buffer for next accumulation - if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { - memset(mConfig.inputCfg.buffer.raw, 0, - mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); - } - } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && - mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { - // If an insert effect is idle and input buffer is different from output buffer, - // accumulate input onto output - sp<EffectChain> chain = mChain.promote(); - if (chain != 0 && chain->activeTrackCnt() != 0) { - size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here - int16_t *in = mConfig.inputCfg.buffer.s16; - int16_t *out = mConfig.outputCfg.buffer.s16; - for (size_t i = 0; i < frameCnt; i++) { - out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); - } - } - } -} - -void AudioFlinger::EffectModule::reset_l() -{ - if (mEffectInterface == NULL) { - return; - } - (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); -} - -status_t AudioFlinger::EffectModule::configure() -{ - if (mEffectInterface == NULL) { - return NO_INIT; - } - - sp<ThreadBase> thread = mThread.promote(); - if (thread == 0) { - return DEAD_OBJECT; - } - - // TODO: handle configuration of effects replacing track process - audio_channel_mask_t channelMask = thread->channelMask(); - - if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { - mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; - } else { - mConfig.inputCfg.channels = channelMask; - } - mConfig.outputCfg.channels = channelMask; - mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; - mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; - mConfig.inputCfg.samplingRate = thread->sampleRate(); - mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; - mConfig.inputCfg.bufferProvider.cookie = NULL; - mConfig.inputCfg.bufferProvider.getBuffer = NULL; - mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; - mConfig.outputCfg.bufferProvider.cookie = NULL; - mConfig.outputCfg.bufferProvider.getBuffer = NULL; - mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; - mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; - // Insert effect: - // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, - // always overwrites output buffer: input buffer == output buffer - // - in other sessions: - // last effect in the chain accumulates in output buffer: input buffer != output buffer - // other effect: overwrites output buffer: input buffer == output buffer - // Auxiliary effect: - // accumulates in output buffer: input buffer != output buffer - // Therefore: accumulate <=> input buffer != output buffer - if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { - mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; - } else { - mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; - } - mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; - mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; - mConfig.inputCfg.buffer.frameCount = thread->frameCount(); - mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; - - ALOGV("configure() %p thread %p buffer %p framecount %d", - this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); - - status_t cmdStatus; - uint32_t size = sizeof(int); - status_t status = (*mEffectInterface)->command(mEffectInterface, - EFFECT_CMD_SET_CONFIG, - sizeof(effect_config_t), - &mConfig, - &size, - &cmdStatus); - if (status == 0) { - status = cmdStatus; - } - - if (status == 0 && - (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { - uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; - effect_param_t *p = (effect_param_t *)buf32; - - p->psize = sizeof(uint32_t); - p->vsize = sizeof(uint32_t); - size = sizeof(int); - *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; - - uint32_t latency = 0; - PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); - if (pbt != NULL) { - latency = pbt->latency_l(); - } - - *((int32_t *)p->data + 1)= latency; - (*mEffectInterface)->command(mEffectInterface, - EFFECT_CMD_SET_PARAM, - sizeof(effect_param_t) + 8, - &buf32, - &size, - &cmdStatus); - } - - mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / - (1000 * mConfig.outputCfg.buffer.frameCount); - - return status; -} - -status_t AudioFlinger::EffectModule::init() -{ - Mutex::Autolock _l(mLock); - if (mEffectInterface == NULL) { - return NO_INIT; - } - status_t cmdStatus; - uint32_t size = sizeof(status_t); - status_t status = (*mEffectInterface)->command(mEffectInterface, - EFFECT_CMD_INIT, - 0, - NULL, - &size, - &cmdStatus); - if (status == 0) { - status = cmdStatus; - } - return status; -} - -status_t AudioFlinger::EffectModule::start() -{ - Mutex::Autolock _l(mLock); - return start_l(); -} - -status_t AudioFlinger::EffectModule::start_l() -{ - if (mEffectInterface == NULL) { - return NO_INIT; - } - status_t cmdStatus; - uint32_t size = sizeof(status_t); - status_t status = (*mEffectInterface)->command(mEffectInterface, - EFFECT_CMD_ENABLE, - 0, - NULL, - &size, - &cmdStatus); - if (status == 0) { - status = cmdStatus; - } - if (status == 0 && - ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || - (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - audio_stream_t *stream = thread->stream(); - if (stream != NULL) { - stream->add_audio_effect(stream, mEffectInterface); - } - } - } - return status; -} - -status_t AudioFlinger::EffectModule::stop() -{ - Mutex::Autolock _l(mLock); - return stop_l(); -} - -status_t AudioFlinger::EffectModule::stop_l() -{ - if (mEffectInterface == NULL) { - return NO_INIT; - } - status_t cmdStatus; - uint32_t size = sizeof(status_t); - status_t status = (*mEffectInterface)->command(mEffectInterface, - EFFECT_CMD_DISABLE, - 0, - NULL, - &size, - &cmdStatus); - if (status == 0) { - status = cmdStatus; - } - if (status == 0 && - ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || - (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { - sp<ThreadBase> thread = mThread.promote(); - if (thread != 0) { - audio_stream_t *stream = thread->stream(); - if (stream != NULL) { - stream->remove_audio_effect(stream, mEffectInterface); - } - } - } - return status; -} - -status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, - uint32_t cmdSize, - void *pCmdData, - uint32_t *replySize, - void *pReplyData) -{ - Mutex::Autolock _l(mLock); - ALOGVV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); - - if (mState == DESTROYED || mEffectInterface == NULL) { - return NO_INIT; - } - status_t status = (*mEffectInterface)->command(mEffectInterface, - cmdCode, - cmdSize, - pCmdData, - replySize, - pReplyData); - if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { - uint32_t size = (replySize == NULL) ? 0 : *replySize; - for (size_t i = 1; i < mHandles.size(); i++) { - EffectHandle *h = mHandles[i]; - if (h != NULL && !h->destroyed_l()) { - h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); - } - } - } - return status; -} - -status_t AudioFlinger::EffectModule::setEnabled(bool enabled) -{ - Mutex::Autolock _l(mLock); - return setEnabled_l(enabled); -} - -// must be called with EffectModule::mLock held -status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) -{ - - ALOGV("setEnabled %p enabled %d", this, enabled); - - if (enabled != isEnabled()) { - status_t status = AudioSystem::setEffectEnabled(mId, enabled); - if (enabled && status != NO_ERROR) { - return status; - } - - switch (mState) { - // going from disabled to enabled - case IDLE: - mState = STARTING; - break; - case STOPPED: - mState = RESTART; - break; - case STOPPING: - mState = ACTIVE; - break; - - // going from enabled to disabled - case RESTART: - mState = STOPPED; - break; - case STARTING: - mState = IDLE; - break; - case ACTIVE: - mState = STOPPING; - break; - case DESTROYED: - return NO_ERROR; // simply ignore as we are being destroyed - } - for (size_t i = 1; i < mHandles.size(); i++) { - EffectHandle *h = mHandles[i]; - if (h != NULL && !h->destroyed_l()) { - h->setEnabled(enabled); - } - } - } - return NO_ERROR; -} - -bool AudioFlinger::EffectModule::isEnabled() const -{ - switch (mState) { - case RESTART: - case STARTING: - case ACTIVE: - return true; - case IDLE: - case STOPPING: - case STOPPED: - case DESTROYED: - default: - return false; - } -} - -bool AudioFlinger::EffectModule::isProcessEnabled() const -{ - switch (mState) { - case RESTART: - case ACTIVE: - case STOPPING: - case STOPPED: - return true; - case IDLE: - case STARTING: - case DESTROYED: - default: - return false; - } -} - -status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) -{ - Mutex::Autolock _l(mLock); - status_t status = NO_ERROR; - - // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume - // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) - if (isProcessEnabled() && - ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || - (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { - status_t cmdStatus; - uint32_t volume[2]; - uint32_t *pVolume = NULL; - uint32_t size = sizeof(volume); - volume[0] = *left; - volume[1] = *right; - if (controller) { - pVolume = volume; - } - status = (*mEffectInterface)->command(mEffectInterface, - EFFECT_CMD_SET_VOLUME, - size, - volume, - &size, - pVolume); - if (controller && status == NO_ERROR && size == sizeof(volume)) { - *left = volume[0]; - *right = volume[1]; - } - } - return status; -} - -status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) -{ - if (device == AUDIO_DEVICE_NONE) { - return NO_ERROR; - } - - Mutex::Autolock _l(mLock); - status_t status = NO_ERROR; - if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { - status_t cmdStatus; - uint32_t size = sizeof(status_t); - uint32_t cmd = audio_is_output_devices(device) ? EFFECT_CMD_SET_DEVICE : - EFFECT_CMD_SET_INPUT_DEVICE; - status = (*mEffectInterface)->command(mEffectInterface, - cmd, - sizeof(uint32_t), - &device, - &size, - &cmdStatus); - } - return status; -} - -status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) -{ - Mutex::Autolock _l(mLock); - status_t status = NO_ERROR; - if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { - status_t cmdStatus; - uint32_t size = sizeof(status_t); - status = (*mEffectInterface)->command(mEffectInterface, - EFFECT_CMD_SET_AUDIO_MODE, - sizeof(audio_mode_t), - &mode, - &size, - &cmdStatus); - if (status == NO_ERROR) { - status = cmdStatus; - } - } - return status; -} - -status_t AudioFlinger::EffectModule::setAudioSource(audio_source_t source) -{ - Mutex::Autolock _l(mLock); - status_t status = NO_ERROR; - if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_SOURCE_MASK) == EFFECT_FLAG_AUDIO_SOURCE_IND) { - uint32_t size = 0; - status = (*mEffectInterface)->command(mEffectInterface, - EFFECT_CMD_SET_AUDIO_SOURCE, - sizeof(audio_source_t), - &source, - &size, - NULL); - } - return status; -} - -void AudioFlinger::EffectModule::setSuspended(bool suspended) -{ - Mutex::Autolock _l(mLock); - mSuspended = suspended; -} - -bool AudioFlinger::EffectModule::suspended() const -{ - Mutex::Autolock _l(mLock); - return mSuspended; -} - -bool AudioFlinger::EffectModule::purgeHandles() -{ - bool enabled = false; - Mutex::Autolock _l(mLock); - for (size_t i = 0; i < mHandles.size(); i++) { - EffectHandle *handle = mHandles[i]; - if (handle != NULL && !handle->destroyed_l()) { - handle->effect().clear(); - if (handle->hasControl()) { - enabled = handle->enabled(); - } - } - } - return enabled; -} - -void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); - result.append(buffer); - - bool locked = tryLock(mLock); - // failed to lock - AudioFlinger is probably deadlocked - if (!locked) { - result.append("\t\tCould not lock Fx mutex:\n"); - } - - result.append("\t\tSession Status State Engine:\n"); - snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", - mSessionId, mStatus, mState, (uint32_t)mEffectInterface); - result.append(buffer); - - result.append("\t\tDescriptor:\n"); - snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", - mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, - mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1], - mDescriptor.uuid.node[2], - mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); - result.append(buffer); - snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", - mDescriptor.type.timeLow, mDescriptor.type.timeMid, - mDescriptor.type.timeHiAndVersion, - mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1], - mDescriptor.type.node[2], - mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); - result.append(buffer); - snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", - mDescriptor.apiVersion, - mDescriptor.flags); - result.append(buffer); - snprintf(buffer, SIZE, "\t\t- name: %s\n", - mDescriptor.name); - result.append(buffer); - snprintf(buffer, SIZE, "\t\t- implementor: %s\n", - mDescriptor.implementor); - result.append(buffer); - - result.append("\t\t- Input configuration:\n"); - result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); - snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", - (uint32_t)mConfig.inputCfg.buffer.raw, - mConfig.inputCfg.buffer.frameCount, - mConfig.inputCfg.samplingRate, - mConfig.inputCfg.channels, - mConfig.inputCfg.format); - result.append(buffer); - - result.append("\t\t- Output configuration:\n"); - result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); - snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", - (uint32_t)mConfig.outputCfg.buffer.raw, - mConfig.outputCfg.buffer.frameCount, - mConfig.outputCfg.samplingRate, - mConfig.outputCfg.channels, - mConfig.outputCfg.format); - result.append(buffer); - - snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); - result.append(buffer); - result.append("\t\t\tPid Priority Ctrl Locked client server\n"); - for (size_t i = 0; i < mHandles.size(); ++i) { - EffectHandle *handle = mHandles[i]; - if (handle != NULL && !handle->destroyed_l()) { - handle->dump(buffer, SIZE); - result.append(buffer); - } - } - - result.append("\n"); - - write(fd, result.string(), result.length()); - - if (locked) { - mLock.unlock(); - } -} - -// ---------------------------------------------------------------------------- -// EffectHandle implementation -// ---------------------------------------------------------------------------- - -#undef LOG_TAG -#define LOG_TAG "AudioFlinger::EffectHandle" - -AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, - const sp<AudioFlinger::Client>& client, - const sp<IEffectClient>& effectClient, - int32_t priority) - : BnEffect(), - mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), - mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) -{ - ALOGV("constructor %p", this); - - if (client == 0) { - return; - } - int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); - mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); - if (mCblkMemory != 0) { - mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); - - if (mCblk != NULL) { - new(mCblk) effect_param_cblk_t(); - mBuffer = (uint8_t *)mCblk + bufOffset; - } - } else { - ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + - sizeof(effect_param_cblk_t)); - return; - } -} - -AudioFlinger::EffectHandle::~EffectHandle() -{ - ALOGV("Destructor %p", this); - - if (mEffect == 0) { - mDestroyed = true; - return; - } - mEffect->lock(); - mDestroyed = true; - mEffect->unlock(); - disconnect(false); -} - -status_t AudioFlinger::EffectHandle::enable() -{ - ALOGV("enable %p", this); - if (!mHasControl) { - return INVALID_OPERATION; - } - if (mEffect == 0) { - return DEAD_OBJECT; - } - - if (mEnabled) { - return NO_ERROR; - } - - mEnabled = true; - - sp<ThreadBase> thread = mEffect->thread().promote(); - if (thread != 0) { - thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); - } - - // checkSuspendOnEffectEnabled() can suspend this same effect when enabled - if (mEffect->suspended()) { - return NO_ERROR; - } - - status_t status = mEffect->setEnabled(true); - if (status != NO_ERROR) { - if (thread != 0) { - thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); - } - mEnabled = false; - } - return status; -} - -status_t AudioFlinger::EffectHandle::disable() -{ - ALOGV("disable %p", this); - if (!mHasControl) { - return INVALID_OPERATION; - } - if (mEffect == 0) { - return DEAD_OBJECT; - } - - if (!mEnabled) { - return NO_ERROR; - } - mEnabled = false; - - if (mEffect->suspended()) { - return NO_ERROR; - } - - status_t status = mEffect->setEnabled(false); - - sp<ThreadBase> thread = mEffect->thread().promote(); - if (thread != 0) { - thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); - } - - return status; -} - -void AudioFlinger::EffectHandle::disconnect() -{ - disconnect(true); -} - -void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) -{ - ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); - if (mEffect == 0) { - return; - } - // restore suspended effects if the disconnected handle was enabled and the last one. - if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { - sp<ThreadBase> thread = mEffect->thread().promote(); - if (thread != 0) { - thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); - } - } - - // release sp on module => module destructor can be called now - mEffect.clear(); - if (mClient != 0) { - if (mCblk != NULL) { - // unlike ~TrackBase(), mCblk is never a local new, so don't delete - mCblk->~effect_param_cblk_t(); // destroy our shared-structure. - } - mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to - // Client destructor must run with AudioFlinger mutex locked - Mutex::Autolock _l(mClient->audioFlinger()->mLock); - mClient.clear(); - } -} - -status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, - uint32_t cmdSize, - void *pCmdData, - uint32_t *replySize, - void *pReplyData) -{ - ALOGVV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", - cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); - - // only get parameter command is permitted for applications not controlling the effect - if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { - return INVALID_OPERATION; - } - if (mEffect == 0) { - return DEAD_OBJECT; - } - if (mClient == 0) { - return INVALID_OPERATION; - } - - // handle commands that are not forwarded transparently to effect engine - if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { - // No need to trylock() here as this function is executed in the binder thread serving a - // particular client process: no risk to block the whole media server process or mixer - // threads if we are stuck here - Mutex::Autolock _l(mCblk->lock); - if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || - mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { - mCblk->serverIndex = 0; - mCblk->clientIndex = 0; - return BAD_VALUE; - } - status_t status = NO_ERROR; - while (mCblk->serverIndex < mCblk->clientIndex) { - int reply; - uint32_t rsize = sizeof(int); - int *p = (int *)(mBuffer + mCblk->serverIndex); - int size = *p++; - if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { - ALOGW("command(): invalid parameter block size"); - break; - } - effect_param_t *param = (effect_param_t *)p; - if (param->psize == 0 || param->vsize == 0) { - ALOGW("command(): null parameter or value size"); - mCblk->serverIndex += size; - continue; - } - uint32_t psize = sizeof(effect_param_t) + - ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + - param->vsize; - status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, - psize, - p, - &rsize, - &reply); - // stop at first error encountered - if (ret != NO_ERROR) { - status = ret; - *(int *)pReplyData = reply; - break; - } else if (reply != NO_ERROR) { - *(int *)pReplyData = reply; - break; - } - mCblk->serverIndex += size; - } - mCblk->serverIndex = 0; - mCblk->clientIndex = 0; - return status; - } else if (cmdCode == EFFECT_CMD_ENABLE) { - *(int *)pReplyData = NO_ERROR; - return enable(); - } else if (cmdCode == EFFECT_CMD_DISABLE) { - *(int *)pReplyData = NO_ERROR; - return disable(); - } - - return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); -} - -void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) -{ - ALOGV("setControl %p control %d", this, hasControl); - - mHasControl = hasControl; - mEnabled = enabled; - - if (signal && mEffectClient != 0) { - mEffectClient->controlStatusChanged(hasControl); - } -} - -void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, - uint32_t cmdSize, - void *pCmdData, - uint32_t replySize, - void *pReplyData) -{ - if (mEffectClient != 0) { - mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); - } -} - - - -void AudioFlinger::EffectHandle::setEnabled(bool enabled) -{ - if (mEffectClient != 0) { - mEffectClient->enableStatusChanged(enabled); - } -} - -status_t AudioFlinger::EffectHandle::onTransact( - uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) -{ - return BnEffect::onTransact(code, data, reply, flags); -} - - -void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) -{ - bool locked = mCblk != NULL && tryLock(mCblk->lock); - - snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", - (mClient == 0) ? getpid_cached : mClient->pid(), - mPriority, - mHasControl, - !locked, - mCblk ? mCblk->clientIndex : 0, - mCblk ? mCblk->serverIndex : 0 - ); - - if (locked) { - mCblk->lock.unlock(); - } -} - -#undef LOG_TAG -#define LOG_TAG "AudioFlinger::EffectChain" - -AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, - int sessionId) - : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), - mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), - mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) -{ - mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); - if (thread == NULL) { - return; - } - mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / - thread->frameCount(); -} - -AudioFlinger::EffectChain::~EffectChain() -{ - if (mOwnInBuffer) { - delete mInBuffer; - } - -} - -// getEffectFromDesc_l() must be called with ThreadBase::mLock held -sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l( - effect_descriptor_t *descriptor) -{ - size_t size = mEffects.size(); - - for (size_t i = 0; i < size; i++) { - if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { - return mEffects[i]; - } - } - return 0; -} - -// getEffectFromId_l() must be called with ThreadBase::mLock held -sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) -{ - size_t size = mEffects.size(); - - for (size_t i = 0; i < size; i++) { - // by convention, return first effect if id provided is 0 (0 is never a valid id) - if (id == 0 || mEffects[i]->id() == id) { - return mEffects[i]; - } - } - return 0; -} - -// getEffectFromType_l() must be called with ThreadBase::mLock held -sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( - const effect_uuid_t *type) -{ - size_t size = mEffects.size(); - - for (size_t i = 0; i < size; i++) { - if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { - return mEffects[i]; - } - } - return 0; -} - -void AudioFlinger::EffectChain::clearInputBuffer() -{ - Mutex::Autolock _l(mLock); - sp<ThreadBase> thread = mThread.promote(); - if (thread == 0) { - ALOGW("clearInputBuffer(): cannot promote mixer thread"); - return; - } - clearInputBuffer_l(thread); -} - -// Must be called with EffectChain::mLock locked -void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) -{ - size_t numSamples = thread->frameCount() * thread->channelCount(); - memset(mInBuffer, 0, numSamples * sizeof(int16_t)); - -} - -// Must be called with EffectChain::mLock locked -void AudioFlinger::EffectChain::process_l() -{ - sp<ThreadBase> thread = mThread.promote(); - if (thread == 0) { - ALOGW("process_l(): cannot promote mixer thread"); - return; - } - bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || - (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); - // always process effects unless no more tracks are on the session and the effect tail - // has been rendered - bool doProcess = true; - if (!isGlobalSession) { - bool tracksOnSession = (trackCnt() != 0); - - if (!tracksOnSession && mTailBufferCount == 0) { - doProcess = false; - } - - if (activeTrackCnt() == 0) { - // if no track is active and the effect tail has not been rendered, - // the input buffer must be cleared here as the mixer process will not do it - if (tracksOnSession || mTailBufferCount > 0) { - clearInputBuffer_l(thread); - if (mTailBufferCount > 0) { - mTailBufferCount--; - } - } - } - } - - size_t size = mEffects.size(); - if (doProcess) { - for (size_t i = 0; i < size; i++) { - mEffects[i]->process(); - } - } - for (size_t i = 0; i < size; i++) { - mEffects[i]->updateState(); - } -} - -// addEffect_l() must be called with PlaybackThread::mLock held -status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) -{ - effect_descriptor_t desc = effect->desc(); - uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; - - Mutex::Autolock _l(mLock); - effect->setChain(this); - sp<ThreadBase> thread = mThread.promote(); - if (thread == 0) { - return NO_INIT; - } - effect->setThread(thread); - - if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { - // Auxiliary effects are inserted at the beginning of mEffects vector as - // they are processed first and accumulated in chain input buffer - mEffects.insertAt(effect, 0); - - // the input buffer for auxiliary effect contains mono samples in - // 32 bit format. This is to avoid saturation in AudoMixer - // accumulation stage. Saturation is done in EffectModule::process() before - // calling the process in effect engine - size_t numSamples = thread->frameCount(); - int32_t *buffer = new int32_t[numSamples]; - memset(buffer, 0, numSamples * sizeof(int32_t)); - effect->setInBuffer((int16_t *)buffer); - // auxiliary effects output samples to chain input buffer for further processing - // by insert effects - effect->setOutBuffer(mInBuffer); - } else { - // Insert effects are inserted at the end of mEffects vector as they are processed - // after track and auxiliary effects. - // Insert effect order as a function of indicated preference: - // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if - // another effect is present - // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the - // last effect claiming first position - // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the - // first effect claiming last position - // else if EFFECT_FLAG_INSERT_ANY insert after first or before last - // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is - // already present - - size_t size = mEffects.size(); - size_t idx_insert = size; - ssize_t idx_insert_first = -1; - ssize_t idx_insert_last = -1; - - for (size_t i = 0; i < size; i++) { - effect_descriptor_t d = mEffects[i]->desc(); - uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; - uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; - if (iMode == EFFECT_FLAG_TYPE_INSERT) { - // check invalid effect chaining combinations - if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || - iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { - ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", - desc.name, d.name); - return INVALID_OPERATION; - } - // remember position of first insert effect and by default - // select this as insert position for new effect - if (idx_insert == size) { - idx_insert = i; - } - // remember position of last insert effect claiming - // first position - if (iPref == EFFECT_FLAG_INSERT_FIRST) { - idx_insert_first = i; - } - // remember position of first insert effect claiming - // last position - if (iPref == EFFECT_FLAG_INSERT_LAST && - idx_insert_last == -1) { - idx_insert_last = i; - } - } - } - - // modify idx_insert from first position if needed - if (insertPref == EFFECT_FLAG_INSERT_LAST) { - if (idx_insert_last != -1) { - idx_insert = idx_insert_last; - } else { - idx_insert = size; - } - } else { - if (idx_insert_first != -1) { - idx_insert = idx_insert_first + 1; - } - } - - // always read samples from chain input buffer - effect->setInBuffer(mInBuffer); - - // if last effect in the chain, output samples to chain - // output buffer, otherwise to chain input buffer - if (idx_insert == size) { - if (idx_insert != 0) { - mEffects[idx_insert-1]->setOutBuffer(mInBuffer); - mEffects[idx_insert-1]->configure(); - } - effect->setOutBuffer(mOutBuffer); - } else { - effect->setOutBuffer(mInBuffer); - } - mEffects.insertAt(effect, idx_insert); - - ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, - idx_insert); - } - effect->configure(); - return NO_ERROR; -} - -// removeEffect_l() must be called with PlaybackThread::mLock held -size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) -{ - Mutex::Autolock _l(mLock); - size_t size = mEffects.size(); - uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; - - for (size_t i = 0; i < size; i++) { - if (effect == mEffects[i]) { - // calling stop here will remove pre-processing effect from the audio HAL. - // This is safe as we hold the EffectChain mutex which guarantees that we are not in - // the middle of a read from audio HAL - if (mEffects[i]->state() == EffectModule::ACTIVE || - mEffects[i]->state() == EffectModule::STOPPING) { - mEffects[i]->stop(); - } - if (type == EFFECT_FLAG_TYPE_AUXILIARY) { - delete[] effect->inBuffer(); - } else { - if (i == size - 1 && i != 0) { - mEffects[i - 1]->setOutBuffer(mOutBuffer); - mEffects[i - 1]->configure(); - } - } - mEffects.removeAt(i); - ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), - this, i); - break; - } - } - - return mEffects.size(); -} - -// setDevice_l() must be called with PlaybackThread::mLock held -void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) -{ - size_t size = mEffects.size(); - for (size_t i = 0; i < size; i++) { - mEffects[i]->setDevice(device); - } -} - -// setMode_l() must be called with PlaybackThread::mLock held -void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) -{ - size_t size = mEffects.size(); - for (size_t i = 0; i < size; i++) { - mEffects[i]->setMode(mode); - } -} - -// setAudioSource_l() must be called with PlaybackThread::mLock held -void AudioFlinger::EffectChain::setAudioSource_l(audio_source_t source) -{ - size_t size = mEffects.size(); - for (size_t i = 0; i < size; i++) { - mEffects[i]->setAudioSource(source); - } -} - -// setVolume_l() must be called with PlaybackThread::mLock held -bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) -{ - uint32_t newLeft = *left; - uint32_t newRight = *right; - bool hasControl = false; - int ctrlIdx = -1; - size_t size = mEffects.size(); - - // first update volume controller - for (size_t i = size; i > 0; i--) { - if (mEffects[i - 1]->isProcessEnabled() && - (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { - ctrlIdx = i - 1; - hasControl = true; - break; - } - } - - if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { - if (hasControl) { - *left = mNewLeftVolume; - *right = mNewRightVolume; - } - return hasControl; - } - - mVolumeCtrlIdx = ctrlIdx; - mLeftVolume = newLeft; - mRightVolume = newRight; - - // second get volume update from volume controller - if (ctrlIdx >= 0) { - mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); - mNewLeftVolume = newLeft; - mNewRightVolume = newRight; - } - // then indicate volume to all other effects in chain. - // Pass altered volume to effects before volume controller - // and requested volume to effects after controller - uint32_t lVol = newLeft; - uint32_t rVol = newRight; - - for (size_t i = 0; i < size; i++) { - if ((int)i == ctrlIdx) { - continue; - } - // this also works for ctrlIdx == -1 when there is no volume controller - if ((int)i > ctrlIdx) { - lVol = *left; - rVol = *right; - } - mEffects[i]->setVolume(&lVol, &rVol, false); - } - *left = newLeft; - *right = newRight; - - return hasControl; -} - -void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) -{ - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); - result.append(buffer); - - bool locked = tryLock(mLock); - // failed to lock - AudioFlinger is probably deadlocked - if (!locked) { - result.append("\tCould not lock mutex:\n"); - } - - result.append("\tNum fx In buffer Out buffer Active tracks:\n"); - snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", - mEffects.size(), - (uint32_t)mInBuffer, - (uint32_t)mOutBuffer, - mActiveTrackCnt); - result.append(buffer); - write(fd, result.string(), result.size()); - - for (size_t i = 0; i < mEffects.size(); ++i) { - sp<EffectModule> effect = mEffects[i]; - if (effect != 0) { - effect->dump(fd, args); - } - } - - if (locked) { - mLock.unlock(); - } -} - -// must be called with ThreadBase::mLock held -void AudioFlinger::EffectChain::setEffectSuspended_l( - const effect_uuid_t *type, bool suspend) +void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) { - sp<SuspendedEffectDesc> desc; - // use effect type UUID timelow as key as there is no real risk of identical - // timeLow fields among effect type UUIDs. - ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); - if (suspend) { - if (index >= 0) { - desc = mSuspendedEffects.valueAt(index); - } else { - desc = new SuspendedEffectDesc(); - desc->mType = *type; - mSuspendedEffects.add(type->timeLow, desc); - ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); - } - if (desc->mRefCount++ == 0) { - sp<EffectModule> effect = getEffectIfEnabled(type); - if (effect != 0) { - desc->mEffect = effect; - effect->setSuspended(true); - effect->setEnabled(false); - } - } - } else { - if (index < 0) { - return; - } - desc = mSuspendedEffects.valueAt(index); - if (desc->mRefCount <= 0) { - ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); - desc->mRefCount = 1; - } - if (--desc->mRefCount == 0) { - ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); - if (desc->mEffect != 0) { - sp<EffectModule> effect = desc->mEffect.promote(); - if (effect != 0) { - effect->setSuspended(false); - effect->lock(); - EffectHandle *handle = effect->controlHandle_l(); - if (handle != NULL && !handle->destroyed_l()) { - effect->setEnabled_l(handle->enabled()); + NBAIO_Source *teeSource = source.get(); + if (teeSource != NULL) { + char teeTime[16]; + struct timeval tv; + gettimeofday(&tv, NULL); + struct tm tm; + localtime_r(&tv.tv_sec, &tm); + strftime(teeTime, sizeof(teeTime), "%T", &tm); + char teePath[64]; + sprintf(teePath, "/data/misc/media/%s_%d.wav", teeTime, id); + int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); + if (teeFd >= 0) { + char wavHeader[44]; + memcpy(wavHeader, + "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0", + sizeof(wavHeader)); + NBAIO_Format format = teeSource->format(); + unsigned channelCount = Format_channelCount(format); + ALOG_ASSERT(channelCount <= FCC_2); + uint32_t sampleRate = Format_sampleRate(format); + wavHeader[22] = channelCount; // number of channels + wavHeader[24] = sampleRate; // sample rate + wavHeader[25] = sampleRate >> 8; + wavHeader[32] = channelCount * 2; // block alignment + write(teeFd, wavHeader, sizeof(wavHeader)); + size_t total = 0; + bool firstRead = true; + for (;;) { +#define TEE_SINK_READ 1024 + short buffer[TEE_SINK_READ * FCC_2]; + size_t count = TEE_SINK_READ; + ssize_t actual = teeSource->read(buffer, count, + AudioBufferProvider::kInvalidPTS); + bool wasFirstRead = firstRead; + firstRead = false; + if (actual <= 0) { + if (actual == (ssize_t) OVERRUN && wasFirstRead) { + continue; } - effect->unlock(); + break; } - desc->mEffect.clear(); + ALOG_ASSERT(actual <= (ssize_t)count); + write(teeFd, buffer, actual * channelCount * sizeof(short)); + total += actual; } - mSuspendedEffects.removeItemsAt(index); - } - } -} - -// must be called with ThreadBase::mLock held -void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) -{ - sp<SuspendedEffectDesc> desc; - - ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); - if (suspend) { - if (index >= 0) { - desc = mSuspendedEffects.valueAt(index); + lseek(teeFd, (off_t) 4, SEEK_SET); + uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; + write(teeFd, &temp, sizeof(temp)); + lseek(teeFd, (off_t) 40, SEEK_SET); + temp = total * channelCount * sizeof(short); + write(teeFd, &temp, sizeof(temp)); + close(teeFd); + fdprintf(fd, "FastMixer tee copied to %s\n", teePath); } else { - desc = new SuspendedEffectDesc(); - mSuspendedEffects.add((int)kKeyForSuspendAll, desc); - ALOGV("setEffectSuspendedAll_l() add entry for 0"); - } - if (desc->mRefCount++ == 0) { - Vector< sp<EffectModule> > effects; - getSuspendEligibleEffects(effects); - for (size_t i = 0; i < effects.size(); i++) { - setEffectSuspended_l(&effects[i]->desc().type, true); - } - } - } else { - if (index < 0) { - return; - } - desc = mSuspendedEffects.valueAt(index); - if (desc->mRefCount <= 0) { - ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); - desc->mRefCount = 1; - } - if (--desc->mRefCount == 0) { - Vector<const effect_uuid_t *> types; - for (size_t i = 0; i < mSuspendedEffects.size(); i++) { - if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { - continue; - } - types.add(&mSuspendedEffects.valueAt(i)->mType); - } - for (size_t i = 0; i < types.size(); i++) { - setEffectSuspended_l(types[i], false); - } - ALOGV("setEffectSuspendedAll_l() remove entry for %08x", - mSuspendedEffects.keyAt(index)); - mSuspendedEffects.removeItem((int)kKeyForSuspendAll); - } - } -} - - -// The volume effect is used for automated tests only -#ifndef OPENSL_ES_H_ -static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, - { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; -const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; -#endif //OPENSL_ES_H_ - -bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) -{ - // auxiliary effects and visualizer are never suspended on output mix - if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && - (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || - (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || - (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { - return false; - } - return true; -} - -void AudioFlinger::EffectChain::getSuspendEligibleEffects( - Vector< sp<AudioFlinger::EffectModule> > &effects) -{ - effects.clear(); - for (size_t i = 0; i < mEffects.size(); i++) { - if (isEffectEligibleForSuspend(mEffects[i]->desc())) { - effects.add(mEffects[i]); - } - } -} - -sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( - const effect_uuid_t *type) -{ - sp<EffectModule> effect = getEffectFromType_l(type); - return effect != 0 && effect->isEnabled() ? effect : 0; -} - -void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, - bool enabled) -{ - ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); - if (enabled) { - if (index < 0) { - // if the effect is not suspend check if all effects are suspended - index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); - if (index < 0) { - return; - } - if (!isEffectEligibleForSuspend(effect->desc())) { - return; - } - setEffectSuspended_l(&effect->desc().type, enabled); - index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); - if (index < 0) { - ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); - return; - } - } - ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", - effect->desc().type.timeLow); - sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); - // if effect is requested to suspended but was not yet enabled, supend it now. - if (desc->mEffect == 0) { - desc->mEffect = effect; - effect->setEnabled(false); - effect->setSuspended(true); - } - } else { - if (index < 0) { - return; + fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); } - ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", - effect->desc().type.timeLow); - sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); - desc->mEffect.clear(); - effect->setSuspended(false); } } -#undef LOG_TAG -#define LOG_TAG "AudioFlinger" - // ---------------------------------------------------------------------------- status_t AudioFlinger::onTransact( diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h index 2541b15..46a8e0f 100644 --- a/services/audioflinger/AudioFlinger.h +++ b/services/audioflinger/AudioFlinger.h @@ -75,6 +75,11 @@ class FastMixer; static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); +#define MAX_GAIN 4096.0f +#define MAX_GAIN_INT 0x1000 + +#define INCLUDING_FROM_AUDIOFLINGER_H + class AudioFlinger : public BinderService<AudioFlinger>, public BnAudioFlinger @@ -283,7 +288,14 @@ private: // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs static nsecs_t mStandbyTimeInNsecs; + // incremented by 2 when screen state changes, bit 0 == 1 means "off" + // AudioFlinger::setParameters() updates, other threads read w/o lock + static uint32_t mScreenState; + // Internal dump utilities. + static const int kDumpLockRetries = 50; + static const int kDumpLockSleepUs = 20000; + static bool dumpTryLock(Mutex& mutex); void dumpPermissionDenial(int fd, const Vector<String16>& args); void dumpClients(int fd, const Vector<String16>& args); void dumpInternals(int fd, const Vector<String16>& args); @@ -348,419 +360,6 @@ private: struct AudioStreamOut; struct AudioStreamIn; - class ThreadBase : public Thread { - public: - - enum type_t { - MIXER, // Thread class is MixerThread - DIRECT, // Thread class is DirectOutputThread - DUPLICATING, // Thread class is DuplicatingThread - RECORD // Thread class is RecordThread - }; - - ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, - audio_devices_t outDevice, audio_devices_t inDevice, type_t type); - virtual ~ThreadBase(); - - void dumpBase(int fd, const Vector<String16>& args); - void dumpEffectChains(int fd, const Vector<String16>& args); - - void clearPowerManager(); - - // base for record and playback - class TrackBase : public ExtendedAudioBufferProvider, public RefBase { - - public: - enum track_state { - IDLE, - TERMINATED, - FLUSHED, - STOPPED, - // next 2 states are currently used for fast tracks only - STOPPING_1, // waiting for first underrun - STOPPING_2, // waiting for presentation complete - RESUMING, - ACTIVE, - PAUSING, - PAUSED - }; - - TrackBase(ThreadBase *thread, - const sp<Client>& client, - uint32_t sampleRate, - audio_format_t format, - audio_channel_mask_t channelMask, - size_t frameCount, - const sp<IMemory>& sharedBuffer, - int sessionId); - virtual ~TrackBase(); - - virtual status_t start(AudioSystem::sync_event_t event, - int triggerSession) = 0; - virtual void stop() = 0; - sp<IMemory> getCblk() const { return mCblkMemory; } - audio_track_cblk_t* cblk() const { return mCblk; } - int sessionId() const { return mSessionId; } - virtual status_t setSyncEvent(const sp<SyncEvent>& event); - - protected: - TrackBase(const TrackBase&); - TrackBase& operator = (const TrackBase&); - - // AudioBufferProvider interface - virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; - virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); - - // ExtendedAudioBufferProvider interface is only needed for Track, - // but putting it in TrackBase avoids the complexity of virtual inheritance - virtual size_t framesReady() const { return SIZE_MAX; } - - audio_format_t format() const { - return mFormat; - } - - uint32_t channelCount() const { return mChannelCount; } - - audio_channel_mask_t channelMask() const { return mChannelMask; } - - uint32_t sampleRate() const; // FIXME inline after cblk sr moved - - // Return a pointer to the start of a contiguous slice of the track buffer. - // Parameter 'offset' is the requested start position, expressed in - // monotonically increasing frame units relative to the track epoch. - // Parameter 'frames' is the requested length, also in frame units. - // Always returns non-NULL. It is the caller's responsibility to - // verify that this will be successful; the result of calling this - // function with invalid 'offset' or 'frames' is undefined. - void* getBuffer(uint32_t offset, uint32_t frames) const; - - bool isStopped() const { - return (mState == STOPPED || mState == FLUSHED); - } - - // for fast tracks only - bool isStopping() const { - return mState == STOPPING_1 || mState == STOPPING_2; - } - bool isStopping_1() const { - return mState == STOPPING_1; - } - bool isStopping_2() const { - return mState == STOPPING_2; - } - - bool isTerminated() const { - return mState == TERMINATED; - } - - bool step(); // mStepCount is an implicit input - void reset(); - - virtual bool isOut() const = 0; // true for Track and TimedTrack, false for RecordTrack, - // this could be a track type if needed later - - const wp<ThreadBase> mThread; - /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const - sp<IMemory> mCblkMemory; - audio_track_cblk_t* mCblk; - void* mBuffer; // start of track buffer, typically in shared memory - void* mBufferEnd; // &mBuffer[mFrameCount * frameSize], where frameSize - // is based on mChannelCount and 16-bit samples - uint32_t mStepCount; // saves AudioBufferProvider::Buffer::frameCount as of - // time of releaseBuffer() for later use by step() - // we don't really need a lock for these - track_state mState; - const uint32_t mSampleRate; // initial sample rate only; for tracks which - // support dynamic rates, the current value is in control block - const audio_format_t mFormat; - const audio_channel_mask_t mChannelMask; - const uint8_t mChannelCount; - const size_t mFrameSize; // AudioFlinger's view of frame size in shared memory, - // where for AudioTrack (but not AudioRecord), - // 8-bit PCM samples are stored as 16-bit - const size_t mFrameCount;// size of track buffer given at createTrack() or - // openRecord(), and then adjusted as needed - bool mStepServerFailed; - const int mSessionId; - Vector < sp<SyncEvent> >mSyncEvents; - }; - - enum { - CFG_EVENT_IO, - CFG_EVENT_PRIO - }; - - class ConfigEvent { - public: - ConfigEvent(int type) : mType(type) {} - virtual ~ConfigEvent() {} - - int type() const { return mType; } - - virtual void dump(char *buffer, size_t size) = 0; - - private: - const int mType; - }; - - class IoConfigEvent : public ConfigEvent { - public: - IoConfigEvent(int event, int param) : - ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {} - virtual ~IoConfigEvent() {} - - int event() const { return mEvent; } - int param() const { return mParam; } - - virtual void dump(char *buffer, size_t size) { - snprintf(buffer, size, "IO event: event %d, param %d\n", mEvent, mParam); - } - - private: - const int mEvent; - const int mParam; - }; - - class PrioConfigEvent : public ConfigEvent { - public: - PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : - ConfigEvent(CFG_EVENT_PRIO), mPid(pid), mTid(tid), mPrio(prio) {} - virtual ~PrioConfigEvent() {} - - pid_t pid() const { return mPid; } - pid_t tid() const { return mTid; } - int32_t prio() const { return mPrio; } - - virtual void dump(char *buffer, size_t size) { - snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); - } - - private: - const pid_t mPid; - const pid_t mTid; - const int32_t mPrio; - }; - - - class PMDeathRecipient : public IBinder::DeathRecipient { - public: - PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} - virtual ~PMDeathRecipient() {} - - // IBinder::DeathRecipient - virtual void binderDied(const wp<IBinder>& who); - - private: - PMDeathRecipient(const PMDeathRecipient&); - PMDeathRecipient& operator = (const PMDeathRecipient&); - - wp<ThreadBase> mThread; - }; - - virtual status_t initCheck() const = 0; - - // static externally-visible - type_t type() const { return mType; } - audio_io_handle_t id() const { return mId;} - - // dynamic externally-visible - uint32_t sampleRate() const { return mSampleRate; } - uint32_t channelCount() const { return mChannelCount; } - audio_channel_mask_t channelMask() const { return mChannelMask; } - audio_format_t format() const { return mFormat; } - // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, - // and returns the normal mix buffer's frame count. - size_t frameCount() const { return mNormalFrameCount; } - // Return's the HAL's frame count i.e. fast mixer buffer size. - size_t frameCountHAL() const { return mFrameCount; } - - // Should be "virtual status_t requestExitAndWait()" and override same - // method in Thread, but Thread::requestExitAndWait() is not yet virtual. - void exit(); - virtual bool checkForNewParameters_l() = 0; - virtual status_t setParameters(const String8& keyValuePairs); - virtual String8 getParameters(const String8& keys) = 0; - virtual void audioConfigChanged_l(int event, int param = 0) = 0; - void sendIoConfigEvent(int event, int param = 0); - void sendIoConfigEvent_l(int event, int param = 0); - void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); - void processConfigEvents(); - - // see note at declaration of mStandby, mOutDevice and mInDevice - bool standby() const { return mStandby; } - audio_devices_t outDevice() const { return mOutDevice; } - audio_devices_t inDevice() const { return mInDevice; } - - virtual audio_stream_t* stream() const = 0; - - sp<EffectHandle> createEffect_l( - const sp<AudioFlinger::Client>& client, - const sp<IEffectClient>& effectClient, - int32_t priority, - int sessionId, - effect_descriptor_t *desc, - int *enabled, - status_t *status); - void disconnectEffect(const sp< EffectModule>& effect, - EffectHandle *handle, - bool unpinIfLast); - - // return values for hasAudioSession (bit field) - enum effect_state { - EFFECT_SESSION = 0x1, // the audio session corresponds to at least one - // effect - TRACK_SESSION = 0x2 // the audio session corresponds to at least one - // track - }; - - // get effect chain corresponding to session Id. - sp<EffectChain> getEffectChain(int sessionId); - // same as getEffectChain() but must be called with ThreadBase mutex locked - sp<EffectChain> getEffectChain_l(int sessionId) const; - // add an effect chain to the chain list (mEffectChains) - virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; - // remove an effect chain from the chain list (mEffectChains) - virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; - // lock all effect chains Mutexes. Must be called before releasing the - // ThreadBase mutex before processing the mixer and effects. This guarantees the - // integrity of the chains during the process. - // Also sets the parameter 'effectChains' to current value of mEffectChains. - void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); - // unlock effect chains after process - void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); - // set audio mode to all effect chains - void setMode(audio_mode_t mode); - // get effect module with corresponding ID on specified audio session - sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); - sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); - // add and effect module. Also creates the effect chain is none exists for - // the effects audio session - status_t addEffect_l(const sp< EffectModule>& effect); - // remove and effect module. Also removes the effect chain is this was the last - // effect - void removeEffect_l(const sp< EffectModule>& effect); - // detach all tracks connected to an auxiliary effect - virtual void detachAuxEffect_l(int effectId) {} - // returns either EFFECT_SESSION if effects on this audio session exist in one - // chain, or TRACK_SESSION if tracks on this audio session exist, or both - virtual uint32_t hasAudioSession(int sessionId) const = 0; - // the value returned by default implementation is not important as the - // strategy is only meaningful for PlaybackThread which implements this method - virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } - - // suspend or restore effect according to the type of effect passed. a NULL - // type pointer means suspend all effects in the session - void setEffectSuspended(const effect_uuid_t *type, - bool suspend, - int sessionId = AUDIO_SESSION_OUTPUT_MIX); - // check if some effects must be suspended/restored when an effect is enabled - // or disabled - void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, - bool enabled, - int sessionId = AUDIO_SESSION_OUTPUT_MIX); - void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, - bool enabled, - int sessionId = AUDIO_SESSION_OUTPUT_MIX); - - virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; - virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; - - - mutable Mutex mLock; - - protected: - - // entry describing an effect being suspended in mSuspendedSessions keyed vector - class SuspendedSessionDesc : public RefBase { - public: - SuspendedSessionDesc() : mRefCount(0) {} - - int mRefCount; // number of active suspend requests - effect_uuid_t mType; // effect type UUID - }; - - void acquireWakeLock(); - void acquireWakeLock_l(); - void releaseWakeLock(); - void releaseWakeLock_l(); - void setEffectSuspended_l(const effect_uuid_t *type, - bool suspend, - int sessionId); - // updated mSuspendedSessions when an effect suspended or restored - void updateSuspendedSessions_l(const effect_uuid_t *type, - bool suspend, - int sessionId); - // check if some effects must be suspended when an effect chain is added - void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); - - virtual void preExit() { } - - friend class AudioFlinger; // for mEffectChains - - const type_t mType; - - // Used by parameters, config events, addTrack_l, exit - Condition mWaitWorkCV; - - const sp<AudioFlinger> mAudioFlinger; - uint32_t mSampleRate; - size_t mFrameCount; // output HAL, direct output, record - size_t mNormalFrameCount; // normal mixer and effects - audio_channel_mask_t mChannelMask; - uint16_t mChannelCount; - size_t mFrameSize; - audio_format_t mFormat; - - // Parameter sequence by client: binder thread calling setParameters(): - // 1. Lock mLock - // 2. Append to mNewParameters - // 3. mWaitWorkCV.signal - // 4. mParamCond.waitRelative with timeout - // 5. read mParamStatus - // 6. mWaitWorkCV.signal - // 7. Unlock - // - // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): - // 1. Lock mLock - // 2. If there is an entry in mNewParameters proceed ... - // 2. Read first entry in mNewParameters - // 3. Process - // 4. Remove first entry from mNewParameters - // 5. Set mParamStatus - // 6. mParamCond.signal - // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) - // 8. Unlock - Condition mParamCond; - Vector<String8> mNewParameters; - status_t mParamStatus; - - Vector<ConfigEvent *> mConfigEvents; - - // These fields are written and read by thread itself without lock or barrier, - // and read by other threads without lock or barrier via standby() , outDevice() - // and inDevice(). - // Because of the absence of a lock or barrier, any other thread that reads - // these fields must use the information in isolation, or be prepared to deal - // with possibility that it might be inconsistent with other information. - bool mStandby; // Whether thread is currently in standby. - audio_devices_t mOutDevice; // output device - audio_devices_t mInDevice; // input device - audio_source_t mAudioSource; // (see audio.h, audio_source_t) - - const audio_io_handle_t mId; - Vector< sp<EffectChain> > mEffectChains; - - static const int kNameLength = 16; // prctl(PR_SET_NAME) limit - char mName[kNameLength]; - sp<IPowerManager> mPowerManager; - sp<IBinder> mWakeLockToken; - const sp<PMDeathRecipient> mDeathRecipient; - // list of suspended effects per session and per type. The first vector is - // keyed by session ID, the second by type UUID timeLow field - KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > - mSuspendedSessions; - }; - struct stream_type_t { stream_type_t() : volume(1.0f), @@ -772,658 +371,10 @@ private: }; // --- PlaybackThread --- - class PlaybackThread : public ThreadBase { - public: - - enum mixer_state { - MIXER_IDLE, // no active tracks - MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready - MIXER_TRACKS_READY // at least one active track, and at least one track has data - // standby mode does not have an enum value - // suspend by audio policy manager is orthogonal to mixer state - }; - - // playback track - class Track : public TrackBase, public VolumeProvider { - public: - Track( PlaybackThread *thread, - const sp<Client>& client, - audio_stream_type_t streamType, - uint32_t sampleRate, - audio_format_t format, - audio_channel_mask_t channelMask, - size_t frameCount, - const sp<IMemory>& sharedBuffer, - int sessionId, - IAudioFlinger::track_flags_t flags); - virtual ~Track(); - - static void appendDumpHeader(String8& result); - void dump(char* buffer, size_t size); - virtual status_t start(AudioSystem::sync_event_t event = - AudioSystem::SYNC_EVENT_NONE, - int triggerSession = 0); - virtual void stop(); - void pause(); - - void flush(); - void destroy(); - void mute(bool); - int name() const { return mName; } - - audio_stream_type_t streamType() const { - return mStreamType; - } - status_t attachAuxEffect(int EffectId); - void setAuxBuffer(int EffectId, int32_t *buffer); - int32_t *auxBuffer() const { return mAuxBuffer; } - void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } - int16_t *mainBuffer() const { return mMainBuffer; } - int auxEffectId() const { return mAuxEffectId; } - - // implement FastMixerState::VolumeProvider interface - virtual uint32_t getVolumeLR(); - - virtual status_t setSyncEvent(const sp<SyncEvent>& event); - - protected: - // for numerous - friend class PlaybackThread; - friend class MixerThread; - friend class DirectOutputThread; - - Track(const Track&); - Track& operator = (const Track&); - - // AudioBufferProvider interface - virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, - int64_t pts = kInvalidPTS); - // releaseBuffer() not overridden - - virtual size_t framesReady() const; - - bool isMuted() const { return mMute; } - bool isPausing() const { - return mState == PAUSING; - } - bool isPaused() const { - return mState == PAUSED; - } - bool isResuming() const { - return mState == RESUMING; - } - bool isReady() const; - void setPaused() { mState = PAUSED; } - void reset(); - - bool isOutputTrack() const { - return (mStreamType == AUDIO_STREAM_CNT); - } - - sp<IMemory> sharedBuffer() const { return mSharedBuffer; } - - // framesWritten is cumulative, never reset, and is shared all tracks - // audioHalFrames is derived from output latency - // FIXME parameters not needed, could get them from the thread - bool presentationComplete(size_t framesWritten, size_t audioHalFrames); - - public: - void triggerEvents(AudioSystem::sync_event_t type); - virtual bool isTimedTrack() const { return false; } - bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } - virtual bool isOut() const; - - protected: - - // written by Track::mute() called by binder thread(s), without a mutex or barrier. - // read by Track::isMuted() called by playback thread, also without a mutex or barrier. - // The lack of mutex or barrier is safe because the mute status is only used by itself. - bool mMute; - - // FILLED state is used for suppressing volume ramp at begin of playing - enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; - mutable uint8_t mFillingUpStatus; - int8_t mRetryCount; - const sp<IMemory> mSharedBuffer; - bool mResetDone; - const audio_stream_type_t mStreamType; - int mName; // track name on the normal mixer, - // allocated statically at track creation time, - // and is even allocated (though unused) for fast tracks - // FIXME don't allocate track name for fast tracks - int16_t *mMainBuffer; - int32_t *mAuxBuffer; - int mAuxEffectId; - bool mHasVolumeController; - size_t mPresentationCompleteFrames; // number of frames written to the - // audio HAL when this track will be fully rendered - // zero means not monitoring - private: - IAudioFlinger::track_flags_t mFlags; - - // The following fields are only for fast tracks, and should be in a subclass - int mFastIndex; // index within FastMixerState::mFastTracks[]; - // either mFastIndex == -1 if not isFastTrack() - // or 0 < mFastIndex < FastMixerState::kMaxFast because - // index 0 is reserved for normal mixer's submix; - // index is allocated statically at track creation time - // but the slot is only used if track is active - FastTrackUnderruns mObservedUnderruns; // Most recently observed value of - // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns - uint32_t mUnderrunCount; // Counter of total number of underruns, never reset - volatile float mCachedVolume; // combined master volume and stream type volume; - // 'volatile' means accessed without lock or - // barrier, but is read/written atomically - }; // end of Track - - class TimedTrack : public Track { - public: - static sp<TimedTrack> create(PlaybackThread *thread, - const sp<Client>& client, - audio_stream_type_t streamType, - uint32_t sampleRate, - audio_format_t format, - audio_channel_mask_t channelMask, - size_t frameCount, - const sp<IMemory>& sharedBuffer, - int sessionId); - virtual ~TimedTrack(); - - class TimedBuffer { - public: - TimedBuffer(); - TimedBuffer(const sp<IMemory>& buffer, int64_t pts); - const sp<IMemory>& buffer() const { return mBuffer; } - int64_t pts() const { return mPTS; } - uint32_t position() const { return mPosition; } - void setPosition(uint32_t pos) { mPosition = pos; } - private: - sp<IMemory> mBuffer; - int64_t mPTS; - uint32_t mPosition; - }; - - // Mixer facing methods. - virtual bool isTimedTrack() const { return true; } - virtual size_t framesReady() const; - - // AudioBufferProvider interface - virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, - int64_t pts); - virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); - - // Client/App facing methods. - status_t allocateTimedBuffer(size_t size, - sp<IMemory>* buffer); - status_t queueTimedBuffer(const sp<IMemory>& buffer, - int64_t pts); - status_t setMediaTimeTransform(const LinearTransform& xform, - TimedAudioTrack::TargetTimeline target); - - private: - TimedTrack(PlaybackThread *thread, - const sp<Client>& client, - audio_stream_type_t streamType, - uint32_t sampleRate, - audio_format_t format, - audio_channel_mask_t channelMask, - size_t frameCount, - const sp<IMemory>& sharedBuffer, - int sessionId); - - void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); - void timedYieldSilence_l(uint32_t numFrames, - AudioBufferProvider::Buffer* buffer); - void trimTimedBufferQueue_l(); - void trimTimedBufferQueueHead_l(const char* logTag); - void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, - const char* logTag); - - uint64_t mLocalTimeFreq; - LinearTransform mLocalTimeToSampleTransform; - LinearTransform mMediaTimeToSampleTransform; - sp<MemoryDealer> mTimedMemoryDealer; - - Vector<TimedBuffer> mTimedBufferQueue; - bool mQueueHeadInFlight; - bool mTrimQueueHeadOnRelease; - uint32_t mFramesPendingInQueue; - - uint8_t* mTimedSilenceBuffer; - uint32_t mTimedSilenceBufferSize; - mutable Mutex mTimedBufferQueueLock; - bool mTimedAudioOutputOnTime; - CCHelper mCCHelper; - - Mutex mMediaTimeTransformLock; - LinearTransform mMediaTimeTransform; - bool mMediaTimeTransformValid; - TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; - }; - - - // playback track, used by DuplicatingThread - class OutputTrack : public Track { - public: - - class Buffer : public AudioBufferProvider::Buffer { - public: - int16_t *mBuffer; - }; - - OutputTrack(PlaybackThread *thread, - DuplicatingThread *sourceThread, - uint32_t sampleRate, - audio_format_t format, - audio_channel_mask_t channelMask, - size_t frameCount); - virtual ~OutputTrack(); - - virtual status_t start(AudioSystem::sync_event_t event = - AudioSystem::SYNC_EVENT_NONE, - int triggerSession = 0); - virtual void stop(); - bool write(int16_t* data, uint32_t frames); - bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } - bool isActive() const { return mActive; } - const wp<ThreadBase>& thread() const { return mThread; } - - private: - - enum { - NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value - }; - - status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, - uint32_t waitTimeMs); - void clearBufferQueue(); - - // Maximum number of pending buffers allocated by OutputTrack::write() - static const uint8_t kMaxOverFlowBuffers = 10; - - Vector < Buffer* > mBufferQueue; - AudioBufferProvider::Buffer mOutBuffer; - bool mActive; - DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() - void* mBuffers; // starting address of buffers in plain memory - }; // end of OutputTrack - - PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, - audio_io_handle_t id, audio_devices_t device, type_t type); - virtual ~PlaybackThread(); - - void dump(int fd, const Vector<String16>& args); - - // Thread virtuals - virtual status_t readyToRun(); - virtual bool threadLoop(); - - // RefBase - virtual void onFirstRef(); - -protected: - // Code snippets that were lifted up out of threadLoop() - virtual void threadLoop_mix() = 0; - virtual void threadLoop_sleepTime() = 0; - virtual void threadLoop_write(); - virtual void threadLoop_standby(); - virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); - - // prepareTracks_l reads and writes mActiveTracks, and returns - // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller - // is responsible for clearing or destroying this Vector later on, when it - // is safe to do so. That will drop the final ref count and destroy the tracks. - virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; - - // ThreadBase virtuals - virtual void preExit(); - -public: - - virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } - - // return estimated latency in milliseconds, as reported by HAL - uint32_t latency() const; - // same, but lock must already be held - uint32_t latency_l() const; - - void setMasterVolume(float value); - void setMasterMute(bool muted); - - void setStreamVolume(audio_stream_type_t stream, float value); - void setStreamMute(audio_stream_type_t stream, bool muted); - - float streamVolume(audio_stream_type_t stream) const; - - sp<Track> createTrack_l( - const sp<AudioFlinger::Client>& client, - audio_stream_type_t streamType, - uint32_t sampleRate, - audio_format_t format, - audio_channel_mask_t channelMask, - size_t frameCount, - const sp<IMemory>& sharedBuffer, - int sessionId, - IAudioFlinger::track_flags_t *flags, - pid_t tid, - status_t *status); - - AudioStreamOut* getOutput() const; - AudioStreamOut* clearOutput(); - virtual audio_stream_t* stream() const; - - // a very large number of suspend() will eventually wraparound, but unlikely - void suspend() { (void) android_atomic_inc(&mSuspended); } - void restore() - { - // if restore() is done without suspend(), get back into - // range so that the next suspend() will operate correctly - if (android_atomic_dec(&mSuspended) <= 0) { - android_atomic_release_store(0, &mSuspended); - } - } - bool isSuspended() const - { return android_atomic_acquire_load(&mSuspended) > 0; } - - virtual String8 getParameters(const String8& keys); - virtual void audioConfigChanged_l(int event, int param = 0); - status_t getRenderPosition(size_t *halFrames, size_t *dspFrames); - int16_t *mixBuffer() const { return mMixBuffer; }; - - virtual void detachAuxEffect_l(int effectId); - status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, - int EffectId); - status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, - int EffectId); - - virtual status_t addEffectChain_l(const sp<EffectChain>& chain); - virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); - virtual uint32_t hasAudioSession(int sessionId) const; - virtual uint32_t getStrategyForSession_l(int sessionId); - - - virtual status_t setSyncEvent(const sp<SyncEvent>& event); - virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; - void invalidateTracks(audio_stream_type_t streamType); - - - protected: - int16_t* mMixBuffer; - - // suspend count, > 0 means suspended. While suspended, the thread continues to pull from - // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle - // concurrent use of both of them, so Audio Policy Service suspends one of the threads to - // workaround that restriction. - // 'volatile' means accessed via atomic operations and no lock. - volatile int32_t mSuspended; - - // FIXME overflows every 6+ hours at 44.1 kHz stereo 16-bit samples - // mFramesWritten would be better, or 64-bit even better - size_t mBytesWritten; - private: - // mMasterMute is in both PlaybackThread and in AudioFlinger. When a - // PlaybackThread needs to find out if master-muted, it checks it's local - // copy rather than the one in AudioFlinger. This optimization saves a lock. - bool mMasterMute; - void setMasterMute_l(bool muted) { mMasterMute = muted; } - protected: - SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> - - // Allocate a track name for a given channel mask. - // Returns name >= 0 if successful, -1 on failure. - virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0; - virtual void deleteTrackName_l(int name) = 0; - - // Time to sleep between cycles when: - virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED - virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE - virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us - // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() - // No sleep in standby mode; waits on a condition - - // Code snippets that are temporarily lifted up out of threadLoop() until the merge - void checkSilentMode_l(); - - // Non-trivial for DUPLICATING only - virtual void saveOutputTracks() { } - virtual void clearOutputTracks() { } - - // Cache various calculated values, at threadLoop() entry and after a parameter change - virtual void cacheParameters_l(); - - virtual uint32_t correctLatency_l(uint32_t latency) const; - - private: - - friend class AudioFlinger; // for numerous - - PlaybackThread(const Client&); - PlaybackThread& operator = (const PlaybackThread&); - - status_t addTrack_l(const sp<Track>& track); - void destroyTrack_l(const sp<Track>& track); - void removeTrack_l(const sp<Track>& track); - - void readOutputParameters(); - - virtual void dumpInternals(int fd, const Vector<String16>& args); - void dumpTracks(int fd, const Vector<String16>& args); - - SortedVector< sp<Track> > mTracks; - // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by - // DuplicatingThread - stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; - AudioStreamOut *mOutput; - - float mMasterVolume; - nsecs_t mLastWriteTime; - int mNumWrites; - int mNumDelayedWrites; - bool mInWrite; - - // FIXME rename these former local variables of threadLoop to standard "m" names - nsecs_t standbyTime; - size_t mixBufferSize; - - // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() - uint32_t activeSleepTime; - uint32_t idleSleepTime; - - uint32_t sleepTime; - - // mixer status returned by prepareTracks_l() - mixer_state mMixerStatus; // current cycle - // previous cycle when in prepareTracks_l() - mixer_state mMixerStatusIgnoringFastTracks; - // FIXME or a separate ready state per track - - // FIXME move these declarations into the specific sub-class that needs them - // MIXER only - uint32_t sleepTimeShift; - - // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value - nsecs_t standbyDelay; - - // MIXER only - nsecs_t maxPeriod; - - // DUPLICATING only - uint32_t writeFrames; - - private: - // The HAL output sink is treated as non-blocking, but current implementation is blocking - sp<NBAIO_Sink> mOutputSink; - // If a fast mixer is present, the blocking pipe sink, otherwise clear - sp<NBAIO_Sink> mPipeSink; - // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink - sp<NBAIO_Sink> mNormalSink; - // For dumpsys - sp<NBAIO_Sink> mTeeSink; - sp<NBAIO_Source> mTeeSource; - uint32_t mScreenState; // cached copy of gScreenState - public: - virtual bool hasFastMixer() const = 0; - virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const - { FastTrackUnderruns dummy; return dummy; } - - protected: - // accessed by both binder threads and within threadLoop(), lock on mutex needed - unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available - - }; - - class MixerThread : public PlaybackThread { - public: - MixerThread(const sp<AudioFlinger>& audioFlinger, - AudioStreamOut* output, - audio_io_handle_t id, - audio_devices_t device, - type_t type = MIXER); - virtual ~MixerThread(); - - // Thread virtuals - - virtual bool checkForNewParameters_l(); - virtual void dumpInternals(int fd, const Vector<String16>& args); - - protected: - virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); - virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); - virtual void deleteTrackName_l(int name); - virtual uint32_t idleSleepTimeUs() const; - virtual uint32_t suspendSleepTimeUs() const; - virtual void cacheParameters_l(); - - // threadLoop snippets - virtual void threadLoop_write(); - virtual void threadLoop_standby(); - virtual void threadLoop_mix(); - virtual void threadLoop_sleepTime(); - virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); - virtual uint32_t correctLatency_l(uint32_t latency) const; - - AudioMixer* mAudioMixer; // normal mixer - private: - // one-time initialization, no locks required - FastMixer* mFastMixer; // non-NULL if there is also a fast mixer - sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread - - // contents are not guaranteed to be consistent, no locks required - FastMixerDumpState mFastMixerDumpState; -#ifdef STATE_QUEUE_DUMP - StateQueueObserverDump mStateQueueObserverDump; - StateQueueMutatorDump mStateQueueMutatorDump; -#endif - AudioWatchdogDump mAudioWatchdogDump; - - // accessible only within the threadLoop(), no locks required - // mFastMixer->sq() // for mutating and pushing state - int32_t mFastMixerFutex; // for cold idle - - public: - virtual bool hasFastMixer() const { return mFastMixer != NULL; } - virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { - ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); - return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; - } - }; - - class DirectOutputThread : public PlaybackThread { - public: - - DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, - audio_io_handle_t id, audio_devices_t device); - virtual ~DirectOutputThread(); - - // Thread virtuals - - virtual bool checkForNewParameters_l(); - - protected: - virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); - virtual void deleteTrackName_l(int name); - virtual uint32_t activeSleepTimeUs() const; - virtual uint32_t idleSleepTimeUs() const; - virtual uint32_t suspendSleepTimeUs() const; - virtual void cacheParameters_l(); - - // threadLoop snippets - virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); - virtual void threadLoop_mix(); - virtual void threadLoop_sleepTime(); - - private: - // volumes last sent to audio HAL with stream->set_volume() - float mLeftVolFloat; - float mRightVolFloat; - - // prepareTracks_l() tells threadLoop_mix() the name of the single active track - sp<Track> mActiveTrack; - public: - virtual bool hasFastMixer() const { return false; } - }; - - class DuplicatingThread : public MixerThread { - public: - DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, - audio_io_handle_t id); - virtual ~DuplicatingThread(); - - // Thread virtuals - void addOutputTrack(MixerThread* thread); - void removeOutputTrack(MixerThread* thread); - uint32_t waitTimeMs() const { return mWaitTimeMs; } - protected: - virtual uint32_t activeSleepTimeUs() const; - - private: - bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); - protected: - // threadLoop snippets - virtual void threadLoop_mix(); - virtual void threadLoop_sleepTime(); - virtual void threadLoop_write(); - virtual void threadLoop_standby(); - virtual void cacheParameters_l(); - - private: - // called from threadLoop, addOutputTrack, removeOutputTrack - virtual void updateWaitTime_l(); - protected: - virtual void saveOutputTracks(); - virtual void clearOutputTracks(); - private: - - uint32_t mWaitTimeMs; - SortedVector < sp<OutputTrack> > outputTracks; - SortedVector < sp<OutputTrack> > mOutputTracks; - public: - virtual bool hasFastMixer() const { return false; } - }; - - PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; - MixerThread *checkMixerThread_l(audio_io_handle_t output) const; - RecordThread *checkRecordThread_l(audio_io_handle_t input) const; - // no range check, AudioFlinger::mLock held - bool streamMute_l(audio_stream_type_t stream) const - { return mStreamTypes[stream].mute; } - // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held - float streamVolume_l(audio_stream_type_t stream) const - { return mStreamTypes[stream].volume; } - void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); - - // allocate an audio_io_handle_t, session ID, or effect ID - uint32_t nextUniqueId(); - status_t moveEffectChain_l(int sessionId, - PlaybackThread *srcThread, - PlaybackThread *dstThread, - bool reRegister); - // return thread associated with primary hardware device, or NULL - PlaybackThread *primaryPlaybackThread_l() const; - audio_devices_t primaryOutputDevice_l() const; +#include "Threads.h" - sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); +#include "Effects.h" // server side of the client's IAudioTrack class TrackHandle : public android::BnAudioTrack { @@ -1449,165 +400,6 @@ public: const sp<PlaybackThread::Track> mTrack; }; - void removeClient_l(pid_t pid); - void removeNotificationClient(pid_t pid); - - - // record thread - class RecordThread : public ThreadBase, public AudioBufferProvider - // derives from AudioBufferProvider interface for use by resampler - { - public: - - // record track - class RecordTrack : public TrackBase { - public: - RecordTrack(RecordThread *thread, - const sp<Client>& client, - uint32_t sampleRate, - audio_format_t format, - audio_channel_mask_t channelMask, - size_t frameCount, - int sessionId); - virtual ~RecordTrack(); - - virtual status_t start(AudioSystem::sync_event_t event, int triggerSession); - virtual void stop(); - - void destroy(); - - // clear the buffer overflow flag - void clearOverflow() { mOverflow = false; } - // set the buffer overflow flag and return previous value - bool setOverflow() { bool tmp = mOverflow; mOverflow = true; - return tmp; } - - static void appendDumpHeader(String8& result); - void dump(char* buffer, size_t size); - - virtual bool isOut() const; - - private: - friend class AudioFlinger; // for mState - - RecordTrack(const RecordTrack&); - RecordTrack& operator = (const RecordTrack&); - - // AudioBufferProvider interface - virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, - int64_t pts = kInvalidPTS); - // releaseBuffer() not overridden - - bool mOverflow; // overflow on most recent attempt to fill client buffer - }; - - RecordThread(const sp<AudioFlinger>& audioFlinger, - AudioStreamIn *input, - uint32_t sampleRate, - audio_channel_mask_t channelMask, - audio_io_handle_t id, - audio_devices_t device, - const sp<NBAIO_Sink>& teeSink); - virtual ~RecordThread(); - - // no addTrack_l ? - void destroyTrack_l(const sp<RecordTrack>& track); - void removeTrack_l(const sp<RecordTrack>& track); - - void dumpInternals(int fd, const Vector<String16>& args); - void dumpTracks(int fd, const Vector<String16>& args); - - // Thread virtuals - virtual bool threadLoop(); - virtual status_t readyToRun(); - - // RefBase - virtual void onFirstRef(); - - virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } - sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( - const sp<AudioFlinger::Client>& client, - uint32_t sampleRate, - audio_format_t format, - audio_channel_mask_t channelMask, - size_t frameCount, - int sessionId, - IAudioFlinger::track_flags_t flags, - pid_t tid, - status_t *status); - - status_t start(RecordTrack* recordTrack, - AudioSystem::sync_event_t event, - int triggerSession); - - // ask the thread to stop the specified track, and - // return true if the caller should then do it's part of the stopping process - bool stop_l(RecordTrack* recordTrack); - - void dump(int fd, const Vector<String16>& args); - AudioStreamIn* clearInput(); - virtual audio_stream_t* stream() const; - - // AudioBufferProvider interface - virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); - virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); - - virtual bool checkForNewParameters_l(); - virtual String8 getParameters(const String8& keys); - virtual void audioConfigChanged_l(int event, int param = 0); - void readInputParameters(); - virtual unsigned int getInputFramesLost(); - - virtual status_t addEffectChain_l(const sp<EffectChain>& chain); - virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); - virtual uint32_t hasAudioSession(int sessionId) const; - - // Return the set of unique session IDs across all tracks. - // The keys are the session IDs, and the associated values are meaningless. - // FIXME replace by Set [and implement Bag/Multiset for other uses]. - KeyedVector<int, bool> sessionIds() const; - - virtual status_t setSyncEvent(const sp<SyncEvent>& event); - virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; - - static void syncStartEventCallback(const wp<SyncEvent>& event); - void handleSyncStartEvent(const sp<SyncEvent>& event); - - private: - void clearSyncStartEvent(); - - // Enter standby if not already in standby, and set mStandby flag - void standby(); - - // Call the HAL standby method unconditionally, and don't change mStandby flag - void inputStandBy(); - - AudioStreamIn *mInput; - SortedVector < sp<RecordTrack> > mTracks; - // mActiveTrack has dual roles: it indicates the current active track, and - // is used together with mStartStopCond to indicate start()/stop() progress - sp<RecordTrack> mActiveTrack; - Condition mStartStopCond; - AudioResampler *mResampler; - int32_t *mRsmpOutBuffer; - int16_t *mRsmpInBuffer; - size_t mRsmpInIndex; - size_t mInputBytes; - const uint32_t mReqChannelCount; - const uint32_t mReqSampleRate; - ssize_t mBytesRead; - // sync event triggering actual audio capture. Frames read before this event will - // be dropped and therefore not read by the application. - sp<SyncEvent> mSyncStartEvent; - // number of captured frames to drop after the start sync event has been received. - // when < 0, maximum frames to drop before starting capture even if sync event is - // not received - ssize_t mFramestoDrop; - - // For dumpsys - const sp<NBAIO_Sink> mTeeSink; - }; - // server side of the client's IAudioRecord class RecordHandle : public android::BnAudioRecord { public: @@ -1625,344 +417,33 @@ public: void stop_nonvirtual(); }; - //--- Audio Effect Management - - // EffectModule and EffectChain classes both have their own mutex to protect - // state changes or resource modifications. Always respect the following order - // if multiple mutexes must be acquired to avoid cross deadlock: - // AudioFlinger -> ThreadBase -> EffectChain -> EffectModule - - // The EffectModule class is a wrapper object controlling the effect engine implementation - // in the effect library. It prevents concurrent calls to process() and command() functions - // from different client threads. It keeps a list of EffectHandle objects corresponding - // to all client applications using this effect and notifies applications of effect state, - // control or parameter changes. It manages the activation state machine to send appropriate - // reset, enable, disable commands to effect engine and provide volume - // ramping when effects are activated/deactivated. - // When controlling an auxiliary effect, the EffectModule also provides an input buffer used by - // the attached track(s) to accumulate their auxiliary channel. - class EffectModule : public RefBase { - public: - EffectModule(ThreadBase *thread, - const wp<AudioFlinger::EffectChain>& chain, - effect_descriptor_t *desc, - int id, - int sessionId); - virtual ~EffectModule(); - - enum effect_state { - IDLE, - RESTART, - STARTING, - ACTIVE, - STOPPING, - STOPPED, - DESTROYED - }; - - int id() const { return mId; } - void process(); - void updateState(); - status_t command(uint32_t cmdCode, - uint32_t cmdSize, - void *pCmdData, - uint32_t *replySize, - void *pReplyData); - - void reset_l(); - status_t configure(); - status_t init(); - effect_state state() const { - return mState; - } - uint32_t status() { - return mStatus; - } - int sessionId() const { - return mSessionId; - } - status_t setEnabled(bool enabled); - status_t setEnabled_l(bool enabled); - bool isEnabled() const; - bool isProcessEnabled() const; - - void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } - int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } - void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } - int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } - void setChain(const wp<EffectChain>& chain) { mChain = chain; } - void setThread(const wp<ThreadBase>& thread) { mThread = thread; } - const wp<ThreadBase>& thread() { return mThread; } - - status_t addHandle(EffectHandle *handle); - size_t disconnect(EffectHandle *handle, bool unpinIfLast); - size_t removeHandle(EffectHandle *handle); - - const effect_descriptor_t& desc() const { return mDescriptor; } - wp<EffectChain>& chain() { return mChain; } - - status_t setDevice(audio_devices_t device); - status_t setVolume(uint32_t *left, uint32_t *right, bool controller); - status_t setMode(audio_mode_t mode); - status_t setAudioSource(audio_source_t source); - status_t start(); - status_t stop(); - void setSuspended(bool suspended); - bool suspended() const; - - EffectHandle* controlHandle_l(); - - bool isPinned() const { return mPinned; } - void unPin() { mPinned = false; } - bool purgeHandles(); - void lock() { mLock.lock(); } - void unlock() { mLock.unlock(); } - - void dump(int fd, const Vector<String16>& args); - - protected: - friend class AudioFlinger; // for mHandles - bool mPinned; - - // Maximum time allocated to effect engines to complete the turn off sequence - static const uint32_t MAX_DISABLE_TIME_MS = 10000; - - EffectModule(const EffectModule&); - EffectModule& operator = (const EffectModule&); - - status_t start_l(); - status_t stop_l(); - -mutable Mutex mLock; // mutex for process, commands and handles list protection - wp<ThreadBase> mThread; // parent thread - wp<EffectChain> mChain; // parent effect chain - const int mId; // this instance unique ID - const int mSessionId; // audio session ID - const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine - effect_config_t mConfig; // input and output audio configuration - effect_handle_t mEffectInterface; // Effect module C API - status_t mStatus; // initialization status - effect_state mState; // current activation state - Vector<EffectHandle *> mHandles; // list of client handles - // First handle in mHandles has highest priority and controls the effect module - uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after - // sending disable command. - uint32_t mDisableWaitCnt; // current process() calls count during disable period. - bool mSuspended; // effect is suspended: temporarily disabled by framework - }; - - // The EffectHandle class implements the IEffect interface. It provides resources - // to receive parameter updates, keeps track of effect control - // ownership and state and has a pointer to the EffectModule object it is controlling. - // There is one EffectHandle object for each application controlling (or using) - // an effect module. - // The EffectHandle is obtained by calling AudioFlinger::createEffect(). - class EffectHandle: public android::BnEffect { - public: - - EffectHandle(const sp<EffectModule>& effect, - const sp<AudioFlinger::Client>& client, - const sp<IEffectClient>& effectClient, - int32_t priority); - virtual ~EffectHandle(); - - // IEffect - virtual status_t enable(); - virtual status_t disable(); - virtual status_t command(uint32_t cmdCode, - uint32_t cmdSize, - void *pCmdData, - uint32_t *replySize, - void *pReplyData); - virtual void disconnect(); - private: - void disconnect(bool unpinIfLast); - public: - virtual sp<IMemory> getCblk() const { return mCblkMemory; } - virtual status_t onTransact(uint32_t code, const Parcel& data, - Parcel* reply, uint32_t flags); - - - // Give or take control of effect module - // - hasControl: true if control is given, false if removed - // - signal: true client app should be signaled of change, false otherwise - // - enabled: state of the effect when control is passed - void setControl(bool hasControl, bool signal, bool enabled); - void commandExecuted(uint32_t cmdCode, - uint32_t cmdSize, - void *pCmdData, - uint32_t replySize, - void *pReplyData); - void setEnabled(bool enabled); - bool enabled() const { return mEnabled; } - - // Getters - int id() const { return mEffect->id(); } - int priority() const { return mPriority; } - bool hasControl() const { return mHasControl; } - sp<EffectModule> effect() const { return mEffect; } - // destroyed_l() must be called with the associated EffectModule mLock held - bool destroyed_l() const { return mDestroyed; } - - void dump(char* buffer, size_t size); - - protected: - friend class AudioFlinger; // for mEffect, mHasControl, mEnabled - EffectHandle(const EffectHandle&); - EffectHandle& operator =(const EffectHandle&); - - sp<EffectModule> mEffect; // pointer to controlled EffectModule - sp<IEffectClient> mEffectClient; // callback interface for client notifications - /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() - sp<IMemory> mCblkMemory; // shared memory for control block - effect_param_cblk_t* mCblk; // control block for deferred parameter setting via - // shared memory - uint8_t* mBuffer; // pointer to parameter area in shared memory - int mPriority; // client application priority to control the effect - bool mHasControl; // true if this handle is controlling the effect - bool mEnabled; // cached enable state: needed when the effect is - // restored after being suspended - bool mDestroyed; // Set to true by destructor. Access with EffectModule - // mLock held - }; - - // the EffectChain class represents a group of effects associated to one audio session. - // There can be any number of EffectChain objects per output mixer thread (PlaybackThread). - // The EffecChain with session ID 0 contains global effects applied to the output mix. - // Effects in this chain can be insert or auxiliary. Effects in other chains (attached to - // tracks) are insert only. The EffectChain maintains an ordered list of effect module, the - // order corresponding in the effect process order. When attached to a track (session ID != 0), - // it also provide it's own input buffer used by the track as accumulation buffer. - class EffectChain : public RefBase { - public: - EffectChain(const wp<ThreadBase>& wThread, int sessionId); - EffectChain(ThreadBase *thread, int sessionId); - virtual ~EffectChain(); - - // special key used for an entry in mSuspendedEffects keyed vector - // corresponding to a suspend all request. - static const int kKeyForSuspendAll = 0; - - // minimum duration during which we force calling effect process when last track on - // a session is stopped or removed to allow effect tail to be rendered - static const int kProcessTailDurationMs = 1000; - - void process_l(); - - void lock() { - mLock.lock(); - } - void unlock() { - mLock.unlock(); - } - - status_t addEffect_l(const sp<EffectModule>& handle); - size_t removeEffect_l(const sp<EffectModule>& handle); - - int sessionId() const { return mSessionId; } - void setSessionId(int sessionId) { mSessionId = sessionId; } - - sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); - sp<EffectModule> getEffectFromId_l(int id); - sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); - bool setVolume_l(uint32_t *left, uint32_t *right); - void setDevice_l(audio_devices_t device); - void setMode_l(audio_mode_t mode); - void setAudioSource_l(audio_source_t source); - - void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { - mInBuffer = buffer; - mOwnInBuffer = ownsBuffer; - } - int16_t *inBuffer() const { - return mInBuffer; - } - void setOutBuffer(int16_t *buffer) { - mOutBuffer = buffer; - } - int16_t *outBuffer() const { - return mOutBuffer; - } - - void incTrackCnt() { android_atomic_inc(&mTrackCnt); } - void decTrackCnt() { android_atomic_dec(&mTrackCnt); } - int32_t trackCnt() const { return android_atomic_acquire_load(&mTrackCnt); } - - void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); - mTailBufferCount = mMaxTailBuffers; } - void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } - int32_t activeTrackCnt() const { return android_atomic_acquire_load(&mActiveTrackCnt); } - - uint32_t strategy() const { return mStrategy; } - void setStrategy(uint32_t strategy) - { mStrategy = strategy; } - - // suspend effect of the given type - void setEffectSuspended_l(const effect_uuid_t *type, - bool suspend); - // suspend all eligible effects - void setEffectSuspendedAll_l(bool suspend); - // check if effects should be suspend or restored when a given effect is enable or disabled - void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, - bool enabled); - - void clearInputBuffer(); + PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; + MixerThread *checkMixerThread_l(audio_io_handle_t output) const; + RecordThread *checkRecordThread_l(audio_io_handle_t input) const; + // no range check, AudioFlinger::mLock held + bool streamMute_l(audio_stream_type_t stream) const + { return mStreamTypes[stream].mute; } + // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held + float streamVolume_l(audio_stream_type_t stream) const + { return mStreamTypes[stream].volume; } + void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); - void dump(int fd, const Vector<String16>& args); + // allocate an audio_io_handle_t, session ID, or effect ID + uint32_t nextUniqueId(); - protected: - friend class AudioFlinger; // for mThread, mEffects - EffectChain(const EffectChain&); - EffectChain& operator =(const EffectChain&); + status_t moveEffectChain_l(int sessionId, + PlaybackThread *srcThread, + PlaybackThread *dstThread, + bool reRegister); + // return thread associated with primary hardware device, or NULL + PlaybackThread *primaryPlaybackThread_l() const; + audio_devices_t primaryOutputDevice_l() const; - class SuspendedEffectDesc : public RefBase { - public: - SuspendedEffectDesc() : mRefCount(0) {} + sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); - int mRefCount; - effect_uuid_t mType; - wp<EffectModule> mEffect; - }; - // get a list of effect modules to suspend when an effect of the type - // passed is enabled. - void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); - - // get an effect module if it is currently enable - sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); - // true if the effect whose descriptor is passed can be suspended - // OEMs can modify the rules implemented in this method to exclude specific effect - // types or implementations from the suspend/restore mechanism. - bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); - - void clearInputBuffer_l(sp<ThreadBase> thread); - - wp<ThreadBase> mThread; // parent mixer thread - Mutex mLock; // mutex protecting effect list - Vector< sp<EffectModule> > mEffects; // list of effect modules - int mSessionId; // audio session ID - int16_t *mInBuffer; // chain input buffer - int16_t *mOutBuffer; // chain output buffer - - // 'volatile' here means these are accessed with atomic operations instead of mutex - volatile int32_t mActiveTrackCnt; // number of active tracks connected - volatile int32_t mTrackCnt; // number of tracks connected - - int32_t mTailBufferCount; // current effect tail buffer count - int32_t mMaxTailBuffers; // maximum effect tail buffers - bool mOwnInBuffer; // true if the chain owns its input buffer - int mVolumeCtrlIdx; // index of insert effect having control over volume - uint32_t mLeftVolume; // previous volume on left channel - uint32_t mRightVolume; // previous volume on right channel - uint32_t mNewLeftVolume; // new volume on left channel - uint32_t mNewRightVolume; // new volume on right channel - uint32_t mStrategy; // strategy for this effect chain - // mSuspendedEffects lists all effects currently suspended in the chain. - // Use effect type UUID timelow field as key. There is no real risk of identical - // timeLow fields among effect type UUIDs. - // Updated by updateSuspendedSessions_l() only. - KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; - }; + void removeClient_l(pid_t pid); + void removeNotificationClient(pid_t pid); class AudioHwDevice { public: @@ -2108,6 +589,7 @@ public: static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); }; +#undef INCLUDING_FROM_AUDIOFLINGER_H // ---------------------------------------------------------------------------- diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp new file mode 100644 index 0000000..74ba59e --- /dev/null +++ b/services/audioflinger/Effects.cpp @@ -0,0 +1,1684 @@ +/* +** +** Copyright 2012, 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 <utils/Log.h> +#include <audio_effects/effect_visualizer.h> +#include <audio_utils/primitives.h> +#include <private/media/AudioEffectShared.h> +#include <media/EffectsFactoryApi.h> + +#include "AudioFlinger.h" +#include "ServiceUtilities.h" + +// ---------------------------------------------------------------------------- + +// Note: the following macro is used for extremely verbose logging message. In +// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to +// 0; but one side effect of this is to turn all LOGV's as well. Some messages +// are so verbose that we want to suppress them even when we have ALOG_ASSERT +// turned on. Do not uncomment the #def below unless you really know what you +// are doing and want to see all of the extremely verbose messages. +//#define VERY_VERY_VERBOSE_LOGGING +#ifdef VERY_VERY_VERBOSE_LOGGING +#define ALOGVV ALOGV +#else +#define ALOGVV(a...) do { } while(0) +#endif + +namespace android { + +// ---------------------------------------------------------------------------- +// EffectModule implementation +// ---------------------------------------------------------------------------- + +#undef LOG_TAG +#define LOG_TAG "AudioFlinger::EffectModule" + +AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, + const wp<AudioFlinger::EffectChain>& chain, + effect_descriptor_t *desc, + int id, + int sessionId) + : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), + mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), + mDescriptor(*desc), + // mConfig is set by configure() and not used before then + mEffectInterface(NULL), + mStatus(NO_INIT), mState(IDLE), + // mMaxDisableWaitCnt is set by configure() and not used before then + // mDisableWaitCnt is set by process() and updateState() and not used before then + mSuspended(false) +{ + ALOGV("Constructor %p", this); + int lStatus; + + // create effect engine from effect factory + mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); + + if (mStatus != NO_ERROR) { + return; + } + lStatus = init(); + if (lStatus < 0) { + mStatus = lStatus; + goto Error; + } + + ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); + return; +Error: + EffectRelease(mEffectInterface); + mEffectInterface = NULL; + ALOGV("Constructor Error %d", mStatus); +} + +AudioFlinger::EffectModule::~EffectModule() +{ + ALOGV("Destructor %p", this); + if (mEffectInterface != NULL) { + if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || + (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + audio_stream_t *stream = thread->stream(); + if (stream != NULL) { + stream->remove_audio_effect(stream, mEffectInterface); + } + } + } + // release effect engine + EffectRelease(mEffectInterface); + } +} + +status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) +{ + status_t status; + + Mutex::Autolock _l(mLock); + int priority = handle->priority(); + size_t size = mHandles.size(); + EffectHandle *controlHandle = NULL; + size_t i; + for (i = 0; i < size; i++) { + EffectHandle *h = mHandles[i]; + if (h == NULL || h->destroyed_l()) { + continue; + } + // first non destroyed handle is considered in control + if (controlHandle == NULL) + controlHandle = h; + if (h->priority() <= priority) { + break; + } + } + // if inserted in first place, move effect control from previous owner to this handle + if (i == 0) { + bool enabled = false; + if (controlHandle != NULL) { + enabled = controlHandle->enabled(); + controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); + } + handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); + status = NO_ERROR; + } else { + status = ALREADY_EXISTS; + } + ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); + mHandles.insertAt(handle, i); + return status; +} + +size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) +{ + Mutex::Autolock _l(mLock); + size_t size = mHandles.size(); + size_t i; + for (i = 0; i < size; i++) { + if (mHandles[i] == handle) { + break; + } + } + if (i == size) { + return size; + } + ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); + + mHandles.removeAt(i); + // if removed from first place, move effect control from this handle to next in line + if (i == 0) { + EffectHandle *h = controlHandle_l(); + if (h != NULL) { + h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); + } + } + + // Prevent calls to process() and other functions on effect interface from now on. + // The effect engine will be released by the destructor when the last strong reference on + // this object is released which can happen after next process is called. + if (mHandles.size() == 0 && !mPinned) { + mState = DESTROYED; + } + + return mHandles.size(); +} + +// must be called with EffectModule::mLock held +AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() +{ + // the first valid handle in the list has control over the module + for (size_t i = 0; i < mHandles.size(); i++) { + EffectHandle *h = mHandles[i]; + if (h != NULL && !h->destroyed_l()) { + return h; + } + } + + return NULL; +} + +size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) +{ + ALOGV("disconnect() %p handle %p", this, handle); + // keep a strong reference on this EffectModule to avoid calling the + // destructor before we exit + sp<EffectModule> keep(this); + { + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + thread->disconnectEffect(keep, handle, unpinIfLast); + } + } + return mHandles.size(); +} + +void AudioFlinger::EffectModule::updateState() { + Mutex::Autolock _l(mLock); + + switch (mState) { + case RESTART: + reset_l(); + // FALL THROUGH + + case STARTING: + // clear auxiliary effect input buffer for next accumulation + if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { + memset(mConfig.inputCfg.buffer.raw, + 0, + mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); + } + start_l(); + mState = ACTIVE; + break; + case STOPPING: + stop_l(); + mDisableWaitCnt = mMaxDisableWaitCnt; + mState = STOPPED; + break; + case STOPPED: + // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the + // turn off sequence. + if (--mDisableWaitCnt == 0) { + reset_l(); + mState = IDLE; + } + break; + default: //IDLE , ACTIVE, DESTROYED + break; + } +} + +void AudioFlinger::EffectModule::process() +{ + Mutex::Autolock _l(mLock); + + if (mState == DESTROYED || mEffectInterface == NULL || + mConfig.inputCfg.buffer.raw == NULL || + mConfig.outputCfg.buffer.raw == NULL) { + return; + } + + if (isProcessEnabled()) { + // do 32 bit to 16 bit conversion for auxiliary effect input buffer + if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { + ditherAndClamp(mConfig.inputCfg.buffer.s32, + mConfig.inputCfg.buffer.s32, + mConfig.inputCfg.buffer.frameCount/2); + } + + // do the actual processing in the effect engine + int ret = (*mEffectInterface)->process(mEffectInterface, + &mConfig.inputCfg.buffer, + &mConfig.outputCfg.buffer); + + // force transition to IDLE state when engine is ready + if (mState == STOPPED && ret == -ENODATA) { + mDisableWaitCnt = 1; + } + + // clear auxiliary effect input buffer for next accumulation + if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { + memset(mConfig.inputCfg.buffer.raw, 0, + mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); + } + } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && + mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { + // If an insert effect is idle and input buffer is different from output buffer, + // accumulate input onto output + sp<EffectChain> chain = mChain.promote(); + if (chain != 0 && chain->activeTrackCnt() != 0) { + size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here + int16_t *in = mConfig.inputCfg.buffer.s16; + int16_t *out = mConfig.outputCfg.buffer.s16; + for (size_t i = 0; i < frameCnt; i++) { + out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); + } + } + } +} + +void AudioFlinger::EffectModule::reset_l() +{ + if (mEffectInterface == NULL) { + return; + } + (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); +} + +status_t AudioFlinger::EffectModule::configure() +{ + if (mEffectInterface == NULL) { + return NO_INIT; + } + + sp<ThreadBase> thread = mThread.promote(); + if (thread == 0) { + return DEAD_OBJECT; + } + + // TODO: handle configuration of effects replacing track process + audio_channel_mask_t channelMask = thread->channelMask(); + + if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { + mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; + } else { + mConfig.inputCfg.channels = channelMask; + } + mConfig.outputCfg.channels = channelMask; + mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; + mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; + mConfig.inputCfg.samplingRate = thread->sampleRate(); + mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; + mConfig.inputCfg.bufferProvider.cookie = NULL; + mConfig.inputCfg.bufferProvider.getBuffer = NULL; + mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; + mConfig.outputCfg.bufferProvider.cookie = NULL; + mConfig.outputCfg.bufferProvider.getBuffer = NULL; + mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; + mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; + // Insert effect: + // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, + // always overwrites output buffer: input buffer == output buffer + // - in other sessions: + // last effect in the chain accumulates in output buffer: input buffer != output buffer + // other effect: overwrites output buffer: input buffer == output buffer + // Auxiliary effect: + // accumulates in output buffer: input buffer != output buffer + // Therefore: accumulate <=> input buffer != output buffer + if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { + mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; + } else { + mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; + } + mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; + mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; + mConfig.inputCfg.buffer.frameCount = thread->frameCount(); + mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; + + ALOGV("configure() %p thread %p buffer %p framecount %d", + this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); + + status_t cmdStatus; + uint32_t size = sizeof(int); + status_t status = (*mEffectInterface)->command(mEffectInterface, + EFFECT_CMD_SET_CONFIG, + sizeof(effect_config_t), + &mConfig, + &size, + &cmdStatus); + if (status == 0) { + status = cmdStatus; + } + + if (status == 0 && + (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { + uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; + effect_param_t *p = (effect_param_t *)buf32; + + p->psize = sizeof(uint32_t); + p->vsize = sizeof(uint32_t); + size = sizeof(int); + *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; + + uint32_t latency = 0; + PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); + if (pbt != NULL) { + latency = pbt->latency_l(); + } + + *((int32_t *)p->data + 1)= latency; + (*mEffectInterface)->command(mEffectInterface, + EFFECT_CMD_SET_PARAM, + sizeof(effect_param_t) + 8, + &buf32, + &size, + &cmdStatus); + } + + mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / + (1000 * mConfig.outputCfg.buffer.frameCount); + + return status; +} + +status_t AudioFlinger::EffectModule::init() +{ + Mutex::Autolock _l(mLock); + if (mEffectInterface == NULL) { + return NO_INIT; + } + status_t cmdStatus; + uint32_t size = sizeof(status_t); + status_t status = (*mEffectInterface)->command(mEffectInterface, + EFFECT_CMD_INIT, + 0, + NULL, + &size, + &cmdStatus); + if (status == 0) { + status = cmdStatus; + } + return status; +} + +status_t AudioFlinger::EffectModule::start() +{ + Mutex::Autolock _l(mLock); + return start_l(); +} + +status_t AudioFlinger::EffectModule::start_l() +{ + if (mEffectInterface == NULL) { + return NO_INIT; + } + status_t cmdStatus; + uint32_t size = sizeof(status_t); + status_t status = (*mEffectInterface)->command(mEffectInterface, + EFFECT_CMD_ENABLE, + 0, + NULL, + &size, + &cmdStatus); + if (status == 0) { + status = cmdStatus; + } + if (status == 0 && + ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || + (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + audio_stream_t *stream = thread->stream(); + if (stream != NULL) { + stream->add_audio_effect(stream, mEffectInterface); + } + } + } + return status; +} + +status_t AudioFlinger::EffectModule::stop() +{ + Mutex::Autolock _l(mLock); + return stop_l(); +} + +status_t AudioFlinger::EffectModule::stop_l() +{ + if (mEffectInterface == NULL) { + return NO_INIT; + } + status_t cmdStatus; + uint32_t size = sizeof(status_t); + status_t status = (*mEffectInterface)->command(mEffectInterface, + EFFECT_CMD_DISABLE, + 0, + NULL, + &size, + &cmdStatus); + if (status == 0) { + status = cmdStatus; + } + if (status == 0 && + ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || + (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + audio_stream_t *stream = thread->stream(); + if (stream != NULL) { + stream->remove_audio_effect(stream, mEffectInterface); + } + } + } + return status; +} + +status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, + uint32_t cmdSize, + void *pCmdData, + uint32_t *replySize, + void *pReplyData) +{ + Mutex::Autolock _l(mLock); + ALOGVV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); + + if (mState == DESTROYED || mEffectInterface == NULL) { + return NO_INIT; + } + status_t status = (*mEffectInterface)->command(mEffectInterface, + cmdCode, + cmdSize, + pCmdData, + replySize, + pReplyData); + if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { + uint32_t size = (replySize == NULL) ? 0 : *replySize; + for (size_t i = 1; i < mHandles.size(); i++) { + EffectHandle *h = mHandles[i]; + if (h != NULL && !h->destroyed_l()) { + h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); + } + } + } + return status; +} + +status_t AudioFlinger::EffectModule::setEnabled(bool enabled) +{ + Mutex::Autolock _l(mLock); + return setEnabled_l(enabled); +} + +// must be called with EffectModule::mLock held +status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) +{ + + ALOGV("setEnabled %p enabled %d", this, enabled); + + if (enabled != isEnabled()) { + status_t status = AudioSystem::setEffectEnabled(mId, enabled); + if (enabled && status != NO_ERROR) { + return status; + } + + switch (mState) { + // going from disabled to enabled + case IDLE: + mState = STARTING; + break; + case STOPPED: + mState = RESTART; + break; + case STOPPING: + mState = ACTIVE; + break; + + // going from enabled to disabled + case RESTART: + mState = STOPPED; + break; + case STARTING: + mState = IDLE; + break; + case ACTIVE: + mState = STOPPING; + break; + case DESTROYED: + return NO_ERROR; // simply ignore as we are being destroyed + } + for (size_t i = 1; i < mHandles.size(); i++) { + EffectHandle *h = mHandles[i]; + if (h != NULL && !h->destroyed_l()) { + h->setEnabled(enabled); + } + } + } + return NO_ERROR; +} + +bool AudioFlinger::EffectModule::isEnabled() const +{ + switch (mState) { + case RESTART: + case STARTING: + case ACTIVE: + return true; + case IDLE: + case STOPPING: + case STOPPED: + case DESTROYED: + default: + return false; + } +} + +bool AudioFlinger::EffectModule::isProcessEnabled() const +{ + switch (mState) { + case RESTART: + case ACTIVE: + case STOPPING: + case STOPPED: + return true; + case IDLE: + case STARTING: + case DESTROYED: + default: + return false; + } +} + +status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) +{ + Mutex::Autolock _l(mLock); + status_t status = NO_ERROR; + + // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume + // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) + if (isProcessEnabled() && + ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || + (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { + status_t cmdStatus; + uint32_t volume[2]; + uint32_t *pVolume = NULL; + uint32_t size = sizeof(volume); + volume[0] = *left; + volume[1] = *right; + if (controller) { + pVolume = volume; + } + status = (*mEffectInterface)->command(mEffectInterface, + EFFECT_CMD_SET_VOLUME, + size, + volume, + &size, + pVolume); + if (controller && status == NO_ERROR && size == sizeof(volume)) { + *left = volume[0]; + *right = volume[1]; + } + } + return status; +} + +status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) +{ + if (device == AUDIO_DEVICE_NONE) { + return NO_ERROR; + } + + Mutex::Autolock _l(mLock); + status_t status = NO_ERROR; + if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { + status_t cmdStatus; + uint32_t size = sizeof(status_t); + uint32_t cmd = audio_is_output_devices(device) ? EFFECT_CMD_SET_DEVICE : + EFFECT_CMD_SET_INPUT_DEVICE; + status = (*mEffectInterface)->command(mEffectInterface, + cmd, + sizeof(uint32_t), + &device, + &size, + &cmdStatus); + } + return status; +} + +status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) +{ + Mutex::Autolock _l(mLock); + status_t status = NO_ERROR; + if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { + status_t cmdStatus; + uint32_t size = sizeof(status_t); + status = (*mEffectInterface)->command(mEffectInterface, + EFFECT_CMD_SET_AUDIO_MODE, + sizeof(audio_mode_t), + &mode, + &size, + &cmdStatus); + if (status == NO_ERROR) { + status = cmdStatus; + } + } + return status; +} + +status_t AudioFlinger::EffectModule::setAudioSource(audio_source_t source) +{ + Mutex::Autolock _l(mLock); + status_t status = NO_ERROR; + if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_SOURCE_MASK) == EFFECT_FLAG_AUDIO_SOURCE_IND) { + uint32_t size = 0; + status = (*mEffectInterface)->command(mEffectInterface, + EFFECT_CMD_SET_AUDIO_SOURCE, + sizeof(audio_source_t), + &source, + &size, + NULL); + } + return status; +} + +void AudioFlinger::EffectModule::setSuspended(bool suspended) +{ + Mutex::Autolock _l(mLock); + mSuspended = suspended; +} + +bool AudioFlinger::EffectModule::suspended() const +{ + Mutex::Autolock _l(mLock); + return mSuspended; +} + +bool AudioFlinger::EffectModule::purgeHandles() +{ + bool enabled = false; + Mutex::Autolock _l(mLock); + for (size_t i = 0; i < mHandles.size(); i++) { + EffectHandle *handle = mHandles[i]; + if (handle != NULL && !handle->destroyed_l()) { + handle->effect().clear(); + if (handle->hasControl()) { + enabled = handle->enabled(); + } + } + } + return enabled; +} + +void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); + result.append(buffer); + + bool locked = AudioFlinger::dumpTryLock(mLock); + // failed to lock - AudioFlinger is probably deadlocked + if (!locked) { + result.append("\t\tCould not lock Fx mutex:\n"); + } + + result.append("\t\tSession Status State Engine:\n"); + snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", + mSessionId, mStatus, mState, (uint32_t)mEffectInterface); + result.append(buffer); + + result.append("\t\tDescriptor:\n"); + snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", + mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, + mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1], + mDescriptor.uuid.node[2], + mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); + result.append(buffer); + snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", + mDescriptor.type.timeLow, mDescriptor.type.timeMid, + mDescriptor.type.timeHiAndVersion, + mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1], + mDescriptor.type.node[2], + mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); + result.append(buffer); + snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", + mDescriptor.apiVersion, + mDescriptor.flags); + result.append(buffer); + snprintf(buffer, SIZE, "\t\t- name: %s\n", + mDescriptor.name); + result.append(buffer); + snprintf(buffer, SIZE, "\t\t- implementor: %s\n", + mDescriptor.implementor); + result.append(buffer); + + result.append("\t\t- Input configuration:\n"); + result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); + snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", + (uint32_t)mConfig.inputCfg.buffer.raw, + mConfig.inputCfg.buffer.frameCount, + mConfig.inputCfg.samplingRate, + mConfig.inputCfg.channels, + mConfig.inputCfg.format); + result.append(buffer); + + result.append("\t\t- Output configuration:\n"); + result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); + snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", + (uint32_t)mConfig.outputCfg.buffer.raw, + mConfig.outputCfg.buffer.frameCount, + mConfig.outputCfg.samplingRate, + mConfig.outputCfg.channels, + mConfig.outputCfg.format); + result.append(buffer); + + snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); + result.append(buffer); + result.append("\t\t\tPid Priority Ctrl Locked client server\n"); + for (size_t i = 0; i < mHandles.size(); ++i) { + EffectHandle *handle = mHandles[i]; + if (handle != NULL && !handle->destroyed_l()) { + handle->dump(buffer, SIZE); + result.append(buffer); + } + } + + result.append("\n"); + + write(fd, result.string(), result.length()); + + if (locked) { + mLock.unlock(); + } +} + +// ---------------------------------------------------------------------------- +// EffectHandle implementation +// ---------------------------------------------------------------------------- + +#undef LOG_TAG +#define LOG_TAG "AudioFlinger::EffectHandle" + +AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, + const sp<AudioFlinger::Client>& client, + const sp<IEffectClient>& effectClient, + int32_t priority) + : BnEffect(), + mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), + mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) +{ + ALOGV("constructor %p", this); + + if (client == 0) { + return; + } + int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); + mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); + if (mCblkMemory != 0) { + mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); + + if (mCblk != NULL) { + new(mCblk) effect_param_cblk_t(); + mBuffer = (uint8_t *)mCblk + bufOffset; + } + } else { + ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + + sizeof(effect_param_cblk_t)); + return; + } +} + +AudioFlinger::EffectHandle::~EffectHandle() +{ + ALOGV("Destructor %p", this); + + if (mEffect == 0) { + mDestroyed = true; + return; + } + mEffect->lock(); + mDestroyed = true; + mEffect->unlock(); + disconnect(false); +} + +status_t AudioFlinger::EffectHandle::enable() +{ + ALOGV("enable %p", this); + if (!mHasControl) { + return INVALID_OPERATION; + } + if (mEffect == 0) { + return DEAD_OBJECT; + } + + if (mEnabled) { + return NO_ERROR; + } + + mEnabled = true; + + sp<ThreadBase> thread = mEffect->thread().promote(); + if (thread != 0) { + thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); + } + + // checkSuspendOnEffectEnabled() can suspend this same effect when enabled + if (mEffect->suspended()) { + return NO_ERROR; + } + + status_t status = mEffect->setEnabled(true); + if (status != NO_ERROR) { + if (thread != 0) { + thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); + } + mEnabled = false; + } + return status; +} + +status_t AudioFlinger::EffectHandle::disable() +{ + ALOGV("disable %p", this); + if (!mHasControl) { + return INVALID_OPERATION; + } + if (mEffect == 0) { + return DEAD_OBJECT; + } + + if (!mEnabled) { + return NO_ERROR; + } + mEnabled = false; + + if (mEffect->suspended()) { + return NO_ERROR; + } + + status_t status = mEffect->setEnabled(false); + + sp<ThreadBase> thread = mEffect->thread().promote(); + if (thread != 0) { + thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); + } + + return status; +} + +void AudioFlinger::EffectHandle::disconnect() +{ + disconnect(true); +} + +void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) +{ + ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); + if (mEffect == 0) { + return; + } + // restore suspended effects if the disconnected handle was enabled and the last one. + if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { + sp<ThreadBase> thread = mEffect->thread().promote(); + if (thread != 0) { + thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); + } + } + + // release sp on module => module destructor can be called now + mEffect.clear(); + if (mClient != 0) { + if (mCblk != NULL) { + // unlike ~TrackBase(), mCblk is never a local new, so don't delete + mCblk->~effect_param_cblk_t(); // destroy our shared-structure. + } + mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to + // Client destructor must run with AudioFlinger mutex locked + Mutex::Autolock _l(mClient->audioFlinger()->mLock); + mClient.clear(); + } +} + +status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, + uint32_t cmdSize, + void *pCmdData, + uint32_t *replySize, + void *pReplyData) +{ + ALOGVV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", + cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); + + // only get parameter command is permitted for applications not controlling the effect + if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { + return INVALID_OPERATION; + } + if (mEffect == 0) { + return DEAD_OBJECT; + } + if (mClient == 0) { + return INVALID_OPERATION; + } + + // handle commands that are not forwarded transparently to effect engine + if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { + // No need to trylock() here as this function is executed in the binder thread serving a + // particular client process: no risk to block the whole media server process or mixer + // threads if we are stuck here + Mutex::Autolock _l(mCblk->lock); + if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || + mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { + mCblk->serverIndex = 0; + mCblk->clientIndex = 0; + return BAD_VALUE; + } + status_t status = NO_ERROR; + while (mCblk->serverIndex < mCblk->clientIndex) { + int reply; + uint32_t rsize = sizeof(int); + int *p = (int *)(mBuffer + mCblk->serverIndex); + int size = *p++; + if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { + ALOGW("command(): invalid parameter block size"); + break; + } + effect_param_t *param = (effect_param_t *)p; + if (param->psize == 0 || param->vsize == 0) { + ALOGW("command(): null parameter or value size"); + mCblk->serverIndex += size; + continue; + } + uint32_t psize = sizeof(effect_param_t) + + ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + + param->vsize; + status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, + psize, + p, + &rsize, + &reply); + // stop at first error encountered + if (ret != NO_ERROR) { + status = ret; + *(int *)pReplyData = reply; + break; + } else if (reply != NO_ERROR) { + *(int *)pReplyData = reply; + break; + } + mCblk->serverIndex += size; + } + mCblk->serverIndex = 0; + mCblk->clientIndex = 0; + return status; + } else if (cmdCode == EFFECT_CMD_ENABLE) { + *(int *)pReplyData = NO_ERROR; + return enable(); + } else if (cmdCode == EFFECT_CMD_DISABLE) { + *(int *)pReplyData = NO_ERROR; + return disable(); + } + + return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); +} + +void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) +{ + ALOGV("setControl %p control %d", this, hasControl); + + mHasControl = hasControl; + mEnabled = enabled; + + if (signal && mEffectClient != 0) { + mEffectClient->controlStatusChanged(hasControl); + } +} + +void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, + uint32_t cmdSize, + void *pCmdData, + uint32_t replySize, + void *pReplyData) +{ + if (mEffectClient != 0) { + mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); + } +} + + + +void AudioFlinger::EffectHandle::setEnabled(bool enabled) +{ + if (mEffectClient != 0) { + mEffectClient->enableStatusChanged(enabled); + } +} + +status_t AudioFlinger::EffectHandle::onTransact( + uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) +{ + return BnEffect::onTransact(code, data, reply, flags); +} + + +void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) +{ + bool locked = mCblk != NULL && AudioFlinger::dumpTryLock(mCblk->lock); + + snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", + (mClient == 0) ? getpid_cached : mClient->pid(), + mPriority, + mHasControl, + !locked, + mCblk ? mCblk->clientIndex : 0, + mCblk ? mCblk->serverIndex : 0 + ); + + if (locked) { + mCblk->lock.unlock(); + } +} + +#undef LOG_TAG +#define LOG_TAG "AudioFlinger::EffectChain" + +AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, + int sessionId) + : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), + mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), + mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) +{ + mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); + if (thread == NULL) { + return; + } + mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / + thread->frameCount(); +} + +AudioFlinger::EffectChain::~EffectChain() +{ + if (mOwnInBuffer) { + delete mInBuffer; + } + +} + +// getEffectFromDesc_l() must be called with ThreadBase::mLock held +sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l( + effect_descriptor_t *descriptor) +{ + size_t size = mEffects.size(); + + for (size_t i = 0; i < size; i++) { + if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { + return mEffects[i]; + } + } + return 0; +} + +// getEffectFromId_l() must be called with ThreadBase::mLock held +sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) +{ + size_t size = mEffects.size(); + + for (size_t i = 0; i < size; i++) { + // by convention, return first effect if id provided is 0 (0 is never a valid id) + if (id == 0 || mEffects[i]->id() == id) { + return mEffects[i]; + } + } + return 0; +} + +// getEffectFromType_l() must be called with ThreadBase::mLock held +sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( + const effect_uuid_t *type) +{ + size_t size = mEffects.size(); + + for (size_t i = 0; i < size; i++) { + if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { + return mEffects[i]; + } + } + return 0; +} + +void AudioFlinger::EffectChain::clearInputBuffer() +{ + Mutex::Autolock _l(mLock); + sp<ThreadBase> thread = mThread.promote(); + if (thread == 0) { + ALOGW("clearInputBuffer(): cannot promote mixer thread"); + return; + } + clearInputBuffer_l(thread); +} + +// Must be called with EffectChain::mLock locked +void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) +{ + size_t numSamples = thread->frameCount() * thread->channelCount(); + memset(mInBuffer, 0, numSamples * sizeof(int16_t)); + +} + +// Must be called with EffectChain::mLock locked +void AudioFlinger::EffectChain::process_l() +{ + sp<ThreadBase> thread = mThread.promote(); + if (thread == 0) { + ALOGW("process_l(): cannot promote mixer thread"); + return; + } + bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || + (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); + // always process effects unless no more tracks are on the session and the effect tail + // has been rendered + bool doProcess = true; + if (!isGlobalSession) { + bool tracksOnSession = (trackCnt() != 0); + + if (!tracksOnSession && mTailBufferCount == 0) { + doProcess = false; + } + + if (activeTrackCnt() == 0) { + // if no track is active and the effect tail has not been rendered, + // the input buffer must be cleared here as the mixer process will not do it + if (tracksOnSession || mTailBufferCount > 0) { + clearInputBuffer_l(thread); + if (mTailBufferCount > 0) { + mTailBufferCount--; + } + } + } + } + + size_t size = mEffects.size(); + if (doProcess) { + for (size_t i = 0; i < size; i++) { + mEffects[i]->process(); + } + } + for (size_t i = 0; i < size; i++) { + mEffects[i]->updateState(); + } +} + +// addEffect_l() must be called with PlaybackThread::mLock held +status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) +{ + effect_descriptor_t desc = effect->desc(); + uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; + + Mutex::Autolock _l(mLock); + effect->setChain(this); + sp<ThreadBase> thread = mThread.promote(); + if (thread == 0) { + return NO_INIT; + } + effect->setThread(thread); + + if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { + // Auxiliary effects are inserted at the beginning of mEffects vector as + // they are processed first and accumulated in chain input buffer + mEffects.insertAt(effect, 0); + + // the input buffer for auxiliary effect contains mono samples in + // 32 bit format. This is to avoid saturation in AudoMixer + // accumulation stage. Saturation is done in EffectModule::process() before + // calling the process in effect engine + size_t numSamples = thread->frameCount(); + int32_t *buffer = new int32_t[numSamples]; + memset(buffer, 0, numSamples * sizeof(int32_t)); + effect->setInBuffer((int16_t *)buffer); + // auxiliary effects output samples to chain input buffer for further processing + // by insert effects + effect->setOutBuffer(mInBuffer); + } else { + // Insert effects are inserted at the end of mEffects vector as they are processed + // after track and auxiliary effects. + // Insert effect order as a function of indicated preference: + // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if + // another effect is present + // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the + // last effect claiming first position + // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the + // first effect claiming last position + // else if EFFECT_FLAG_INSERT_ANY insert after first or before last + // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is + // already present + + size_t size = mEffects.size(); + size_t idx_insert = size; + ssize_t idx_insert_first = -1; + ssize_t idx_insert_last = -1; + + for (size_t i = 0; i < size; i++) { + effect_descriptor_t d = mEffects[i]->desc(); + uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; + uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; + if (iMode == EFFECT_FLAG_TYPE_INSERT) { + // check invalid effect chaining combinations + if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || + iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { + ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", + desc.name, d.name); + return INVALID_OPERATION; + } + // remember position of first insert effect and by default + // select this as insert position for new effect + if (idx_insert == size) { + idx_insert = i; + } + // remember position of last insert effect claiming + // first position + if (iPref == EFFECT_FLAG_INSERT_FIRST) { + idx_insert_first = i; + } + // remember position of first insert effect claiming + // last position + if (iPref == EFFECT_FLAG_INSERT_LAST && + idx_insert_last == -1) { + idx_insert_last = i; + } + } + } + + // modify idx_insert from first position if needed + if (insertPref == EFFECT_FLAG_INSERT_LAST) { + if (idx_insert_last != -1) { + idx_insert = idx_insert_last; + } else { + idx_insert = size; + } + } else { + if (idx_insert_first != -1) { + idx_insert = idx_insert_first + 1; + } + } + + // always read samples from chain input buffer + effect->setInBuffer(mInBuffer); + + // if last effect in the chain, output samples to chain + // output buffer, otherwise to chain input buffer + if (idx_insert == size) { + if (idx_insert != 0) { + mEffects[idx_insert-1]->setOutBuffer(mInBuffer); + mEffects[idx_insert-1]->configure(); + } + effect->setOutBuffer(mOutBuffer); + } else { + effect->setOutBuffer(mInBuffer); + } + mEffects.insertAt(effect, idx_insert); + + ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, + idx_insert); + } + effect->configure(); + return NO_ERROR; +} + +// removeEffect_l() must be called with PlaybackThread::mLock held +size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) +{ + Mutex::Autolock _l(mLock); + size_t size = mEffects.size(); + uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; + + for (size_t i = 0; i < size; i++) { + if (effect == mEffects[i]) { + // calling stop here will remove pre-processing effect from the audio HAL. + // This is safe as we hold the EffectChain mutex which guarantees that we are not in + // the middle of a read from audio HAL + if (mEffects[i]->state() == EffectModule::ACTIVE || + mEffects[i]->state() == EffectModule::STOPPING) { + mEffects[i]->stop(); + } + if (type == EFFECT_FLAG_TYPE_AUXILIARY) { + delete[] effect->inBuffer(); + } else { + if (i == size - 1 && i != 0) { + mEffects[i - 1]->setOutBuffer(mOutBuffer); + mEffects[i - 1]->configure(); + } + } + mEffects.removeAt(i); + ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), + this, i); + break; + } + } + + return mEffects.size(); +} + +// setDevice_l() must be called with PlaybackThread::mLock held +void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) +{ + size_t size = mEffects.size(); + for (size_t i = 0; i < size; i++) { + mEffects[i]->setDevice(device); + } +} + +// setMode_l() must be called with PlaybackThread::mLock held +void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) +{ + size_t size = mEffects.size(); + for (size_t i = 0; i < size; i++) { + mEffects[i]->setMode(mode); + } +} + +// setAudioSource_l() must be called with PlaybackThread::mLock held +void AudioFlinger::EffectChain::setAudioSource_l(audio_source_t source) +{ + size_t size = mEffects.size(); + for (size_t i = 0; i < size; i++) { + mEffects[i]->setAudioSource(source); + } +} + +// setVolume_l() must be called with PlaybackThread::mLock held +bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) +{ + uint32_t newLeft = *left; + uint32_t newRight = *right; + bool hasControl = false; + int ctrlIdx = -1; + size_t size = mEffects.size(); + + // first update volume controller + for (size_t i = size; i > 0; i--) { + if (mEffects[i - 1]->isProcessEnabled() && + (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { + ctrlIdx = i - 1; + hasControl = true; + break; + } + } + + if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { + if (hasControl) { + *left = mNewLeftVolume; + *right = mNewRightVolume; + } + return hasControl; + } + + mVolumeCtrlIdx = ctrlIdx; + mLeftVolume = newLeft; + mRightVolume = newRight; + + // second get volume update from volume controller + if (ctrlIdx >= 0) { + mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); + mNewLeftVolume = newLeft; + mNewRightVolume = newRight; + } + // then indicate volume to all other effects in chain. + // Pass altered volume to effects before volume controller + // and requested volume to effects after controller + uint32_t lVol = newLeft; + uint32_t rVol = newRight; + + for (size_t i = 0; i < size; i++) { + if ((int)i == ctrlIdx) { + continue; + } + // this also works for ctrlIdx == -1 when there is no volume controller + if ((int)i > ctrlIdx) { + lVol = *left; + rVol = *right; + } + mEffects[i]->setVolume(&lVol, &rVol, false); + } + *left = newLeft; + *right = newRight; + + return hasControl; +} + +void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); + result.append(buffer); + + bool locked = AudioFlinger::dumpTryLock(mLock); + // failed to lock - AudioFlinger is probably deadlocked + if (!locked) { + result.append("\tCould not lock mutex:\n"); + } + + result.append("\tNum fx In buffer Out buffer Active tracks:\n"); + snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", + mEffects.size(), + (uint32_t)mInBuffer, + (uint32_t)mOutBuffer, + mActiveTrackCnt); + result.append(buffer); + write(fd, result.string(), result.size()); + + for (size_t i = 0; i < mEffects.size(); ++i) { + sp<EffectModule> effect = mEffects[i]; + if (effect != 0) { + effect->dump(fd, args); + } + } + + if (locked) { + mLock.unlock(); + } +} + +// must be called with ThreadBase::mLock held +void AudioFlinger::EffectChain::setEffectSuspended_l( + const effect_uuid_t *type, bool suspend) +{ + sp<SuspendedEffectDesc> desc; + // use effect type UUID timelow as key as there is no real risk of identical + // timeLow fields among effect type UUIDs. + ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); + if (suspend) { + if (index >= 0) { + desc = mSuspendedEffects.valueAt(index); + } else { + desc = new SuspendedEffectDesc(); + desc->mType = *type; + mSuspendedEffects.add(type->timeLow, desc); + ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); + } + if (desc->mRefCount++ == 0) { + sp<EffectModule> effect = getEffectIfEnabled(type); + if (effect != 0) { + desc->mEffect = effect; + effect->setSuspended(true); + effect->setEnabled(false); + } + } + } else { + if (index < 0) { + return; + } + desc = mSuspendedEffects.valueAt(index); + if (desc->mRefCount <= 0) { + ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); + desc->mRefCount = 1; + } + if (--desc->mRefCount == 0) { + ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); + if (desc->mEffect != 0) { + sp<EffectModule> effect = desc->mEffect.promote(); + if (effect != 0) { + effect->setSuspended(false); + effect->lock(); + EffectHandle *handle = effect->controlHandle_l(); + if (handle != NULL && !handle->destroyed_l()) { + effect->setEnabled_l(handle->enabled()); + } + effect->unlock(); + } + desc->mEffect.clear(); + } + mSuspendedEffects.removeItemsAt(index); + } + } +} + +// must be called with ThreadBase::mLock held +void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) +{ + sp<SuspendedEffectDesc> desc; + + ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); + if (suspend) { + if (index >= 0) { + desc = mSuspendedEffects.valueAt(index); + } else { + desc = new SuspendedEffectDesc(); + mSuspendedEffects.add((int)kKeyForSuspendAll, desc); + ALOGV("setEffectSuspendedAll_l() add entry for 0"); + } + if (desc->mRefCount++ == 0) { + Vector< sp<EffectModule> > effects; + getSuspendEligibleEffects(effects); + for (size_t i = 0; i < effects.size(); i++) { + setEffectSuspended_l(&effects[i]->desc().type, true); + } + } + } else { + if (index < 0) { + return; + } + desc = mSuspendedEffects.valueAt(index); + if (desc->mRefCount <= 0) { + ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); + desc->mRefCount = 1; + } + if (--desc->mRefCount == 0) { + Vector<const effect_uuid_t *> types; + for (size_t i = 0; i < mSuspendedEffects.size(); i++) { + if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { + continue; + } + types.add(&mSuspendedEffects.valueAt(i)->mType); + } + for (size_t i = 0; i < types.size(); i++) { + setEffectSuspended_l(types[i], false); + } + ALOGV("setEffectSuspendedAll_l() remove entry for %08x", + mSuspendedEffects.keyAt(index)); + mSuspendedEffects.removeItem((int)kKeyForSuspendAll); + } + } +} + + +// The volume effect is used for automated tests only +#ifndef OPENSL_ES_H_ +static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, + { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; +const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; +#endif //OPENSL_ES_H_ + +bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) +{ + // auxiliary effects and visualizer are never suspended on output mix + if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && + (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || + (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || + (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { + return false; + } + return true; +} + +void AudioFlinger::EffectChain::getSuspendEligibleEffects( + Vector< sp<AudioFlinger::EffectModule> > &effects) +{ + effects.clear(); + for (size_t i = 0; i < mEffects.size(); i++) { + if (isEffectEligibleForSuspend(mEffects[i]->desc())) { + effects.add(mEffects[i]); + } + } +} + +sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( + const effect_uuid_t *type) +{ + sp<EffectModule> effect = getEffectFromType_l(type); + return effect != 0 && effect->isEnabled() ? effect : 0; +} + +void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, + bool enabled) +{ + ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); + if (enabled) { + if (index < 0) { + // if the effect is not suspend check if all effects are suspended + index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); + if (index < 0) { + return; + } + if (!isEffectEligibleForSuspend(effect->desc())) { + return; + } + setEffectSuspended_l(&effect->desc().type, enabled); + index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); + if (index < 0) { + ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); + return; + } + } + ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", + effect->desc().type.timeLow); + sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); + // if effect is requested to suspended but was not yet enabled, supend it now. + if (desc->mEffect == 0) { + desc->mEffect = effect; + effect->setEnabled(false); + effect->setSuspended(true); + } + } else { + if (index < 0) { + return; + } + ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", + effect->desc().type.timeLow); + sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); + desc->mEffect.clear(); + effect->setSuspended(false); + } +} + +}; // namespace android diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h new file mode 100644 index 0000000..91303ee --- /dev/null +++ b/services/audioflinger/Effects.h @@ -0,0 +1,359 @@ +/* +** +** Copyright 2012, 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. +*/ + +#ifndef INCLUDING_FROM_AUDIOFLINGER_H + #error This header file should only be included from AudioFlinger.h +#endif + +//--- Audio Effect Management + +// EffectModule and EffectChain classes both have their own mutex to protect +// state changes or resource modifications. Always respect the following order +// if multiple mutexes must be acquired to avoid cross deadlock: +// AudioFlinger -> ThreadBase -> EffectChain -> EffectModule + +// The EffectModule class is a wrapper object controlling the effect engine implementation +// in the effect library. It prevents concurrent calls to process() and command() functions +// from different client threads. It keeps a list of EffectHandle objects corresponding +// to all client applications using this effect and notifies applications of effect state, +// control or parameter changes. It manages the activation state machine to send appropriate +// reset, enable, disable commands to effect engine and provide volume +// ramping when effects are activated/deactivated. +// When controlling an auxiliary effect, the EffectModule also provides an input buffer used by +// the attached track(s) to accumulate their auxiliary channel. +class EffectModule : public RefBase { +public: + EffectModule(ThreadBase *thread, + const wp<AudioFlinger::EffectChain>& chain, + effect_descriptor_t *desc, + int id, + int sessionId); + virtual ~EffectModule(); + + enum effect_state { + IDLE, + RESTART, + STARTING, + ACTIVE, + STOPPING, + STOPPED, + DESTROYED + }; + + int id() const { return mId; } + void process(); + void updateState(); + status_t command(uint32_t cmdCode, + uint32_t cmdSize, + void *pCmdData, + uint32_t *replySize, + void *pReplyData); + + void reset_l(); + status_t configure(); + status_t init(); + effect_state state() const { + return mState; + } + uint32_t status() { + return mStatus; + } + int sessionId() const { + return mSessionId; + } + status_t setEnabled(bool enabled); + status_t setEnabled_l(bool enabled); + bool isEnabled() const; + bool isProcessEnabled() const; + + void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; } + int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; } + void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; } + int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; } + void setChain(const wp<EffectChain>& chain) { mChain = chain; } + void setThread(const wp<ThreadBase>& thread) { mThread = thread; } + const wp<ThreadBase>& thread() { return mThread; } + + status_t addHandle(EffectHandle *handle); + size_t disconnect(EffectHandle *handle, bool unpinIfLast); + size_t removeHandle(EffectHandle *handle); + + const effect_descriptor_t& desc() const { return mDescriptor; } + wp<EffectChain>& chain() { return mChain; } + + status_t setDevice(audio_devices_t device); + status_t setVolume(uint32_t *left, uint32_t *right, bool controller); + status_t setMode(audio_mode_t mode); + status_t setAudioSource(audio_source_t source); + status_t start(); + status_t stop(); + void setSuspended(bool suspended); + bool suspended() const; + + EffectHandle* controlHandle_l(); + + bool isPinned() const { return mPinned; } + void unPin() { mPinned = false; } + bool purgeHandles(); + void lock() { mLock.lock(); } + void unlock() { mLock.unlock(); } + + void dump(int fd, const Vector<String16>& args); + +protected: + friend class AudioFlinger; // for mHandles + bool mPinned; + + // Maximum time allocated to effect engines to complete the turn off sequence + static const uint32_t MAX_DISABLE_TIME_MS = 10000; + + EffectModule(const EffectModule&); + EffectModule& operator = (const EffectModule&); + + status_t start_l(); + status_t stop_l(); + +mutable Mutex mLock; // mutex for process, commands and handles list protection + wp<ThreadBase> mThread; // parent thread + wp<EffectChain> mChain; // parent effect chain + const int mId; // this instance unique ID + const int mSessionId; // audio session ID + const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine + effect_config_t mConfig; // input and output audio configuration + effect_handle_t mEffectInterface; // Effect module C API + status_t mStatus; // initialization status + effect_state mState; // current activation state + Vector<EffectHandle *> mHandles; // list of client handles + // First handle in mHandles has highest priority and controls the effect module + uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after + // sending disable command. + uint32_t mDisableWaitCnt; // current process() calls count during disable period. + bool mSuspended; // effect is suspended: temporarily disabled by framework +}; + +// The EffectHandle class implements the IEffect interface. It provides resources +// to receive parameter updates, keeps track of effect control +// ownership and state and has a pointer to the EffectModule object it is controlling. +// There is one EffectHandle object for each application controlling (or using) +// an effect module. +// The EffectHandle is obtained by calling AudioFlinger::createEffect(). +class EffectHandle: public android::BnEffect { +public: + + EffectHandle(const sp<EffectModule>& effect, + const sp<AudioFlinger::Client>& client, + const sp<IEffectClient>& effectClient, + int32_t priority); + virtual ~EffectHandle(); + + // IEffect + virtual status_t enable(); + virtual status_t disable(); + virtual status_t command(uint32_t cmdCode, + uint32_t cmdSize, + void *pCmdData, + uint32_t *replySize, + void *pReplyData); + virtual void disconnect(); +private: + void disconnect(bool unpinIfLast); +public: + virtual sp<IMemory> getCblk() const { return mCblkMemory; } + virtual status_t onTransact(uint32_t code, const Parcel& data, + Parcel* reply, uint32_t flags); + + + // Give or take control of effect module + // - hasControl: true if control is given, false if removed + // - signal: true client app should be signaled of change, false otherwise + // - enabled: state of the effect when control is passed + void setControl(bool hasControl, bool signal, bool enabled); + void commandExecuted(uint32_t cmdCode, + uint32_t cmdSize, + void *pCmdData, + uint32_t replySize, + void *pReplyData); + void setEnabled(bool enabled); + bool enabled() const { return mEnabled; } + + // Getters + int id() const { return mEffect->id(); } + int priority() const { return mPriority; } + bool hasControl() const { return mHasControl; } + sp<EffectModule> effect() const { return mEffect; } + // destroyed_l() must be called with the associated EffectModule mLock held + bool destroyed_l() const { return mDestroyed; } + + void dump(char* buffer, size_t size); + +protected: + friend class AudioFlinger; // for mEffect, mHasControl, mEnabled + EffectHandle(const EffectHandle&); + EffectHandle& operator =(const EffectHandle&); + + sp<EffectModule> mEffect; // pointer to controlled EffectModule + sp<IEffectClient> mEffectClient; // callback interface for client notifications + /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect() + sp<IMemory> mCblkMemory; // shared memory for control block + effect_param_cblk_t* mCblk; // control block for deferred parameter setting via + // shared memory + uint8_t* mBuffer; // pointer to parameter area in shared memory + int mPriority; // client application priority to control the effect + bool mHasControl; // true if this handle is controlling the effect + bool mEnabled; // cached enable state: needed when the effect is + // restored after being suspended + bool mDestroyed; // Set to true by destructor. Access with EffectModule + // mLock held +}; + +// the EffectChain class represents a group of effects associated to one audio session. +// There can be any number of EffectChain objects per output mixer thread (PlaybackThread). +// The EffecChain with session ID 0 contains global effects applied to the output mix. +// Effects in this chain can be insert or auxiliary. Effects in other chains (attached to +// tracks) are insert only. The EffectChain maintains an ordered list of effect module, the +// order corresponding in the effect process order. When attached to a track (session ID != 0), +// it also provide it's own input buffer used by the track as accumulation buffer. +class EffectChain : public RefBase { +public: + EffectChain(const wp<ThreadBase>& wThread, int sessionId); + EffectChain(ThreadBase *thread, int sessionId); + virtual ~EffectChain(); + + // special key used for an entry in mSuspendedEffects keyed vector + // corresponding to a suspend all request. + static const int kKeyForSuspendAll = 0; + + // minimum duration during which we force calling effect process when last track on + // a session is stopped or removed to allow effect tail to be rendered + static const int kProcessTailDurationMs = 1000; + + void process_l(); + + void lock() { + mLock.lock(); + } + void unlock() { + mLock.unlock(); + } + + status_t addEffect_l(const sp<EffectModule>& handle); + size_t removeEffect_l(const sp<EffectModule>& handle); + + int sessionId() const { return mSessionId; } + void setSessionId(int sessionId) { mSessionId = sessionId; } + + sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor); + sp<EffectModule> getEffectFromId_l(int id); + sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type); + bool setVolume_l(uint32_t *left, uint32_t *right); + void setDevice_l(audio_devices_t device); + void setMode_l(audio_mode_t mode); + void setAudioSource_l(audio_source_t source); + + void setInBuffer(int16_t *buffer, bool ownsBuffer = false) { + mInBuffer = buffer; + mOwnInBuffer = ownsBuffer; + } + int16_t *inBuffer() const { + return mInBuffer; + } + void setOutBuffer(int16_t *buffer) { + mOutBuffer = buffer; + } + int16_t *outBuffer() const { + return mOutBuffer; + } + + void incTrackCnt() { android_atomic_inc(&mTrackCnt); } + void decTrackCnt() { android_atomic_dec(&mTrackCnt); } + int32_t trackCnt() const { return android_atomic_acquire_load(&mTrackCnt); } + + void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt); + mTailBufferCount = mMaxTailBuffers; } + void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); } + int32_t activeTrackCnt() const { return android_atomic_acquire_load(&mActiveTrackCnt); } + + uint32_t strategy() const { return mStrategy; } + void setStrategy(uint32_t strategy) + { mStrategy = strategy; } + + // suspend effect of the given type + void setEffectSuspended_l(const effect_uuid_t *type, + bool suspend); + // suspend all eligible effects + void setEffectSuspendedAll_l(bool suspend); + // check if effects should be suspend or restored when a given effect is enable or disabled + void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, + bool enabled); + + void clearInputBuffer(); + + void dump(int fd, const Vector<String16>& args); + +protected: + friend class AudioFlinger; // for mThread, mEffects + EffectChain(const EffectChain&); + EffectChain& operator =(const EffectChain&); + + class SuspendedEffectDesc : public RefBase { + public: + SuspendedEffectDesc() : mRefCount(0) {} + + int mRefCount; + effect_uuid_t mType; + wp<EffectModule> mEffect; + }; + + // get a list of effect modules to suspend when an effect of the type + // passed is enabled. + void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects); + + // get an effect module if it is currently enable + sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type); + // true if the effect whose descriptor is passed can be suspended + // OEMs can modify the rules implemented in this method to exclude specific effect + // types or implementations from the suspend/restore mechanism. + bool isEffectEligibleForSuspend(const effect_descriptor_t& desc); + + void clearInputBuffer_l(sp<ThreadBase> thread); + + wp<ThreadBase> mThread; // parent mixer thread + Mutex mLock; // mutex protecting effect list + Vector< sp<EffectModule> > mEffects; // list of effect modules + int mSessionId; // audio session ID + int16_t *mInBuffer; // chain input buffer + int16_t *mOutBuffer; // chain output buffer + + // 'volatile' here means these are accessed with atomic operations instead of mutex + volatile int32_t mActiveTrackCnt; // number of active tracks connected + volatile int32_t mTrackCnt; // number of tracks connected + + int32_t mTailBufferCount; // current effect tail buffer count + int32_t mMaxTailBuffers; // maximum effect tail buffers + bool mOwnInBuffer; // true if the chain owns its input buffer + int mVolumeCtrlIdx; // index of insert effect having control over volume + uint32_t mLeftVolume; // previous volume on left channel + uint32_t mRightVolume; // previous volume on right channel + uint32_t mNewLeftVolume; // new volume on left channel + uint32_t mNewRightVolume; // new volume on right channel + uint32_t mStrategy; // strategy for this effect chain + // mSuspendedEffects lists all effects currently suspended in the chain. + // Use effect type UUID timelow field as key. There is no real risk of identical + // timeLow fields among effect type UUIDs. + // Updated by updateSuspendedSessions_l() only. + KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects; +}; diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h new file mode 100644 index 0000000..b898924 --- /dev/null +++ b/services/audioflinger/PlaybackTracks.h @@ -0,0 +1,285 @@ +/* +** +** Copyright 2012, 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. +*/ + +#ifndef INCLUDING_FROM_AUDIOFLINGER_H + #error This header file should only be included from AudioFlinger.h +#endif + +// playback track +class Track : public TrackBase, public VolumeProvider { +public: + Track( PlaybackThread *thread, + const sp<Client>& client, + audio_stream_type_t streamType, + uint32_t sampleRate, + audio_format_t format, + audio_channel_mask_t channelMask, + size_t frameCount, + const sp<IMemory>& sharedBuffer, + int sessionId, + IAudioFlinger::track_flags_t flags); + virtual ~Track(); + + static void appendDumpHeader(String8& result); + void dump(char* buffer, size_t size); + virtual status_t start(AudioSystem::sync_event_t event = + AudioSystem::SYNC_EVENT_NONE, + int triggerSession = 0); + virtual void stop(); + void pause(); + + void flush(); + void destroy(); + void mute(bool); + int name() const { return mName; } + + audio_stream_type_t streamType() const { + return mStreamType; + } + status_t attachAuxEffect(int EffectId); + void setAuxBuffer(int EffectId, int32_t *buffer); + int32_t *auxBuffer() const { return mAuxBuffer; } + void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; } + int16_t *mainBuffer() const { return mMainBuffer; } + int auxEffectId() const { return mAuxEffectId; } + +// implement FastMixerState::VolumeProvider interface + virtual uint32_t getVolumeLR(); + + virtual status_t setSyncEvent(const sp<SyncEvent>& event); + +protected: + // for numerous + friend class PlaybackThread; + friend class MixerThread; + friend class DirectOutputThread; + + Track(const Track&); + Track& operator = (const Track&); + + // AudioBufferProvider interface + virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, + int64_t pts = kInvalidPTS); + // releaseBuffer() not overridden + + virtual size_t framesReady() const; + + bool isMuted() const { return mMute; } + bool isPausing() const { + return mState == PAUSING; + } + bool isPaused() const { + return mState == PAUSED; + } + bool isResuming() const { + return mState == RESUMING; + } + bool isReady() const; + void setPaused() { mState = PAUSED; } + void reset(); + + bool isOutputTrack() const { + return (mStreamType == AUDIO_STREAM_CNT); + } + + sp<IMemory> sharedBuffer() const { return mSharedBuffer; } + + // framesWritten is cumulative, never reset, and is shared all tracks + // audioHalFrames is derived from output latency + // FIXME parameters not needed, could get them from the thread + bool presentationComplete(size_t framesWritten, size_t audioHalFrames); + +public: + void triggerEvents(AudioSystem::sync_event_t type); + virtual bool isTimedTrack() const { return false; } + bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; } + virtual bool isOut() const; + +protected: + + // written by Track::mute() called by binder thread(s), without a mutex or barrier. + // read by Track::isMuted() called by playback thread, also without a mutex or barrier. + // The lack of mutex or barrier is safe because the mute status is only used by itself. + bool mMute; + + // FILLED state is used for suppressing volume ramp at begin of playing + enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; + mutable uint8_t mFillingUpStatus; + int8_t mRetryCount; + const sp<IMemory> mSharedBuffer; + bool mResetDone; + const audio_stream_type_t mStreamType; + int mName; // track name on the normal mixer, + // allocated statically at track creation time, + // and is even allocated (though unused) for fast tracks + // FIXME don't allocate track name for fast tracks + int16_t *mMainBuffer; + int32_t *mAuxBuffer; + int mAuxEffectId; + bool mHasVolumeController; + size_t mPresentationCompleteFrames; // number of frames written to the + // audio HAL when this track will be fully rendered + // zero means not monitoring +private: + IAudioFlinger::track_flags_t mFlags; + + // The following fields are only for fast tracks, and should be in a subclass + int mFastIndex; // index within FastMixerState::mFastTracks[]; + // either mFastIndex == -1 if not isFastTrack() + // or 0 < mFastIndex < FastMixerState::kMaxFast because + // index 0 is reserved for normal mixer's submix; + // index is allocated statically at track creation time + // but the slot is only used if track is active + FastTrackUnderruns mObservedUnderruns; // Most recently observed value of + // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns + uint32_t mUnderrunCount; // Counter of total number of underruns, never reset + volatile float mCachedVolume; // combined master volume and stream type volume; + // 'volatile' means accessed without lock or + // barrier, but is read/written atomically +}; // end of Track + +class TimedTrack : public Track { + public: + static sp<TimedTrack> create(PlaybackThread *thread, + const sp<Client>& client, + audio_stream_type_t streamType, + uint32_t sampleRate, + audio_format_t format, + audio_channel_mask_t channelMask, + size_t frameCount, + const sp<IMemory>& sharedBuffer, + int sessionId); + virtual ~TimedTrack(); + + class TimedBuffer { + public: + TimedBuffer(); + TimedBuffer(const sp<IMemory>& buffer, int64_t pts); + const sp<IMemory>& buffer() const { return mBuffer; } + int64_t pts() const { return mPTS; } + uint32_t position() const { return mPosition; } + void setPosition(uint32_t pos) { mPosition = pos; } + private: + sp<IMemory> mBuffer; + int64_t mPTS; + uint32_t mPosition; + }; + + // Mixer facing methods. + virtual bool isTimedTrack() const { return true; } + virtual size_t framesReady() const; + + // AudioBufferProvider interface + virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, + int64_t pts); + virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); + + // Client/App facing methods. + status_t allocateTimedBuffer(size_t size, + sp<IMemory>* buffer); + status_t queueTimedBuffer(const sp<IMemory>& buffer, + int64_t pts); + status_t setMediaTimeTransform(const LinearTransform& xform, + TimedAudioTrack::TargetTimeline target); + + private: + TimedTrack(PlaybackThread *thread, + const sp<Client>& client, + audio_stream_type_t streamType, + uint32_t sampleRate, + audio_format_t format, + audio_channel_mask_t channelMask, + size_t frameCount, + const sp<IMemory>& sharedBuffer, + int sessionId); + + void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer); + void timedYieldSilence_l(uint32_t numFrames, + AudioBufferProvider::Buffer* buffer); + void trimTimedBufferQueue_l(); + void trimTimedBufferQueueHead_l(const char* logTag); + void updateFramesPendingAfterTrim_l(const TimedBuffer& buf, + const char* logTag); + + uint64_t mLocalTimeFreq; + LinearTransform mLocalTimeToSampleTransform; + LinearTransform mMediaTimeToSampleTransform; + sp<MemoryDealer> mTimedMemoryDealer; + + Vector<TimedBuffer> mTimedBufferQueue; + bool mQueueHeadInFlight; + bool mTrimQueueHeadOnRelease; + uint32_t mFramesPendingInQueue; + + uint8_t* mTimedSilenceBuffer; + uint32_t mTimedSilenceBufferSize; + mutable Mutex mTimedBufferQueueLock; + bool mTimedAudioOutputOnTime; + CCHelper mCCHelper; + + Mutex mMediaTimeTransformLock; + LinearTransform mMediaTimeTransform; + bool mMediaTimeTransformValid; + TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget; +}; + + +// playback track, used by DuplicatingThread +class OutputTrack : public Track { +public: + + class Buffer : public AudioBufferProvider::Buffer { + public: + int16_t *mBuffer; + }; + + OutputTrack(PlaybackThread *thread, + DuplicatingThread *sourceThread, + uint32_t sampleRate, + audio_format_t format, + audio_channel_mask_t channelMask, + size_t frameCount); + virtual ~OutputTrack(); + + virtual status_t start(AudioSystem::sync_event_t event = + AudioSystem::SYNC_EVENT_NONE, + int triggerSession = 0); + virtual void stop(); + bool write(int16_t* data, uint32_t frames); + bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } + bool isActive() const { return mActive; } + const wp<ThreadBase>& thread() const { return mThread; } + +private: + + enum { + NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value + }; + + status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, + uint32_t waitTimeMs); + void clearBufferQueue(); + + // Maximum number of pending buffers allocated by OutputTrack::write() + static const uint8_t kMaxOverFlowBuffers = 10; + + Vector < Buffer* > mBufferQueue; + AudioBufferProvider::Buffer mOutBuffer; + bool mActive; + DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() + void* mBuffers; // starting address of buffers in plain memory +}; // end of OutputTrack diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h new file mode 100644 index 0000000..fe681d7 --- /dev/null +++ b/services/audioflinger/RecordTracks.h @@ -0,0 +1,62 @@ +/* +** +** Copyright 2012, 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. +*/ + +#ifndef INCLUDING_FROM_AUDIOFLINGER_H + #error This header file should only be included from AudioFlinger.h +#endif + +// record track +class RecordTrack : public TrackBase { +public: + RecordTrack(RecordThread *thread, + const sp<Client>& client, + uint32_t sampleRate, + audio_format_t format, + audio_channel_mask_t channelMask, + size_t frameCount, + int sessionId); + virtual ~RecordTrack(); + + virtual status_t start(AudioSystem::sync_event_t event, int triggerSession); + virtual void stop(); + + void destroy(); + + // clear the buffer overflow flag + void clearOverflow() { mOverflow = false; } + // set the buffer overflow flag and return previous value + bool setOverflow() { bool tmp = mOverflow; mOverflow = true; + return tmp; } + + static void appendDumpHeader(String8& result); + void dump(char* buffer, size_t size); + + virtual bool isOut() const; + +private: + friend class AudioFlinger; // for mState + + RecordTrack(const RecordTrack&); + RecordTrack& operator = (const RecordTrack&); + + // AudioBufferProvider interface + virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, + int64_t pts = kInvalidPTS); + // releaseBuffer() not overridden + + bool mOverflow; // overflow on most recent attempt to fill client buffer +}; diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp new file mode 100644 index 0000000..1ceb850 --- /dev/null +++ b/services/audioflinger/Threads.cpp @@ -0,0 +1,4426 @@ +/* +** +** Copyright 2012, 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 <fcntl.h> +#include <sys/stat.h> +#include <cutils/properties.h> +#include <cutils/compiler.h> +#include <utils/Log.h> + +#include <private/media/AudioTrackShared.h> +#include <hardware/audio.h> +#include <audio_effects/effect_ns.h> +#include <audio_effects/effect_aec.h> +#include <audio_utils/primitives.h> + +// NBAIO implementations +#include <media/nbaio/AudioStreamOutSink.h> +#include <media/nbaio/MonoPipe.h> +#include <media/nbaio/MonoPipeReader.h> +#include <media/nbaio/Pipe.h> +#include <media/nbaio/PipeReader.h> +#include <media/nbaio/SourceAudioBufferProvider.h> + +#include <powermanager/PowerManager.h> + +#include <common_time/cc_helper.h> +#include <common_time/local_clock.h> + +#include "AudioFlinger.h" +#include "AudioMixer.h" +#include "FastMixer.h" +#include "ServiceUtilities.h" +#include "SchedulingPolicyService.h" + +#undef ADD_BATTERY_DATA + +#ifdef ADD_BATTERY_DATA +#include <media/IMediaPlayerService.h> +#include <media/IMediaDeathNotifier.h> +#endif + +// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds +#ifdef DEBUG_CPU_USAGE +#include <cpustats/CentralTendencyStatistics.h> +#include <cpustats/ThreadCpuUsage.h> +#endif + +// ---------------------------------------------------------------------------- + +// Note: the following macro is used for extremely verbose logging message. In +// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to +// 0; but one side effect of this is to turn all LOGV's as well. Some messages +// are so verbose that we want to suppress them even when we have ALOG_ASSERT +// turned on. Do not uncomment the #def below unless you really know what you +// are doing and want to see all of the extremely verbose messages. +//#define VERY_VERY_VERBOSE_LOGGING +#ifdef VERY_VERY_VERBOSE_LOGGING +#define ALOGVV ALOGV +#else +#define ALOGVV(a...) do { } while(0) +#endif + +namespace android { + +// 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; + +// don't warn about blocked writes or record buffer overflows more often than this +static const nsecs_t kWarningThrottleNs = seconds(5); + +// RecordThread loop sleep time upon application overrun or audio HAL read error +static const int kRecordThreadSleepUs = 5000; + +// maximum time to wait for setParameters to complete +static const nsecs_t kSetParametersTimeoutNs = seconds(2); + +// minimum sleep time for the mixer thread loop when tracks are active but in underrun +static const uint32_t kMinThreadSleepTimeUs = 5000; +// maximum divider applied to the active sleep time in the mixer thread loop +static const uint32_t kMaxThreadSleepTimeShift = 2; + +// minimum normal mix buffer size, expressed in milliseconds rather than frames +static const uint32_t kMinNormalMixBufferSizeMs = 20; +// maximum normal mix buffer size +static const uint32_t kMaxNormalMixBufferSizeMs = 24; + +// Whether to use fast mixer +static const enum { + FastMixer_Never, // never initialize or use: for debugging only + FastMixer_Always, // always initialize and use, even if not needed: for debugging only + // normal mixer multiplier is 1 + FastMixer_Static, // initialize if needed, then use all the time if initialized, + // multiplier is calculated based on min & max normal mixer buffer size + FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, + // multiplier is calculated based on min & max normal mixer buffer size + // FIXME for FastMixer_Dynamic: + // Supporting this option will require fixing HALs that can't handle large writes. + // For example, one HAL implementation returns an error from a large write, + // and another HAL implementation corrupts memory, possibly in the sample rate converter. + // We could either fix the HAL implementations, or provide a wrapper that breaks + // up large writes into smaller ones, and the wrapper would need to deal with scheduler. +} kUseFastMixer = FastMixer_Static; + +// Priorities for requestPriority +static const int kPriorityAudioApp = 2; +static const int kPriorityFastMixer = 3; + +// IAudioFlinger::createTrack() reports back to client the total size of shared memory area +// for the track. The client then sub-divides this into smaller buffers for its use. +// Currently the client uses double-buffering by default, but doesn't tell us about that. +// So for now we just assume that client is double-buffered. +// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or +// N-buffering, so AudioFlinger could allocate the right amount of memory. +// See the client's minBufCount and mNotificationFramesAct calculations for details. +static const int kFastTrackMultiplier = 2; + +// ---------------------------------------------------------------------------- + +#ifdef ADD_BATTERY_DATA +// To collect the amplifier usage +static void addBatteryData(uint32_t params) { + sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); + if (service == NULL) { + // it already logged + return; + } + + service->addBatteryData(params); +} +#endif + + +// ---------------------------------------------------------------------------- +// CPU Stats +// ---------------------------------------------------------------------------- + +class CpuStats { +public: + CpuStats(); + void sample(const String8 &title); +#ifdef DEBUG_CPU_USAGE +private: + ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns + CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns + + CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles + + int mCpuNum; // thread's current CPU number + int mCpukHz; // frequency of thread's current CPU in kHz +#endif +}; + +CpuStats::CpuStats() +#ifdef DEBUG_CPU_USAGE + : mCpuNum(-1), mCpukHz(-1) +#endif +{ +} + +void CpuStats::sample(const String8 &title) { +#ifdef DEBUG_CPU_USAGE + // get current thread's delta CPU time in wall clock ns + double wcNs; + bool valid = mCpuUsage.sampleAndEnable(wcNs); + + // record sample for wall clock statistics + if (valid) { + mWcStats.sample(wcNs); + } + + // get the current CPU number + int cpuNum = sched_getcpu(); + + // get the current CPU frequency in kHz + int cpukHz = mCpuUsage.getCpukHz(cpuNum); + + // check if either CPU number or frequency changed + if (cpuNum != mCpuNum || cpukHz != mCpukHz) { + mCpuNum = cpuNum; + mCpukHz = cpukHz; + // ignore sample for purposes of cycles + valid = false; + } + + // if no change in CPU number or frequency, then record sample for cycle statistics + if (valid && mCpukHz > 0) { + double cycles = wcNs * cpukHz * 0.000001; + mHzStats.sample(cycles); + } + + unsigned n = mWcStats.n(); + // mCpuUsage.elapsed() is expensive, so don't call it every loop + if ((n & 127) == 1) { + long long elapsed = mCpuUsage.elapsed(); + if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { + double perLoop = elapsed / (double) n; + double perLoop100 = perLoop * 0.01; + double perLoop1k = perLoop * 0.001; + double mean = mWcStats.mean(); + double stddev = mWcStats.stddev(); + double minimum = mWcStats.minimum(); + double maximum = mWcStats.maximum(); + double meanCycles = mHzStats.mean(); + double stddevCycles = mHzStats.stddev(); + double minCycles = mHzStats.minimum(); + double maxCycles = mHzStats.maximum(); + mCpuUsage.resetElapsed(); + mWcStats.reset(); + mHzStats.reset(); + ALOGD("CPU usage for %s over past %.1f secs\n" + " (%u mixer loops at %.1f mean ms per loop):\n" + " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" + " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" + " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", + title.string(), + elapsed * .000000001, n, perLoop * .000001, + mean * .001, + stddev * .001, + minimum * .001, + maximum * .001, + mean / perLoop100, + stddev / perLoop100, + minimum / perLoop100, + maximum / perLoop100, + meanCycles / perLoop1k, + stddevCycles / perLoop1k, + minCycles / perLoop1k, + maxCycles / perLoop1k); + + } + } +#endif +}; + +// ---------------------------------------------------------------------------- +// ThreadBase +// ---------------------------------------------------------------------------- + +AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, + audio_devices_t outDevice, audio_devices_t inDevice, type_t type) + : Thread(false /*canCallJava*/), + mType(type), + mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), + // mChannelMask + mChannelCount(0), + mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), + mParamStatus(NO_ERROR), + mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), + mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), + // mName will be set by concrete (non-virtual) subclass + mDeathRecipient(new PMDeathRecipient(this)) +{ +} + +AudioFlinger::ThreadBase::~ThreadBase() +{ + mParamCond.broadcast(); + // do not lock the mutex in destructor + releaseWakeLock_l(); + if (mPowerManager != 0) { + sp<IBinder> binder = mPowerManager->asBinder(); + binder->unlinkToDeath(mDeathRecipient); + } +} + +void AudioFlinger::ThreadBase::exit() +{ + ALOGV("ThreadBase::exit"); + // do any cleanup required for exit to succeed + preExit(); + { + // This lock prevents the following race in thread (uniprocessor for illustration): + // if (!exitPending()) { + // // context switch from here to exit() + // // exit() calls requestExit(), what exitPending() observes + // // exit() calls signal(), which is dropped since no waiters + // // context switch back from exit() to here + // mWaitWorkCV.wait(...); + // // now thread is hung + // } + AutoMutex lock(mLock); + requestExit(); + mWaitWorkCV.broadcast(); + } + // When Thread::requestExitAndWait is made virtual and this method is renamed to + // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" + requestExitAndWait(); +} + +status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) +{ + status_t status; + + ALOGV("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, kSetParametersTimeoutNs) == NO_ERROR) { + status = mParamStatus; + mWaitWorkCV.signal(); + } else { + status = TIMED_OUT; + } + return status; +} + +void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) +{ + Mutex::Autolock _l(mLock); + sendIoConfigEvent_l(event, param); +} + +// sendIoConfigEvent_l() must be called with ThreadBase::mLock held +void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) +{ + IoConfigEvent *ioEvent = new IoConfigEvent(event, param); + mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); + ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, + param); + mWaitWorkCV.signal(); +} + +// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held +void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) +{ + PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); + mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); + ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", + mConfigEvents.size(), pid, tid, prio); + mWaitWorkCV.signal(); +} + +void AudioFlinger::ThreadBase::processConfigEvents() +{ + mLock.lock(); + while (!mConfigEvents.isEmpty()) { + ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); + ConfigEvent *event = mConfigEvents[0]; + mConfigEvents.removeAt(0); + // release mLock before locking AudioFlinger mLock: lock order is always + // AudioFlinger then ThreadBase to avoid cross deadlock + mLock.unlock(); + switch(event->type()) { + case CFG_EVENT_PRIO: { + PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); + int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); + if (err != 0) { + ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " + "error %d", + prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); + } + } break; + case CFG_EVENT_IO: { + IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); + mAudioFlinger->mLock.lock(); + audioConfigChanged_l(ioEvent->event(), ioEvent->param()); + mAudioFlinger->mLock.unlock(); + } break; + default: + ALOGE("processConfigEvents() unknown event type %d", event->type()); + break; + } + delete event; + mLock.lock(); + } + mLock.unlock(); +} + +void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + bool locked = AudioFlinger::dumpTryLock(mLock); + if (!locked) { + snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); + write(fd, buffer, strlen(buffer)); + } + + snprintf(buffer, SIZE, "io handle: %d\n", mId); + result.append(buffer); + snprintf(buffer, SIZE, "TID: %d\n", getTid()); + result.append(buffer); + snprintf(buffer, SIZE, "standby: %d\n", mStandby); + result.append(buffer); + snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); + result.append(buffer); + snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); + result.append(buffer); + snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); + result.append(buffer); + snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); + result.append(buffer); + snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); + result.append(buffer); + snprintf(buffer, SIZE, "Format: %d\n", mFormat); + result.append(buffer); + snprintf(buffer, SIZE, "Frame size: %u\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); + for (size_t i = 0; i < mConfigEvents.size(); i++) { + mConfigEvents[i]->dump(buffer, SIZE); + result.append(buffer); + } + result.append("\n"); + + write(fd, result.string(), result.size()); + + if (locked) { + mLock.unlock(); + } +} + +void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); + write(fd, buffer, strlen(buffer)); + + for (size_t i = 0; i < mEffectChains.size(); ++i) { + sp<EffectChain> chain = mEffectChains[i]; + if (chain != 0) { + chain->dump(fd, args); + } + } +} + +void AudioFlinger::ThreadBase::acquireWakeLock() +{ + Mutex::Autolock _l(mLock); + acquireWakeLock_l(); +} + +void AudioFlinger::ThreadBase::acquireWakeLock_l() +{ + if (mPowerManager == 0) { + // use checkService() to avoid blocking if power service is not up yet + sp<IBinder> binder = + defaultServiceManager()->checkService(String16("power")); + if (binder == 0) { + ALOGW("Thread %s cannot connect to the power manager service", mName); + } else { + mPowerManager = interface_cast<IPowerManager>(binder); + binder->linkToDeath(mDeathRecipient); + } + } + if (mPowerManager != 0) { + sp<IBinder> binder = new BBinder(); + status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, + binder, + String16(mName)); + if (status == NO_ERROR) { + mWakeLockToken = binder; + } + ALOGV("acquireWakeLock_l() %s status %d", mName, status); + } +} + +void AudioFlinger::ThreadBase::releaseWakeLock() +{ + Mutex::Autolock _l(mLock); + releaseWakeLock_l(); +} + +void AudioFlinger::ThreadBase::releaseWakeLock_l() +{ + if (mWakeLockToken != 0) { + ALOGV("releaseWakeLock_l() %s", mName); + if (mPowerManager != 0) { + mPowerManager->releaseWakeLock(mWakeLockToken, 0); + } + mWakeLockToken.clear(); + } +} + +void AudioFlinger::ThreadBase::clearPowerManager() +{ + Mutex::Autolock _l(mLock); + releaseWakeLock_l(); + mPowerManager.clear(); +} + +void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) +{ + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + thread->clearPowerManager(); + } + ALOGW("power manager service died !!!"); +} + +void AudioFlinger::ThreadBase::setEffectSuspended( + const effect_uuid_t *type, bool suspend, int sessionId) +{ + Mutex::Autolock _l(mLock); + setEffectSuspended_l(type, suspend, sessionId); +} + +void AudioFlinger::ThreadBase::setEffectSuspended_l( + const effect_uuid_t *type, bool suspend, int sessionId) +{ + sp<EffectChain> chain = getEffectChain_l(sessionId); + if (chain != 0) { + if (type != NULL) { + chain->setEffectSuspended_l(type, suspend); + } else { + chain->setEffectSuspendedAll_l(suspend); + } + } + + updateSuspendedSessions_l(type, suspend, sessionId); +} + +void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) +{ + ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); + if (index < 0) { + return; + } + + const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = + mSuspendedSessions.valueAt(index); + + for (size_t i = 0; i < sessionEffects.size(); i++) { + sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); + for (int j = 0; j < desc->mRefCount; j++) { + if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { + chain->setEffectSuspendedAll_l(true); + } else { + ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", + desc->mType.timeLow); + chain->setEffectSuspended_l(&desc->mType, true); + } + } + } +} + +void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, + bool suspend, + int sessionId) +{ + ssize_t index = mSuspendedSessions.indexOfKey(sessionId); + + KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; + + if (suspend) { + if (index >= 0) { + sessionEffects = mSuspendedSessions.valueAt(index); + } else { + mSuspendedSessions.add(sessionId, sessionEffects); + } + } else { + if (index < 0) { + return; + } + sessionEffects = mSuspendedSessions.valueAt(index); + } + + + int key = EffectChain::kKeyForSuspendAll; + if (type != NULL) { + key = type->timeLow; + } + index = sessionEffects.indexOfKey(key); + + sp<SuspendedSessionDesc> desc; + if (suspend) { + if (index >= 0) { + desc = sessionEffects.valueAt(index); + } else { + desc = new SuspendedSessionDesc(); + if (type != NULL) { + desc->mType = *type; + } + sessionEffects.add(key, desc); + ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); + } + desc->mRefCount++; + } else { + if (index < 0) { + return; + } + desc = sessionEffects.valueAt(index); + if (--desc->mRefCount == 0) { + ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); + sessionEffects.removeItemsAt(index); + if (sessionEffects.isEmpty()) { + ALOGV("updateSuspendedSessions_l() restore removing session %d", + sessionId); + mSuspendedSessions.removeItem(sessionId); + } + } + } + if (!sessionEffects.isEmpty()) { + mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); + } +} + +void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, + bool enabled, + int sessionId) +{ + Mutex::Autolock _l(mLock); + checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); +} + +void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, + bool enabled, + int sessionId) +{ + if (mType != RECORD) { + // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on + // another session. This gives the priority to well behaved effect control panels + // and applications not using global effects. + // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect + // global effects + if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { + setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); + } + } + + sp<EffectChain> chain = getEffectChain_l(sessionId); + if (chain != 0) { + chain->checkSuspendOnEffectEnabled(effect, enabled); + } +} + +// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held +sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( + const sp<AudioFlinger::Client>& client, + const sp<IEffectClient>& effectClient, + int32_t priority, + int sessionId, + effect_descriptor_t *desc, + int *enabled, + status_t *status + ) +{ + sp<EffectModule> effect; + sp<EffectHandle> handle; + status_t lStatus; + sp<EffectChain> chain; + bool chainCreated = false; + bool effectCreated = false; + bool effectRegistered = false; + + lStatus = initCheck(); + if (lStatus != NO_ERROR) { + ALOGW("createEffect_l() Audio driver not initialized."); + goto Exit; + } + + // Do not allow effects with session ID 0 on direct output or duplicating threads + // TODO: add rule for hw accelerated effects on direct outputs with non PCM format + if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { + ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", + desc->name, sessionId); + lStatus = BAD_VALUE; + goto Exit; + } + // Only Pre processor effects are allowed on input threads and only on input threads + if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { + ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", + desc->name, desc->flags, mType); + lStatus = BAD_VALUE; + goto Exit; + } + + ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); + + { // scope for mLock + Mutex::Autolock _l(mLock); + + // check for existing effect chain with the requested audio session + chain = getEffectChain_l(sessionId); + if (chain == 0) { + // create a new chain for this session + ALOGV("createEffect_l() new effect chain for session %d", sessionId); + chain = new EffectChain(this, sessionId); + addEffectChain_l(chain); + chain->setStrategy(getStrategyForSession_l(sessionId)); + chainCreated = true; + } else { + effect = chain->getEffectFromDesc_l(desc); + } + + ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); + + if (effect == 0) { + int id = mAudioFlinger->nextUniqueId(); + // Check CPU and memory usage + lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); + if (lStatus != NO_ERROR) { + goto Exit; + } + effectRegistered = true; + // create a new effect module if none present in the chain + effect = new EffectModule(this, chain, desc, id, sessionId); + lStatus = effect->status(); + if (lStatus != NO_ERROR) { + goto Exit; + } + lStatus = chain->addEffect_l(effect); + if (lStatus != NO_ERROR) { + goto Exit; + } + effectCreated = true; + + effect->setDevice(mOutDevice); + effect->setDevice(mInDevice); + effect->setMode(mAudioFlinger->getMode()); + effect->setAudioSource(mAudioSource); + } + // create effect handle and connect it to effect module + handle = new EffectHandle(effect, client, effectClient, priority); + lStatus = effect->addHandle(handle.get()); + if (enabled != NULL) { + *enabled = (int)effect->isEnabled(); + } + } + +Exit: + if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { + Mutex::Autolock _l(mLock); + if (effectCreated) { + chain->removeEffect_l(effect); + } + if (effectRegistered) { + AudioSystem::unregisterEffect(effect->id()); + } + if (chainCreated) { + removeEffectChain_l(chain); + } + handle.clear(); + } + + if (status != NULL) { + *status = lStatus; + } + return handle; +} + +sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) +{ + Mutex::Autolock _l(mLock); + return getEffect_l(sessionId, effectId); +} + +sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) +{ + sp<EffectChain> chain = getEffectChain_l(sessionId); + return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; +} + +// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and +// PlaybackThread::mLock held +status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) +{ + // check for existing effect chain with the requested audio session + int sessionId = effect->sessionId(); + sp<EffectChain> chain = getEffectChain_l(sessionId); + bool chainCreated = false; + + if (chain == 0) { + // create a new chain for this session + ALOGV("addEffect_l() new effect chain for session %d", sessionId); + chain = new EffectChain(this, sessionId); + addEffectChain_l(chain); + chain->setStrategy(getStrategyForSession_l(sessionId)); + chainCreated = true; + } + ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); + + if (chain->getEffectFromId_l(effect->id()) != 0) { + ALOGW("addEffect_l() %p effect %s already present in chain %p", + this, effect->desc().name, chain.get()); + return BAD_VALUE; + } + + status_t status = chain->addEffect_l(effect); + if (status != NO_ERROR) { + if (chainCreated) { + removeEffectChain_l(chain); + } + return status; + } + + effect->setDevice(mOutDevice); + effect->setDevice(mInDevice); + effect->setMode(mAudioFlinger->getMode()); + effect->setAudioSource(mAudioSource); + return NO_ERROR; +} + +void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { + + ALOGV("removeEffect_l() %p effect %p", this, effect.get()); + effect_descriptor_t desc = effect->desc(); + if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { + detachAuxEffect_l(effect->id()); + } + + sp<EffectChain> chain = effect->chain().promote(); + if (chain != 0) { + // remove effect chain if removing last effect + if (chain->removeEffect_l(effect) == 0) { + removeEffectChain_l(chain); + } + } else { + ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); + } +} + +void AudioFlinger::ThreadBase::lockEffectChains_l( + Vector< sp<AudioFlinger::EffectChain> >& effectChains) +{ + effectChains = mEffectChains; + for (size_t i = 0; i < mEffectChains.size(); i++) { + mEffectChains[i]->lock(); + } +} + +void AudioFlinger::ThreadBase::unlockEffectChains( + const Vector< sp<AudioFlinger::EffectChain> >& effectChains) +{ + for (size_t i = 0; i < effectChains.size(); i++) { + effectChains[i]->unlock(); + } +} + +sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) +{ + Mutex::Autolock _l(mLock); + return getEffectChain_l(sessionId); +} + +sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const +{ + size_t size = mEffectChains.size(); + for (size_t i = 0; i < size; i++) { + if (mEffectChains[i]->sessionId() == sessionId) { + return mEffectChains[i]; + } + } + return 0; +} + +void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) +{ + Mutex::Autolock _l(mLock); + size_t size = mEffectChains.size(); + for (size_t i = 0; i < size; i++) { + mEffectChains[i]->setMode_l(mode); + } +} + +void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, + EffectHandle *handle, + bool unpinIfLast) { + + Mutex::Autolock _l(mLock); + ALOGV("disconnectEffect() %p effect %p", this, effect.get()); + // delete the effect module if removing last handle on it + if (effect->removeHandle(handle) == 0) { + if (!effect->isPinned() || unpinIfLast) { + removeEffect_l(effect); + AudioSystem::unregisterEffect(effect->id()); + } + } +} + +// ---------------------------------------------------------------------------- +// Playback +// ---------------------------------------------------------------------------- + +AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, + AudioStreamOut* output, + audio_io_handle_t id, + audio_devices_t device, + type_t type) + : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), + mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), + // mStreamTypes[] initialized in constructor body + mOutput(output), + mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), + mMixerStatus(MIXER_IDLE), + mMixerStatusIgnoringFastTracks(MIXER_IDLE), + standbyDelay(AudioFlinger::mStandbyTimeInNsecs), + mScreenState(AudioFlinger::mScreenState), + // index 0 is reserved for normal mixer's submix + mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) +{ + snprintf(mName, kNameLength, "AudioOut_%X", id); + + // Assumes constructor is called by AudioFlinger with it's mLock held, but + // it would be safer to explicitly pass initial masterVolume/masterMute as + // parameter. + // + // If the HAL we are using has support for master volume or master mute, + // then do not attenuate or mute during mixing (just leave the volume at 1.0 + // and the mute set to false). + mMasterVolume = audioFlinger->masterVolume_l(); + mMasterMute = audioFlinger->masterMute_l(); + if (mOutput && mOutput->audioHwDev) { + if (mOutput->audioHwDev->canSetMasterVolume()) { + mMasterVolume = 1.0; + } + + if (mOutput->audioHwDev->canSetMasterMute()) { + mMasterMute = false; + } + } + + readOutputParameters(); + + // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor + // There is no AUDIO_STREAM_MIN, and ++ operator does not compile + for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; + stream = (audio_stream_type_t) (stream + 1)) { + mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); + mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); + } + // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, + // because mAudioFlinger doesn't have one to copy from +} + +AudioFlinger::PlaybackThread::~PlaybackThread() +{ + delete [] mMixBuffer; +} + +void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) +{ + dumpInternals(fd, args); + dumpTracks(fd, args); + dumpEffectChains(fd, args); +} + +void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + result.appendFormat("Output thread %p stream volumes in dB:\n ", this); + for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { + const stream_type_t *st = &mStreamTypes[i]; + if (i > 0) { + result.appendFormat(", "); + } + result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); + if (st->mute) { + result.append("M"); + } + } + result.append("\n"); + write(fd, result.string(), result.length()); + result.clear(); + + snprintf(buffer, SIZE, "Output thread %p tracks\n", this); + result.append(buffer); + Track::appendDumpHeader(result); + 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); + Track::appendDumpHeader(result); + for (size_t i = 0; i < mActiveTracks.size(); ++i) { + sp<Track> track = mActiveTracks[i].promote(); + if (track != 0) { + track->dump(buffer, SIZE); + result.append(buffer); + } + } + write(fd, result.string(), result.size()); + + // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. + FastTrackUnderruns underruns = getFastTrackUnderruns(0); + fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", + underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); +} + +void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); + result.append(buffer); + snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", + ns2ms(systemTime() - mLastWriteTime)); + result.append(buffer); + snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); + result.append(buffer); + snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); + result.append(buffer); + snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); + result.append(buffer); + snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); + result.append(buffer); + snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); + result.append(buffer); + write(fd, result.string(), result.size()); + fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); + + dumpBase(fd, args); +} + +// Thread virtuals +status_t AudioFlinger::PlaybackThread::readyToRun() +{ + status_t status = initCheck(); + if (status == NO_ERROR) { + ALOGI("AudioFlinger's thread %p ready to run", this); + } else { + ALOGE("No working audio driver found."); + } + return status; +} + +void AudioFlinger::PlaybackThread::onFirstRef() +{ + run(mName, ANDROID_PRIORITY_URGENT_AUDIO); +} + +// ThreadBase virtuals +void AudioFlinger::PlaybackThread::preExit() +{ + ALOGV(" preExit()"); + // FIXME this is using hard-coded strings but in the future, this functionality will be + // converted to use audio HAL extensions required to support tunneling + mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); +} + +// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held +sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( + const sp<AudioFlinger::Client>& client, + audio_stream_type_t streamType, + uint32_t sampleRate, + audio_format_t format, + audio_channel_mask_t channelMask, + size_t frameCount, + const sp<IMemory>& sharedBuffer, + int sessionId, + IAudioFlinger::track_flags_t *flags, + pid_t tid, + status_t *status) +{ + sp<Track> track; + status_t lStatus; + + bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; + + // client expresses a preference for FAST, but we get the final say + if (*flags & IAudioFlinger::TRACK_FAST) { + if ( + // not timed + (!isTimed) && + // either of these use cases: + ( + // use case 1: shared buffer with any frame count + ( + (sharedBuffer != 0) + ) || + // use case 2: callback handler and frame count is default or at least as large as HAL + ( + (tid != -1) && + ((frameCount == 0) || + (frameCount >= (mFrameCount * kFastTrackMultiplier))) + ) + ) && + // PCM data + audio_is_linear_pcm(format) && + // mono or stereo + ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || + (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && +#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE + // hardware sample rate + (sampleRate == mSampleRate) && +#endif + // normal mixer has an associated fast mixer + hasFastMixer() && + // there are sufficient fast track slots available + (mFastTrackAvailMask != 0) + // FIXME test that MixerThread for this fast track has a capable output HAL + // FIXME add a permission test also? + ) { + // if frameCount not specified, then it defaults to fast mixer (HAL) frame count + if (frameCount == 0) { + frameCount = mFrameCount * kFastTrackMultiplier; + } + ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", + frameCount, mFrameCount); + } else { + ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " + "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " + "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", + isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, + audio_is_linear_pcm(format), + channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); + *flags &= ~IAudioFlinger::TRACK_FAST; + // For compatibility with AudioTrack calculation, buffer depth is forced + // to be at least 2 x the normal mixer frame count and cover audio hardware latency. + // This is probably too conservative, but legacy application code may depend on it. + // If you change this calculation, also review the start threshold which is related. + uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); + uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); + if (minBufCount < 2) { + minBufCount = 2; + } + size_t minFrameCount = mNormalFrameCount * minBufCount; + if (frameCount < minFrameCount) { + frameCount = minFrameCount; + } + } + } + + if (mType == DIRECT) { + if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { + if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { + ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " + "for output %p with format %d", + sampleRate, format, channelMask, mOutput, mFormat); + lStatus = BAD_VALUE; + goto Exit; + } + } + } else { + // Resampler implementation limits input sampling rate to 2 x output sampling rate. + if (sampleRate > mSampleRate*2) { + ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); + lStatus = BAD_VALUE; + goto Exit; + } + } + + lStatus = initCheck(); + if (lStatus != NO_ERROR) { + ALOGE("Audio driver not initialized."); + goto Exit; + } + + { // scope for mLock + Mutex::Autolock _l(mLock); + + // all tracks in same audio session must share the same routing strategy otherwise + // conflicts will happen when tracks are moved from one output to another by audio policy + // manager + uint32_t strategy = AudioSystem::getStrategyForStream(streamType); + for (size_t i = 0; i < mTracks.size(); ++i) { + sp<Track> t = mTracks[i]; + if (t != 0 && !t->isOutputTrack()) { + uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); + if (sessionId == t->sessionId() && strategy != actual) { + ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", + strategy, actual); + lStatus = BAD_VALUE; + goto Exit; + } + } + } + + if (!isTimed) { + track = new Track(this, client, streamType, sampleRate, format, + channelMask, frameCount, sharedBuffer, sessionId, *flags); + } else { + track = TimedTrack::create(this, client, streamType, sampleRate, format, + channelMask, frameCount, sharedBuffer, sessionId); + } + if (track == 0 || track->getCblk() == NULL || track->name() < 0) { + lStatus = NO_MEMORY; + goto Exit; + } + mTracks.add(track); + + sp<EffectChain> chain = getEffectChain_l(sessionId); + if (chain != 0) { + ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); + track->setMainBuffer(chain->inBuffer()); + chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); + chain->incTrackCnt(); + } + + if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { + pid_t callingPid = IPCThreadState::self()->getCallingPid(); + // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, + // so ask activity manager to do this on our behalf + sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); + } + } + + lStatus = NO_ERROR; + +Exit: + if (status) { + *status = lStatus; + } + return track; +} + +uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const +{ + return latency; +} + +uint32_t AudioFlinger::PlaybackThread::latency() const +{ + Mutex::Autolock _l(mLock); + return latency_l(); +} +uint32_t AudioFlinger::PlaybackThread::latency_l() const +{ + if (initCheck() == NO_ERROR) { + return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); + } else { + return 0; + } +} + +void AudioFlinger::PlaybackThread::setMasterVolume(float value) +{ + Mutex::Autolock _l(mLock); + // Don't apply master volume in SW if our HAL can do it for us. + if (mOutput && mOutput->audioHwDev && + mOutput->audioHwDev->canSetMasterVolume()) { + mMasterVolume = 1.0; + } else { + mMasterVolume = value; + } +} + +void AudioFlinger::PlaybackThread::setMasterMute(bool muted) +{ + Mutex::Autolock _l(mLock); + // Don't apply master mute in SW if our HAL can do it for us. + if (mOutput && mOutput->audioHwDev && + mOutput->audioHwDev->canSetMasterMute()) { + mMasterMute = false; + } else { + mMasterMute = muted; + } +} + +void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) +{ + Mutex::Autolock _l(mLock); + mStreamTypes[stream].volume = value; +} + +void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) +{ + Mutex::Autolock _l(mLock); + mStreamTypes[stream].mute = muted; +} + +float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const +{ + Mutex::Autolock _l(mLock); + return mStreamTypes[stream].volume; +} + +// 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; + track->mPresentationCompleteFrames = 0; + mActiveTracks.add(track); + if (track->mainBuffer() != mMixBuffer) { + sp<EffectChain> chain = getEffectChain_l(track->sessionId()); + if (chain != 0) { + ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), + track->sessionId()); + chain->incActiveTrackCnt(); + } + } + + status = NO_ERROR; + } + + ALOGV("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; + // active tracks are removed by threadLoop() + if (mActiveTracks.indexOf(track) < 0) { + removeTrack_l(track); + } +} + +void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) +{ + track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); + mTracks.remove(track); + deleteTrackName_l(track->name()); + // redundant as track is about to be destroyed, for dumpsys only + track->mName = -1; + if (track->isFastTrack()) { + int index = track->mFastIndex; + ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); + ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); + mFastTrackAvailMask |= 1 << index; + // redundant as track is about to be destroyed, for dumpsys only + track->mFastIndex = -1; + } + sp<EffectChain> chain = getEffectChain_l(track->sessionId()); + if (chain != 0) { + chain->decTrackCnt(); + } +} + +String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) +{ + String8 out_s8 = String8(""); + char *s; + + Mutex::Autolock _l(mLock); + if (initCheck() != NO_ERROR) { + return out_s8; + } + + s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); + out_s8 = String8(s); + free(s); + return out_s8; +} + +// audioConfigChanged_l() must be called with AudioFlinger::mLock held +void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { + AudioSystem::OutputDescriptor desc; + void *param2 = NULL; + + ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, + param); + + switch (event) { + case AudioSystem::OUTPUT_OPENED: + case AudioSystem::OUTPUT_CONFIG_CHANGED: + desc.channels = mChannelMask; + desc.samplingRate = mSampleRate; + desc.format = mFormat; + desc.frameCount = mNormalFrameCount; // FIXME see + // AudioFlinger::frameCount(audio_io_handle_t) + desc.latency = latency(); + param2 = &desc; + break; + + case AudioSystem::STREAM_CONFIG_CHANGED: + param2 = ¶m; + case AudioSystem::OUTPUT_CLOSED: + default: + break; + } + mAudioFlinger->audioConfigChanged_l(event, mId, param2); +} + +void AudioFlinger::PlaybackThread::readOutputParameters() +{ + mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); + mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); + mChannelCount = (uint16_t)popcount(mChannelMask); + mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); + mFrameSize = audio_stream_frame_size(&mOutput->stream->common); + mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; + if (mFrameCount & 15) { + ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", + mFrameCount); + } + + // Calculate size of normal mix buffer relative to the HAL output buffer size + double multiplier = 1.0; + if (mType == MIXER && (kUseFastMixer == FastMixer_Static || + kUseFastMixer == FastMixer_Dynamic)) { + size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; + size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; + // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer + minNormalFrameCount = (minNormalFrameCount + 15) & ~15; + maxNormalFrameCount = maxNormalFrameCount & ~15; + if (maxNormalFrameCount < minNormalFrameCount) { + maxNormalFrameCount = minNormalFrameCount; + } + multiplier = (double) minNormalFrameCount / (double) mFrameCount; + if (multiplier <= 1.0) { + multiplier = 1.0; + } else if (multiplier <= 2.0) { + if (2 * mFrameCount <= maxNormalFrameCount) { + multiplier = 2.0; + } else { + multiplier = (double) maxNormalFrameCount / (double) mFrameCount; + } + } else { + // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL + // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast + // track, but we sometimes have to do this to satisfy the maximum frame count + // constraint) + // FIXME this rounding up should not be done if no HAL SRC + uint32_t truncMult = (uint32_t) multiplier; + if ((truncMult & 1)) { + if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { + ++truncMult; + } + } + multiplier = (double) truncMult; + } + } + mNormalFrameCount = multiplier * mFrameCount; + // round up to nearest 16 frames to satisfy AudioMixer + mNormalFrameCount = (mNormalFrameCount + 15) & ~15; + ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, + mNormalFrameCount); + + delete[] mMixBuffer; + mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; + memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); + + // force reconfiguration of effect chains and engines to take new buffer size and audio + // parameters into account + // Note that mLock is not held when readOutputParameters() is called from the constructor + // but in this case nothing is done below as no audio sessions have effect yet so it doesn't + // matter. + // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains + Vector< sp<EffectChain> > effectChains = mEffectChains; + for (size_t i = 0; i < effectChains.size(); i ++) { + mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); + } +} + + +status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) +{ + if (halFrames == NULL || dspFrames == NULL) { + return BAD_VALUE; + } + Mutex::Autolock _l(mLock); + if (initCheck() != NO_ERROR) { + return INVALID_OPERATION; + } + size_t framesWritten = mBytesWritten / mFrameSize; + *halFrames = framesWritten; + + if (isSuspended()) { + // return an estimation of rendered frames when the output is suspended + size_t latencyFrames = (latency_l() * mSampleRate) / 1000; + *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; + return NO_ERROR; + } else { + return mOutput->stream->get_render_position(mOutput->stream, dspFrames); + } +} + +uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const +{ + Mutex::Autolock _l(mLock); + uint32_t result = 0; + if (getEffectChain_l(sessionId) != 0) { + result = EFFECT_SESSION; + } + + for (size_t i = 0; i < mTracks.size(); ++i) { + sp<Track> track = mTracks[i]; + if (sessionId == track->sessionId() && + !(track->mCblk->flags & CBLK_INVALID)) { + result |= TRACK_SESSION; + break; + } + } + + return result; +} + +uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) +{ + // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that + // it is moved to correct output by audio policy manager when A2DP is connected or disconnected + if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { + return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); + } + for (size_t i = 0; i < mTracks.size(); i++) { + sp<Track> track = mTracks[i]; + if (sessionId == track->sessionId() && + !(track->mCblk->flags & CBLK_INVALID)) { + return AudioSystem::getStrategyForStream(track->streamType()); + } + } + return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); +} + + +AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const +{ + Mutex::Autolock _l(mLock); + return mOutput; +} + +AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() +{ + Mutex::Autolock _l(mLock); + AudioStreamOut *output = mOutput; + mOutput = NULL; + // FIXME FastMixer might also have a raw ptr to mOutputSink; + // must push a NULL and wait for ack + mOutputSink.clear(); + mPipeSink.clear(); + mNormalSink.clear(); + return output; +} + +// this method must always be called either with ThreadBase mLock held or inside the thread loop +audio_stream_t* AudioFlinger::PlaybackThread::stream() const +{ + if (mOutput == NULL) { + return NULL; + } + return &mOutput->stream->common; +} + +uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const +{ + return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); +} + +status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) +{ + if (!isValidSyncEvent(event)) { + return BAD_VALUE; + } + + Mutex::Autolock _l(mLock); + + for (size_t i = 0; i < mTracks.size(); ++i) { + sp<Track> track = mTracks[i]; + if (event->triggerSession() == track->sessionId()) { + (void) track->setSyncEvent(event); + return NO_ERROR; + } + } + + return NAME_NOT_FOUND; +} + +bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const +{ + return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; +} + +void AudioFlinger::PlaybackThread::threadLoop_removeTracks( + const Vector< sp<Track> >& tracksToRemove) +{ + size_t count = tracksToRemove.size(); + if (CC_UNLIKELY(count)) { + for (size_t i = 0 ; i < count ; i++) { + const sp<Track>& track = tracksToRemove.itemAt(i); + if ((track->sharedBuffer() != 0) && + (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { + AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); + } + } + } + +} + +void AudioFlinger::PlaybackThread::checkSilentMode_l() +{ + if (!mMasterMute) { + char value[PROPERTY_VALUE_MAX]; + if (property_get("ro.audio.silent", value, "0") > 0) { + char *endptr; + unsigned long ul = strtoul(value, &endptr, 0); + if (*endptr == '\0' && ul != 0) { + ALOGD("Silence is golden"); + // The setprop command will not allow a property to be changed after + // the first time it is set, so we don't have to worry about un-muting. + setMasterMute_l(true); + } + } + } +} + +// shared by MIXER and DIRECT, overridden by DUPLICATING +void AudioFlinger::PlaybackThread::threadLoop_write() +{ + // FIXME rewrite to reduce number of system calls + mLastWriteTime = systemTime(); + mInWrite = true; + int bytesWritten; + + // If an NBAIO sink is present, use it to write the normal mixer's submix + if (mNormalSink != 0) { +#define mBitShift 2 // FIXME + size_t count = mixBufferSize >> mBitShift; +#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) + Tracer::traceBegin(ATRACE_TAG, "write"); +#endif + // update the setpoint when AudioFlinger::mScreenState changes + uint32_t screenState = AudioFlinger::mScreenState; + if (screenState != mScreenState) { + mScreenState = screenState; + MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); + if (pipe != NULL) { + pipe->setAvgFrames((mScreenState & 1) ? + (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); + } + } + ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); +#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) + Tracer::traceEnd(ATRACE_TAG); +#endif + if (framesWritten > 0) { + bytesWritten = framesWritten << mBitShift; + } else { + bytesWritten = framesWritten; + } + // otherwise use the HAL / AudioStreamOut directly + } else { + // Direct output thread. + bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); + } + + if (bytesWritten > 0) { + mBytesWritten += mixBufferSize; + } + mNumWrites++; + mInWrite = false; +} + +/* +The derived values that are cached: + - mixBufferSize from frame count * frame size + - activeSleepTime from activeSleepTimeUs() + - idleSleepTime from idleSleepTimeUs() + - standbyDelay from mActiveSleepTimeUs (DIRECT only) + - maxPeriod from frame count and sample rate (MIXER only) + +The parameters that affect these derived values are: + - frame count + - frame size + - sample rate + - device type: A2DP or not + - device latency + - format: PCM or not + - active sleep time + - idle sleep time +*/ + +void AudioFlinger::PlaybackThread::cacheParameters_l() +{ + mixBufferSize = mNormalFrameCount * mFrameSize; + activeSleepTime = activeSleepTimeUs(); + idleSleepTime = idleSleepTimeUs(); +} + +void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) +{ + ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", + this, streamType, mTracks.size()); + Mutex::Autolock _l(mLock); + + size_t size = mTracks.size(); + for (size_t i = 0; i < size; i++) { + sp<Track> t = mTracks[i]; + if (t->streamType() == streamType) { + android_atomic_or(CBLK_INVALID, &t->mCblk->flags); + t->mCblk->cv.signal(); + } + } +} + +status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) +{ + int session = chain->sessionId(); + int16_t *buffer = mMixBuffer; + bool ownsBuffer = false; + + ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); + if (session > 0) { + // Only one effect chain can be present in direct output thread and it uses + // the mix buffer as input + if (mType != DIRECT) { + size_t numSamples = mNormalFrameCount * mChannelCount; + buffer = new int16_t[numSamples]; + memset(buffer, 0, numSamples * sizeof(int16_t)); + ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); + ownsBuffer = true; + } + + // Attach all tracks with same session ID to this chain. + for (size_t i = 0; i < mTracks.size(); ++i) { + sp<Track> track = mTracks[i]; + if (session == track->sessionId()) { + ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), + buffer); + track->setMainBuffer(buffer); + chain->incTrackCnt(); + } + } + + // indicate all active tracks in the chain + for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { + sp<Track> track = mActiveTracks[i].promote(); + if (track == 0) { + continue; + } + if (session == track->sessionId()) { + ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); + chain->incActiveTrackCnt(); + } + } + } + + chain->setInBuffer(buffer, ownsBuffer); + chain->setOutBuffer(mMixBuffer); + // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect + // chains list in order to be processed last as it contains output stage effects + // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before + // session AUDIO_SESSION_OUTPUT_STAGE to be processed + // after track specific effects and before output stage + // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and + // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX + // Effect chain for other sessions are inserted at beginning of effect + // chains list to be processed before output mix effects. Relative order between other + // sessions is not important + size_t size = mEffectChains.size(); + size_t i = 0; + for (i = 0; i < size; i++) { + if (mEffectChains[i]->sessionId() < session) { + break; + } + } + mEffectChains.insertAt(chain, i); + checkSuspendOnAddEffectChain_l(chain); + + return NO_ERROR; +} + +size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) +{ + int session = chain->sessionId(); + + ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); + + for (size_t i = 0; i < mEffectChains.size(); i++) { + if (chain == mEffectChains[i]) { + mEffectChains.removeAt(i); + // detach all active tracks from the chain + for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { + sp<Track> track = mActiveTracks[i].promote(); + if (track == 0) { + continue; + } + if (session == track->sessionId()) { + ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", + chain.get(), session); + chain->decActiveTrackCnt(); + } + } + + // detach all tracks with same session ID from this chain + for (size_t i = 0; i < mTracks.size(); ++i) { + sp<Track> track = mTracks[i]; + if (session == track->sessionId()) { + track->setMainBuffer(mMixBuffer); + chain->decTrackCnt(); + } + } + break; + } + } + return mEffectChains.size(); +} + +status_t AudioFlinger::PlaybackThread::attachAuxEffect( + const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) +{ + Mutex::Autolock _l(mLock); + return attachAuxEffect_l(track, EffectId); +} + +status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( + const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) +{ + status_t status = NO_ERROR; + + if (EffectId == 0) { + track->setAuxBuffer(0, NULL); + } else { + // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX + sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); + if (effect != 0) { + if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { + track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); + } else { + status = INVALID_OPERATION; + } + } else { + status = BAD_VALUE; + } + } + return status; +} + +void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) +{ + for (size_t i = 0; i < mTracks.size(); ++i) { + sp<Track> track = mTracks[i]; + if (track->auxEffectId() == effectId) { + attachAuxEffect_l(track, 0); + } + } +} + +bool AudioFlinger::PlaybackThread::threadLoop() +{ + Vector< sp<Track> > tracksToRemove; + + standbyTime = systemTime(); + + // MIXER + nsecs_t lastWarning = 0; + + // DUPLICATING + // FIXME could this be made local to while loop? + writeFrames = 0; + + cacheParameters_l(); + sleepTime = idleSleepTime; + + if (mType == MIXER) { + sleepTimeShift = 0; + } + + CpuStats cpuStats; + const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); + + acquireWakeLock(); + + while (!exitPending()) + { + cpuStats.sample(myName); + + Vector< sp<EffectChain> > effectChains; + + processConfigEvents(); + + { // scope for mLock + + Mutex::Autolock _l(mLock); + + if (checkForNewParameters_l()) { + cacheParameters_l(); + } + + saveOutputTracks(); + + // put audio hardware into standby after short delay + if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || + isSuspended())) { + if (!mStandby) { + + threadLoop_standby(); + + mStandby = true; + } + + if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { + // we're about to wait, flush the binder command buffer + IPCThreadState::self()->flushCommands(); + + clearOutputTracks(); + + if (exitPending()) { + break; + } + + releaseWakeLock_l(); + // wait until we have something to do... + ALOGV("%s going to sleep", myName.string()); + mWaitWorkCV.wait(mLock); + ALOGV("%s waking up", myName.string()); + acquireWakeLock_l(); + + mMixerStatus = MIXER_IDLE; + mMixerStatusIgnoringFastTracks = MIXER_IDLE; + mBytesWritten = 0; + + checkSilentMode_l(); + + standbyTime = systemTime() + standbyDelay; + sleepTime = idleSleepTime; + if (mType == MIXER) { + sleepTimeShift = 0; + } + + continue; + } + } + + // mMixerStatusIgnoringFastTracks is also updated internally + mMixerStatus = prepareTracks_l(&tracksToRemove); + + // prevent any changes in effect chain list and in each effect chain + // during mixing and effect process as the audio buffers could be deleted + // or modified if an effect is created or deleted + lockEffectChains_l(effectChains); + } + + if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { + threadLoop_mix(); + } else { + threadLoop_sleepTime(); + } + + if (isSuspended()) { + sleepTime = suspendSleepTimeUs(); + mBytesWritten += mixBufferSize; + } + + // only process effects if we're going to write + if (sleepTime == 0) { + for (size_t i = 0; i < effectChains.size(); i ++) { + effectChains[i]->process_l(); + } + } + + // enable changes in effect chain + unlockEffectChains(effectChains); + + // sleepTime == 0 means we must write to audio hardware + if (sleepTime == 0) { + + threadLoop_write(); + +if (mType == MIXER) { + // write blocked detection + nsecs_t now = systemTime(); + nsecs_t delta = now - mLastWriteTime; + if (!mStandby && delta > maxPeriod) { + mNumDelayedWrites++; + if ((now - lastWarning) > kWarningThrottleNs) { +#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) + ScopedTrace st(ATRACE_TAG, "underrun"); +#endif + ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", + ns2ms(delta), mNumDelayedWrites, this); + lastWarning = now; + } + } +} + + mStandby = false; + } else { + usleep(sleepTime); + } + + // Finally let go of removed track(s), without the lock held + // since we can't guarantee the destructors won't acquire that + // same lock. This will also mutate and push a new fast mixer state. + threadLoop_removeTracks(tracksToRemove); + tracksToRemove.clear(); + + // FIXME I don't understand the need for this here; + // it was in the original code but maybe the + // assignment in saveOutputTracks() makes this unnecessary? + clearOutputTracks(); + + // Effect chains will be actually deleted here if they were removed from + // mEffectChains list during mixing or effects processing + effectChains.clear(); + + // FIXME Note that the above .clear() is no longer necessary since effectChains + // is now local to this block, but will keep it for now (at least until merge done). + } + + // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... + if (mType == MIXER || mType == DIRECT) { + // put output stream into standby mode + if (!mStandby) { + mOutput->stream->common.standby(&mOutput->stream->common); + } + } + + releaseWakeLock(); + + ALOGV("Thread %p type %d exiting", this, mType); + return false; +} + + +// ---------------------------------------------------------------------------- + +AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, + audio_io_handle_t id, audio_devices_t device, type_t type) + : PlaybackThread(audioFlinger, output, id, device, type), + // mAudioMixer below + // mFastMixer below + mFastMixerFutex(0) + // mOutputSink below + // mPipeSink below + // mNormalSink below +{ + ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); + ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " + "mFrameCount=%d, mNormalFrameCount=%d", + mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, + mNormalFrameCount); + mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); + + // FIXME - Current mixer implementation only supports stereo output + if (mChannelCount != FCC_2) { + ALOGE("Invalid audio hardware channel count %d", mChannelCount); + } + + // create an NBAIO sink for the HAL output stream, and negotiate + mOutputSink = new AudioStreamOutSink(output->stream); + size_t numCounterOffers = 0; + const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; + ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); + ALOG_ASSERT(index == 0); + + // initialize fast mixer depending on configuration + bool initFastMixer; + switch (kUseFastMixer) { + case FastMixer_Never: + initFastMixer = false; + break; + case FastMixer_Always: + initFastMixer = true; + break; + case FastMixer_Static: + case FastMixer_Dynamic: + initFastMixer = mFrameCount < mNormalFrameCount; + break; + } + if (initFastMixer) { + + // create a MonoPipe to connect our submix to FastMixer + NBAIO_Format format = mOutputSink->format(); + // This pipe depth compensates for scheduling latency of the normal mixer thread. + // When it wakes up after a maximum latency, it runs a few cycles quickly before + // finally blocking. Note the pipe implementation rounds up the request to a power of 2. + MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); + const NBAIO_Format offers[1] = {format}; + size_t numCounterOffers = 0; + ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); + ALOG_ASSERT(index == 0); + monoPipe->setAvgFrames((mScreenState & 1) ? + (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); + mPipeSink = monoPipe; + +#ifdef TEE_SINK_FRAMES + // create a Pipe to archive a copy of FastMixer's output for dumpsys + Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); + numCounterOffers = 0; + index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); + ALOG_ASSERT(index == 0); + mTeeSink = teeSink; + PipeReader *teeSource = new PipeReader(*teeSink); + numCounterOffers = 0; + index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); + ALOG_ASSERT(index == 0); + mTeeSource = teeSource; +#endif + + // create fast mixer and configure it initially with just one fast track for our submix + mFastMixer = new FastMixer(); + FastMixerStateQueue *sq = mFastMixer->sq(); +#ifdef STATE_QUEUE_DUMP + sq->setObserverDump(&mStateQueueObserverDump); + sq->setMutatorDump(&mStateQueueMutatorDump); +#endif + FastMixerState *state = sq->begin(); + FastTrack *fastTrack = &state->mFastTracks[0]; + // wrap the source side of the MonoPipe to make it an AudioBufferProvider + fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); + fastTrack->mVolumeProvider = NULL; + fastTrack->mGeneration++; + state->mFastTracksGen++; + state->mTrackMask = 1; + // fast mixer will use the HAL output sink + state->mOutputSink = mOutputSink.get(); + state->mOutputSinkGen++; + state->mFrameCount = mFrameCount; + state->mCommand = FastMixerState::COLD_IDLE; + // already done in constructor initialization list + //mFastMixerFutex = 0; + state->mColdFutexAddr = &mFastMixerFutex; + state->mColdGen++; + state->mDumpState = &mFastMixerDumpState; + state->mTeeSink = mTeeSink.get(); + sq->end(); + sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); + + // start the fast mixer + mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); + pid_t tid = mFastMixer->getTid(); + int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); + if (err != 0) { + ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", + kPriorityFastMixer, getpid_cached, tid, err); + } + +#ifdef AUDIO_WATCHDOG + // create and start the watchdog + mAudioWatchdog = new AudioWatchdog(); + mAudioWatchdog->setDump(&mAudioWatchdogDump); + mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); + tid = mAudioWatchdog->getTid(); + err = requestPriority(getpid_cached, tid, kPriorityFastMixer); + if (err != 0) { + ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", + kPriorityFastMixer, getpid_cached, tid, err); + } +#endif + + } else { + mFastMixer = NULL; + } + + switch (kUseFastMixer) { + case FastMixer_Never: + case FastMixer_Dynamic: + mNormalSink = mOutputSink; + break; + case FastMixer_Always: + mNormalSink = mPipeSink; + break; + case FastMixer_Static: + mNormalSink = initFastMixer ? mPipeSink : mOutputSink; + break; + } +} + +AudioFlinger::MixerThread::~MixerThread() +{ + if (mFastMixer != NULL) { + FastMixerStateQueue *sq = mFastMixer->sq(); + FastMixerState *state = sq->begin(); + if (state->mCommand == FastMixerState::COLD_IDLE) { + int32_t old = android_atomic_inc(&mFastMixerFutex); + if (old == -1) { + __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); + } + } + state->mCommand = FastMixerState::EXIT; + sq->end(); + sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); + mFastMixer->join(); + // Though the fast mixer thread has exited, it's state queue is still valid. + // We'll use that extract the final state which contains one remaining fast track + // corresponding to our sub-mix. + state = sq->begin(); + ALOG_ASSERT(state->mTrackMask == 1); + FastTrack *fastTrack = &state->mFastTracks[0]; + ALOG_ASSERT(fastTrack->mBufferProvider != NULL); + delete fastTrack->mBufferProvider; + sq->end(false /*didModify*/); + delete mFastMixer; +#ifdef AUDIO_WATCHDOG + if (mAudioWatchdog != 0) { + mAudioWatchdog->requestExit(); + mAudioWatchdog->requestExitAndWait(); + mAudioWatchdog.clear(); + } +#endif + } + delete mAudioMixer; +} + + +uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const +{ + if (mFastMixer != NULL) { + MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); + latency += (pipe->getAvgFrames() * 1000) / mSampleRate; + } + return latency; +} + + +void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) +{ + PlaybackThread::threadLoop_removeTracks(tracksToRemove); +} + +void AudioFlinger::MixerThread::threadLoop_write() +{ + // FIXME we should only do one push per cycle; confirm this is true + // Start the fast mixer if it's not already running + if (mFastMixer != NULL) { + FastMixerStateQueue *sq = mFastMixer->sq(); + FastMixerState *state = sq->begin(); + if (state->mCommand != FastMixerState::MIX_WRITE && + (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { + if (state->mCommand == FastMixerState::COLD_IDLE) { + int32_t old = android_atomic_inc(&mFastMixerFutex); + if (old == -1) { + __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); + } +#ifdef AUDIO_WATCHDOG + if (mAudioWatchdog != 0) { + mAudioWatchdog->resume(); + } +#endif + } + state->mCommand = FastMixerState::MIX_WRITE; + sq->end(); + sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); + if (kUseFastMixer == FastMixer_Dynamic) { + mNormalSink = mPipeSink; + } + } else { + sq->end(false /*didModify*/); + } + } + PlaybackThread::threadLoop_write(); +} + +void AudioFlinger::MixerThread::threadLoop_standby() +{ + // Idle the fast mixer if it's currently running + if (mFastMixer != NULL) { + FastMixerStateQueue *sq = mFastMixer->sq(); + FastMixerState *state = sq->begin(); + if (!(state->mCommand & FastMixerState::IDLE)) { + state->mCommand = FastMixerState::COLD_IDLE; + state->mColdFutexAddr = &mFastMixerFutex; + state->mColdGen++; + mFastMixerFutex = 0; + sq->end(); + // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now + sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); + if (kUseFastMixer == FastMixer_Dynamic) { + mNormalSink = mOutputSink; + } +#ifdef AUDIO_WATCHDOG + if (mAudioWatchdog != 0) { + mAudioWatchdog->pause(); + } +#endif + } else { + sq->end(false /*didModify*/); + } + } + PlaybackThread::threadLoop_standby(); +} + +// shared by MIXER and DIRECT, overridden by DUPLICATING +void AudioFlinger::PlaybackThread::threadLoop_standby() +{ + ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); + mOutput->stream->common.standby(&mOutput->stream->common); +} + +void AudioFlinger::MixerThread::threadLoop_mix() +{ + // obtain the presentation timestamp of the next output buffer + int64_t pts; + status_t status = INVALID_OPERATION; + + if (mNormalSink != 0) { + status = mNormalSink->getNextWriteTimestamp(&pts); + } else { + status = mOutputSink->getNextWriteTimestamp(&pts); + } + + if (status != NO_ERROR) { + pts = AudioBufferProvider::kInvalidPTS; + } + + // mix buffers... + mAudioMixer->process(pts); + // increase sleep time progressively when application underrun condition clears. + // Only increase sleep time if the mixer is ready for two consecutive times to avoid + // that a steady state of alternating ready/not ready conditions keeps the sleep time + // such that we would underrun the audio HAL. + if ((sleepTime == 0) && (sleepTimeShift > 0)) { + sleepTimeShift--; + } + sleepTime = 0; + standbyTime = systemTime() + standbyDelay; + //TODO: delay standby when effects have a tail +} + +void AudioFlinger::MixerThread::threadLoop_sleepTime() +{ + // 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 (mMixerStatus == MIXER_TRACKS_ENABLED) { + sleepTime = activeSleepTime >> sleepTimeShift; + if (sleepTime < kMinThreadSleepTimeUs) { + sleepTime = kMinThreadSleepTimeUs; + } + // reduce sleep time in case of consecutive application underruns to avoid + // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer + // duration we would end up writing less data than needed by the audio HAL if + // the condition persists. + if (sleepTimeShift < kMaxThreadSleepTimeShift) { + sleepTimeShift++; + } + } else { + sleepTime = idleSleepTime; + } + } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { + memset (mMixBuffer, 0, mixBufferSize); + sleepTime = 0; + ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), + "anticipated start"); + } + // TODO add standby time extension fct of effect tail +} + +// prepareTracks_l() must be called with ThreadBase::mLock held +AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( + Vector< sp<Track> > *tracksToRemove) +{ + + mixer_state mixerStatus = MIXER_IDLE; + // find out which tracks need to be processed + size_t count = mActiveTracks.size(); + size_t mixedTracks = 0; + size_t tracksWithEffect = 0; + // counts only _active_ fast tracks + size_t fastTracks = 0; + uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset + + float masterVolume = mMasterVolume; + bool masterMute = mMasterMute; + + if (masterMute) { + masterVolume = 0; + } + // Delegate master volume control to effect in output mix effect chain if needed + sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); + if (chain != 0) { + uint32_t v = (uint32_t)(masterVolume * (1 << 24)); + chain->setVolume_l(&v, &v); + masterVolume = (float)((v + (1 << 23)) >> 24); + chain.clear(); + } + + // prepare a new state to push + FastMixerStateQueue *sq = NULL; + FastMixerState *state = NULL; + bool didModify = false; + FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; + if (mFastMixer != NULL) { + sq = mFastMixer->sq(); + state = sq->begin(); + } + + for (size_t i=0 ; i<count ; i++) { + sp<Track> t = mActiveTracks[i].promote(); + if (t == 0) { + continue; + } + + // this const just means the local variable doesn't change + Track* const track = t.get(); + + // process fast tracks + if (track->isFastTrack()) { + + // It's theoretically possible (though unlikely) for a fast track to be created + // and then removed within the same normal mix cycle. This is not a problem, as + // the track never becomes active so it's fast mixer slot is never touched. + // The converse, of removing an (active) track and then creating a new track + // at the identical fast mixer slot within the same normal mix cycle, + // is impossible because the slot isn't marked available until the end of each cycle. + int j = track->mFastIndex; + ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); + ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); + FastTrack *fastTrack = &state->mFastTracks[j]; + + // Determine whether the track is currently in underrun condition, + // and whether it had a recent underrun. + FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; + FastTrackUnderruns underruns = ftDump->mUnderruns; + uint32_t recentFull = (underruns.mBitFields.mFull - + track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; + uint32_t recentPartial = (underruns.mBitFields.mPartial - + track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; + uint32_t recentEmpty = (underruns.mBitFields.mEmpty - + track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; + uint32_t recentUnderruns = recentPartial + recentEmpty; + track->mObservedUnderruns = underruns; + // don't count underruns that occur while stopping or pausing + // or stopped which can occur when flush() is called while active + if (!(track->isStopping() || track->isPausing() || track->isStopped())) { + track->mUnderrunCount += recentUnderruns; + } + + // This is similar to the state machine for normal tracks, + // with a few modifications for fast tracks. + bool isActive = true; + switch (track->mState) { + case TrackBase::STOPPING_1: + // track stays active in STOPPING_1 state until first underrun + if (recentUnderruns > 0) { + track->mState = TrackBase::STOPPING_2; + } + break; + case TrackBase::PAUSING: + // ramp down is not yet implemented + track->setPaused(); + break; + case TrackBase::RESUMING: + // ramp up is not yet implemented + track->mState = TrackBase::ACTIVE; + break; + case TrackBase::ACTIVE: + if (recentFull > 0 || recentPartial > 0) { + // track has provided at least some frames recently: reset retry count + track->mRetryCount = kMaxTrackRetries; + } + if (recentUnderruns == 0) { + // no recent underruns: stay active + break; + } + // there has recently been an underrun of some kind + if (track->sharedBuffer() == 0) { + // were any of the recent underruns "empty" (no frames available)? + if (recentEmpty == 0) { + // no, then ignore the partial underruns as they are allowed indefinitely + break; + } + // there has recently been an "empty" underrun: decrement the retry counter + if (--(track->mRetryCount) > 0) { + break; + } + // indicate to client process that the track was disabled because of underrun; + // it will then automatically call start() when data is available + android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); + // remove from active list, but state remains ACTIVE [confusing but true] + isActive = false; + break; + } + // fall through + case TrackBase::STOPPING_2: + case TrackBase::PAUSED: + case TrackBase::TERMINATED: + case TrackBase::STOPPED: + case TrackBase::FLUSHED: // flush() while active + // Check for presentation complete if track is inactive + // We have consumed all the buffers of this track. + // This would be incomplete if we auto-paused on underrun + { + size_t audioHALFrames = + (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; + size_t framesWritten = mBytesWritten / mFrameSize; + if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { + // track stays in active list until presentation is complete + break; + } + } + if (track->isStopping_2()) { + track->mState = TrackBase::STOPPED; + } + if (track->isStopped()) { + // Can't reset directly, as fast mixer is still polling this track + // track->reset(); + // So instead mark this track as needing to be reset after push with ack + resetMask |= 1 << i; + } + isActive = false; + break; + case TrackBase::IDLE: + default: + LOG_FATAL("unexpected track state %d", track->mState); + } + + if (isActive) { + // was it previously inactive? + if (!(state->mTrackMask & (1 << j))) { + ExtendedAudioBufferProvider *eabp = track; + VolumeProvider *vp = track; + fastTrack->mBufferProvider = eabp; + fastTrack->mVolumeProvider = vp; + fastTrack->mSampleRate = track->mSampleRate; + fastTrack->mChannelMask = track->mChannelMask; + fastTrack->mGeneration++; + state->mTrackMask |= 1 << j; + didModify = true; + // no acknowledgement required for newly active tracks + } + // cache the combined master volume and stream type volume for fast mixer; this + // lacks any synchronization or barrier so VolumeProvider may read a stale value + track->mCachedVolume = track->isMuted() ? + 0 : masterVolume * mStreamTypes[track->streamType()].volume; + ++fastTracks; + } else { + // was it previously active? + if (state->mTrackMask & (1 << j)) { + fastTrack->mBufferProvider = NULL; + fastTrack->mGeneration++; + state->mTrackMask &= ~(1 << j); + didModify = true; + // If any fast tracks were removed, we must wait for acknowledgement + // because we're about to decrement the last sp<> on those tracks. + block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; + } else { + LOG_FATAL("fast track %d should have been active", j); + } + tracksToRemove->add(track); + // Avoids a misleading display in dumpsys + track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; + } + continue; + } + + { // local variable scope to avoid goto warning + + 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 + int name = track->name(); + // make sure that we have enough frames to mix one full buffer. + // enforce this condition only once to enable draining the buffer in case the client + // app does not call stop() and relies on underrun to stop: + // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed + // during last round + uint32_t minFrames = 1; + if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && + (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { + if (t->sampleRate() == mSampleRate) { + minFrames = mNormalFrameCount; + } else { + // +1 for rounding and +1 for additional sample needed for interpolation + minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; + // add frames already consumed but not yet released by the resampler + // because cblk->framesReady() will include these frames + minFrames += mAudioMixer->getUnreleasedFrames(track->name()); + // the minimum track buffer size is normally twice the number of frames necessary + // to fill one buffer and the resampler should not leave more than one buffer worth + // of unreleased frames after each pass, but just in case... + ALOG_ASSERT(minFrames <= cblk->frameCount); + } + } + if ((track->framesReady() >= minFrames) && track->isReady() && + !track->isPaused() && !track->isTerminated()) + { + ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, + this); + + mixedTracks++; + + // track->mainBuffer() != mMixBuffer means there is an effect chain + // connected to the track + chain.clear(); + if (track->mainBuffer() != mMixBuffer) { + chain = getEffectChain_l(track->sessionId()); + // Delegate volume control to effect in track effect chain if needed + if (chain != 0) { + tracksWithEffect++; + } else { + ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " + "session %d", + name, track->sessionId()); + } + } + + + int param = AudioMixer::VOLUME; + if (track->mFillingUpStatus == Track::FS_FILLED) { + // no ramp for the first volume setting + track->mFillingUpStatus = Track::FS_ACTIVE; + if (track->mState == TrackBase::RESUMING) { + track->mState = TrackBase::ACTIVE; + param = AudioMixer::RAMP_VOLUME; + } + mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); + } else if (cblk->server != 0) { + // If the track is stopped before the first frame was mixed, + // do not apply ramp + param = AudioMixer::RAMP_VOLUME; + } + + // compute volume for this track + uint32_t vl, vr, va; + if (track->isMuted() || track->isPausing() || + mStreamTypes[track->streamType()].mute) { + vl = vr = va = 0; + if (track->isPausing()) { + track->setPaused(); + } + } else { + + // read original volumes with volume control + float typeVolume = mStreamTypes[track->streamType()].volume; + float v = masterVolume * typeVolume; + uint32_t vlr = cblk->getVolumeLR(); + vl = vlr & 0xFFFF; + vr = vlr >> 16; + // track volumes come from shared memory, so can't be trusted and must be clamped + if (vl > MAX_GAIN_INT) { + ALOGV("Track left volume out of range: %04X", vl); + vl = MAX_GAIN_INT; + } + if (vr > MAX_GAIN_INT) { + ALOGV("Track right volume out of range: %04X", vr); + vr = MAX_GAIN_INT; + } + // now apply the master volume and stream type volume + vl = (uint32_t)(v * vl) << 12; + vr = (uint32_t)(v * vr) << 12; + // assuming master volume and stream type volume each go up to 1.0, + // vl and vr are now in 8.24 format + + uint16_t sendLevel = cblk->getSendLevel_U4_12(); + // send level comes from shared memory and so may be corrupt + if (sendLevel > MAX_GAIN_INT) { + ALOGV("Track send level out of range: %04X", sendLevel); + sendLevel = MAX_GAIN_INT; + } + va = (uint32_t)(v * sendLevel); + } + // Delegate volume control to effect in track effect chain if needed + if (chain != 0 && chain->setVolume_l(&vl, &vr)) { + // Do not ramp volume if volume is controlled by effect + param = AudioMixer::VOLUME; + track->mHasVolumeController = true; + } else { + // force no volume ramp when volume controller was just disabled or removed + // from effect chain to avoid volume spike + if (track->mHasVolumeController) { + param = AudioMixer::VOLUME; + } + track->mHasVolumeController = false; + } + + // Convert volumes from 8.24 to 4.12 format + // This additional clamping is needed in case chain->setVolume_l() overshot + vl = (vl + (1 << 11)) >> 12; + if (vl > MAX_GAIN_INT) { + vl = MAX_GAIN_INT; + } + vr = (vr + (1 << 11)) >> 12; + if (vr > MAX_GAIN_INT) { + vr = MAX_GAIN_INT; + } + + if (va > MAX_GAIN_INT) { + va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - + } + + // XXX: these things DON'T need to be done each time + mAudioMixer->setBufferProvider(name, track); + mAudioMixer->enable(name); + + mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); + mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); + mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); + mAudioMixer->setParameter( + name, + AudioMixer::TRACK, + AudioMixer::FORMAT, (void *)track->format()); + mAudioMixer->setParameter( + name, + AudioMixer::TRACK, + AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); + mAudioMixer->setParameter( + name, + AudioMixer::RESAMPLE, + AudioMixer::SAMPLE_RATE, + (void *)(cblk->sampleRate)); + mAudioMixer->setParameter( + name, + AudioMixer::TRACK, + AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); + mAudioMixer->setParameter( + name, + AudioMixer::TRACK, + AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); + + // reset retry count + track->mRetryCount = kMaxTrackRetries; + + // If one track is ready, set the mixer ready if: + // - the mixer was not ready during previous round OR + // - no other track is not ready + if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || + mixerStatus != MIXER_TRACKS_ENABLED) { + mixerStatus = MIXER_TRACKS_READY; + } + } else { + // clear effect chain input buffer if an active track underruns to avoid sending + // previous audio buffer again to effects + chain = getEffectChain_l(track->sessionId()); + if (chain != 0) { + chain->clearInputBuffer(); + } + + ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, + cblk->server, this); + if ((track->sharedBuffer() != 0) || track->isTerminated() || + track->isStopped() || track->isPaused()) { + // We have consumed all the buffers of this track. + // Remove it from the list of active tracks. + // TODO: use actual buffer filling status instead of latency when available from + // audio HAL + size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; + size_t framesWritten = mBytesWritten / mFrameSize; + if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { + if (track->isStopped()) { + track->reset(); + } + tracksToRemove->add(track); + } + } else { + track->mUnderrunCount++; + // No buffers for this track. Give it a few chances to + // fill a buffer, then remove it from active list. + if (--(track->mRetryCount) <= 0) { + ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); + tracksToRemove->add(track); + // indicate to client process that the track was disabled because of underrun; + // it will then automatically call start() when data is available + android_atomic_or(CBLK_DISABLED, &cblk->flags); + // If one track is not ready, mark the mixer also not ready if: + // - the mixer was ready during previous round OR + // - no other track is ready + } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || + mixerStatus != MIXER_TRACKS_READY) { + mixerStatus = MIXER_TRACKS_ENABLED; + } + } + mAudioMixer->disable(name); + } + + } // local variable scope to avoid goto warning +track_is_ready: ; + + } + + // Push the new FastMixer state if necessary + bool pauseAudioWatchdog = false; + if (didModify) { + state->mFastTracksGen++; + // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle + if (kUseFastMixer == FastMixer_Dynamic && + state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { + state->mCommand = FastMixerState::COLD_IDLE; + state->mColdFutexAddr = &mFastMixerFutex; + state->mColdGen++; + mFastMixerFutex = 0; + if (kUseFastMixer == FastMixer_Dynamic) { + mNormalSink = mOutputSink; + } + // If we go into cold idle, need to wait for acknowledgement + // so that fast mixer stops doing I/O. + block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; + pauseAudioWatchdog = true; + } + sq->end(); + } + if (sq != NULL) { + sq->end(didModify); + sq->push(block); + } +#ifdef AUDIO_WATCHDOG + if (pauseAudioWatchdog && mAudioWatchdog != 0) { + mAudioWatchdog->pause(); + } +#endif + + // Now perform the deferred reset on fast tracks that have stopped + while (resetMask != 0) { + size_t i = __builtin_ctz(resetMask); + ALOG_ASSERT(i < count); + resetMask &= ~(1 << i); + sp<Track> t = mActiveTracks[i].promote(); + if (t == 0) { + continue; + } + Track* track = t.get(); + ALOG_ASSERT(track->isFastTrack() && track->isStopped()); + track->reset(); + } + + // remove all the tracks that need to be... + count = tracksToRemove->size(); + if (CC_UNLIKELY(count)) { + for (size_t i=0 ; i<count ; i++) { + const sp<Track>& track = tracksToRemove->itemAt(i); + mActiveTracks.remove(track); + if (track->mainBuffer() != mMixBuffer) { + chain = getEffectChain_l(track->sessionId()); + if (chain != 0) { + ALOGV("stopping track on chain %p for session Id: %d", chain.get(), + track->sessionId()); + chain->decActiveTrackCnt(); + } + } + if (track->isTerminated()) { + removeTrack_l(track); + } + } + } + + // mix buffer must be cleared if all tracks are connected to an + // effect chain as in this case the mixer will not write to + // mix buffer and track effects will accumulate into it + if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || + (mixedTracks == 0 && fastTracks > 0)) { + // FIXME as a performance optimization, should remember previous zero status + memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); + } + + // if any fast tracks, then status is ready + mMixerStatusIgnoringFastTracks = mixerStatus; + if (fastTracks > 0) { + mixerStatus = MIXER_TRACKS_READY; + } + return mixerStatus; +} + +// getTrackName_l() must be called with ThreadBase::mLock held +int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) +{ + return mAudioMixer->getTrackName(channelMask, sessionId); +} + +// deleteTrackName_l() must be called with ThreadBase::mLock held +void AudioFlinger::MixerThread::deleteTrackName_l(int name) +{ + ALOGV("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() +{ + // if !&IDLE, holds the FastMixer state to restore after new parameters processed + FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; + bool reconfig = false; + + while (!mNewParameters.isEmpty()) { + + if (mFastMixer != NULL) { + FastMixerStateQueue *sq = mFastMixer->sq(); + FastMixerState *state = sq->begin(); + if (!(state->mCommand & FastMixerState::IDLE)) { + previousCommand = state->mCommand; + state->mCommand = FastMixerState::HOT_IDLE; + sq->end(); + sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); + } else { + sq->end(false /*didModify*/); + } + } + + 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 ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { + status = BAD_VALUE; + } else { + reconfig = true; + } + } + if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { + if (value != AUDIO_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 guaranteed + // if frame count is changed after track creation + if (!mTracks.isEmpty()) { + status = INVALID_OPERATION; + } else { + reconfig = true; + } + } + if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { +#ifdef ADD_BATTERY_DATA + // when changing the audio output device, call addBatteryData to notify + // the change + if (mOutDevice != value) { + uint32_t params = 0; + // check whether speaker is on + if (value & AUDIO_DEVICE_OUT_SPEAKER) { + params |= IMediaPlayerService::kBatteryDataSpeakerOn; + } + + audio_devices_t deviceWithoutSpeaker + = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; + // check if any other device (except speaker) is on + if (value & deviceWithoutSpeaker ) { + params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; + } + + if (params != 0) { + addBatteryData(params); + } + } +#endif + + // forward device change to effects that have requested to be + // aware of attached audio device. + mOutDevice = value; + for (size_t i = 0; i < mEffectChains.size(); i++) { + mEffectChains[i]->setDevice_l(mOutDevice); + } + } + + if (status == NO_ERROR) { + status = mOutput->stream->common.set_parameters(&mOutput->stream->common, + keyValuePair.string()); + if (!mStandby && status == INVALID_OPERATION) { + mOutput->stream->common.standby(&mOutput->stream->common); + mStandby = true; + mBytesWritten = 0; + status = mOutput->stream->common.set_parameters(&mOutput->stream->common, + keyValuePair.string()); + } + if (status == NO_ERROR && reconfig) { + delete mAudioMixer; + // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) + mAudioMixer = NULL; + readOutputParameters(); + mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); + for (size_t i = 0; i < mTracks.size() ; i++) { + int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); + 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(); + } + } + sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); + } + } + + mNewParameters.removeAt(0); + + mParamStatus = status; + mParamCond.signal(); + // wait for condition with time out in case the thread calling ThreadBase::setParameters() + // already timed out waiting for the status and will never signal the condition. + mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); + } + + if (!(previousCommand & FastMixerState::IDLE)) { + ALOG_ASSERT(mFastMixer != NULL); + FastMixerStateQueue *sq = mFastMixer->sq(); + FastMixerState *state = sq->begin(); + ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); + state->mCommand = previousCommand; + sq->end(); + sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); + } + + return reconfig; +} + + +void 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()); + + // Make a non-atomic copy of fast mixer dump state so it won't change underneath us + FastMixerDumpState copy = mFastMixerDumpState; + copy.dump(fd); + +#ifdef STATE_QUEUE_DUMP + // Similar for state queue + StateQueueObserverDump observerCopy = mStateQueueObserverDump; + observerCopy.dump(fd); + StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; + mutatorCopy.dump(fd); +#endif + + // Write the tee output to a .wav file + dumpTee(fd, mTeeSource, mId); + +#ifdef AUDIO_WATCHDOG + if (mAudioWatchdog != 0) { + // Make a non-atomic copy of audio watchdog dump so it won't change underneath us + AudioWatchdogDump wdCopy = mAudioWatchdogDump; + wdCopy.dump(fd); + } +#endif +} + +uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const +{ + return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; +} + +uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const +{ + return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); +} + +void AudioFlinger::MixerThread::cacheParameters_l() +{ + PlaybackThread::cacheParameters_l(); + + // 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 + // increase threshold again due to low power audio mode. The way this warning + // threshold is calculated and its usefulness should be reconsidered anyway. + maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; +} + +// ---------------------------------------------------------------------------- + +AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, + AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) + : PlaybackThread(audioFlinger, output, id, device, DIRECT) + // mLeftVolFloat, mRightVolFloat +{ +} + +AudioFlinger::DirectOutputThread::~DirectOutputThread() +{ +} + +AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( + Vector< sp<Track> > *tracksToRemove +) +{ + sp<Track> trackToRemove; + + mixer_state mixerStatus = MIXER_IDLE; + + // find out which tracks need to be processed + if (mActiveTracks.size() != 0) { + sp<Track> t = mActiveTracks[0].promote(); + // The track died recently + if (t == 0) { + return MIXER_IDLE; + } + + 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 + uint32_t minFrames; + if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { + minFrames = mNormalFrameCount; + } else { + minFrames = 1; + } + if ((track->framesReady() >= minFrames) && track->isReady() && + !track->isPaused() && !track->isTerminated()) + { + ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); + + if (track->mFillingUpStatus == Track::FS_FILLED) { + track->mFillingUpStatus = Track::FS_ACTIVE; + mLeftVolFloat = mRightVolFloat = 0; + if (track->mState == TrackBase::RESUMING) { + track->mState = TrackBase::ACTIVE; + } + } + + // compute volume for this track + float left, right; + if (track->isMuted() || mMasterMute || track->isPausing() || + mStreamTypes[track->streamType()].mute) { + left = right = 0; + if (track->isPausing()) { + track->setPaused(); + } + } else { + float typeVolume = mStreamTypes[track->streamType()].volume; + float v = mMasterVolume * typeVolume; + uint32_t vlr = cblk->getVolumeLR(); + float v_clamped = v * (vlr & 0xFFFF); + if (v_clamped > MAX_GAIN) { + v_clamped = MAX_GAIN; + } + left = v_clamped/MAX_GAIN; + v_clamped = v * (vlr >> 16); + if (v_clamped > MAX_GAIN) { + v_clamped = MAX_GAIN; + } + right = v_clamped/MAX_GAIN; + } + + if (left != mLeftVolFloat || right != mRightVolFloat) { + mLeftVolFloat = left; + mRightVolFloat = right; + + // Convert volumes from float to 8.24 + uint32_t vl = (uint32_t)(left * (1 << 24)); + uint32_t vr = (uint32_t)(right * (1 << 24)); + + // Delegate volume control to effect in track effect chain if needed + // only one effect chain can be present on DirectOutputThread, so if + // there is one, the track is connected to it + if (!mEffectChains.isEmpty()) { + // Do not ramp volume if volume is controlled by effect + mEffectChains[0]->setVolume_l(&vl, &vr); + left = (float)vl / (1 << 24); + right = (float)vr / (1 << 24); + } + mOutput->stream->set_volume(mOutput->stream, left, right); + } + + // reset retry count + track->mRetryCount = kMaxTrackRetriesDirect; + mActiveTrack = t; + mixerStatus = MIXER_TRACKS_READY; + } else { + // clear effect chain input buffer if an active track underruns to avoid sending + // previous audio buffer again to effects + if (!mEffectChains.isEmpty()) { + mEffectChains[0]->clearInputBuffer(); + } + + ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); + if ((track->sharedBuffer() != 0) || track->isTerminated() || + track->isStopped() || track->isPaused()) { + // We have consumed all the buffers of this track. + // Remove it from the list of active tracks. + // TODO: implement behavior for compressed audio + size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; + size_t framesWritten = mBytesWritten / mFrameSize; + if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { + if (track->isStopped()) { + track->reset(); + } + 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) { + ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); + trackToRemove = track; + } else { + mixerStatus = MIXER_TRACKS_ENABLED; + } + } + } + } + + // FIXME merge this with similar code for removing multiple tracks + // remove all the tracks that need to be... + if (CC_UNLIKELY(trackToRemove != 0)) { + tracksToRemove->add(trackToRemove); + mActiveTracks.remove(trackToRemove); + if (!mEffectChains.isEmpty()) { + ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), + trackToRemove->sessionId()); + mEffectChains[0]->decActiveTrackCnt(); + } + if (trackToRemove->isTerminated()) { + removeTrack_l(trackToRemove); + } + } + + return mixerStatus; +} + +void AudioFlinger::DirectOutputThread::threadLoop_mix() +{ + AudioBufferProvider::Buffer buffer; + size_t frameCount = mFrameCount; + int8_t *curBuf = (int8_t *)mMixBuffer; + // output audio to hardware + while (frameCount) { + buffer.frameCount = frameCount; + mActiveTrack->getNextBuffer(&buffer); + if (CC_UNLIKELY(buffer.raw == NULL)) { + memset(curBuf, 0, frameCount * mFrameSize); + break; + } + memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); + frameCount -= buffer.frameCount; + curBuf += buffer.frameCount * mFrameSize; + mActiveTrack->releaseBuffer(&buffer); + } + sleepTime = 0; + standbyTime = systemTime() + standbyDelay; + mActiveTrack.clear(); + +} + +void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() +{ + if (sleepTime == 0) { + if (mMixerStatus == MIXER_TRACKS_ENABLED) { + sleepTime = activeSleepTime; + } else { + sleepTime = idleSleepTime; + } + } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { + memset(mMixBuffer, 0, mFrameCount * mFrameSize); + sleepTime = 0; + } +} + +// getTrackName_l() must be called with ThreadBase::mLock held +int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, + int sessionId) +{ + 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->stream->common.set_parameters(&mOutput->stream->common, + keyValuePair.string()); + if (!mStandby && status == INVALID_OPERATION) { + mOutput->stream->common.standby(&mOutput->stream->common); + mStandby = true; + mBytesWritten = 0; + status = mOutput->stream->common.set_parameters(&mOutput->stream->common, + keyValuePair.string()); + } + if (status == NO_ERROR && reconfig) { + readOutputParameters(); + sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); + } + } + + mNewParameters.removeAt(0); + + mParamStatus = status; + mParamCond.signal(); + // wait for condition with time out in case the thread calling ThreadBase::setParameters() + // already timed out waiting for the status and will never signal the condition. + mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); + } + return reconfig; +} + +uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const +{ + uint32_t time; + if (audio_is_linear_pcm(mFormat)) { + time = PlaybackThread::activeSleepTimeUs(); + } else { + time = 10000; + } + return time; +} + +uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const +{ + uint32_t time; + if (audio_is_linear_pcm(mFormat)) { + time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; + } else { + time = 10000; + } + return time; +} + +uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const +{ + uint32_t time; + if (audio_is_linear_pcm(mFormat)) { + time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); + } else { + time = 10000; + } + return time; +} + +void AudioFlinger::DirectOutputThread::cacheParameters_l() +{ + PlaybackThread::cacheParameters_l(); + + // use shorter standby delay as on normal output to release + // hardware resources as soon as possible + standbyDelay = microseconds(activeSleepTime*2); +} + +// ---------------------------------------------------------------------------- + +AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, + AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) + : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), + DUPLICATING), + mWaitTimeMs(UINT_MAX) +{ + addOutputTrack(mainThread); +} + +AudioFlinger::DuplicatingThread::~DuplicatingThread() +{ + for (size_t i = 0; i < mOutputTracks.size(); i++) { + mOutputTracks[i]->destroy(); + } +} + +void AudioFlinger::DuplicatingThread::threadLoop_mix() +{ + // mix buffers... + if (outputsReady(outputTracks)) { + mAudioMixer->process(AudioBufferProvider::kInvalidPTS); + } else { + memset(mMixBuffer, 0, mixBufferSize); + } + sleepTime = 0; + writeFrames = mNormalFrameCount; + standbyTime = systemTime() + standbyDelay; +} + +void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() +{ + if (sleepTime == 0) { + if (mMixerStatus == MIXER_TRACKS_ENABLED) { + sleepTime = activeSleepTime; + } else { + sleepTime = idleSleepTime; + } + } else if (mBytesWritten != 0) { + if (mMixerStatus == MIXER_TRACKS_ENABLED) { + writeFrames = mNormalFrameCount; + memset(mMixBuffer, 0, mixBufferSize); + } else { + // flush remaining overflow buffers in output tracks + writeFrames = 0; + } + sleepTime = 0; + } +} + +void AudioFlinger::DuplicatingThread::threadLoop_write() +{ + for (size_t i = 0; i < outputTracks.size(); i++) { + outputTracks[i]->write(mMixBuffer, writeFrames); + } + mBytesWritten += mixBufferSize; +} + +void AudioFlinger::DuplicatingThread::threadLoop_standby() +{ + // DuplicatingThread implements standby by stopping all tracks + for (size_t i = 0; i < outputTracks.size(); i++) { + outputTracks[i]->stop(); + } +} + +void AudioFlinger::DuplicatingThread::saveOutputTracks() +{ + outputTracks = mOutputTracks; +} + +void AudioFlinger::DuplicatingThread::clearOutputTracks() +{ + outputTracks.clear(); +} + +void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) +{ + Mutex::Autolock _l(mLock); + // FIXME explain this formula + size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); + OutputTrack *outputTrack = new OutputTrack(thread, + this, + mSampleRate, + mFormat, + mChannelMask, + frameCount); + if (outputTrack->cblk() != NULL) { + thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); + mOutputTracks.add(outputTrack); + ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); + updateWaitTime_l(); + } +} + +void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) +{ + Mutex::Autolock _l(mLock); + for (size_t i = 0; i < mOutputTracks.size(); i++) { + if (mOutputTracks[i]->thread() == thread) { + mOutputTracks[i]->destroy(); + mOutputTracks.removeAt(i); + updateWaitTime_l(); + return; + } + } + ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); +} + +// caller must hold mLock +void AudioFlinger::DuplicatingThread::updateWaitTime_l() +{ + mWaitTimeMs = UINT_MAX; + for (size_t i = 0; i < mOutputTracks.size(); i++) { + sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); + if (strong != 0) { + uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); + if (waitTimeMs < mWaitTimeMs) { + mWaitTimeMs = waitTimeMs; + } + } + } +} + + +bool AudioFlinger::DuplicatingThread::outputsReady( + const SortedVector< sp<OutputTrack> > &outputTracks) +{ + for (size_t i = 0; i < outputTracks.size(); i++) { + sp<ThreadBase> thread = outputTracks[i]->thread().promote(); + if (thread == 0) { + ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", + outputTracks[i].get()); + return false; + } + PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); + // see note at standby() declaration + if (playbackThread->standby() && !playbackThread->isSuspended()) { + ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), + thread.get()); + return false; + } + } + return true; +} + +uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const +{ + return (mWaitTimeMs * 1000) / 2; +} + +void AudioFlinger::DuplicatingThread::cacheParameters_l() +{ + // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first + updateWaitTime_l(); + + MixerThread::cacheParameters_l(); +} + +// ---------------------------------------------------------------------------- +// Record +// ---------------------------------------------------------------------------- + +AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, + AudioStreamIn *input, + uint32_t sampleRate, + audio_channel_mask_t channelMask, + audio_io_handle_t id, + audio_devices_t device, + const sp<NBAIO_Sink>& teeSink) : + ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), + mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), + // mRsmpInIndex and mInputBytes set by readInputParameters() + mReqChannelCount(popcount(channelMask)), + mReqSampleRate(sampleRate), + // mBytesRead is only meaningful while active, and so is cleared in start() + // (but might be better to also clear here for dump?) + mTeeSink(teeSink) +{ + snprintf(mName, kNameLength, "AudioIn_%X", id); + + readInputParameters(); + +} + + +AudioFlinger::RecordThread::~RecordThread() +{ + delete[] mRsmpInBuffer; + delete mResampler; + delete[] mRsmpOutBuffer; +} + +void AudioFlinger::RecordThread::onFirstRef() +{ + run(mName, PRIORITY_URGENT_AUDIO); +} + +status_t AudioFlinger::RecordThread::readyToRun() +{ + status_t status = initCheck(); + ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); + return status; +} + +bool AudioFlinger::RecordThread::threadLoop() +{ + AudioBufferProvider::Buffer buffer; + sp<RecordTrack> activeTrack; + Vector< sp<EffectChain> > effectChains; + + nsecs_t lastWarning = 0; + + inputStandBy(); + acquireWakeLock(); + + // used to verify we've read at least once before evaluating how many bytes were read + bool readOnce = false; + + // start recording + while (!exitPending()) { + + processConfigEvents(); + + { // scope for mLock + Mutex::Autolock _l(mLock); + checkForNewParameters_l(); + if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { + standby(); + + if (exitPending()) { + break; + } + + releaseWakeLock_l(); + ALOGV("RecordThread: loop stopping"); + // go to sleep + mWaitWorkCV.wait(mLock); + ALOGV("RecordThread: loop starting"); + acquireWakeLock_l(); + continue; + } + if (mActiveTrack != 0) { + if (mActiveTrack->mState == TrackBase::PAUSING) { + standby(); + mActiveTrack.clear(); + mStartStopCond.broadcast(); + } else if (mActiveTrack->mState == TrackBase::RESUMING) { + if (mReqChannelCount != mActiveTrack->channelCount()) { + mActiveTrack.clear(); + mStartStopCond.broadcast(); + } else if (readOnce) { + // 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; + } else if (mActiveTrack->mState == TrackBase::TERMINATED) { + removeTrack_l(mActiveTrack); + mActiveTrack.clear(); + } + } + lockEffectChains_l(effectChains); + } + + if (mActiveTrack != 0) { + if (mActiveTrack->mState != TrackBase::ACTIVE && + mActiveTrack->mState != TrackBase::RESUMING) { + unlockEffectChains(effectChains); + usleep(kRecordThreadSleepUs); + continue; + } + for (size_t i = 0; i < effectChains.size(); i ++) { + effectChains[i]->process_l(); + } + + buffer.frameCount = mFrameCount; + if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { + readOnce = true; + size_t framesOut = buffer.frameCount; + if (mResampler == NULL) { + // 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->mFrameSize; + if (framesIn > framesOut) + framesIn = framesOut; + mRsmpInIndex += framesIn; + framesOut -= framesIn; + if (mChannelCount == mReqChannelCount || + mFormat != AUDIO_FORMAT_PCM_16_BIT) { + memcpy(dst, src, framesIn * mFrameSize); + } else { + if (mChannelCount == 1) { + upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, + (int16_t *)src, framesIn); + } else { + downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, + (int16_t *)src, framesIn); + } + } + } + if (framesOut && mFrameCount == mRsmpInIndex) { + void *readInto; + if (framesOut == mFrameCount && + (mChannelCount == mReqChannelCount || + mFormat != AUDIO_FORMAT_PCM_16_BIT)) { + readInto = buffer.raw; + framesOut = 0; + } else { + readInto = mRsmpInBuffer; + mRsmpInIndex = 0; + } + mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); + if (mBytesRead <= 0) { + if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) + { + ALOGE("Error reading audio input"); + // Force input into standby so that it tries to + // recover at next read attempt + inputStandBy(); + usleep(kRecordThreadSleepUs); + } + mRsmpInIndex = mFrameCount; + framesOut = 0; + buffer.frameCount = 0; + } else if (mTeeSink != 0) { + (void) mTeeSink->write(readInto, + mBytesRead >> Format_frameBitShift(mTeeSink->format())); + } + } + } + } 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 /* AudioBufferProvider* */); + // 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) { + ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); + // the resampler always outputs stereo samples: + // do post stereo to mono conversion + downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, + framesOut); + } else { + ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); + } + + } + if (mFramestoDrop == 0) { + mActiveTrack->releaseBuffer(&buffer); + } else { + if (mFramestoDrop > 0) { + mFramestoDrop -= buffer.frameCount; + if (mFramestoDrop <= 0) { + clearSyncStartEvent(); + } + } else { + mFramestoDrop += buffer.frameCount; + if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || + mSyncStartEvent->isCancelled()) { + ALOGW("Synced record %s, session %d, trigger session %d", + (mFramestoDrop >= 0) ? "timed out" : "cancelled", + mActiveTrack->sessionId(), + (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); + clearSyncStartEvent(); + } + } + } + mActiveTrack->clearOverflow(); + } + // client isn't retrieving buffers fast enough + else { + if (!mActiveTrack->setOverflow()) { + nsecs_t now = systemTime(); + if ((now - lastWarning) > kWarningThrottleNs) { + ALOGW("RecordThread: buffer overflow"); + lastWarning = now; + } + } + // 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(kRecordThreadSleepUs); + } + } + // enable changes in effect chain + unlockEffectChains(effectChains); + effectChains.clear(); + } + + standby(); + + { + Mutex::Autolock _l(mLock); + mActiveTrack.clear(); + mStartStopCond.broadcast(); + } + + releaseWakeLock(); + + ALOGV("RecordThread %p exiting", this); + return false; +} + +void AudioFlinger::RecordThread::standby() +{ + if (!mStandby) { + inputStandBy(); + mStandby = true; + } +} + +void AudioFlinger::RecordThread::inputStandBy() +{ + mInput->stream->common.standby(&mInput->stream->common); +} + +sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( + const sp<AudioFlinger::Client>& client, + uint32_t sampleRate, + audio_format_t format, + audio_channel_mask_t channelMask, + size_t frameCount, + int sessionId, + IAudioFlinger::track_flags_t flags, + pid_t tid, + status_t *status) +{ + sp<RecordTrack> track; + status_t lStatus; + + lStatus = initCheck(); + if (lStatus != NO_ERROR) { + ALOGE("Audio driver not initialized."); + goto Exit; + } + + // FIXME use flags and tid similar to createTrack_l() + + { // scope for mLock + Mutex::Autolock _l(mLock); + + track = new RecordTrack(this, client, sampleRate, + format, channelMask, frameCount, sessionId); + + if (track->getCblk() == 0) { + lStatus = NO_MEMORY; + goto Exit; + } + mTracks.add(track); + + // disable AEC and NS if the device is a BT SCO headset supporting those pre processings + bool suspend = audio_is_bluetooth_sco_device(mInDevice) && + mAudioFlinger->btNrecIsOff(); + setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); + setEffectSuspended_l(FX_IID_NS, suspend, sessionId); + } + lStatus = NO_ERROR; + +Exit: + if (status) { + *status = lStatus; + } + return track; +} + +status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, + AudioSystem::sync_event_t event, + int triggerSession) +{ + ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); + sp<ThreadBase> strongMe = this; + status_t status = NO_ERROR; + + if (event == AudioSystem::SYNC_EVENT_NONE) { + clearSyncStartEvent(); + } else if (event != AudioSystem::SYNC_EVENT_SAME) { + mSyncStartEvent = mAudioFlinger->createSyncEvent(event, + triggerSession, + recordTrack->sessionId(), + syncStartEventCallback, + this); + // Sync event can be cancelled by the trigger session if the track is not in a + // compatible state in which case we start record immediately + if (mSyncStartEvent->isCancelled()) { + clearSyncStartEvent(); + } else { + // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs + mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); + } + } + + { + 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(); + clearSyncStartEvent(); + return status; + } + mRsmpInIndex = mFrameCount; + mBytesRead = 0; + if (mResampler != NULL) { + mResampler->reset(); + } + mActiveTrack->mState = TrackBase::RESUMING; + // signal thread to start + ALOGV("Signal record thread"); + mWaitWorkCV.broadcast(); + // do not wait for mStartStopCond if exiting + if (exitPending()) { + mActiveTrack.clear(); + status = INVALID_OPERATION; + goto startError; + } + mStartStopCond.wait(mLock); + if (mActiveTrack == 0) { + ALOGV("Record failed to start"); + status = BAD_VALUE; + goto startError; + } + ALOGV("Record started OK"); + return status; + } +startError: + AudioSystem::stopInput(mId); + clearSyncStartEvent(); + return status; +} + +void AudioFlinger::RecordThread::clearSyncStartEvent() +{ + if (mSyncStartEvent != 0) { + mSyncStartEvent->cancel(); + } + mSyncStartEvent.clear(); + mFramestoDrop = 0; +} + +void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) +{ + sp<SyncEvent> strongEvent = event.promote(); + + if (strongEvent != 0) { + RecordThread *me = (RecordThread *)strongEvent->cookie(); + me->handleSyncStartEvent(strongEvent); + } +} + +void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) +{ + if (event == mSyncStartEvent) { + // TODO: use actual buffer filling status instead of 2 buffers when info is available + // from audio HAL + mFramestoDrop = mFrameCount * 2; + } +} + +bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { + ALOGV("RecordThread::stop"); + if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { + return false; + } + recordTrack->mState = TrackBase::PAUSING; + // do not wait for mStartStopCond if exiting + if (exitPending()) { + return true; + } + mStartStopCond.wait(mLock); + // if we have been restarted, recordTrack == mActiveTrack.get() here + if (exitPending() || recordTrack != mActiveTrack.get()) { + ALOGV("Record stopped OK"); + return true; + } + return false; +} + +bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const +{ + return false; +} + +status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) +{ +#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future + if (!isValidSyncEvent(event)) { + return BAD_VALUE; + } + + int eventSession = event->triggerSession(); + status_t ret = NAME_NOT_FOUND; + + Mutex::Autolock _l(mLock); + + for (size_t i = 0; i < mTracks.size(); i++) { + sp<RecordTrack> track = mTracks[i]; + if (eventSession == track->sessionId()) { + (void) track->setSyncEvent(event); + ret = NO_ERROR; + } + } + return ret; +#else + return BAD_VALUE; +#endif +} + +// destroyTrack_l() must be called with ThreadBase::mLock held +void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) +{ + track->mState = TrackBase::TERMINATED; + // active tracks are removed by threadLoop() + if (mActiveTrack != track) { + removeTrack_l(track); + } +} + +void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) +{ + mTracks.remove(track); + // need anything related to effects here? +} + +void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) +{ + dumpInternals(fd, args); + dumpTracks(fd, args); + dumpEffectChains(fd, args); +} + +void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); + result.append(buffer); + + if (mActiveTrack != 0) { + 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 != NULL)); + result.append(buffer); + snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); + result.append(buffer); + snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); + result.append(buffer); + } else { + result.append("No active record client\n"); + } + + write(fd, result.string(), result.size()); + + dumpBase(fd, args); +} + +void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, "Input thread %p tracks\n", this); + result.append(buffer); + RecordTrack::appendDumpHeader(result); + for (size_t i = 0; i < mTracks.size(); ++i) { + sp<RecordTrack> track = mTracks[i]; + if (track != 0) { + track->dump(buffer, SIZE); + result.append(buffer); + } + } + + if (mActiveTrack != 0) { + snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); + result.append(buffer); + RecordTrack::appendDumpHeader(result); + mActiveTrack->dump(buffer, SIZE); + result.append(buffer); + + } + write(fd, result.string(), result.size()); +} + +// AudioBufferProvider interface +status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) +{ + size_t framesReq = buffer->frameCount; + size_t framesReady = mFrameCount - mRsmpInIndex; + int channelCount; + + if (framesReady == 0) { + mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); + if (mBytesRead <= 0) { + if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { + ALOGE("RecordThread::getNextBuffer() Error reading audio input"); + // Force input into standby so that it tries to + // recover at next read attempt + inputStandBy(); + usleep(kRecordThreadSleepUs); + } + buffer->raw = NULL; + 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; +} + +// AudioBufferProvider interface +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; + audio_format_t reqFormat = mFormat; + uint32_t reqSamplingRate = mReqSampleRate; + uint32_t 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 = (audio_format_t) value; + reconfig = true; + } + if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { + reqChannelCount = 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 guaranteed + // if frame count is changed after track creation + if (mActiveTrack != 0) { + status = INVALID_OPERATION; + } else { + reconfig = true; + } + } + if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { + // forward device change to effects that have requested to be + // aware of attached audio device. + for (size_t i = 0; i < mEffectChains.size(); i++) { + mEffectChains[i]->setDevice_l(value); + } + + // store input device and output device but do not forward output device to audio HAL. + // Note that status is ignored by the caller for output device + // (see AudioFlinger::setParameters() + if (audio_is_output_devices(value)) { + mOutDevice = value; + status = BAD_VALUE; + } else { + mInDevice = value; + // disable AEC and NS if the device is a BT SCO headset supporting those + // pre processings + if (mTracks.size() > 0) { + bool suspend = audio_is_bluetooth_sco_device(mInDevice) && + mAudioFlinger->btNrecIsOff(); + for (size_t i = 0; i < mTracks.size(); i++) { + sp<RecordTrack> track = mTracks[i]; + setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); + setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); + } + } + } + } + if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && + mAudioSource != (audio_source_t)value) { + // forward device change to effects that have requested to be + // aware of attached audio device. + for (size_t i = 0; i < mEffectChains.size(); i++) { + mEffectChains[i]->setAudioSource_l((audio_source_t)value); + } + mAudioSource = (audio_source_t)value; + } + if (status == NO_ERROR) { + status = mInput->stream->common.set_parameters(&mInput->stream->common, + keyValuePair.string()); + if (status == INVALID_OPERATION) { + inputStandBy(); + status = mInput->stream->common.set_parameters(&mInput->stream->common, + keyValuePair.string()); + } + if (reconfig) { + if (status == BAD_VALUE && + reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && + reqFormat == AUDIO_FORMAT_PCM_16_BIT && + ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) + <= (2 * reqSamplingRate)) && + popcount(mInput->stream->common.get_channels(&mInput->stream->common)) + <= FCC_2 && + (reqChannelCount <= FCC_2)) { + status = NO_ERROR; + } + if (status == NO_ERROR) { + readInputParameters(); + sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); + } + } + } + + mNewParameters.removeAt(0); + + mParamStatus = status; + mParamCond.signal(); + // wait for condition with time out in case the thread calling ThreadBase::setParameters() + // already timed out waiting for the status and will never signal the condition. + mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); + } + return reconfig; +} + +String8 AudioFlinger::RecordThread::getParameters(const String8& keys) +{ + char *s; + String8 out_s8 = String8(); + + Mutex::Autolock _l(mLock); + if (initCheck() != NO_ERROR) { + return out_s8; + } + + s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); + out_s8 = String8(s); + free(s); + return out_s8; +} + +void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { + AudioSystem::OutputDescriptor desc; + void *param2 = NULL; + + switch (event) { + case AudioSystem::INPUT_OPENED: + case AudioSystem::INPUT_CONFIG_CHANGED: + desc.channels = mChannelMask; + desc.samplingRate = mSampleRate; + desc.format = mFormat; + desc.frameCount = mFrameCount; + desc.latency = 0; + param2 = &desc; + break; + + case AudioSystem::INPUT_CLOSED: + default: + break; + } + mAudioFlinger->audioConfigChanged_l(event, mId, param2); +} + +void AudioFlinger::RecordThread::readInputParameters() +{ + delete mRsmpInBuffer; + // mRsmpInBuffer is always assigned a new[] below + delete mRsmpOutBuffer; + mRsmpOutBuffer = NULL; + delete mResampler; + mResampler = NULL; + + mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); + mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); + mChannelCount = (uint16_t)popcount(mChannelMask); + mFormat = mInput->stream->common.get_format(&mInput->stream->common); + mFrameSize = audio_stream_frame_size(&mInput->stream->common); + mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); + mFrameCount = mInputBytes / mFrameSize; + mNormalFrameCount = mFrameCount; // not used by record, but used by input effects + mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; + + if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) + { + int channelCount; + // optimization: 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() +{ + Mutex::Autolock _l(mLock); + if (initCheck() != NO_ERROR) { + return 0; + } + + return mInput->stream->get_input_frames_lost(mInput->stream); +} + +uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const +{ + Mutex::Autolock _l(mLock); + uint32_t result = 0; + if (getEffectChain_l(sessionId) != 0) { + result = EFFECT_SESSION; + } + + for (size_t i = 0; i < mTracks.size(); ++i) { + if (sessionId == mTracks[i]->sessionId()) { + result |= TRACK_SESSION; + break; + } + } + + return result; +} + +KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const +{ + KeyedVector<int, bool> ids; + Mutex::Autolock _l(mLock); + for (size_t j = 0; j < mTracks.size(); ++j) { + sp<RecordThread::RecordTrack> track = mTracks[j]; + int sessionId = track->sessionId(); + if (ids.indexOfKey(sessionId) < 0) { + ids.add(sessionId, true); + } + } + return ids; +} + +AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() +{ + Mutex::Autolock _l(mLock); + AudioStreamIn *input = mInput; + mInput = NULL; + return input; +} + +// this method must always be called either with ThreadBase mLock held or inside the thread loop +audio_stream_t* AudioFlinger::RecordThread::stream() const +{ + if (mInput == NULL) { + return NULL; + } + return &mInput->stream->common; +} + +status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) +{ + // only one chain per input thread + if (mEffectChains.size() != 0) { + return INVALID_OPERATION; + } + ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); + + chain->setInBuffer(NULL); + chain->setOutBuffer(NULL); + + checkSuspendOnAddEffectChain_l(chain); + + mEffectChains.add(chain); + + return NO_ERROR; +} + +size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) +{ + ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); + ALOGW_IF(mEffectChains.size() != 1, + "removeEffectChain_l() %p invalid chain size %d on thread %p", + chain.get(), mEffectChains.size(), this); + if (mEffectChains.size() == 1) { + mEffectChains.removeAt(0); + } + return 0; +} + +}; // namespace android diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h new file mode 100644 index 0000000..06a1c8c --- /dev/null +++ b/services/audioflinger/Threads.h @@ -0,0 +1,801 @@ +/* +** +** Copyright 2012, 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. +*/ + +#ifndef INCLUDING_FROM_AUDIOFLINGER_H + #error This header file should only be included from AudioFlinger.h +#endif + +class ThreadBase : public Thread { +public: + +#include "TrackBase.h" + + enum type_t { + MIXER, // Thread class is MixerThread + DIRECT, // Thread class is DirectOutputThread + DUPLICATING, // Thread class is DuplicatingThread + RECORD // Thread class is RecordThread + }; + + ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, + audio_devices_t outDevice, audio_devices_t inDevice, type_t type); + virtual ~ThreadBase(); + + void dumpBase(int fd, const Vector<String16>& args); + void dumpEffectChains(int fd, const Vector<String16>& args); + + void clearPowerManager(); + + // base for record and playback + enum { + CFG_EVENT_IO, + CFG_EVENT_PRIO + }; + + class ConfigEvent { + public: + ConfigEvent(int type) : mType(type) {} + virtual ~ConfigEvent() {} + + int type() const { return mType; } + + virtual void dump(char *buffer, size_t size) = 0; + + private: + const int mType; + }; + + class IoConfigEvent : public ConfigEvent { + public: + IoConfigEvent(int event, int param) : + ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {} + virtual ~IoConfigEvent() {} + + int event() const { return mEvent; } + int param() const { return mParam; } + + virtual void dump(char *buffer, size_t size) { + snprintf(buffer, size, "IO event: event %d, param %d\n", mEvent, mParam); + } + + private: + const int mEvent; + const int mParam; + }; + + class PrioConfigEvent : public ConfigEvent { + public: + PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) : + ConfigEvent(CFG_EVENT_PRIO), mPid(pid), mTid(tid), mPrio(prio) {} + virtual ~PrioConfigEvent() {} + + pid_t pid() const { return mPid; } + pid_t tid() const { return mTid; } + int32_t prio() const { return mPrio; } + + virtual void dump(char *buffer, size_t size) { + snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio); + } + + private: + const pid_t mPid; + const pid_t mTid; + const int32_t mPrio; + }; + + + class PMDeathRecipient : public IBinder::DeathRecipient { + public: + PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {} + virtual ~PMDeathRecipient() {} + + // IBinder::DeathRecipient + virtual void binderDied(const wp<IBinder>& who); + + private: + PMDeathRecipient(const PMDeathRecipient&); + PMDeathRecipient& operator = (const PMDeathRecipient&); + + wp<ThreadBase> mThread; + }; + + virtual status_t initCheck() const = 0; + + // static externally-visible + type_t type() const { return mType; } + audio_io_handle_t id() const { return mId;} + + // dynamic externally-visible + uint32_t sampleRate() const { return mSampleRate; } + uint32_t channelCount() const { return mChannelCount; } + audio_channel_mask_t channelMask() const { return mChannelMask; } + audio_format_t format() const { return mFormat; } + // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, + // and returns the normal mix buffer's frame count. + size_t frameCount() const { return mNormalFrameCount; } + // Return's the HAL's frame count i.e. fast mixer buffer size. + size_t frameCountHAL() const { return mFrameCount; } + + // Should be "virtual status_t requestExitAndWait()" and override same + // method in Thread, but Thread::requestExitAndWait() is not yet virtual. + void exit(); + virtual bool checkForNewParameters_l() = 0; + virtual status_t setParameters(const String8& keyValuePairs); + virtual String8 getParameters(const String8& keys) = 0; + virtual void audioConfigChanged_l(int event, int param = 0) = 0; + void sendIoConfigEvent(int event, int param = 0); + void sendIoConfigEvent_l(int event, int param = 0); + void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio); + void processConfigEvents(); + + // see note at declaration of mStandby, mOutDevice and mInDevice + bool standby() const { return mStandby; } + audio_devices_t outDevice() const { return mOutDevice; } + audio_devices_t inDevice() const { return mInDevice; } + + virtual audio_stream_t* stream() const = 0; + + sp<EffectHandle> createEffect_l( + const sp<AudioFlinger::Client>& client, + const sp<IEffectClient>& effectClient, + int32_t priority, + int sessionId, + effect_descriptor_t *desc, + int *enabled, + status_t *status); + void disconnectEffect(const sp< EffectModule>& effect, + EffectHandle *handle, + bool unpinIfLast); + + // return values for hasAudioSession (bit field) + enum effect_state { + EFFECT_SESSION = 0x1, // the audio session corresponds to at least one + // effect + TRACK_SESSION = 0x2 // the audio session corresponds to at least one + // track + }; + + // get effect chain corresponding to session Id. + sp<EffectChain> getEffectChain(int sessionId); + // same as getEffectChain() but must be called with ThreadBase mutex locked + sp<EffectChain> getEffectChain_l(int sessionId) const; + // add an effect chain to the chain list (mEffectChains) + virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0; + // remove an effect chain from the chain list (mEffectChains) + virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0; + // lock all effect chains Mutexes. Must be called before releasing the + // ThreadBase mutex before processing the mixer and effects. This guarantees the + // integrity of the chains during the process. + // Also sets the parameter 'effectChains' to current value of mEffectChains. + void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains); + // unlock effect chains after process + void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains); + // set audio mode to all effect chains + void setMode(audio_mode_t mode); + // get effect module with corresponding ID on specified audio session + sp<AudioFlinger::EffectModule> getEffect(int sessionId, int effectId); + sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId); + // add and effect module. Also creates the effect chain is none exists for + // the effects audio session + status_t addEffect_l(const sp< EffectModule>& effect); + // remove and effect module. Also removes the effect chain is this was the last + // effect + void removeEffect_l(const sp< EffectModule>& effect); + // detach all tracks connected to an auxiliary effect + virtual void detachAuxEffect_l(int effectId) {} + // returns either EFFECT_SESSION if effects on this audio session exist in one + // chain, or TRACK_SESSION if tracks on this audio session exist, or both + virtual uint32_t hasAudioSession(int sessionId) const = 0; + // the value returned by default implementation is not important as the + // strategy is only meaningful for PlaybackThread which implements this method + virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; } + + // suspend or restore effect according to the type of effect passed. a NULL + // type pointer means suspend all effects in the session + void setEffectSuspended(const effect_uuid_t *type, + bool suspend, + int sessionId = AUDIO_SESSION_OUTPUT_MIX); + // check if some effects must be suspended/restored when an effect is enabled + // or disabled + void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, + bool enabled, + int sessionId = AUDIO_SESSION_OUTPUT_MIX); + void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, + bool enabled, + int sessionId = AUDIO_SESSION_OUTPUT_MIX); + + virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0; + virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0; + + + mutable Mutex mLock; + +protected: + + // entry describing an effect being suspended in mSuspendedSessions keyed vector + class SuspendedSessionDesc : public RefBase { + public: + SuspendedSessionDesc() : mRefCount(0) {} + + int mRefCount; // number of active suspend requests + effect_uuid_t mType; // effect type UUID + }; + + void acquireWakeLock(); + void acquireWakeLock_l(); + void releaseWakeLock(); + void releaseWakeLock_l(); + void setEffectSuspended_l(const effect_uuid_t *type, + bool suspend, + int sessionId); + // updated mSuspendedSessions when an effect suspended or restored + void updateSuspendedSessions_l(const effect_uuid_t *type, + bool suspend, + int sessionId); + // check if some effects must be suspended when an effect chain is added + void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain); + + virtual void preExit() { } + + friend class AudioFlinger; // for mEffectChains + + const type_t mType; + + // Used by parameters, config events, addTrack_l, exit + Condition mWaitWorkCV; + + const sp<AudioFlinger> mAudioFlinger; + uint32_t mSampleRate; + size_t mFrameCount; // output HAL, direct output, record + size_t mNormalFrameCount; // normal mixer and effects + audio_channel_mask_t mChannelMask; + uint16_t mChannelCount; + size_t mFrameSize; + audio_format_t mFormat; + + // Parameter sequence by client: binder thread calling setParameters(): + // 1. Lock mLock + // 2. Append to mNewParameters + // 3. mWaitWorkCV.signal + // 4. mParamCond.waitRelative with timeout + // 5. read mParamStatus + // 6. mWaitWorkCV.signal + // 7. Unlock + // + // Parameter sequence by server: threadLoop calling checkForNewParameters_l(): + // 1. Lock mLock + // 2. If there is an entry in mNewParameters proceed ... + // 2. Read first entry in mNewParameters + // 3. Process + // 4. Remove first entry from mNewParameters + // 5. Set mParamStatus + // 6. mParamCond.signal + // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus) + // 8. Unlock + Condition mParamCond; + Vector<String8> mNewParameters; + status_t mParamStatus; + + Vector<ConfigEvent *> mConfigEvents; + + // These fields are written and read by thread itself without lock or barrier, + // and read by other threads without lock or barrier via standby() , outDevice() + // and inDevice(). + // Because of the absence of a lock or barrier, any other thread that reads + // these fields must use the information in isolation, or be prepared to deal + // with possibility that it might be inconsistent with other information. + bool mStandby; // Whether thread is currently in standby. + audio_devices_t mOutDevice; // output device + audio_devices_t mInDevice; // input device + audio_source_t mAudioSource; // (see audio.h, audio_source_t) + + const audio_io_handle_t mId; + Vector< sp<EffectChain> > mEffectChains; + + static const int kNameLength = 16; // prctl(PR_SET_NAME) limit + char mName[kNameLength]; + sp<IPowerManager> mPowerManager; + sp<IBinder> mWakeLockToken; + const sp<PMDeathRecipient> mDeathRecipient; + // list of suspended effects per session and per type. The first vector is + // keyed by session ID, the second by type UUID timeLow field + KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > + mSuspendedSessions; +}; + +// --- PlaybackThread --- +class PlaybackThread : public ThreadBase { +public: + +#include "PlaybackTracks.h" + + enum mixer_state { + MIXER_IDLE, // no active tracks + MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready + MIXER_TRACKS_READY // at least one active track, and at least one track has data + // standby mode does not have an enum value + // suspend by audio policy manager is orthogonal to mixer state + }; + + PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, + audio_io_handle_t id, audio_devices_t device, type_t type); + virtual ~PlaybackThread(); + + void dump(int fd, const Vector<String16>& args); + + // Thread virtuals + virtual status_t readyToRun(); + virtual bool threadLoop(); + + // RefBase + virtual void onFirstRef(); + +protected: + // Code snippets that were lifted up out of threadLoop() + virtual void threadLoop_mix() = 0; + virtual void threadLoop_sleepTime() = 0; + virtual void threadLoop_write(); + virtual void threadLoop_standby(); + virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); + + // prepareTracks_l reads and writes mActiveTracks, and returns + // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller + // is responsible for clearing or destroying this Vector later on, when it + // is safe to do so. That will drop the final ref count and destroy the tracks. + virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0; + + // ThreadBase virtuals + virtual void preExit(); + +public: + + virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } + + // return estimated latency in milliseconds, as reported by HAL + uint32_t latency() const; + // same, but lock must already be held + uint32_t latency_l() const; + + void setMasterVolume(float value); + void setMasterMute(bool muted); + + void setStreamVolume(audio_stream_type_t stream, float value); + void setStreamMute(audio_stream_type_t stream, bool muted); + + float streamVolume(audio_stream_type_t stream) const; + + sp<Track> createTrack_l( + const sp<AudioFlinger::Client>& client, + audio_stream_type_t streamType, + uint32_t sampleRate, + audio_format_t format, + audio_channel_mask_t channelMask, + size_t frameCount, + const sp<IMemory>& sharedBuffer, + int sessionId, + IAudioFlinger::track_flags_t *flags, + pid_t tid, + status_t *status); + + AudioStreamOut* getOutput() const; + AudioStreamOut* clearOutput(); + virtual audio_stream_t* stream() const; + + // a very large number of suspend() will eventually wraparound, but unlikely + void suspend() { (void) android_atomic_inc(&mSuspended); } + void restore() + { + // if restore() is done without suspend(), get back into + // range so that the next suspend() will operate correctly + if (android_atomic_dec(&mSuspended) <= 0) { + android_atomic_release_store(0, &mSuspended); + } + } + bool isSuspended() const + { return android_atomic_acquire_load(&mSuspended) > 0; } + + virtual String8 getParameters(const String8& keys); + virtual void audioConfigChanged_l(int event, int param = 0); + status_t getRenderPosition(size_t *halFrames, size_t *dspFrames); + int16_t *mixBuffer() const { return mMixBuffer; }; + + virtual void detachAuxEffect_l(int effectId); + status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, + int EffectId); + status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, + int EffectId); + + virtual status_t addEffectChain_l(const sp<EffectChain>& chain); + virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); + virtual uint32_t hasAudioSession(int sessionId) const; + virtual uint32_t getStrategyForSession_l(int sessionId); + + + virtual status_t setSyncEvent(const sp<SyncEvent>& event); + virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; + void invalidateTracks(audio_stream_type_t streamType); + + +protected: + int16_t* mMixBuffer; + + // suspend count, > 0 means suspended. While suspended, the thread continues to pull from + // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle + // concurrent use of both of them, so Audio Policy Service suspends one of the threads to + // workaround that restriction. + // 'volatile' means accessed via atomic operations and no lock. + volatile int32_t mSuspended; + + // FIXME overflows every 6+ hours at 44.1 kHz stereo 16-bit samples + // mFramesWritten would be better, or 64-bit even better + size_t mBytesWritten; +private: + // mMasterMute is in both PlaybackThread and in AudioFlinger. When a + // PlaybackThread needs to find out if master-muted, it checks it's local + // copy rather than the one in AudioFlinger. This optimization saves a lock. + bool mMasterMute; + void setMasterMute_l(bool muted) { mMasterMute = muted; } +protected: + SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<> + + // Allocate a track name for a given channel mask. + // Returns name >= 0 if successful, -1 on failure. + virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0; + virtual void deleteTrackName_l(int name) = 0; + + // Time to sleep between cycles when: + virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED + virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE + virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us + // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() + // No sleep in standby mode; waits on a condition + + // Code snippets that are temporarily lifted up out of threadLoop() until the merge + void checkSilentMode_l(); + + // Non-trivial for DUPLICATING only + virtual void saveOutputTracks() { } + virtual void clearOutputTracks() { } + + // Cache various calculated values, at threadLoop() entry and after a parameter change + virtual void cacheParameters_l(); + + virtual uint32_t correctLatency_l(uint32_t latency) const; + +private: + + friend class AudioFlinger; // for numerous + + PlaybackThread(const Client&); + PlaybackThread& operator = (const PlaybackThread&); + + status_t addTrack_l(const sp<Track>& track); + void destroyTrack_l(const sp<Track>& track); + void removeTrack_l(const sp<Track>& track); + + void readOutputParameters(); + + virtual void dumpInternals(int fd, const Vector<String16>& args); + void dumpTracks(int fd, const Vector<String16>& args); + + SortedVector< sp<Track> > mTracks; + // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by + // DuplicatingThread + stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1]; + AudioStreamOut *mOutput; + + float mMasterVolume; + nsecs_t mLastWriteTime; + int mNumWrites; + int mNumDelayedWrites; + bool mInWrite; + + // FIXME rename these former local variables of threadLoop to standard "m" names + nsecs_t standbyTime; + size_t mixBufferSize; + + // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() + uint32_t activeSleepTime; + uint32_t idleSleepTime; + + uint32_t sleepTime; + + // mixer status returned by prepareTracks_l() + mixer_state mMixerStatus; // current cycle + // previous cycle when in prepareTracks_l() + mixer_state mMixerStatusIgnoringFastTracks; + // FIXME or a separate ready state per track + + // FIXME move these declarations into the specific sub-class that needs them + // MIXER only + uint32_t sleepTimeShift; + + // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value + nsecs_t standbyDelay; + + // MIXER only + nsecs_t maxPeriod; + + // DUPLICATING only + uint32_t writeFrames; + +private: + // The HAL output sink is treated as non-blocking, but current implementation is blocking + sp<NBAIO_Sink> mOutputSink; + // If a fast mixer is present, the blocking pipe sink, otherwise clear + sp<NBAIO_Sink> mPipeSink; + // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink + sp<NBAIO_Sink> mNormalSink; + // For dumpsys + sp<NBAIO_Sink> mTeeSink; + sp<NBAIO_Source> mTeeSource; + uint32_t mScreenState; // cached copy of gScreenState +public: + virtual bool hasFastMixer() const = 0; + virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const + { FastTrackUnderruns dummy; return dummy; } + +protected: + // accessed by both binder threads and within threadLoop(), lock on mutex needed + unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available + +}; + +class MixerThread : public PlaybackThread { +public: + MixerThread(const sp<AudioFlinger>& audioFlinger, + AudioStreamOut* output, + audio_io_handle_t id, + audio_devices_t device, + type_t type = MIXER); + virtual ~MixerThread(); + + // Thread virtuals + + virtual bool checkForNewParameters_l(); + virtual void dumpInternals(int fd, const Vector<String16>& args); + +protected: + virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); + virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); + virtual void deleteTrackName_l(int name); + virtual uint32_t idleSleepTimeUs() const; + virtual uint32_t suspendSleepTimeUs() const; + virtual void cacheParameters_l(); + + // threadLoop snippets + virtual void threadLoop_write(); + virtual void threadLoop_standby(); + virtual void threadLoop_mix(); + virtual void threadLoop_sleepTime(); + virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove); + virtual uint32_t correctLatency_l(uint32_t latency) const; + + AudioMixer* mAudioMixer; // normal mixer +private: + // one-time initialization, no locks required + FastMixer* mFastMixer; // non-NULL if there is also a fast mixer + sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread + + // contents are not guaranteed to be consistent, no locks required + FastMixerDumpState mFastMixerDumpState; +#ifdef STATE_QUEUE_DUMP + StateQueueObserverDump mStateQueueObserverDump; + StateQueueMutatorDump mStateQueueMutatorDump; +#endif + AudioWatchdogDump mAudioWatchdogDump; + + // accessible only within the threadLoop(), no locks required + // mFastMixer->sq() // for mutating and pushing state + int32_t mFastMixerFutex; // for cold idle + +public: + virtual bool hasFastMixer() const { return mFastMixer != NULL; } + virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { + ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks); + return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; + } +}; + +class DirectOutputThread : public PlaybackThread { +public: + + DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, + audio_io_handle_t id, audio_devices_t device); + virtual ~DirectOutputThread(); + + // Thread virtuals + + virtual bool checkForNewParameters_l(); + +protected: + virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId); + virtual void deleteTrackName_l(int name); + virtual uint32_t activeSleepTimeUs() const; + virtual uint32_t idleSleepTimeUs() const; + virtual uint32_t suspendSleepTimeUs() const; + virtual void cacheParameters_l(); + + // threadLoop snippets + virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove); + virtual void threadLoop_mix(); + virtual void threadLoop_sleepTime(); + +private: + // volumes last sent to audio HAL with stream->set_volume() + float mLeftVolFloat; + float mRightVolFloat; + + // prepareTracks_l() tells threadLoop_mix() the name of the single active track + sp<Track> mActiveTrack; +public: + virtual bool hasFastMixer() const { return false; } +}; + +class DuplicatingThread : public MixerThread { +public: + DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, + audio_io_handle_t id); + virtual ~DuplicatingThread(); + + // Thread virtuals + void addOutputTrack(MixerThread* thread); + void removeOutputTrack(MixerThread* thread); + uint32_t waitTimeMs() const { return mWaitTimeMs; } +protected: + virtual uint32_t activeSleepTimeUs() const; + +private: + bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks); +protected: + // threadLoop snippets + virtual void threadLoop_mix(); + virtual void threadLoop_sleepTime(); + virtual void threadLoop_write(); + virtual void threadLoop_standby(); + virtual void cacheParameters_l(); + +private: + // called from threadLoop, addOutputTrack, removeOutputTrack + virtual void updateWaitTime_l(); +protected: + virtual void saveOutputTracks(); + virtual void clearOutputTracks(); +private: + + uint32_t mWaitTimeMs; + SortedVector < sp<OutputTrack> > outputTracks; + SortedVector < sp<OutputTrack> > mOutputTracks; +public: + virtual bool hasFastMixer() const { return false; } +}; + + +// record thread +class RecordThread : public ThreadBase, public AudioBufferProvider + // derives from AudioBufferProvider interface for use by resampler +{ +public: + +#include "RecordTracks.h" + + RecordThread(const sp<AudioFlinger>& audioFlinger, + AudioStreamIn *input, + uint32_t sampleRate, + audio_channel_mask_t channelMask, + audio_io_handle_t id, + audio_devices_t device, + const sp<NBAIO_Sink>& teeSink); + virtual ~RecordThread(); + + // no addTrack_l ? + void destroyTrack_l(const sp<RecordTrack>& track); + void removeTrack_l(const sp<RecordTrack>& track); + + void dumpInternals(int fd, const Vector<String16>& args); + void dumpTracks(int fd, const Vector<String16>& args); + + // Thread virtuals + virtual bool threadLoop(); + virtual status_t readyToRun(); + + // RefBase + virtual void onFirstRef(); + + virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } + sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l( + const sp<AudioFlinger::Client>& client, + uint32_t sampleRate, + audio_format_t format, + audio_channel_mask_t channelMask, + size_t frameCount, + int sessionId, + IAudioFlinger::track_flags_t flags, + pid_t tid, + status_t *status); + + status_t start(RecordTrack* recordTrack, + AudioSystem::sync_event_t event, + int triggerSession); + + // ask the thread to stop the specified track, and + // return true if the caller should then do it's part of the stopping process + bool stop_l(RecordTrack* recordTrack); + + void dump(int fd, const Vector<String16>& args); + AudioStreamIn* clearInput(); + virtual audio_stream_t* stream() const; + + // AudioBufferProvider interface + virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts); + virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); + + virtual bool checkForNewParameters_l(); + virtual String8 getParameters(const String8& keys); + virtual void audioConfigChanged_l(int event, int param = 0); + void readInputParameters(); + virtual unsigned int getInputFramesLost(); + + virtual status_t addEffectChain_l(const sp<EffectChain>& chain); + virtual size_t removeEffectChain_l(const sp<EffectChain>& chain); + virtual uint32_t hasAudioSession(int sessionId) const; + + // Return the set of unique session IDs across all tracks. + // The keys are the session IDs, and the associated values are meaningless. + // FIXME replace by Set [and implement Bag/Multiset for other uses]. + KeyedVector<int, bool> sessionIds() const; + + virtual status_t setSyncEvent(const sp<SyncEvent>& event); + virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const; + + static void syncStartEventCallback(const wp<SyncEvent>& event); + void handleSyncStartEvent(const sp<SyncEvent>& event); + +private: + void clearSyncStartEvent(); + + // Enter standby if not already in standby, and set mStandby flag + void standby(); + + // Call the HAL standby method unconditionally, and don't change mStandby flag + void inputStandBy(); + + AudioStreamIn *mInput; + SortedVector < sp<RecordTrack> > mTracks; + // mActiveTrack has dual roles: it indicates the current active track, and + // is used together with mStartStopCond to indicate start()/stop() progress + sp<RecordTrack> mActiveTrack; + Condition mStartStopCond; + AudioResampler *mResampler; + int32_t *mRsmpOutBuffer; + int16_t *mRsmpInBuffer; + size_t mRsmpInIndex; + size_t mInputBytes; + const uint32_t mReqChannelCount; + const uint32_t mReqSampleRate; + ssize_t mBytesRead; + // sync event triggering actual audio capture. Frames read before this event will + // be dropped and therefore not read by the application. + sp<SyncEvent> mSyncStartEvent; + // number of captured frames to drop after the start sync event has been received. + // when < 0, maximum frames to drop before starting capture even if sync event is + // not received + ssize_t mFramestoDrop; + + // For dumpsys + const sp<NBAIO_Sink> mTeeSink; +}; diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h new file mode 100644 index 0000000..17de49b --- /dev/null +++ b/services/audioflinger/TrackBase.h @@ -0,0 +1,139 @@ +/* +** +** Copyright 2012, 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. +*/ + +#ifndef INCLUDING_FROM_AUDIOFLINGER_H + #error This header file should only be included from AudioFlinger.h +#endif + +// base for record and playback +class TrackBase : public ExtendedAudioBufferProvider, public RefBase { + +public: + enum track_state { + IDLE, + TERMINATED, + FLUSHED, + STOPPED, + // next 2 states are currently used for fast tracks only + STOPPING_1, // waiting for first underrun + STOPPING_2, // waiting for presentation complete + RESUMING, + ACTIVE, + PAUSING, + PAUSED + }; + + TrackBase(ThreadBase *thread, + const sp<Client>& client, + uint32_t sampleRate, + audio_format_t format, + audio_channel_mask_t channelMask, + size_t frameCount, + const sp<IMemory>& sharedBuffer, + int sessionId); + virtual ~TrackBase(); + + virtual status_t start(AudioSystem::sync_event_t event, + int triggerSession) = 0; + virtual void stop() = 0; + sp<IMemory> getCblk() const { return mCblkMemory; } + audio_track_cblk_t* cblk() const { return mCblk; } + int sessionId() const { return mSessionId; } + virtual status_t setSyncEvent(const sp<SyncEvent>& event); + +protected: + TrackBase(const TrackBase&); + TrackBase& operator = (const TrackBase&); + + // AudioBufferProvider interface + virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; + virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); + + // ExtendedAudioBufferProvider interface is only needed for Track, + // but putting it in TrackBase avoids the complexity of virtual inheritance + virtual size_t framesReady() const { return SIZE_MAX; } + + audio_format_t format() const { + return mFormat; + } + + uint32_t channelCount() const { return mChannelCount; } + + audio_channel_mask_t channelMask() const { return mChannelMask; } + + uint32_t sampleRate() const; // FIXME inline after cblk sr moved + + // Return a pointer to the start of a contiguous slice of the track buffer. + // Parameter 'offset' is the requested start position, expressed in + // monotonically increasing frame units relative to the track epoch. + // Parameter 'frames' is the requested length, also in frame units. + // Always returns non-NULL. It is the caller's responsibility to + // verify that this will be successful; the result of calling this + // function with invalid 'offset' or 'frames' is undefined. + void* getBuffer(uint32_t offset, uint32_t frames) const; + + bool isStopped() const { + return (mState == STOPPED || mState == FLUSHED); + } + + // for fast tracks only + bool isStopping() const { + return mState == STOPPING_1 || mState == STOPPING_2; + } + bool isStopping_1() const { + return mState == STOPPING_1; + } + bool isStopping_2() const { + return mState == STOPPING_2; + } + + bool isTerminated() const { + return mState == TERMINATED; + } + + bool step(); // mStepCount is an implicit input + void reset(); + + virtual bool isOut() const = 0; // true for Track and TimedTrack, false for RecordTrack, + // this could be a track type if needed later + + const wp<ThreadBase> mThread; + /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const + sp<IMemory> mCblkMemory; + audio_track_cblk_t* mCblk; + void* mBuffer; // start of track buffer, typically in shared memory + void* mBufferEnd; // &mBuffer[mFrameCount * frameSize], where frameSize + // is based on mChannelCount and 16-bit samples + uint32_t mStepCount; // saves AudioBufferProvider::Buffer::frameCount as of + // time of releaseBuffer() for later use by step() + // we don't really need a lock for these + track_state mState; + const uint32_t mSampleRate; // initial sample rate only; for tracks which + // support dynamic rates, the current value is in control block + const audio_format_t mFormat; + const audio_channel_mask_t mChannelMask; + const uint8_t mChannelCount; + const size_t mFrameSize; // AudioFlinger's view of frame size in shared memory, + // where for AudioTrack (but not AudioRecord), + // 8-bit PCM samples are stored as 16-bit + const size_t mFrameCount;// size of track buffer given at createTrack() or + // openRecord(), and then adjusted as needed + + bool mStepServerFailed; + const int mSessionId; + Vector < sp<SyncEvent> >mSyncEvents; +}; diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp new file mode 100644 index 0000000..2c6ba8b --- /dev/null +++ b/services/audioflinger/Tracks.cpp @@ -0,0 +1,1789 @@ +/* +** +** Copyright 2012, 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 <cutils/compiler.h> +#include <utils/Log.h> + +#include <private/media/AudioTrackShared.h> + +#include <common_time/cc_helper.h> +#include <common_time/local_clock.h> + +#include "AudioMixer.h" +#include "AudioFlinger.h" +#include "ServiceUtilities.h" + +// ---------------------------------------------------------------------------- + +// Note: the following macro is used for extremely verbose logging message. In +// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to +// 0; but one side effect of this is to turn all LOGV's as well. Some messages +// are so verbose that we want to suppress them even when we have ALOG_ASSERT +// turned on. Do not uncomment the #def below unless you really know what you +// are doing and want to see all of the extremely verbose messages. +//#define VERY_VERY_VERBOSE_LOGGING +#ifdef VERY_VERY_VERBOSE_LOGGING +#define ALOGVV ALOGV +#else +#define ALOGVV(a...) do { } while(0) +#endif + +namespace android { + +// ---------------------------------------------------------------------------- +// TrackBase +// ---------------------------------------------------------------------------- + +// TrackBase constructor must be called with AudioFlinger::mLock held +AudioFlinger::ThreadBase::TrackBase::TrackBase( + ThreadBase *thread, + const sp<Client>& client, + uint32_t sampleRate, + audio_format_t format, + audio_channel_mask_t channelMask, + size_t frameCount, + const sp<IMemory>& sharedBuffer, + int sessionId) + : RefBase(), + mThread(thread), + mClient(client), + mCblk(NULL), + // mBuffer + // mBufferEnd + mStepCount(0), + mState(IDLE), + mSampleRate(sampleRate), + mFormat(format), + mChannelMask(channelMask), + mChannelCount(popcount(channelMask)), + mFrameSize(audio_is_linear_pcm(format) ? + mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), + mFrameCount(frameCount), + mStepServerFailed(false), + mSessionId(sessionId) +{ + // client == 0 implies sharedBuffer == 0 + ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); + + ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), + sharedBuffer->size()); + + // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); + size_t size = sizeof(audio_track_cblk_t); + size_t bufferSize = frameCount * mFrameSize; + if (sharedBuffer == 0) { + size += bufferSize; + } + + if (client != 0) { + mCblkMemory = client->heap()->allocate(size); + if (mCblkMemory != 0) { + mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); + // can't assume mCblk != NULL + } else { + ALOGE("not enough memory for AudioTrack size=%u", size); + client->heap()->dump("AudioTrack"); + return; + } + } else { + mCblk = (audio_track_cblk_t *)(new uint8_t[size]); + // assume mCblk != NULL + } + + // construct the shared structure in-place. + if (mCblk != NULL) { + new(mCblk) audio_track_cblk_t(); + // clear all buffers + mCblk->frameCount_ = frameCount; + mCblk->sampleRate = sampleRate; +// uncomment the following lines to quickly test 32-bit wraparound +// mCblk->user = 0xffff0000; +// mCblk->server = 0xffff0000; +// mCblk->userBase = 0xffff0000; +// mCblk->serverBase = 0xffff0000; + if (sharedBuffer == 0) { + mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); + memset(mBuffer, 0, bufferSize); + // Force underrun condition to avoid false underrun callback until first data is + // written to buffer (other flags are cleared) + mCblk->flags = CBLK_UNDERRUN; + } else { + mBuffer = sharedBuffer->pointer(); + } + mBufferEnd = (uint8_t *)mBuffer + bufferSize; + } +} + +AudioFlinger::ThreadBase::TrackBase::~TrackBase() +{ + if (mCblk != NULL) { + if (mClient == 0) { + delete mCblk; + } else { + mCblk->~audio_track_cblk_t(); // destroy our shared-structure. + } + } + mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to + if (mClient != 0) { + // Client destructor must run with AudioFlinger mutex locked + Mutex::Autolock _l(mClient->audioFlinger()->mLock); + // If the client's reference count drops to zero, the associated destructor + // must run with AudioFlinger lock held. Thus the explicit clear() rather than + // relying on the automatic clear() at end of scope. + mClient.clear(); + } +} + +// AudioBufferProvider interface +// getNextBuffer() = 0; +// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack +void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) +{ + buffer->raw = NULL; + mStepCount = buffer->frameCount; + // FIXME See note at getNextBuffer() + (void) step(); // ignore return value of step() + buffer->frameCount = 0; +} + +bool AudioFlinger::ThreadBase::TrackBase::step() { + bool result; + audio_track_cblk_t* cblk = this->cblk(); + + result = cblk->stepServer(mStepCount, mFrameCount, isOut()); + if (!result) { + ALOGV("stepServer failed acquiring cblk mutex"); + mStepServerFailed = true; + } + 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; + mStepServerFailed = false; + ALOGV("TrackBase::reset"); +} + +uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const { + return mCblk->sampleRate; +} + +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) * mFrameSize; + int8_t *bufferEnd = bufferStart + frames * mFrameSize; + + // Check validity of returned pointer in case the track control block would have been corrupted. + ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), + "TrackBase::getBuffer buffer out of range:\n" + " start: %p, end %p , mBuffer %p mBufferEnd %p\n" + " server %u, serverBase %u, user %u, userBase %u, frameSize %u", + bufferStart, bufferEnd, mBuffer, mBufferEnd, + cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize); + + return bufferStart; +} + +status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) +{ + mSyncEvents.add(event); + return NO_ERROR; +} + +// ---------------------------------------------------------------------------- +// Playback +// ---------------------------------------------------------------------------- + +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(); +} + +sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { + return mTrack->getCblk(); +} + +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(); +} + +status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) +{ + return mTrack->attachAuxEffect(EffectId); +} + +status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, + sp<IMemory>* buffer) { + if (!mTrack->isTimedTrack()) + return INVALID_OPERATION; + + PlaybackThread::TimedTrack* tt = + reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); + return tt->allocateTimedBuffer(size, buffer); +} + +status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, + int64_t pts) { + if (!mTrack->isTimedTrack()) + return INVALID_OPERATION; + + PlaybackThread::TimedTrack* tt = + reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); + return tt->queueTimedBuffer(buffer, pts); +} + +status_t AudioFlinger::TrackHandle::setMediaTimeTransform( + const LinearTransform& xform, int target) { + + if (!mTrack->isTimedTrack()) + return INVALID_OPERATION; + + PlaybackThread::TimedTrack* tt = + reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); + return tt->setMediaTimeTransform( + xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); +} + +status_t AudioFlinger::TrackHandle::onTransact( + uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) +{ + return BnAudioTrack::onTransact(code, data, reply, flags); +} + +// ---------------------------------------------------------------------------- + +// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held +AudioFlinger::PlaybackThread::Track::Track( + PlaybackThread *thread, + const sp<Client>& client, + audio_stream_type_t streamType, + uint32_t sampleRate, + audio_format_t format, + audio_channel_mask_t channelMask, + size_t frameCount, + const sp<IMemory>& sharedBuffer, + int sessionId, + IAudioFlinger::track_flags_t flags) + : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, + sessionId), + mMute(false), + mFillingUpStatus(FS_INVALID), + // mRetryCount initialized later when needed + mSharedBuffer(sharedBuffer), + mStreamType(streamType), + mName(-1), // see note below + mMainBuffer(thread->mixBuffer()), + mAuxBuffer(NULL), + mAuxEffectId(0), mHasVolumeController(false), + mPresentationCompleteFrames(0), + mFlags(flags), + mFastIndex(-1), + mUnderrunCount(0), + mCachedVolume(1.0) +{ + if (mCblk != NULL) { + // to avoid leaking a track name, do not allocate one unless there is an mCblk + mName = thread->getTrackName_l(channelMask, sessionId); + mCblk->mName = mName; + if (mName < 0) { + ALOGE("no more track names available"); + return; + } + // only allocate a fast track index if we were able to allocate a normal track name + if (flags & IAudioFlinger::TRACK_FAST) { + ALOG_ASSERT(thread->mFastTrackAvailMask != 0); + int i = __builtin_ctz(thread->mFastTrackAvailMask); + ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); + // FIXME This is too eager. We allocate a fast track index before the + // fast track becomes active. Since fast tracks are a scarce resource, + // this means we are potentially denying other more important fast tracks from + // being created. It would be better to allocate the index dynamically. + mFastIndex = i; + mCblk->mName = i; + // Read the initial underruns because this field is never cleared by the fast mixer + mObservedUnderruns = thread->getFastTrackUnderruns(i); + thread->mFastTrackAvailMask &= ~(1 << i); + } + } + ALOGV("Track constructor name %d, calling pid %d", mName, + IPCThreadState::self()->getCallingPid()); +} + +AudioFlinger::PlaybackThread::Track::~Track() +{ + ALOGV("PlaybackThread::Track destructor"); +} + +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 + // destructor 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(), mStreamType, mSessionId); + +#ifdef ADD_BATTERY_DATA + // to track the speaker usage + addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); +#endif + } + AudioSystem::releaseOutput(thread->id()); + } + Mutex::Autolock _l(thread->mLock); + PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); + playbackThread->destroyTrack_l(this); + } + } +} + +/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) +{ + result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S M F SRate " + "L dB R dB Server User Main buf Aux Buf Flags Underruns\n"); +} + +void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) +{ + uint32_t vlr = mCblk->getVolumeLR(); + if (isFastTrack()) { + sprintf(buffer, " F %2d", mFastIndex); + } else { + sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); + } + track_state state = mState; + char stateChar; + switch (state) { + case IDLE: + stateChar = 'I'; + break; + case TERMINATED: + stateChar = 'T'; + break; + case STOPPING_1: + stateChar = 's'; + break; + case STOPPING_2: + stateChar = '5'; + break; + case STOPPED: + stateChar = 'S'; + break; + case RESUMING: + stateChar = 'R'; + break; + case ACTIVE: + stateChar = 'A'; + break; + case PAUSING: + stateChar = 'p'; + break; + case PAUSED: + stateChar = 'P'; + break; + case FLUSHED: + stateChar = 'F'; + break; + default: + stateChar = '?'; + break; + } + char nowInUnderrun; + switch (mObservedUnderruns.mBitFields.mMostRecent) { + case UNDERRUN_FULL: + nowInUnderrun = ' '; + break; + case UNDERRUN_PARTIAL: + nowInUnderrun = '<'; + break; + case UNDERRUN_EMPTY: + nowInUnderrun = '*'; + break; + default: + nowInUnderrun = '?'; + break; + } + snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " + "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", + (mClient == 0) ? getpid_cached : mClient->pid(), + mStreamType, + mFormat, + mChannelMask, + mSessionId, + mStepCount, + mFrameCount, + stateChar, + mMute, + mFillingUpStatus, + mCblk->sampleRate, + 20.0 * log10((vlr & 0xFFFF) / 4096.0), + 20.0 * log10((vlr >> 16) / 4096.0), + mCblk->server, + mCblk->user, + (int)mMainBuffer, + (int)mAuxBuffer, + mCblk->flags, + mUnderrunCount, + nowInUnderrun); +} + +// AudioBufferProvider interface +status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( + AudioBufferProvider::Buffer* buffer, int64_t pts) +{ + 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 (mStepServerFailed) { + // FIXME When called by fast mixer, this takes a mutex with tryLock(). + // Since the fast mixer is higher priority than client callback thread, + // it does not result in priority inversion for client. + // But a non-blocking solution would be preferable to avoid + // fast mixer being unable to tryLock(), and + // to avoid the extra context switches if the client wakes up, + // discovers the mutex is locked, then has to wait for fast mixer to unlock. + if (!step()) goto getNextBuffer_exit; + ALOGV("stepServer recovered"); + mStepServerFailed = false; + } + + // FIXME Same as above + framesReady = cblk->framesReadyOut(); + + if (CC_LIKELY(framesReady)) { + uint32_t s = cblk->server; + uint32_t bufferEnd = cblk->serverBase + mFrameCount; + + bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; + if (framesReq > framesReady) { + framesReq = framesReady; + } + if (framesReq > bufferEnd - s) { + framesReq = bufferEnd - s; + } + + buffer->raw = getBuffer(s, framesReq); + buffer->frameCount = framesReq; + return NO_ERROR; + } + +getNextBuffer_exit: + buffer->raw = NULL; + buffer->frameCount = 0; + ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); + return NOT_ENOUGH_DATA; +} + +// Note that framesReady() takes a mutex on the control block using tryLock(). +// This could result in priority inversion if framesReady() is called by the normal mixer, +// as the normal mixer thread runs at lower +// priority than the client's callback thread: there is a short window within framesReady() +// during which the normal mixer could be preempted, and the client callback would block. +// Another problem can occur if framesReady() is called by the fast mixer: +// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. +// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. +size_t AudioFlinger::PlaybackThread::Track::framesReady() const { + return mCblk->framesReadyOut(); +} + +// Don't call for fast tracks; the framesReady() could result in priority inversion +bool AudioFlinger::PlaybackThread::Track::isReady() const { + if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { + return true; + } + + if (framesReady() >= mFrameCount || + (mCblk->flags & CBLK_FORCEREADY)) { + mFillingUpStatus = FS_FILLED; + android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); + return true; + } + return false; +} + +status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, + int triggerSession) +{ + status_t status = NO_ERROR; + ALOGV("start(%d), calling pid %d session %d", + mName, IPCThreadState::self()->getCallingPid(), mSessionId); + + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + Mutex::Autolock _l(thread->mLock); + track_state state = mState; + // here the track could be either new, or restarted + // in both cases "unstop" the track + if (mState == PAUSED) { + mState = TrackBase::RESUMING; + ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); + } else { + mState = TrackBase::ACTIVE; + ALOGV("? => ACTIVE (%d) on thread %p", mName, this); + } + + if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { + thread->mLock.unlock(); + status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); + thread->mLock.lock(); + +#ifdef ADD_BATTERY_DATA + // to track the speaker usage + if (status == NO_ERROR) { + addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); + } +#endif + } + if (status == NO_ERROR) { + PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); + playbackThread->addTrack_l(this); + } else { + mState = state; + triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); + } + } else { + status = BAD_VALUE; + } + return status; +} + +void AudioFlinger::PlaybackThread::Track::stop() +{ + ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + Mutex::Autolock _l(thread->mLock); + track_state state = mState; + if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { + // If the track is not active (PAUSED and buffers full), flush buffers + PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); + if (playbackThread->mActiveTracks.indexOf(this) < 0) { + reset(); + mState = STOPPED; + } else if (!isFastTrack()) { + mState = STOPPED; + } else { + // prepareTracks_l() will set state to STOPPING_2 after next underrun, + // and then to STOPPED and reset() when presentation is complete + mState = STOPPING_1; + } + ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, + playbackThread); + } + if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { + thread->mLock.unlock(); + AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); + thread->mLock.lock(); + +#ifdef ADD_BATTERY_DATA + // to track the speaker usage + addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); +#endif + } + } +} + +void AudioFlinger::PlaybackThread::Track::pause() +{ + ALOGV("pause(%d), calling pid %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; + ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); + if (!isOutputTrack()) { + thread->mLock.unlock(); + AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); + thread->mLock.lock(); + +#ifdef ADD_BATTERY_DATA + // to track the speaker usage + addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); +#endif + } + } + } +} + +void AudioFlinger::PlaybackThread::Track::flush() +{ + ALOGV("flush(%d)", mName); + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + Mutex::Autolock _l(thread->mLock); + if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && + mState != PAUSING && mState != IDLE && mState != FLUSHED) { + return; + } + // No point remaining in PAUSED state after a flush => go to + // FLUSHED state + mState = FLUSHED; + // do not reset the track if it is still in the process of being stopped or paused. + // this will be done by prepareTracks_l() when the track is stopped. + // prepareTracks_l() will see mState == FLUSHED, then + // remove from active track list, reset(), and trigger presentation complete + PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); + if (playbackThread->mActiveTracks.indexOf(this) < 0) { + reset(); + } + } +} + +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 + android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); + android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); + mFillingUpStatus = FS_FILLING; + mResetDone = true; + if (mState == FLUSHED) { + mState = IDLE; + } + } +} + +void AudioFlinger::PlaybackThread::Track::mute(bool muted) +{ + mMute = muted; +} + +status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) +{ + status_t status = DEAD_OBJECT; + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); + sp<AudioFlinger> af = mClient->audioFlinger(); + + Mutex::Autolock _l(af->mLock); + + sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); + + if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { + Mutex::Autolock _dl(playbackThread->mLock); + Mutex::Autolock _sl(srcThread->mLock); + sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); + if (chain == 0) { + return INVALID_OPERATION; + } + + sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); + if (effect == 0) { + return INVALID_OPERATION; + } + srcThread->removeEffect_l(effect); + playbackThread->addEffect_l(effect); + // removeEffect_l() has stopped the effect if it was active so it must be restarted + if (effect->state() == EffectModule::ACTIVE || + effect->state() == EffectModule::STOPPING) { + effect->start(); + } + + sp<EffectChain> dstChain = effect->chain().promote(); + if (dstChain == 0) { + srcThread->addEffect_l(effect); + return INVALID_OPERATION; + } + AudioSystem::unregisterEffect(effect->id()); + AudioSystem::registerEffect(&effect->desc(), + srcThread->id(), + dstChain->strategy(), + AUDIO_SESSION_OUTPUT_MIX, + effect->id()); + } + status = playbackThread->attachAuxEffect(this, EffectId); + } + return status; +} + +void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) +{ + mAuxEffectId = EffectId; + mAuxBuffer = buffer; +} + +bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, + size_t audioHalFrames) +{ + // a track is considered presented when the total number of frames written to audio HAL + // corresponds to the number of frames written when presentationComplete() is called for the + // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. + if (mPresentationCompleteFrames == 0) { + mPresentationCompleteFrames = framesWritten + audioHalFrames; + ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", + mPresentationCompleteFrames, audioHalFrames); + } + if (framesWritten >= mPresentationCompleteFrames) { + ALOGV("presentationComplete() session %d complete: framesWritten %d", + mSessionId, framesWritten); + triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); + return true; + } + return false; +} + +void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) +{ + for (int i = 0; i < (int)mSyncEvents.size(); i++) { + if (mSyncEvents[i]->type() == type) { + mSyncEvents[i]->trigger(); + mSyncEvents.removeAt(i); + i--; + } + } +} + +// implement VolumeBufferProvider interface + +uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() +{ + // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs + ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); + uint32_t vlr = mCblk->getVolumeLR(); + uint32_t vl = vlr & 0xFFFF; + uint32_t vr = vlr >> 16; + // track volumes come from shared memory, so can't be trusted and must be clamped + if (vl > MAX_GAIN_INT) { + vl = MAX_GAIN_INT; + } + if (vr > MAX_GAIN_INT) { + vr = MAX_GAIN_INT; + } + // now apply the cached master volume and stream type volume; + // this is trusted but lacks any synchronization or barrier so may be stale + float v = mCachedVolume; + vl *= v; + vr *= v; + // re-combine into U4.16 + vlr = (vr << 16) | (vl & 0xFFFF); + // FIXME look at mute, pause, and stop flags + return vlr; +} + +status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) +{ + if (mState == TERMINATED || mState == PAUSED || + ((framesReady() == 0) && ((mSharedBuffer != 0) || + (mState == STOPPED)))) { + ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", + mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); + event->cancel(); + return INVALID_OPERATION; + } + (void) TrackBase::setSyncEvent(event); + return NO_ERROR; +} + +bool AudioFlinger::PlaybackThread::Track::isOut() const +{ + return true; +} + +// ---------------------------------------------------------------------------- + +sp<AudioFlinger::PlaybackThread::TimedTrack> +AudioFlinger::PlaybackThread::TimedTrack::create( + PlaybackThread *thread, + const sp<Client>& client, + audio_stream_type_t streamType, + uint32_t sampleRate, + audio_format_t format, + audio_channel_mask_t channelMask, + size_t frameCount, + const sp<IMemory>& sharedBuffer, + int sessionId) { + if (!client->reserveTimedTrack()) + return 0; + + return new TimedTrack( + thread, client, streamType, sampleRate, format, channelMask, frameCount, + sharedBuffer, sessionId); +} + +AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( + PlaybackThread *thread, + const sp<Client>& client, + audio_stream_type_t streamType, + uint32_t sampleRate, + audio_format_t format, + audio_channel_mask_t channelMask, + size_t frameCount, + const sp<IMemory>& sharedBuffer, + int sessionId) + : Track(thread, client, streamType, sampleRate, format, channelMask, + frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), + mQueueHeadInFlight(false), + mTrimQueueHeadOnRelease(false), + mFramesPendingInQueue(0), + mTimedSilenceBuffer(NULL), + mTimedSilenceBufferSize(0), + mTimedAudioOutputOnTime(false), + mMediaTimeTransformValid(false) +{ + LocalClock lc; + mLocalTimeFreq = lc.getLocalFreq(); + + mLocalTimeToSampleTransform.a_zero = 0; + mLocalTimeToSampleTransform.b_zero = 0; + mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; + mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; + LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, + &mLocalTimeToSampleTransform.a_to_b_denom); + + mMediaTimeToSampleTransform.a_zero = 0; + mMediaTimeToSampleTransform.b_zero = 0; + mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; + mMediaTimeToSampleTransform.a_to_b_denom = 1000000; + LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, + &mMediaTimeToSampleTransform.a_to_b_denom); +} + +AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { + mClient->releaseTimedTrack(); + delete [] mTimedSilenceBuffer; +} + +status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( + size_t size, sp<IMemory>* buffer) { + + Mutex::Autolock _l(mTimedBufferQueueLock); + + trimTimedBufferQueue_l(); + + // lazily initialize the shared memory heap for timed buffers + if (mTimedMemoryDealer == NULL) { + const int kTimedBufferHeapSize = 512 << 10; + + mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, + "AudioFlingerTimed"); + if (mTimedMemoryDealer == NULL) + return NO_MEMORY; + } + + sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); + if (newBuffer == NULL) { + newBuffer = mTimedMemoryDealer->allocate(size); + if (newBuffer == NULL) + return NO_MEMORY; + } + + *buffer = newBuffer; + return NO_ERROR; +} + +// caller must hold mTimedBufferQueueLock +void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { + int64_t mediaTimeNow; + { + Mutex::Autolock mttLock(mMediaTimeTransformLock); + if (!mMediaTimeTransformValid) + return; + + int64_t targetTimeNow; + status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) + ? mCCHelper.getCommonTime(&targetTimeNow) + : mCCHelper.getLocalTime(&targetTimeNow); + + if (OK != res) + return; + + if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, + &mediaTimeNow)) { + return; + } + } + + size_t trimEnd; + for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { + int64_t bufEnd; + + if ((trimEnd + 1) < mTimedBufferQueue.size()) { + // We have a next buffer. Just use its PTS as the PTS of the frame + // following the last frame in this buffer. If the stream is sparse + // (ie, there are deliberate gaps left in the stream which should be + // filled with silence by the TimedAudioTrack), then this can result + // in one extra buffer being left un-trimmed when it could have + // been. In general, this is not typical, and we would rather + // optimized away the TS calculation below for the more common case + // where PTSes are contiguous. + bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); + } else { + // We have no next buffer. Compute the PTS of the frame following + // the last frame in this buffer by computing the duration of of + // this frame in media time units and adding it to the PTS of the + // buffer. + int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() + / mFrameSize; + + if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, + &bufEnd)) { + ALOGE("Failed to convert frame count of %lld to media time" + " duration" " (scale factor %d/%u) in %s", + frameCount, + mMediaTimeToSampleTransform.a_to_b_numer, + mMediaTimeToSampleTransform.a_to_b_denom, + __PRETTY_FUNCTION__); + break; + } + bufEnd += mTimedBufferQueue[trimEnd].pts(); + } + + if (bufEnd > mediaTimeNow) + break; + + // Is the buffer we want to use in the middle of a mix operation right + // now? If so, don't actually trim it. Just wait for the releaseBuffer + // from the mixer which should be coming back shortly. + if (!trimEnd && mQueueHeadInFlight) { + mTrimQueueHeadOnRelease = true; + } + } + + size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; + if (trimStart < trimEnd) { + // Update the bookkeeping for framesReady() + for (size_t i = trimStart; i < trimEnd; ++i) { + updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); + } + + // Now actually remove the buffers from the queue. + mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); + } +} + +void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( + const char* logTag) { + ALOG_ASSERT(mTimedBufferQueue.size() > 0, + "%s called (reason \"%s\"), but timed buffer queue has no" + " elements to trim.", __FUNCTION__, logTag); + + updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); + mTimedBufferQueue.removeAt(0); +} + +void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( + const TimedBuffer& buf, + const char* logTag) { + uint32_t bufBytes = buf.buffer()->size(); + uint32_t consumedAlready = buf.position(); + + ALOG_ASSERT(consumedAlready <= bufBytes, + "Bad bookkeeping while updating frames pending. Timed buffer is" + " only %u bytes long, but claims to have consumed %u" + " bytes. (update reason: \"%s\")", + bufBytes, consumedAlready, logTag); + + uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; + ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, + "Bad bookkeeping while updating frames pending. Should have at" + " least %u queued frames, but we think we have only %u. (update" + " reason: \"%s\")", + bufFrames, mFramesPendingInQueue, logTag); + + mFramesPendingInQueue -= bufFrames; +} + +status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( + const sp<IMemory>& buffer, int64_t pts) { + + { + Mutex::Autolock mttLock(mMediaTimeTransformLock); + if (!mMediaTimeTransformValid) + return INVALID_OPERATION; + } + + Mutex::Autolock _l(mTimedBufferQueueLock); + + uint32_t bufFrames = buffer->size() / mFrameSize; + mFramesPendingInQueue += bufFrames; + mTimedBufferQueue.add(TimedBuffer(buffer, pts)); + + return NO_ERROR; +} + +status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( + const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { + + ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", + xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, + target); + + if (!(target == TimedAudioTrack::LOCAL_TIME || + target == TimedAudioTrack::COMMON_TIME)) { + return BAD_VALUE; + } + + Mutex::Autolock lock(mMediaTimeTransformLock); + mMediaTimeTransform = xform; + mMediaTimeTransformTarget = target; + mMediaTimeTransformValid = true; + + return NO_ERROR; +} + +#define min(a, b) ((a) < (b) ? (a) : (b)) + +// implementation of getNextBuffer for tracks whose buffers have timestamps +status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( + AudioBufferProvider::Buffer* buffer, int64_t pts) +{ + if (pts == AudioBufferProvider::kInvalidPTS) { + buffer->raw = NULL; + buffer->frameCount = 0; + mTimedAudioOutputOnTime = false; + return INVALID_OPERATION; + } + + Mutex::Autolock _l(mTimedBufferQueueLock); + + ALOG_ASSERT(!mQueueHeadInFlight, + "getNextBuffer called without releaseBuffer!"); + + while (true) { + + // if we have no timed buffers, then fail + if (mTimedBufferQueue.isEmpty()) { + buffer->raw = NULL; + buffer->frameCount = 0; + return NOT_ENOUGH_DATA; + } + + TimedBuffer& head = mTimedBufferQueue.editItemAt(0); + + // calculate the PTS of the head of the timed buffer queue expressed in + // local time + int64_t headLocalPTS; + { + Mutex::Autolock mttLock(mMediaTimeTransformLock); + + ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); + + if (mMediaTimeTransform.a_to_b_denom == 0) { + // the transform represents a pause, so yield silence + timedYieldSilence_l(buffer->frameCount, buffer); + return NO_ERROR; + } + + int64_t transformedPTS; + if (!mMediaTimeTransform.doForwardTransform(head.pts(), + &transformedPTS)) { + // the transform failed. this shouldn't happen, but if it does + // then just drop this buffer + ALOGW("timedGetNextBuffer transform failed"); + buffer->raw = NULL; + buffer->frameCount = 0; + trimTimedBufferQueueHead_l("getNextBuffer; no transform"); + return NO_ERROR; + } + + if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { + if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, + &headLocalPTS)) { + buffer->raw = NULL; + buffer->frameCount = 0; + return INVALID_OPERATION; + } + } else { + headLocalPTS = transformedPTS; + } + } + + // adjust the head buffer's PTS to reflect the portion of the head buffer + // that has already been consumed + int64_t effectivePTS = headLocalPTS + + ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate()); + + // Calculate the delta in samples between the head of the input buffer + // queue and the start of the next output buffer that will be written. + // If the transformation fails because of over or underflow, it means + // that the sample's position in the output stream is so far out of + // whack that it should just be dropped. + int64_t sampleDelta; + if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { + ALOGV("*** head buffer is too far from PTS: dropped buffer"); + trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" + " mix"); + continue; + } + if (!mLocalTimeToSampleTransform.doForwardTransform( + (effectivePTS - pts) << 32, &sampleDelta)) { + ALOGV("*** too late during sample rate transform: dropped buffer"); + trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); + continue; + } + + ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" + " sampleDelta=[%d.%08x]", + head.pts(), head.position(), pts, + static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + + (sampleDelta >> 32)), + static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); + + // if the delta between the ideal placement for the next input sample and + // the current output position is within this threshold, then we will + // concatenate the next input samples to the previous output + const int64_t kSampleContinuityThreshold = + (static_cast<int64_t>(sampleRate()) << 32) / 250; + + // if this is the first buffer of audio that we're emitting from this track + // then it should be almost exactly on time. + const int64_t kSampleStartupThreshold = 1LL << 32; + + if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || + (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { + // the next input is close enough to being on time, so concatenate it + // with the last output + timedYieldSamples_l(buffer); + + ALOGVV("*** on time: head.pos=%d frameCount=%u", + head.position(), buffer->frameCount); + return NO_ERROR; + } + + // Looks like our output is not on time. Reset our on timed status. + // Next time we mix samples from our input queue, then should be within + // the StartupThreshold. + mTimedAudioOutputOnTime = false; + if (sampleDelta > 0) { + // the gap between the current output position and the proper start of + // the next input sample is too big, so fill it with silence + uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; + + timedYieldSilence_l(framesUntilNextInput, buffer); + ALOGV("*** silence: frameCount=%u", buffer->frameCount); + return NO_ERROR; + } else { + // the next input sample is late + uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); + size_t onTimeSamplePosition = + head.position() + lateFrames * mFrameSize; + + if (onTimeSamplePosition > head.buffer()->size()) { + // all the remaining samples in the head are too late, so + // drop it and move on + ALOGV("*** too late: dropped buffer"); + trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); + continue; + } else { + // skip over the late samples + head.setPosition(onTimeSamplePosition); + + // yield the available samples + timedYieldSamples_l(buffer); + + ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); + return NO_ERROR; + } + } + } +} + +// Yield samples from the timed buffer queue head up to the given output +// buffer's capacity. +// +// Caller must hold mTimedBufferQueueLock +void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( + AudioBufferProvider::Buffer* buffer) { + + const TimedBuffer& head = mTimedBufferQueue[0]; + + buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + + head.position()); + + uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / + mFrameSize); + size_t framesRequested = buffer->frameCount; + buffer->frameCount = min(framesLeftInHead, framesRequested); + + mQueueHeadInFlight = true; + mTimedAudioOutputOnTime = true; +} + +// Yield samples of silence up to the given output buffer's capacity +// +// Caller must hold mTimedBufferQueueLock +void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( + uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { + + // lazily allocate a buffer filled with silence + if (mTimedSilenceBufferSize < numFrames * mFrameSize) { + delete [] mTimedSilenceBuffer; + mTimedSilenceBufferSize = numFrames * mFrameSize; + mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; + memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); + } + + buffer->raw = mTimedSilenceBuffer; + size_t framesRequested = buffer->frameCount; + buffer->frameCount = min(numFrames, framesRequested); + + mTimedAudioOutputOnTime = false; +} + +// AudioBufferProvider interface +void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( + AudioBufferProvider::Buffer* buffer) { + + Mutex::Autolock _l(mTimedBufferQueueLock); + + // If the buffer which was just released is part of the buffer at the head + // of the queue, be sure to update the amt of the buffer which has been + // consumed. If the buffer being returned is not part of the head of the + // queue, its either because the buffer is part of the silence buffer, or + // because the head of the timed queue was trimmed after the mixer called + // getNextBuffer but before the mixer called releaseBuffer. + if (buffer->raw == mTimedSilenceBuffer) { + ALOG_ASSERT(!mQueueHeadInFlight, + "Queue head in flight during release of silence buffer!"); + goto done; + } + + ALOG_ASSERT(mQueueHeadInFlight, + "TimedTrack::releaseBuffer of non-silence buffer, but no queue" + " head in flight."); + + if (mTimedBufferQueue.size()) { + TimedBuffer& head = mTimedBufferQueue.editItemAt(0); + + void* start = head.buffer()->pointer(); + void* end = reinterpret_cast<void*>( + reinterpret_cast<uint8_t*>(head.buffer()->pointer()) + + head.buffer()->size()); + + ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), + "released buffer not within the head of the timed buffer" + " queue; qHead = [%p, %p], released buffer = %p", + start, end, buffer->raw); + + head.setPosition(head.position() + + (buffer->frameCount * mFrameSize)); + mQueueHeadInFlight = false; + + ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, + "Bad bookkeeping during releaseBuffer! Should have at" + " least %u queued frames, but we think we have only %u", + buffer->frameCount, mFramesPendingInQueue); + + mFramesPendingInQueue -= buffer->frameCount; + + if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) + || mTrimQueueHeadOnRelease) { + trimTimedBufferQueueHead_l("releaseBuffer"); + mTrimQueueHeadOnRelease = false; + } + } else { + LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" + " buffers in the timed buffer queue"); + } + +done: + buffer->raw = 0; + buffer->frameCount = 0; +} + +size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { + Mutex::Autolock _l(mTimedBufferQueueLock); + return mFramesPendingInQueue; +} + +AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() + : mPTS(0), mPosition(0) {} + +AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( + const sp<IMemory>& buffer, int64_t pts) + : mBuffer(buffer), mPTS(pts), mPosition(0) {} + + +// ---------------------------------------------------------------------------- + +AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( + PlaybackThread *playbackThread, + DuplicatingThread *sourceThread, + uint32_t sampleRate, + audio_format_t format, + audio_channel_mask_t channelMask, + size_t frameCount) + : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, + NULL, 0, IAudioFlinger::TRACK_DEFAULT), + mActive(false), mSourceThread(sourceThread), mBuffers(NULL) +{ + + if (mCblk != NULL) { + mBuffers = (char*)mCblk + sizeof(audio_track_cblk_t); + mOutBuffer.frameCount = 0; + playbackThread->mTracks.add(this); + ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mBuffers %p, " \ + "mCblk->frameCount %d, mCblk->sampleRate %u, mChannelMask 0x%08x mBufferEnd %p", + mCblk, mBuffer, mBuffers, + mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); + } else { + ALOGW("Error creating output track on thread %p", playbackThread); + } +} + +AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() +{ + clearBufferQueue(); +} + +status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, + int triggerSession) +{ + status_t status = Track::start(event, triggerSession); + if (status != NO_ERROR) { + return status; + } + + mActive = true; + mRetryCount = 127; + return status; +} + +void AudioFlinger::PlaybackThread::OutputTrack::stop() +{ + Track::stop(); + clearBufferQueue(); + mOutBuffer.frameCount = 0; + mActive = false; +} + +bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) +{ + Buffer *pInBuffer; + Buffer inBuffer; + uint32_t channelCount = mChannelCount; + 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 (mFrameCount > frames) { + if (mBufferQueue.size() < kMaxOverFlowBuffers) { + uint32_t startFrames = (mFrameCount - frames); + pInBuffer = new Buffer; + pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; + pInBuffer->frameCount = startFrames; + pInBuffer->i16 = pInBuffer->mBuffer; + memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); + mBufferQueue.add(pInBuffer); + } else { + ALOGW ("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)NO_MORE_BUFFERS) { + ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, + mThread.unsafe_get()); + outputBufferFull = true; + break; + } + uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); + if (waitTimeLeftMs >= waitTimeMs) { + waitTimeLeftMs -= waitTimeMs; + } else { + waitTimeLeftMs = 0; + } + } + + uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : + pInBuffer->frameCount; + memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); + mCblk->stepUserOut(outFrames, mFrameCount); + pInBuffer->frameCount -= outFrames; + pInBuffer->i16 += outFrames * channelCount; + mOutBuffer.frameCount -= outFrames; + mOutBuffer.i16 += outFrames * channelCount; + + if (pInBuffer->frameCount == 0) { + if (mBufferQueue.size()) { + mBufferQueue.removeAt(0); + delete [] pInBuffer->mBuffer; + delete pInBuffer; + ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, + mThread.unsafe_get(), mBufferQueue.size()); + } else { + break; + } + } + } + + // If we could not write all frames, allocate a buffer and queue it for next time. + if (inBuffer.frameCount) { + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0 && !thread->standby()) { + if (mBufferQueue.size() < kMaxOverFlowBuffers) { + pInBuffer = new Buffer; + pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; + pInBuffer->frameCount = inBuffer.frameCount; + pInBuffer->i16 = pInBuffer->mBuffer; + memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * + sizeof(int16_t)); + mBufferQueue.add(pInBuffer); + ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, + mThread.unsafe_get(), mBufferQueue.size()); + } else { + ALOGW("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 < mFrameCount) { + frames = mFrameCount - mCblk->user; + pInBuffer = new Buffer; + pInBuffer->mBuffer = new int16_t[frames * channelCount]; + pInBuffer->frameCount = frames; + pInBuffer->i16 = pInBuffer->mBuffer; + memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); + mBufferQueue.add(pInBuffer); + } else if (mActive) { + stop(); + } + } + + return outputBufferFull; +} + +status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( + AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) +{ + int active; + status_t result; + audio_track_cblk_t* cblk = mCblk; + uint32_t framesReq = buffer->frameCount; + + ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); + buffer->frameCount = 0; + + uint32_t framesAvail = cblk->framesAvailableOut(mFrameCount); + + + if (framesAvail == 0) { + Mutex::Autolock _l(cblk->lock); + goto start_loop_here; + while (framesAvail == 0) { + active = mActive; + if (CC_UNLIKELY(!active)) { + ALOGV("Not active and NO_MORE_BUFFERS"); + return NO_MORE_BUFFERS; + } + result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); + if (result != NO_ERROR) { + return NO_MORE_BUFFERS; + } + // read the server count again + start_loop_here: + framesAvail = cblk->framesAvailableOut_l(mFrameCount); + } + } + +// if (framesAvail < framesReq) { +// return NO_MORE_BUFFERS; +// } + + if (framesReq > framesAvail) { + framesReq = framesAvail; + } + + uint32_t u = cblk->user; + uint32_t bufferEnd = cblk->userBase + mFrameCount; + + if (framesReq > bufferEnd - u) { + framesReq = bufferEnd - u; + } + + buffer->frameCount = framesReq; + buffer->raw = cblk->buffer(mBuffers, mFrameSize, u); + return NO_ERROR; +} + + +void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() +{ + size_t size = mBufferQueue.size(); + + for (size_t i = 0; i < size; i++) { + Buffer *pBuffer = mBufferQueue.itemAt(i); + delete [] pBuffer->mBuffer; + delete pBuffer; + } + mBufferQueue.clear(); +} + + +// ---------------------------------------------------------------------------- +// Record +// ---------------------------------------------------------------------------- + +AudioFlinger::RecordHandle::RecordHandle( + const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) + : BnAudioRecord(), + mRecordTrack(recordTrack) +{ +} + +AudioFlinger::RecordHandle::~RecordHandle() { + stop_nonvirtual(); + mRecordTrack->destroy(); +} + +sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { + return mRecordTrack->getCblk(); +} + +status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, + int triggerSession) { + ALOGV("RecordHandle::start()"); + return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); +} + +void AudioFlinger::RecordHandle::stop() { + stop_nonvirtual(); +} + +void AudioFlinger::RecordHandle::stop_nonvirtual() { + ALOGV("RecordHandle::stop()"); + mRecordTrack->stop(); +} + +status_t AudioFlinger::RecordHandle::onTransact( + uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) +{ + return BnAudioRecord::onTransact(code, data, reply, flags); +} + +// ---------------------------------------------------------------------------- + +// RecordTrack constructor must be called with AudioFlinger::mLock held +AudioFlinger::RecordThread::RecordTrack::RecordTrack( + RecordThread *thread, + const sp<Client>& client, + uint32_t sampleRate, + audio_format_t format, + audio_channel_mask_t channelMask, + size_t frameCount, + int sessionId) + : TrackBase(thread, client, sampleRate, format, + channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), + mOverflow(false) +{ + ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); +} + +AudioFlinger::RecordThread::RecordTrack::~RecordTrack() +{ + ALOGV("%s", __func__); +} + +// AudioBufferProvider interface +status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, + int64_t pts) +{ + 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 (mStepServerFailed) { + if (!step()) { + goto getNextBuffer_exit; + } + ALOGV("stepServer recovered"); + mStepServerFailed = false; + } + + // FIXME lock is not actually held, so overrun is possible + framesAvail = cblk->framesAvailableIn_l(mFrameCount); + + if (CC_LIKELY(framesAvail)) { + uint32_t s = cblk->server; + uint32_t bufferEnd = cblk->serverBase + mFrameCount; + + if (framesReq > framesAvail) { + framesReq = framesAvail; + } + if (framesReq > bufferEnd - s) { + framesReq = bufferEnd - s; + } + + buffer->raw = getBuffer(s, framesReq); + buffer->frameCount = framesReq; + return NO_ERROR; + } + +getNextBuffer_exit: + buffer->raw = NULL; + buffer->frameCount = 0; + return NOT_ENOUGH_DATA; +} + +status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, + int triggerSession) +{ + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + RecordThread *recordThread = (RecordThread *)thread.get(); + return recordThread->start(this, event, triggerSession); + } else { + return BAD_VALUE; + } +} + +void AudioFlinger::RecordThread::RecordTrack::stop() +{ + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + RecordThread *recordThread = (RecordThread *)thread.get(); + recordThread->mLock.lock(); + bool doStop = recordThread->stop_l(this); + if (doStop) { + TrackBase::reset(); + // Force overrun condition to avoid false overrun callback until first data is + // read from buffer + android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); + } + recordThread->mLock.unlock(); + if (doStop) { + AudioSystem::stopInput(recordThread->id()); + } + } +} + +void AudioFlinger::RecordThread::RecordTrack::destroy() +{ + // see comments at AudioFlinger::PlaybackThread::Track::destroy() + sp<RecordTrack> keep(this); + { + sp<ThreadBase> thread = mThread.promote(); + if (thread != 0) { + if (mState == ACTIVE || mState == RESUMING) { + AudioSystem::stopInput(thread->id()); + } + AudioSystem::releaseInput(thread->id()); + Mutex::Autolock _l(thread->mLock); + RecordThread *recordThread = (RecordThread *) thread.get(); + recordThread->destroyTrack_l(this); + } + } +} + + +/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) +{ + result.append(" Clien Fmt Chn mask Session Step S SRate Serv User FrameCount\n"); +} + +void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) +{ + snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x %05d\n", + (mClient == 0) ? getpid_cached : mClient->pid(), + mFormat, + mChannelMask, + mSessionId, + mStepCount, + mState, + mCblk->sampleRate, + mCblk->server, + mCblk->user, + mFrameCount); +} + +bool AudioFlinger::RecordThread::RecordTrack::isOut() const +{ + return false; +} + +}; // namespace android |