diff options
Diffstat (limited to 'services/audioflinger/Threads.cpp')
| -rw-r--r-- | services/audioflinger/Threads.cpp | 1316 |
1 files changed, 1055 insertions, 261 deletions
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index 0b88c0e..47dcca6 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -20,11 +20,12 @@ //#define LOG_NDEBUG 0 #define ATRACE_TAG ATRACE_TAG_AUDIO +#include "Configuration.h" #include <math.h> #include <fcntl.h> #include <sys/stat.h> #include <cutils/properties.h> -#include <cutils/compiler.h> +#include <media/AudioParameter.h> #include <utils/Log.h> #include <utils/Trace.h> @@ -53,14 +54,11 @@ #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> @@ -111,6 +109,9 @@ static const uint32_t kMinNormalMixBufferSizeMs = 20; // maximum normal mix buffer size static const uint32_t kMaxNormalMixBufferSizeMs = 24; +// Offloaded output thread standby delay: allows track transition without going to standby +static const nsecs_t kOffloadStandbyDelayNs = seconds(1); + // Whether to use fast mixer static const enum { FastMixer_Never, // never initialize or use: for debugging only @@ -139,7 +140,7 @@ static const int kPriorityFastMixer = 3; // 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; +static const int kFastTrackMultiplier = 1; // ---------------------------------------------------------------------------- @@ -267,10 +268,9 @@ AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio 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), + mAudioFlinger(audioFlinger), + // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are + // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() mParamStatus(NO_ERROR), mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), @@ -281,6 +281,12 @@ AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio AudioFlinger::ThreadBase::~ThreadBase() { + // mConfigEvents should be empty, but just in case it isn't, free the memory it owns + for (size_t i = 0; i < mConfigEvents.size(); i++) { + delete mConfigEvents[i]; + } + mConfigEvents.clear(); + mParamCond.broadcast(); // do not lock the mutex in destructor releaseWakeLock_l(); @@ -420,9 +426,7 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 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); + snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); result.append(buffer); snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); result.append(buffer); @@ -472,13 +476,32 @@ void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& } } -void AudioFlinger::ThreadBase::acquireWakeLock() +void AudioFlinger::ThreadBase::acquireWakeLock(int uid) { Mutex::Autolock _l(mLock); - acquireWakeLock_l(); + acquireWakeLock_l(uid); } -void AudioFlinger::ThreadBase::acquireWakeLock_l() +String16 AudioFlinger::ThreadBase::getWakeLockTag() +{ + switch (mType) { + case MIXER: + return String16("AudioMix"); + case DIRECT: + return String16("AudioDirectOut"); + case DUPLICATING: + return String16("AudioDup"); + case RECORD: + return String16("AudioIn"); + case OFFLOAD: + return String16("AudioOffload"); + default: + ALOG_ASSERT(false); + return String16("AudioUnknown"); + } +} + +void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) { if (mPowerManager == 0) { // use checkService() to avoid blocking if power service is not up yet @@ -493,9 +516,19 @@ void AudioFlinger::ThreadBase::acquireWakeLock_l() } if (mPowerManager != 0) { sp<IBinder> binder = new BBinder(); - status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, - binder, - String16(mName)); + status_t status; + if (uid >= 0) { + status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, + binder, + getWakeLockTag(), + String16("media"), + uid); + } else { + status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, + binder, + getWakeLockTag(), + String16("media")); + } if (status == NO_ERROR) { mWakeLockToken = binder; } @@ -697,14 +730,22 @@ sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 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; + // Allow global effects only on offloaded and mixer threads + if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { + switch (mType) { + case MIXER: + case OFFLOAD: + break; + case DIRECT: + case DUPLICATING: + case RECORD: + default: + ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); + 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", @@ -747,6 +788,8 @@ sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( if (lStatus != NO_ERROR) { goto Exit; } + effect->setOffloaded(mType == OFFLOAD, mId); + lStatus = chain->addEffect_l(effect); if (lStatus != NO_ERROR) { goto Exit; @@ -808,6 +851,10 @@ status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) sp<EffectChain> chain = getEffectChain_l(sessionId); bool chainCreated = false; + ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), + "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", + this, effect->desc().name, effect->desc().flags); + if (chain == 0) { // create a new chain for this session ALOGV("addEffect_l() new effect chain for session %d", sessionId); @@ -824,6 +871,8 @@ status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) return BAD_VALUE; } + effect->setOffloaded(mType == OFFLOAD, mId); + status_t status = chain->addEffect_l(effect); if (status != NO_ERROR) { if (chainCreated) { @@ -926,16 +975,25 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge audio_devices_t device, type_t type) : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), - mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), + mNormalFrameCount(0), mMixBuffer(NULL), + mAllocMixBuffer(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), + mBytesRemaining(0), + mCurrentWriteLength(0), + mUseAsyncWrite(false), + mWriteAckSequence(0), + mDrainSequence(0), + mSignalPending(false), mScreenState(AudioFlinger::mScreenState), // index 0 is reserved for normal mixer's submix - mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) + mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), + // mLatchD, mLatchQ, + mLatchDValid(false), mLatchQValid(false) { snprintf(mName, kNameLength, "AudioOut_%X", id); mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); @@ -975,7 +1033,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge AudioFlinger::PlaybackThread::~PlaybackThread() { mAudioFlinger->unregisterWriter(mNBLogWriter); - delete [] mMixBuffer; + delete [] mAllocMixBuffer; } void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) @@ -1043,6 +1101,8 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); result.append(buffer); + snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); + result.append(buffer); snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); result.append(buffer); @@ -1181,7 +1241,22 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac goto Exit; } } + } else if (mType == OFFLOAD) { + if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { + ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" + "for output %p with format %d", + sampleRate, format, channelMask, mOutput, mFormat); + lStatus = BAD_VALUE; + goto Exit; + } } else { + if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { + ALOGE("createTrack_l() Bad parameter: format %d \"" + "for output %p with format %d", + format, mOutput, mFormat); + lStatus = BAD_VALUE; + goto Exit; + } // 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); @@ -1227,6 +1302,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac lStatus = NO_MEMORY; goto Exit; } + mTracks.add(track); sp<EffectChain> chain = getEffectChain_l(sessionId); @@ -1301,12 +1377,14 @@ void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, f { Mutex::Autolock _l(mLock); mStreamTypes[stream].volume = value; + broadcast_l(); } void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) { Mutex::Autolock _l(mLock); mStreamTypes[stream].mute = muted; + broadcast_l(); } float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const @@ -1326,7 +1404,31 @@ status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) // 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; + if (!track->isOutputTrack()) { + TrackBase::track_state state = track->mState; + mLock.unlock(); + status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); + mLock.lock(); + // abort track was stopped/paused while we released the lock + if (state != track->mState) { + if (status == NO_ERROR) { + mLock.unlock(); + AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); + mLock.lock(); + } + return INVALID_OPERATION; + } + // abort if start is rejected by audio policy manager + if (status != NO_ERROR) { + return PERMISSION_DENIED; + } +#ifdef ADD_BATTERY_DATA + // to track the speaker usage + addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); +#endif + } + + track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; track->mResetDone = false; track->mPresentationCompleteFrames = 0; mActiveTracks.add(track); @@ -1340,20 +1442,25 @@ status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) status = NO_ERROR; } - ALOGV("mWaitWorkCV.broadcast"); - mWaitWorkCV.broadcast(); + ALOGV("signal playback thread"); + broadcast_l(); return status; } -// destroyTrack_l() must be called with ThreadBase::mLock held -void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) +bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) { - track->mState = TrackBase::TERMINATED; + track->terminate(); // active tracks are removed by threadLoop() - if (mActiveTracks.indexOf(track) < 0) { + bool trackActive = (mActiveTracks.indexOf(track) >= 0); + track->mState = TrackBase::STOPPED; + if (!trackActive) { removeTrack_l(track); + } else if (track->isFastTrack() || track->isOffloaded()) { + track->mState = TrackBase::STOPPING_1; } + + return trackActive; } void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) @@ -1377,18 +1484,25 @@ void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) } } -String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) +void AudioFlinger::PlaybackThread::broadcast_l() { - String8 out_s8 = String8(""); - char *s; + // Thread could be blocked waiting for async + // so signal it to handle state changes immediately + // If threadLoop is currently unlocked a signal of mWaitWorkCV will + // be lost so we also flag to prevent it blocking on mWaitWorkCV + mSignalPending = true; + mWaitWorkCV.broadcast(); +} +String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) +{ Mutex::Autolock _l(mLock); if (initCheck() != NO_ERROR) { - return out_s8; + return String8(); } - s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); - out_s8 = String8(s); + char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); + const String8 out_s8(s); free(s); return out_s8; } @@ -1404,7 +1518,7 @@ void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { switch (event) { case AudioSystem::OUTPUT_OPENED: case AudioSystem::OUTPUT_CONFIG_CHANGED: - desc.channels = mChannelMask; + desc.channelMask = mChannelMask; desc.samplingRate = mSampleRate; desc.format = mFormat; desc.frameCount = mNormalFrameCount; // FIXME see @@ -1422,12 +1536,80 @@ void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { mAudioFlinger->audioConfigChanged_l(event, mId, param2); } +void AudioFlinger::PlaybackThread::writeCallback() +{ + ALOG_ASSERT(mCallbackThread != 0); + mCallbackThread->resetWriteBlocked(); +} + +void AudioFlinger::PlaybackThread::drainCallback() +{ + ALOG_ASSERT(mCallbackThread != 0); + mCallbackThread->resetDraining(); +} + +void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) +{ + Mutex::Autolock _l(mLock); + // reject out of sequence requests + if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { + mWriteAckSequence &= ~1; + mWaitWorkCV.signal(); + } +} + +void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) +{ + Mutex::Autolock _l(mLock); + // reject out of sequence requests + if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { + mDrainSequence &= ~1; + mWaitWorkCV.signal(); + } +} + +// static +int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, + void *param, + void *cookie) +{ + AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; + ALOGV("asyncCallback() event %d", event); + switch (event) { + case STREAM_CBK_EVENT_WRITE_READY: + me->writeCallback(); + break; + case STREAM_CBK_EVENT_DRAIN_READY: + me->drainCallback(); + break; + default: + ALOGW("asyncCallback() unknown event %d", event); + break; + } + return 0; +} + void AudioFlinger::PlaybackThread::readOutputParameters() { + // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); - mChannelCount = (uint16_t)popcount(mChannelMask); + if (!audio_is_output_channel(mChannelMask)) { + LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); + } + if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { + LOG_FATAL("HAL channel mask %#x not supported for mixed output; " + "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); + } + mChannelCount = popcount(mChannelMask); mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); + if (!audio_is_valid_format(mFormat)) { + LOG_FATAL("HAL format %d not valid for output", mFormat); + } + if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { + LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", + mFormat); + } mFrameSize = audio_stream_frame_size(&mOutput->stream->common); mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; if (mFrameCount & 15) { @@ -1435,6 +1617,15 @@ void AudioFlinger::PlaybackThread::readOutputParameters() mFrameCount); } + if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && + (mOutput->stream->set_callback != NULL)) { + if (mOutput->stream->set_callback(mOutput->stream, + AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { + mUseAsyncWrite = true; + mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); + } + } + // Calculate size of normal mix buffer relative to the HAL output buffer size double multiplier = 1.0; if (mType == MIXER && (kUseFastMixer == FastMixer_Static || @@ -1477,9 +1668,11 @@ void AudioFlinger::PlaybackThread::readOutputParameters() 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)); + delete[] mAllocMixBuffer; + size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; + mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; + mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); + memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); // force reconfiguration of effect chains and engines to take new buffer size and audio // parameters into account @@ -1613,16 +1806,21 @@ void AudioFlinger::PlaybackThread::threadLoop_removeTracks( const Vector< sp<Track> >& tracksToRemove) { size_t count = tracksToRemove.size(); - if (CC_UNLIKELY(count)) { + if (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)) { + if (!track->isOutputTrack()) { AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); +#ifdef ADD_BATTERY_DATA + // to track the speaker usage + addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); +#endif + if (track->isTerminated()) { + AudioSystem::releaseOutput(mId); + } } } } - } void AudioFlinger::PlaybackThread::checkSilentMode_l() @@ -1643,17 +1841,18 @@ void AudioFlinger::PlaybackThread::checkSilentMode_l() } // shared by MIXER and DIRECT, overridden by DUPLICATING -void AudioFlinger::PlaybackThread::threadLoop_write() +ssize_t AudioFlinger::PlaybackThread::threadLoop_write() { // FIXME rewrite to reduce number of system calls mLastWriteTime = systemTime(); mInWrite = true; - int bytesWritten; + ssize_t 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; + size_t count = mBytesRemaining >> mBitShift; + size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; ATRACE_BEGIN("write"); // update the setpoint when AudioFlinger::mScreenState changes uint32_t screenState = AudioFlinger::mScreenState; @@ -1665,24 +1864,70 @@ void AudioFlinger::PlaybackThread::threadLoop_write() (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); } } - ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); + ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); ATRACE_END(); if (framesWritten > 0) { bytesWritten = framesWritten << mBitShift; } else { bytesWritten = framesWritten; } + status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); + if (status == NO_ERROR) { + size_t totalFramesWritten = mNormalSink->framesWritten(); + if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { + mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; + mLatchDValid = true; + } + } // otherwise use the HAL / AudioStreamOut directly } else { - // Direct output thread. - bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); + // Direct output and offload threads + size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); + if (mUseAsyncWrite) { + ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); + mWriteAckSequence += 2; + mWriteAckSequence |= 1; + ALOG_ASSERT(mCallbackThread != 0); + mCallbackThread->setWriteBlocked(mWriteAckSequence); + } + // FIXME We should have an implementation of timestamps for direct output threads. + // They are used e.g for multichannel PCM playback over HDMI. + bytesWritten = mOutput->stream->write(mOutput->stream, + mMixBuffer + offset, mBytesRemaining); + if (mUseAsyncWrite && + ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { + // do not wait for async callback in case of error of full write + mWriteAckSequence &= ~1; + ALOG_ASSERT(mCallbackThread != 0); + mCallbackThread->setWriteBlocked(mWriteAckSequence); + } } - if (bytesWritten > 0) { - mBytesWritten += mixBufferSize; - } mNumWrites++; mInWrite = false; + + return bytesWritten; +} + +void AudioFlinger::PlaybackThread::threadLoop_drain() +{ + if (mOutput->stream->drain) { + ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); + if (mUseAsyncWrite) { + ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); + mDrainSequence |= 1; + ALOG_ASSERT(mCallbackThread != 0); + mCallbackThread->setDraining(mDrainSequence); + } + mOutput->stream->drain(mOutput->stream, + (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY + : AUDIO_DRAIN_ALL); + } +} + +void AudioFlinger::PlaybackThread::threadLoop_exit() +{ + // Default implementation has nothing to do } /* @@ -1713,7 +1958,7 @@ void AudioFlinger::PlaybackThread::cacheParameters_l() void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) { - ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", + ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", this, streamType, mTracks.size()); Mutex::Autolock _l(mLock); @@ -1899,6 +2144,8 @@ bool AudioFlinger::PlaybackThread::threadLoop() // and then that string will be logged at the next convenient opportunity. const char *logString = NULL; + checkSilentMode_l(); + while (!exitPending()) { cpuStats.sample(myName); @@ -1917,16 +2164,38 @@ bool AudioFlinger::PlaybackThread::threadLoop() logString = NULL; } + if (mLatchDValid) { + mLatchQ = mLatchD; + mLatchDValid = false; + mLatchQValid = true; + } + if (checkForNewParameters_l()) { cacheParameters_l(); } saveOutputTracks(); + if (mSignalPending) { + // A signal was raised while we were unlocked + mSignalPending = false; + } else if (waitingAsyncCallback_l()) { + if (exitPending()) { + break; + } + releaseWakeLock_l(); + ALOGV("wait async completion"); + mWaitWorkCV.wait(mLock); + ALOGV("async completion/wake"); + acquireWakeLock_l(); + standbyTime = systemTime() + standbyDelay; + sleepTime = 0; - // put audio hardware into standby after short delay - if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || - isSuspended())) { - if (!mStandby) { + continue; + } + if ((!mActiveTracks.size() && systemTime() > standbyTime) || + isSuspended()) { + // put audio hardware into standby after short delay + if (shouldStandby_l()) { threadLoop_standby(); @@ -1953,7 +2222,7 @@ bool AudioFlinger::PlaybackThread::threadLoop() mMixerStatus = MIXER_IDLE; mMixerStatusIgnoringFastTracks = MIXER_IDLE; mBytesWritten = 0; - + mBytesRemaining = 0; checkSilentMode_l(); standbyTime = systemTime() + standbyDelay; @@ -1965,7 +2234,6 @@ bool AudioFlinger::PlaybackThread::threadLoop() continue; } } - // mMixerStatusIgnoringFastTracks is also updated internally mMixerStatus = prepareTracks_l(&tracksToRemove); @@ -1975,19 +2243,40 @@ bool AudioFlinger::PlaybackThread::threadLoop() lockEffectChains_l(effectChains); } - if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { - threadLoop_mix(); - } else { - threadLoop_sleepTime(); - } + if (mBytesRemaining == 0) { + mCurrentWriteLength = 0; + if (mMixerStatus == MIXER_TRACKS_READY) { + // threadLoop_mix() sets mCurrentWriteLength + threadLoop_mix(); + } else if ((mMixerStatus != MIXER_DRAIN_TRACK) + && (mMixerStatus != MIXER_DRAIN_ALL)) { + // threadLoop_sleepTime sets sleepTime to 0 if data + // must be written to HAL + threadLoop_sleepTime(); + if (sleepTime == 0) { + mCurrentWriteLength = mixBufferSize; + } + } + mBytesRemaining = mCurrentWriteLength; + if (isSuspended()) { + sleepTime = suspendSleepTimeUs(); + // simulate write to HAL when suspended + mBytesWritten += mixBufferSize; + mBytesRemaining = 0; + } - if (isSuspended()) { - sleepTime = suspendSleepTimeUs(); - mBytesWritten += mixBufferSize; + // only process effects if we're going to write + if (sleepTime == 0 && mType != OFFLOAD) { + for (size_t i = 0; i < effectChains.size(); i ++) { + effectChains[i]->process_l(); + } + } } - - // only process effects if we're going to write - if (sleepTime == 0) { + // Process effect chains for offloaded thread even if no audio + // was read from audio track: process only updates effect state + // and thus does have to be synchronized with audio writes but may have + // to be called while waiting for async write callback + if (mType == OFFLOAD) { for (size_t i = 0; i < effectChains.size(); i ++) { effectChains[i]->process_l(); } @@ -1996,29 +2285,40 @@ bool AudioFlinger::PlaybackThread::threadLoop() // enable changes in effect chain unlockEffectChains(effectChains); - // sleepTime == 0 means we must write to audio hardware - if (sleepTime == 0) { - - threadLoop_write(); - + if (!waitingAsyncCallback()) { + // sleepTime == 0 means we must write to audio hardware + if (sleepTime == 0) { + if (mBytesRemaining) { + ssize_t ret = threadLoop_write(); + if (ret < 0) { + mBytesRemaining = 0; + } else { + mBytesWritten += ret; + mBytesRemaining -= ret; + } + } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || + (mMixerStatus == MIXER_DRAIN_ALL)) { + threadLoop_drain(); + } if (mType == MIXER) { - // write blocked detection - nsecs_t now = systemTime(); - nsecs_t delta = now - mLastWriteTime; - if (!mStandby && delta > maxPeriod) { - mNumDelayedWrites++; - if ((now - lastWarning) > kWarningThrottleNs) { - ATRACE_NAME("underrun"); - ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", - ns2ms(delta), mNumDelayedWrites, this); - lastWarning = now; + // write blocked detection + nsecs_t now = systemTime(); + nsecs_t delta = now - mLastWriteTime; + if (!mStandby && delta > maxPeriod) { + mNumDelayedWrites++; + if ((now - lastWarning) > kWarningThrottleNs) { + ATRACE_NAME("underrun"); + ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", + ns2ms(delta), mNumDelayedWrites, this); + lastWarning = now; + } } - } } - mStandby = false; - } else { - usleep(sleepTime); + mStandby = false; + } else { + usleep(sleepTime); + } } // Finally let go of removed track(s), without the lock held @@ -2040,8 +2340,10 @@ if (mType == MIXER) { // is now local to this block, but will keep it for now (at least until merge done). } + threadLoop_exit(); + // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... - if (mType == MIXER || mType == DIRECT) { + if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { // put output stream into standby mode if (!mStandby) { mOutput->stream->common.standby(&mOutput->stream->common); @@ -2054,7 +2356,45 @@ if (mType == MIXER) { return false; } +// removeTracks_l() must be called with ThreadBase::mLock held +void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) +{ + size_t count = tracksToRemove.size(); + if (count) { + for (size_t i=0 ; i<count ; i++) { + const sp<Track>& track = tracksToRemove.itemAt(i); + mActiveTracks.remove(track); + ALOGV("removeTracks_l removing track on session %d", track->sessionId()); + sp<EffectChain> 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); + } + } + } + +} +status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) +{ + if (mNormalSink != 0) { + return mNormalSink->getTimestamp(timestamp); + } + if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { + uint64_t position64; + int ret = mOutput->stream->get_presentation_position( + mOutput->stream, &position64, ×tamp.mTime); + if (ret == 0) { + timestamp.mPosition = (uint32_t)position64; + return NO_ERROR; + } + } + return INVALID_OPERATION; +} // ---------------------------------------------------------------------------- AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, @@ -2068,7 +2408,7 @@ AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, Aud // mNormalSink below { ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); - ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " + ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " "mFrameCount=%d, mNormalFrameCount=%d", mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, mNormalFrameCount); @@ -2258,7 +2598,7 @@ void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> PlaybackThread::threadLoop_removeTracks(tracksToRemove); } -void AudioFlinger::MixerThread::threadLoop_write() +ssize_t 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 @@ -2279,6 +2619,8 @@ void AudioFlinger::MixerThread::threadLoop_write() #endif } state->mCommand = FastMixerState::MIX_WRITE; + mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? + FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); sq->end(); sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); if (kUseFastMixer == FastMixer_Dynamic) { @@ -2288,7 +2630,7 @@ void AudioFlinger::MixerThread::threadLoop_write() sq->end(false /*didModify*/); } } - PlaybackThread::threadLoop_write(); + return PlaybackThread::threadLoop_write(); } void AudioFlinger::MixerThread::threadLoop_standby() @@ -2320,11 +2662,41 @@ void AudioFlinger::MixerThread::threadLoop_standby() PlaybackThread::threadLoop_standby(); } +// Empty implementation for standard mixer +// Overridden for offloaded playback +void AudioFlinger::PlaybackThread::flushOutput_l() +{ +} + +bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() +{ + return false; +} + +bool AudioFlinger::PlaybackThread::shouldStandby_l() +{ + return !mStandby; +} + +bool AudioFlinger::PlaybackThread::waitingAsyncCallback() +{ + Mutex::Autolock _l(mLock); + return waitingAsyncCallback_l(); +} + // 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); + if (mUseAsyncWrite != 0) { + // discard any pending drain or write ack by incrementing sequence + mWriteAckSequence = (mWriteAckSequence + 2) & ~1; + mDrainSequence = (mDrainSequence + 2) & ~1; + ALOG_ASSERT(mCallbackThread != 0); + mCallbackThread->setWriteBlocked(mWriteAckSequence); + mCallbackThread->setDraining(mDrainSequence); + } } void AudioFlinger::MixerThread::threadLoop_mix() @@ -2345,6 +2717,7 @@ void AudioFlinger::MixerThread::threadLoop_mix() // mix buffers... mAudioMixer->process(pts); + mCurrentWriteLength = mixBufferSize; // 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 @@ -2426,7 +2799,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac } for (size_t i=0 ; i<count ; i++) { - sp<Track> t = mActiveTracks[i].promote(); + const sp<Track> t = mActiveTracks[i].promote(); if (t == 0) { continue; } @@ -2462,8 +2835,10 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac 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; + if (!(track->isStopping() || track->isPausing() || track->isStopped()) && + recentUnderruns > 0) { + // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun + track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); } // This is similar to the state machine for normal tracks, @@ -2472,7 +2847,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac switch (track->mState) { case TrackBase::STOPPING_1: // track stays active in STOPPING_1 state until first underrun - if (recentUnderruns > 0) { + if (recentUnderruns > 0 || track->isTerminated()) { track->mState = TrackBase::STOPPING_2; } break; @@ -2506,7 +2881,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac } // 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); + android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); // remove from active list, but state remains ACTIVE [confusing but true] isActive = false; break; @@ -2514,7 +2889,6 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac // 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 @@ -2595,28 +2969,39 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac // 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 + size_t desiredFrames; + uint32_t sr = track->sampleRate(); + if (sr == mSampleRate) { + desiredFrames = mNormalFrameCount; + } else { + // +1 for rounding and +1 for additional sample needed for interpolation + desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; + // add frames already consumed but not yet released by the resampler + // because cblk->framesReady() will include these frames + desiredFrames += 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(desiredFrames <= cblk->frameCount_); + } 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_); - } + minFrames = desiredFrames; } - if ((track->framesReady() >= minFrames) && track->isReady() && + // It's not safe to call framesReady() for a static buffer track, so assume it's ready + size_t framesReady; + if (track->sharedBuffer() == 0) { + framesReady = track->framesReady(); + } else if (track->isStopped()) { + framesReady = 0; + } else { + framesReady = 1; + } + if ((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); + ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); mixedTracks++; @@ -2645,7 +3030,8 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac param = AudioMixer::RAMP_VOLUME; } mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); - } else if (cblk->server != 0) { + // FIXME should not make a decision based on mServer + } else if (cblk->mServer != 0) { // If the track is stopped before the first frame was mixed, // do not apply ramp param = AudioMixer::RAMP_VOLUME; @@ -2663,7 +3049,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac // read original volumes with volume control float typeVolume = mStreamTypes[track->streamType()].volume; float v = masterVolume * typeVolume; - ServerProxy *proxy = track->mServerProxy; + AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; uint32_t vlr = proxy->getVolumeLR(); vl = vlr & 0xFFFF; vr = vlr >> 16; @@ -2690,6 +3076,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac } 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 @@ -2736,7 +3123,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); // limit track sample rate to 2 x output sample rate, which changes at re-configuration uint32_t maxSampleRate = mSampleRate * 2; - uint32_t reqSampleRate = track->mServerProxy->getSampleRate(); + uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); if (reqSampleRate == 0) { reqSampleRate = mSampleRate; } else if (reqSampleRate > maxSampleRate) { @@ -2767,6 +3154,9 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac mixerStatus = MIXER_TRACKS_READY; } } else { + if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { + track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); + } // clear effect chain input buffer if an active track underruns to avoid sending // previous audio buffer again to effects chain = getEffectChain_l(track->sessionId()); @@ -2774,8 +3164,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac chain->clearInputBuffer(); } - ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, - cblk->server, this); + ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); if ((track->sharedBuffer() != 0) || track->isTerminated() || track->isStopped() || track->isPaused()) { // We have consumed all the buffers of this track. @@ -2791,7 +3180,6 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac 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) { @@ -2799,7 +3187,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac 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); + android_atomic_or(CBLK_DISABLED, &cblk->mFlags); // 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 @@ -2861,30 +3249,13 @@ track_is_ready: ; } // 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); - } - } - } + removeTracks_l(*tracksToRemove); // 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)) { + if ((mBytesRemaining == 0) && ((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)); } @@ -2948,7 +3319,7 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l() } } if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { - if (value != AUDIO_CHANNEL_OUT_STEREO) { + if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { status = BAD_VALUE; } else { reconfig = true; @@ -3009,10 +3380,8 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l() keyValuePair.string()); } if (status == NO_ERROR && reconfig) { - delete mAudioMixer; - // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) - mAudioMixer = NULL; readOutputParameters(); + delete mAudioMixer; mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); for (size_t i = 0; i < mTracks.size() ; i++) { int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); @@ -3061,7 +3430,7 @@ void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& ar 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; + const FastMixerDumpState copy(mFastMixerDumpState); copy.dump(fd); #ifdef STATE_QUEUE_DUMP @@ -3116,10 +3485,63 @@ AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& aud { } +AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, + AudioStreamOut* output, audio_io_handle_t id, uint32_t device, + ThreadBase::type_t type) + : PlaybackThread(audioFlinger, output, id, device, type) + // mLeftVolFloat, mRightVolFloat +{ +} + AudioFlinger::DirectOutputThread::~DirectOutputThread() { } +void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) +{ + audio_track_cblk_t* cblk = track->cblk(); + float left, right; + + if (mMasterMute || mStreamTypes[track->streamType()].mute) { + left = right = 0; + } else { + float typeVolume = mStreamTypes[track->streamType()].volume; + float v = mMasterVolume * typeVolume; + AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; + uint32_t vlr = proxy->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 (lastTrack) { + 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()) { + mEffectChains[0]->setVolume_l(&vl, &vr); + left = (float)vl / (1 << 24); + right = (float)vr / (1 << 24); + } + if (mOutput->stream->set_volume) { + mOutput->stream->set_volume(mOutput->stream, left, right); + } + } + } +} + + AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( Vector< sp<Track> > *tracksToRemove ) @@ -3146,66 +3568,29 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep } else { minFrames = 1; } + // Only consider last track started for volume and mixer state control. + // This is the last entry in mActiveTracks unless a track underruns. + // As we only care about the transition phase between two tracks on a + // direct output, it is not a problem to ignore the underrun case. + bool last = (i == (count - 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); + ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); if (track->mFillingUpStatus == Track::FS_FILLED) { track->mFillingUpStatus = Track::FS_ACTIVE; - mLeftVolFloat = mRightVolFloat = 0; + // make sure processVolume_l() will apply new volume even if 0 + mLeftVolFloat = mRightVolFloat = -1.0; if (track->mState == TrackBase::RESUMING) { track->mState = TrackBase::ACTIVE; } } // compute volume for this track - float left, right; - if (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 = track->mServerProxy->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; - } - // Only consider last track started for volume and mixer state control. - // This is the last entry in mActiveTracks unless a track underruns. - // As we only care about the transition phase between two tracks on a - // direct output, it is not a problem to ignore the underrun case. - if (i == (count - 1)) { - 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); - } - + processVolume_l(track, last); + if (last) { // reset retry count track->mRetryCount = kMaxTrackRetriesDirect; mActiveTrack = t; @@ -3218,7 +3603,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep mEffectChains[0]->clearInputBuffer(); } - ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); + ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); if ((track->sharedBuffer() != 0) || track->isTerminated() || track->isStopped() || track->isPaused()) { // We have consumed all the buffers of this track. @@ -3239,7 +3624,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep if (--(track->mRetryCount) <= 0) { ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); tracksToRemove->add(track); - } else if (i == (count -1)){ + } else if (last) { mixerStatus = MIXER_TRACKS_ENABLED; } } @@ -3247,35 +3632,21 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep } // 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 (!mEffectChains.isEmpty()) { - ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), - track->sessionId()); - mEffectChains[0]->decActiveTrackCnt(); - } - if (track->isTerminated()) { - removeTrack_l(track); - } - } - } + removeTracks_l(*tracksToRemove); 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) { + AudioBufferProvider::Buffer buffer; buffer.frameCount = frameCount; mActiveTrack->getNextBuffer(&buffer); - if (CC_UNLIKELY(buffer.raw == NULL)) { + if (buffer.raw == NULL) { memset(curBuf, 0, frameCount * mFrameSize); break; } @@ -3284,10 +3655,10 @@ void AudioFlinger::DirectOutputThread::threadLoop_mix() curBuf += buffer.frameCount * mFrameSize; mActiveTrack->releaseBuffer(&buffer); } + mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; sleepTime = 0; standbyTime = systemTime() + standbyDelay; mActiveTrack.clear(); - } void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() @@ -3403,7 +3774,358 @@ void AudioFlinger::DirectOutputThread::cacheParameters_l() // use shorter standby delay as on normal output to release // hardware resources as soon as possible - standbyDelay = microseconds(activeSleepTime*2); + if (audio_is_linear_pcm(mFormat)) { + standbyDelay = microseconds(activeSleepTime*2); + } else { + standbyDelay = kOffloadStandbyDelayNs; + } +} + +// ---------------------------------------------------------------------------- + +AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( + const wp<AudioFlinger::PlaybackThread>& playbackThread) + : Thread(false /*canCallJava*/), + mPlaybackThread(playbackThread), + mWriteAckSequence(0), + mDrainSequence(0) +{ +} + +AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() +{ +} + +void AudioFlinger::AsyncCallbackThread::onFirstRef() +{ + run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); +} + +bool AudioFlinger::AsyncCallbackThread::threadLoop() +{ + while (!exitPending()) { + uint32_t writeAckSequence; + uint32_t drainSequence; + + { + Mutex::Autolock _l(mLock); + mWaitWorkCV.wait(mLock); + if (exitPending()) { + break; + } + ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", + mWriteAckSequence, mDrainSequence); + writeAckSequence = mWriteAckSequence; + mWriteAckSequence &= ~1; + drainSequence = mDrainSequence; + mDrainSequence &= ~1; + } + { + sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); + if (playbackThread != 0) { + if (writeAckSequence & 1) { + playbackThread->resetWriteBlocked(writeAckSequence >> 1); + } + if (drainSequence & 1) { + playbackThread->resetDraining(drainSequence >> 1); + } + } + } + } + return false; +} + +void AudioFlinger::AsyncCallbackThread::exit() +{ + ALOGV("AsyncCallbackThread::exit"); + Mutex::Autolock _l(mLock); + requestExit(); + mWaitWorkCV.broadcast(); +} + +void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) +{ + Mutex::Autolock _l(mLock); + // bit 0 is cleared + mWriteAckSequence = sequence << 1; +} + +void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() +{ + Mutex::Autolock _l(mLock); + // ignore unexpected callbacks + if (mWriteAckSequence & 2) { + mWriteAckSequence |= 1; + mWaitWorkCV.signal(); + } +} + +void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) +{ + Mutex::Autolock _l(mLock); + // bit 0 is cleared + mDrainSequence = sequence << 1; +} + +void AudioFlinger::AsyncCallbackThread::resetDraining() +{ + Mutex::Autolock _l(mLock); + // ignore unexpected callbacks + if (mDrainSequence & 2) { + mDrainSequence |= 1; + mWaitWorkCV.signal(); + } +} + + +// ---------------------------------------------------------------------------- +AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, + AudioStreamOut* output, audio_io_handle_t id, uint32_t device) + : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), + mHwPaused(false), + mFlushPending(false), + mPausedBytesRemaining(0) +{ +} + +AudioFlinger::OffloadThread::~OffloadThread() +{ + mPreviousTrack.clear(); +} + +void AudioFlinger::OffloadThread::threadLoop_exit() +{ + if (mFlushPending || mHwPaused) { + // If a flush is pending or track was paused, just discard buffered data + flushHw_l(); + } else { + mMixerStatus = MIXER_DRAIN_ALL; + threadLoop_drain(); + } + mCallbackThread->exit(); + PlaybackThread::threadLoop_exit(); +} + +AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( + Vector< sp<Track> > *tracksToRemove +) +{ + size_t count = mActiveTracks.size(); + + mixer_state mixerStatus = MIXER_IDLE; + bool doHwPause = false; + bool doHwResume = false; + + ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); + + // find out which tracks need to be processed + for (size_t i = 0; i < count; i++) { + sp<Track> t = mActiveTracks[i].promote(); + // The track died recently + if (t == 0) { + continue; + } + Track* const track = t.get(); + audio_track_cblk_t* cblk = track->cblk(); + if (mPreviousTrack != NULL) { + if (t != mPreviousTrack) { + // Flush any data still being written from last track + mBytesRemaining = 0; + if (mPausedBytesRemaining) { + // Last track was paused so we also need to flush saved + // mixbuffer state and invalidate track so that it will + // re-submit that unwritten data when it is next resumed + mPausedBytesRemaining = 0; + // Invalidate is a bit drastic - would be more efficient + // to have a flag to tell client that some of the + // previously written data was lost + mPreviousTrack->invalidate(); + } + } + } + mPreviousTrack = t; + bool last = (i == (count - 1)); + if (track->isPausing()) { + track->setPaused(); + if (last) { + if (!mHwPaused) { + doHwPause = true; + mHwPaused = true; + } + // If we were part way through writing the mixbuffer to + // the HAL we must save this until we resume + // BUG - this will be wrong if a different track is made active, + // in that case we want to discard the pending data in the + // mixbuffer and tell the client to present it again when the + // track is resumed + mPausedWriteLength = mCurrentWriteLength; + mPausedBytesRemaining = mBytesRemaining; + mBytesRemaining = 0; // stop writing + } + tracksToRemove->add(track); + } else if (track->framesReady() && track->isReady() && + !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { + ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); + if (track->mFillingUpStatus == Track::FS_FILLED) { + track->mFillingUpStatus = Track::FS_ACTIVE; + // make sure processVolume_l() will apply new volume even if 0 + mLeftVolFloat = mRightVolFloat = -1.0; + if (track->mState == TrackBase::RESUMING) { + track->mState = TrackBase::ACTIVE; + if (last) { + if (mPausedBytesRemaining) { + // Need to continue write that was interrupted + mCurrentWriteLength = mPausedWriteLength; + mBytesRemaining = mPausedBytesRemaining; + mPausedBytesRemaining = 0; + } + if (mHwPaused) { + doHwResume = true; + mHwPaused = false; + // threadLoop_mix() will handle the case that we need to + // resume an interrupted write + } + // enable write to audio HAL + sleepTime = 0; + } + } + } + + if (last) { + // reset retry count + track->mRetryCount = kMaxTrackRetriesOffload; + mActiveTrack = t; + mixerStatus = MIXER_TRACKS_READY; + } + } else { + ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); + if (track->isStopping_1()) { + // Hardware buffer can hold a large amount of audio so we must + // wait for all current track's data to drain before we say + // that the track is stopped. + if (mBytesRemaining == 0) { + // Only start draining when all data in mixbuffer + // has been written + ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); + track->mState = TrackBase::STOPPING_2; // so presentation completes after drain + if (last) { + sleepTime = 0; + standbyTime = systemTime() + standbyDelay; + mixerStatus = MIXER_DRAIN_TRACK; + mDrainSequence += 2; + if (mHwPaused) { + // It is possible to move from PAUSED to STOPPING_1 without + // a resume so we must ensure hardware is running + mOutput->stream->resume(mOutput->stream); + mHwPaused = false; + } + } + } + } else if (track->isStopping_2()) { + // Drain has completed, signal presentation complete + if (!(mDrainSequence & 1) || !last) { + track->mState = TrackBase::STOPPED; + size_t audioHALFrames = + (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; + size_t framesWritten = + mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); + track->presentationComplete(framesWritten, audioHALFrames); + track->reset(); + tracksToRemove->add(track); + } + } else { + // No buffers for this track. Give it a few chances to + // fill a buffer, then remove it from active list. + if (--(track->mRetryCount) <= 0) { + ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", + track->name()); + tracksToRemove->add(track); + } else if (last){ + mixerStatus = MIXER_TRACKS_ENABLED; + } + } + } + // compute volume for this track + processVolume_l(track, last); + } + + // make sure the pause/flush/resume sequence is executed in the right order. + // If a flush is pending and a track is active but the HW is not paused, force a HW pause + // before flush and then resume HW. This can happen in case of pause/flush/resume + // if resume is received before pause is executed. + if (doHwPause || (mFlushPending && !mHwPaused && (count != 0))) { + mOutput->stream->pause(mOutput->stream); + if (!doHwPause) { + doHwResume = true; + } + } + if (mFlushPending) { + flushHw_l(); + mFlushPending = false; + } + if (doHwResume) { + mOutput->stream->resume(mOutput->stream); + } + + // remove all the tracks that need to be... + removeTracks_l(*tracksToRemove); + + return mixerStatus; +} + +void AudioFlinger::OffloadThread::flushOutput_l() +{ + mFlushPending = true; +} + +// must be called with thread mutex locked +bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() +{ + ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", + mWriteAckSequence, mDrainSequence); + if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { + return true; + } + return false; +} + +// must be called with thread mutex locked +bool AudioFlinger::OffloadThread::shouldStandby_l() +{ + bool TrackPaused = false; + + // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack + // after a timeout and we will enter standby then. + if (mTracks.size() > 0) { + TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); + } + + return !mStandby && !TrackPaused; +} + + +bool AudioFlinger::OffloadThread::waitingAsyncCallback() +{ + Mutex::Autolock _l(mLock); + return waitingAsyncCallback_l(); +} + +void AudioFlinger::OffloadThread::flushHw_l() +{ + mOutput->stream->flush(mOutput->stream); + // Flush anything still waiting in the mixbuffer + mCurrentWriteLength = 0; + mBytesRemaining = 0; + mPausedWriteLength = 0; + mPausedBytesRemaining = 0; + if (mUseAsyncWrite) { + // discard any pending drain or write ack by incrementing sequence + mWriteAckSequence = (mWriteAckSequence + 2) & ~1; + mDrainSequence = (mDrainSequence + 2) & ~1; + ALOG_ASSERT(mCallbackThread != 0); + mCallbackThread->setWriteBlocked(mWriteAckSequence); + mCallbackThread->setDraining(mDrainSequence); + } } // ---------------------------------------------------------------------------- @@ -3434,6 +4156,7 @@ void AudioFlinger::DuplicatingThread::threadLoop_mix() } sleepTime = 0; writeFrames = mNormalFrameCount; + mCurrentWriteLength = mixBufferSize; standbyTime = systemTime() + standbyDelay; } @@ -3457,12 +4180,12 @@ void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() } } -void AudioFlinger::DuplicatingThread::threadLoop_write() +ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() { for (size_t i = 0; i < outputTracks.size(); i++) { outputTracks[i]->write(mMixBuffer, writeFrames); } - mBytesWritten += mixBufferSize; + return (ssize_t)mixBufferSize; } void AudioFlinger::DuplicatingThread::threadLoop_standby() @@ -3583,7 +4306,7 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, ) : ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), - // mRsmpInIndex and mInputBytes set by readInputParameters() + // mRsmpInIndex and mBufferSize set by readInputParameters() mReqChannelCount(popcount(channelMask)), mReqSampleRate(sampleRate) // mBytesRead is only meaningful while active, and so is cleared in start() @@ -3595,7 +4318,7 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, snprintf(mName, kNameLength, "AudioIn_%X", id); readInputParameters(); - + mClientUid = IPCThreadState::self()->getCallingUid(); } @@ -3627,7 +4350,7 @@ bool AudioFlinger::RecordThread::threadLoop() nsecs_t lastWarning = 0; inputStandBy(); - acquireWakeLock(); + acquireWakeLock(mClientUid); // used to verify we've read at least once before evaluating how many bytes were read bool readOnce = false; @@ -3652,11 +4375,14 @@ bool AudioFlinger::RecordThread::threadLoop() // go to sleep mWaitWorkCV.wait(mLock); ALOGV("RecordThread: loop starting"); - acquireWakeLock_l(); + acquireWakeLock_l(mClientUid); continue; } if (mActiveTrack != 0) { - if (mActiveTrack->mState == TrackBase::PAUSING) { + if (mActiveTrack->isTerminated()) { + removeTrack_l(mActiveTrack); + mActiveTrack.clear(); + } else if (mActiveTrack->mState == TrackBase::PAUSING) { standby(); mActiveTrack.clear(); mStartStopCond.broadcast(); @@ -3675,11 +4401,9 @@ bool AudioFlinger::RecordThread::threadLoop() mStartStopCond.broadcast(); } mStandby = false; - } else if (mActiveTrack->mState == TrackBase::TERMINATED) { - removeTrack_l(mActiveTrack); - mActiveTrack.clear(); } } + lockEffectChains_l(effectChains); } @@ -3695,7 +4419,8 @@ bool AudioFlinger::RecordThread::threadLoop() } buffer.frameCount = mFrameCount; - if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { + status_t status = mActiveTrack->getNextBuffer(&buffer); + if (status == NO_ERROR) { readOnce = true; size_t framesOut = buffer.frameCount; if (mResampler == NULL) { @@ -3710,8 +4435,7 @@ bool AudioFlinger::RecordThread::threadLoop() framesIn = framesOut; mRsmpInIndex += framesIn; framesOut -= framesIn; - if (mChannelCount == mReqChannelCount || - mFormat != AUDIO_FORMAT_PCM_16_BIT) { + if (mChannelCount == mReqChannelCount) { memcpy(dst, src, framesIn * mFrameSize); } else { if (mChannelCount == 1) { @@ -3725,9 +4449,7 @@ bool AudioFlinger::RecordThread::threadLoop() } if (framesOut && mFrameCount == mRsmpInIndex) { void *readInto; - if (framesOut == mFrameCount && - (mChannelCount == mReqChannelCount || - mFormat != AUDIO_FORMAT_PCM_16_BIT)) { + if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { readInto = buffer.raw; framesOut = 0; } else { @@ -3735,7 +4457,7 @@ bool AudioFlinger::RecordThread::threadLoop() mRsmpInIndex = 0; } mBytesRead = mInput->stream->read(mInput->stream, readInto, - mInputBytes); + mBufferSize); if (mBytesRead <= 0) { if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { @@ -3760,7 +4482,8 @@ bool AudioFlinger::RecordThread::threadLoop() } else { // resampling - memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); + // resampler accumulates, but we only have one source track + memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); // alter output frame count as if we were expecting stereo samples if (mChannelCount == 1 && mReqChannelCount == 1) { framesOut >>= 1; @@ -3770,6 +4493,7 @@ bool AudioFlinger::RecordThread::threadLoop() // 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) { + // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); // the resampler always outputs stereo samples: // do post stereo to mono conversion @@ -3778,6 +4502,7 @@ bool AudioFlinger::RecordThread::threadLoop() } else { ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); } + // now done with mRsmpOutBuffer } if (mFramestoDrop == 0) { @@ -3826,6 +4551,10 @@ bool AudioFlinger::RecordThread::threadLoop() { Mutex::Autolock _l(mLock); + for (size_t i = 0; i < mTracks.size(); i++) { + sp<RecordTrack> track = mTracks[i]; + track->invalidate(); + } mActiveTrack.clear(); mStartStopCond.broadcast(); } @@ -3856,7 +4585,7 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR audio_channel_mask_t channelMask, size_t frameCount, int sessionId, - IAudioFlinger::track_flags_t flags, + IAudioFlinger::track_flags_t *flags, pid_t tid, status_t *status) { @@ -3865,9 +4594,59 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR lStatus = initCheck(); if (lStatus != NO_ERROR) { - ALOGE("Audio driver not initialized."); + ALOGE("createRecordTrack_l() audio driver not initialized"); goto Exit; } + // client expresses a preference for FAST, but we get the final say + if (*flags & IAudioFlinger::TRACK_FAST) { + if ( + // use case: callback handler and frame count is default or at least as large as HAL + ( + (tid != -1) && + ((frameCount == 0) || + (frameCount >= (mFrameCount * kFastTrackMultiplier))) + ) && + // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) + // mono or stereo + ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || + (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && + // hardware sample rate + (sampleRate == mSampleRate) && + // record thread has an associated fast recorder + hasFastRecorder() + // FIXME test that RecordThread for this fast track has a capable output HAL + // FIXME add a permission test also? + ) { + // if frameCount not specified, then it defaults to fast recorder (HAL) frame count + if (frameCount == 0) { + frameCount = mFrameCount * kFastTrackMultiplier; + } + ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", + frameCount, mFrameCount); + } else { + ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " + "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " + "hasFastRecorder=%d tid=%d", + frameCount, mFrameCount, format, + audio_is_linear_pcm(format), + channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); + *flags &= ~IAudioFlinger::TRACK_FAST; + // For compatibility with AudioRecord calculation, buffer depth is forced + // to be at least 2 x the record thread 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 = 50; // FIXME mInput->stream->get_latency(mInput->stream); + size_t mNormalFrameCount = 2048; // FIXME + uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); + if (minBufCount < 2) { + minBufCount = 2; + } + size_t minFrameCount = mNormalFrameCount * minBufCount; + if (frameCount < minFrameCount) { + frameCount = minFrameCount; + } + } + } // FIXME use flags and tid similar to createTrack_l() @@ -3878,7 +4657,9 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR format, channelMask, frameCount, sessionId); if (track->getCblk() == 0) { + ALOGE("createRecordTrack_l() no control block"); lStatus = NO_MEMORY; + track.clear(); goto Exit; } mTracks.add(track); @@ -3888,6 +4669,13 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createR mAudioFlinger->btNrecIsOff(); setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); setEffectSuspended_l(FX_IID_NS, suspend, sessionId); + + 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; @@ -3969,6 +4757,7 @@ status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrac ALOGV("Record started OK"); return status; } + startError: AudioSystem::stopInput(mId); clearSyncStartEvent(); @@ -4003,8 +4792,9 @@ void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event } } -bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { +bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { ALOGV("RecordThread::stop"); + AutoMutex _l(mLock); if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { return false; } @@ -4055,7 +4845,8 @@ status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) // destroyTrack_l() must be called with ThreadBase::mLock held void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) { - track->mState = TrackBase::TERMINATED; + track->terminate(); + track->mState = TrackBase::STOPPED; // active tracks are removed by threadLoop() if (mActiveTrack != track) { removeTrack_l(track); @@ -4087,7 +4878,7 @@ void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& a if (mActiveTrack != 0) { snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); result.append(buffer); - snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); + snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); result.append(buffer); snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); result.append(buffer); @@ -4140,7 +4931,7 @@ status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* int channelCount; if (framesReady == 0) { - mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); + mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); if (mBytesRead <= 0) { if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { ALOGE("RecordThread::getNextBuffer() Error reading audio input"); @@ -4196,8 +4987,12 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() reconfig = true; } if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { - reqFormat = (audio_format_t) value; - reconfig = true; + if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { + status = BAD_VALUE; + } else { + reqFormat = (audio_format_t) value; + reconfig = true; + } } if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { reqChannelCount = popcount(value); @@ -4289,16 +5084,13 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l() String8 AudioFlinger::RecordThread::getParameters(const String8& keys) { - char *s; - String8 out_s8 = String8(); - Mutex::Autolock _l(mLock); if (initCheck() != NO_ERROR) { - return out_s8; + return String8(); } - s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); - out_s8 = String8(s); + char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); + const String8 out_s8(s); free(s); return out_s8; } @@ -4310,7 +5102,7 @@ void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { switch (event) { case AudioSystem::INPUT_OPENED: case AudioSystem::INPUT_CONFIG_CHANGED: - desc.channels = mChannelMask; + desc.channelMask = mChannelMask; desc.samplingRate = mSampleRate; desc.format = mFormat; desc.frameCount = mFrameCount; @@ -4336,12 +5128,14 @@ void AudioFlinger::RecordThread::readInputParameters() mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); - mChannelCount = (uint16_t)popcount(mChannelMask); + mChannelCount = popcount(mChannelMask); mFormat = mInput->stream->common.get_format(&mInput->stream->common); + if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { + ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); + } 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 + mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); + mFrameCount = mBufferSize / mFrameSize; mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) @@ -4357,7 +5151,7 @@ void AudioFlinger::RecordThread::readInputParameters() mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); mResampler->setSampleRate(mSampleRate); mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); - mRsmpOutBuffer = new int32_t[mFrameCount * 2]; + mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; // optmization: if mono to mono, alter input frame count as if we were inputing // stereo samples |