/* ** ** 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 #define ATRACE_TAG ATRACE_TAG_AUDIO #include "Configuration.h" #include #include #include #include #include #include #include "AudioFlinger.h" #include #include #include #include #include // ---------------------------------------------------------------------------- // 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 { using media::VolumeShaper; // ---------------------------------------------------------------------------- // TrackBase // ---------------------------------------------------------------------------- #undef LOG_TAG #define LOG_TAG "AF::TrackBase" static volatile int32_t nextTrackId = 55; // TrackBase constructor must be called with AudioFlinger::mLock held AudioFlinger::ThreadBase::TrackBase::TrackBase( ThreadBase *thread, const sp& client, const audio_attributes_t& attr, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, size_t bufferSize, audio_session_t sessionId, pid_t creatorPid, uid_t clientUid, bool isOut, alloc_type alloc, track_type type, audio_port_handle_t portId, std::string metricsId) : RefBase(), mThread(thread), mClient(client), mCblk(NULL), // mBuffer, mBufferSize mState(IDLE), mAttr(attr), mSampleRate(sampleRate), mFormat(format), mChannelMask(channelMask), mChannelCount(isOut ? audio_channel_count_from_out_mask(channelMask) : audio_channel_count_from_in_mask(channelMask)), mFrameSize(audio_has_proportional_frames(format) ? mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), mFrameCount(frameCount), mSessionId(sessionId), mIsOut(isOut), mId(android_atomic_inc(&nextTrackId)), mTerminated(false), mType(type), mThreadIoHandle(thread ? thread->id() : AUDIO_IO_HANDLE_NONE), mPortId(portId), mIsInvalid(false), mTrackMetrics(std::move(metricsId), isOut), mCreatorPid(creatorPid) { const uid_t callingUid = IPCThreadState::self()->getCallingUid(); if (!isAudioServerOrMediaServerUid(callingUid) || clientUid == AUDIO_UID_INVALID) { ALOGW_IF(clientUid != AUDIO_UID_INVALID && clientUid != callingUid, "%s(%d): uid %d tried to pass itself off as %d", __func__, mId, callingUid, clientUid); clientUid = callingUid; } // clientUid contains the uid of the app that is responsible for this track, so we can blame // battery usage on it. mUid = clientUid; // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); size_t minBufferSize = buffer == NULL ? roundup(frameCount) : frameCount; // check overflow when computing bufferSize due to multiplication by mFrameSize. if (minBufferSize < frameCount // roundup rounds down for values above UINT_MAX / 2 || mFrameSize == 0 // format needs to be correct || minBufferSize > SIZE_MAX / mFrameSize) { android_errorWriteLog(0x534e4554, "34749571"); return; } minBufferSize *= mFrameSize; if (buffer == nullptr) { bufferSize = minBufferSize; // allocated here. } else if (minBufferSize > bufferSize) { android_errorWriteLog(0x534e4554, "38340117"); return; } size_t size = sizeof(audio_track_cblk_t); if (buffer == NULL && alloc == ALLOC_CBLK) { // check overflow when computing allocation size for streaming tracks. if (size > SIZE_MAX - bufferSize) { android_errorWriteLog(0x534e4554, "34749571"); return; } size += bufferSize; } if (client != 0) { mCblkMemory = client->heap()->allocate(size); if (mCblkMemory == 0 || (mCblk = static_cast(mCblkMemory->unsecurePointer())) == NULL) { ALOGE("%s(%d): not enough memory for AudioTrack size=%zu", __func__, mId, size); client->heap()->dump("AudioTrack"); mCblkMemory.clear(); return; } } else { mCblk = (audio_track_cblk_t *) malloc(size); if (mCblk == NULL) { ALOGE("%s(%d): not enough memory for AudioTrack size=%zu", __func__, mId, size); return; } } // construct the shared structure in-place. if (mCblk != NULL) { new(mCblk) audio_track_cblk_t(); switch (alloc) { case ALLOC_READONLY: { const sp roHeap(thread->readOnlyHeap()); if (roHeap == 0 || (mBufferMemory = roHeap->allocate(bufferSize)) == 0 || (mBuffer = mBufferMemory->unsecurePointer()) == NULL) { ALOGE("%s(%d): not enough memory for read-only buffer size=%zu", __func__, mId, bufferSize); if (roHeap != 0) { roHeap->dump("buffer"); } mCblkMemory.clear(); mBufferMemory.clear(); return; } memset(mBuffer, 0, bufferSize); } break; case ALLOC_PIPE: mBufferMemory = thread->pipeMemory(); // mBuffer is the virtual address as seen from current process (mediaserver), // and should normally be coming from mBufferMemory->unsecurePointer(). // However in this case the TrackBase does not reference the buffer directly. // It should references the buffer via the pipe. // Therefore, to detect incorrect usage of the buffer, we set mBuffer to NULL. mBuffer = NULL; bufferSize = 0; break; case ALLOC_CBLK: // clear all buffers if (buffer == NULL) { mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); memset(mBuffer, 0, bufferSize); } else { mBuffer = buffer; #if 0 mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic #endif } break; case ALLOC_LOCAL: mBuffer = calloc(1, bufferSize); break; case ALLOC_NONE: mBuffer = buffer; break; default: LOG_ALWAYS_FATAL("%s(%d): invalid allocation type: %d", __func__, mId, (int)alloc); } mBufferSize = bufferSize; #ifdef TEE_SINK mTee.set(sampleRate, mChannelCount, format, NBAIO_Tee::TEE_FLAG_TRACK); #endif } } status_t AudioFlinger::ThreadBase::TrackBase::initCheck() const { status_t status; if (mType == TYPE_OUTPUT || mType == TYPE_PATCH) { status = cblk() != NULL ? NO_ERROR : NO_MEMORY; } else { status = getCblk() != 0 ? NO_ERROR : NO_MEMORY; } return status; } AudioFlinger::ThreadBase::TrackBase::~TrackBase() { // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference mServerProxy.clear(); releaseCblk(); mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to if (mClient != 0) { // Client destructor must run with AudioFlinger client mutex locked Mutex::Autolock _l(mClient->audioFlinger()->mClientLock); // 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(); } // flush the binder command buffer IPCThreadState::self()->flushCommands(); } // AudioBufferProvider interface // getNextBuffer() = 0; // This implementation of releaseBuffer() is used by Track and RecordTrack void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) { #ifdef TEE_SINK mTee.write(buffer->raw, buffer->frameCount); #endif ServerProxy::Buffer buf; buf.mFrameCount = buffer->frameCount; buf.mRaw = buffer->raw; buffer->frameCount = 0; buffer->raw = NULL; mServerProxy->releaseBuffer(&buf); } status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp& event) { mSyncEvents.add(event); return NO_ERROR; } AudioFlinger::ThreadBase::PatchTrackBase::PatchTrackBase(sp proxy, const ThreadBase& thread, const Timeout& timeout) : mProxy(proxy) { if (timeout) { setPeerTimeout(*timeout); } else { // Double buffer mixer uint64_t mixBufferNs = ((uint64_t)2 * thread.frameCount() * 1000000000) / thread.sampleRate(); setPeerTimeout(std::chrono::nanoseconds{mixBufferNs}); } } void AudioFlinger::ThreadBase::PatchTrackBase::setPeerTimeout(std::chrono::nanoseconds timeout) { mPeerTimeout.tv_sec = timeout.count() / std::nano::den; mPeerTimeout.tv_nsec = timeout.count() % std::nano::den; } // ---------------------------------------------------------------------------- // Playback // ---------------------------------------------------------------------------- #undef LOG_TAG #define LOG_TAG "AF::TrackHandle" AudioFlinger::TrackHandle::TrackHandle(const sp& 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 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::pause() { mTrack->pause(); } status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) { return mTrack->attachAuxEffect(EffectId); } status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) { return mTrack->setParameters(keyValuePairs); } status_t AudioFlinger::TrackHandle::selectPresentation(int presentationId, int programId) { return mTrack->selectPresentation(presentationId, programId); } VolumeShaper::Status AudioFlinger::TrackHandle::applyVolumeShaper( const sp& configuration, const sp& operation) { return mTrack->applyVolumeShaper(configuration, operation); } sp AudioFlinger::TrackHandle::getVolumeShaperState(int id) { return mTrack->getVolumeShaperState(id); } status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp) { return mTrack->getTimestamp(timestamp); } void AudioFlinger::TrackHandle::signal() { return mTrack->signal(); } status_t AudioFlinger::TrackHandle::onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { return BnAudioTrack::onTransact(code, data, reply, flags); } // ---------------------------------------------------------------------------- // AppOp for audio playback // ------------------------------- // static sp AudioFlinger::PlaybackThread::OpPlayAudioMonitor::createIfNeeded( uid_t uid, const audio_attributes_t& attr, int id, audio_stream_type_t streamType) { if (isServiceUid(uid)) { Vector packages; getPackagesForUid(uid, packages); if (packages.isEmpty()) { ALOGD("OpPlayAudio: not muting track:%d usage:%d for service UID %d", id, attr.usage, uid); return nullptr; } } // stream type has been filtered by audio policy to indicate whether it can be muted if (streamType == AUDIO_STREAM_ENFORCED_AUDIBLE) { ALOGD("OpPlayAudio: not muting track:%d usage:%d ENFORCED_AUDIBLE", id, attr.usage); return nullptr; } if ((attr.flags & AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY) == AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY) { ALOGD("OpPlayAudio: not muting track:%d flags %#x have FLAG_BYPASS_INTERRUPTION_POLICY", id, attr.flags); return nullptr; } return new OpPlayAudioMonitor(uid, attr.usage, id); } AudioFlinger::PlaybackThread::OpPlayAudioMonitor::OpPlayAudioMonitor( uid_t uid, audio_usage_t usage, int id) : mHasOpPlayAudio(true), mUid(uid), mUsage((int32_t) usage), mId(id) { } AudioFlinger::PlaybackThread::OpPlayAudioMonitor::~OpPlayAudioMonitor() { if (mOpCallback != 0) { mAppOpsManager.stopWatchingMode(mOpCallback); } mOpCallback.clear(); } void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::onFirstRef() { getPackagesForUid(mUid, mPackages); checkPlayAudioForUsage(); if (!mPackages.isEmpty()) { mOpCallback = new PlayAudioOpCallback(this); mAppOpsManager.startWatchingMode(AppOpsManager::OP_PLAY_AUDIO, mPackages[0], mOpCallback); } } bool AudioFlinger::PlaybackThread::OpPlayAudioMonitor::hasOpPlayAudio() const { return mHasOpPlayAudio.load(); } // Note this method is never called (and never to be) for audio server / patch record track // - not called from constructor due to check on UID, // - not called from PlayAudioOpCallback because the callback is not installed in this case void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::checkPlayAudioForUsage() { if (mPackages.isEmpty()) { mHasOpPlayAudio.store(false); } else { bool hasIt = true; for (const String16& packageName : mPackages) { const int32_t mode = mAppOpsManager.checkAudioOpNoThrow(AppOpsManager::OP_PLAY_AUDIO, mUsage, mUid, packageName); if (mode != AppOpsManager::MODE_ALLOWED) { hasIt = false; break; } } ALOGD("OpPlayAudio: track:%d usage:%d %smuted", mId, mUsage, hasIt ? "not " : ""); mHasOpPlayAudio.store(hasIt); } } AudioFlinger::PlaybackThread::OpPlayAudioMonitor::PlayAudioOpCallback::PlayAudioOpCallback( const wp& monitor) : mMonitor(monitor) { } void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::PlayAudioOpCallback::opChanged(int32_t op, const String16& packageName) { // we only have uid, so we need to check all package names anyway UNUSED(packageName); if (op != AppOpsManager::OP_PLAY_AUDIO) { return; } sp monitor = mMonitor.promote(); if (monitor != NULL) { monitor->checkPlayAudioForUsage(); } } // static void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::getPackagesForUid( uid_t uid, Vector& packages) { PermissionController permissionController; permissionController.getPackagesForUid(uid, packages); } // ---------------------------------------------------------------------------- #undef LOG_TAG #define LOG_TAG "AF::Track" // Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held AudioFlinger::PlaybackThread::Track::Track( PlaybackThread *thread, const sp& client, audio_stream_type_t streamType, const audio_attributes_t& attr, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, size_t bufferSize, const sp& sharedBuffer, audio_session_t sessionId, pid_t creatorPid, uid_t uid, audio_output_flags_t flags, track_type type, audio_port_handle_t portId, size_t frameCountToBeReady) : TrackBase(thread, client, attr, sampleRate, format, channelMask, frameCount, // TODO: Using unsecurePointer() has some associated security pitfalls // (see declaration for details). // Either document why it is safe in this case or address the // issue (e.g. by copying). (sharedBuffer != 0) ? sharedBuffer->unsecurePointer() : buffer, (sharedBuffer != 0) ? sharedBuffer->size() : bufferSize, sessionId, creatorPid, uid, true /*isOut*/, (type == TYPE_PATCH) ? ( buffer == NULL ? ALLOC_LOCAL : ALLOC_NONE) : ALLOC_CBLK, type, portId, std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(portId)), mFillingUpStatus(FS_INVALID), // mRetryCount initialized later when needed mSharedBuffer(sharedBuffer), mStreamType(streamType), mMainBuffer(thread->sinkBuffer()), mAuxBuffer(NULL), mAuxEffectId(0), mHasVolumeController(false), mPresentationCompleteFrames(0), mFrameMap(16 /* sink-frame-to-track-frame map memory */), mVolumeHandler(new media::VolumeHandler(sampleRate)), mOpPlayAudioMonitor(OpPlayAudioMonitor::createIfNeeded(uid, attr, id(), streamType)), // mSinkTimestamp mFrameCountToBeReady(frameCountToBeReady), mFastIndex(-1), mCachedVolume(1.0), /* The track might not play immediately after being active, similarly as if its volume was 0. * When the track starts playing, its volume will be computed. */ mFinalVolume(0.f), mResumeToStopping(false), mFlushHwPending(false), mFlags(flags) { // client == 0 implies sharedBuffer == 0 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); ALOGV_IF(sharedBuffer != 0, "%s(%d): sharedBuffer: %p, size: %zu", __func__, mId, sharedBuffer->unsecurePointer(), sharedBuffer->size()); if (mCblk == NULL) { return; } if (!thread->isTrackAllowed_l(channelMask, format, sessionId, uid)) { ALOGE("%s(%d): no more tracks available", __func__, mId); releaseCblk(); // this makes the track invalid. return; } if (sharedBuffer == 0) { mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, mFrameSize, !isExternalTrack(), sampleRate); } else { mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, mFrameSize, sampleRate); } mServerProxy = mAudioTrackServerProxy; // only allocate a fast track index if we were able to allocate a normal track name if (flags & AUDIO_OUTPUT_FLAG_FAST) { // FIXME: Not calling framesReadyIsCalledByMultipleThreads() exposes a potential // race with setSyncEvent(). However, if we call it, we cannot properly start // static fast tracks (SoundPool) immediately after stopping. //mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); ALOG_ASSERT(thread->mFastTrackAvailMask != 0); int i = __builtin_ctz(thread->mFastTrackAvailMask); ALOG_ASSERT(0 < i && i < (int)FastMixerState::sMaxFastTracks); // 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; thread->mFastTrackAvailMask &= ~(1 << i); } mServerLatencySupported = thread->type() == ThreadBase::MIXER || thread->type() == ThreadBase::DUPLICATING; #ifdef TEE_SINK mTee.setId(std::string("_") + std::to_string(mThreadIoHandle) + "_" + std::to_string(mId) + "_T"); #endif if (channelMask & AUDIO_CHANNEL_HAPTIC_ALL) { mAudioVibrationController = new AudioVibrationController(this); mExternalVibration = new os::ExternalVibration( mUid, "" /* pkg */, mAttr, mAudioVibrationController); } // Once this item is logged by the server, the client can add properties. mTrackMetrics.logConstructor(creatorPid, uid); } AudioFlinger::PlaybackThread::Track::~Track() { ALOGV("%s(%d)", __func__, mId); // The destructor would clear mSharedBuffer, // but it will not push the decremented reference count, // leaving the client's IMemory dangling indefinitely. // This prevents that leak. if (mSharedBuffer != 0) { mSharedBuffer.clear(); } } status_t AudioFlinger::PlaybackThread::Track::initCheck() const { status_t status = TrackBase::initCheck(); if (status == NO_ERROR && mCblk == nullptr) { status = NO_MEMORY; } return status; } 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 keep(this); { // scope for mLock bool wasActive = false; sp thread = mThread.promote(); if (thread != 0) { Mutex::Autolock _l(thread->mLock); PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); wasActive = playbackThread->destroyTrack_l(this); } if (isExternalTrack() && !wasActive) { AudioSystem::releaseOutput(mPortId); } } forEachTeePatchTrack([](auto patchTrack) { patchTrack->destroy(); }); } void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) { result.appendFormat("Type Id Active Client Session Port Id S Flags " " Format Chn mask SRate " "ST Usg CT " " G db L dB R dB VS dB " " Server FrmCnt FrmRdy F Underruns Flushed" "%s\n", isServerLatencySupported() ? " Latency" : ""); } void AudioFlinger::PlaybackThread::Track::appendDump(String8& result, bool active) { char trackType; switch (mType) { case TYPE_DEFAULT: case TYPE_OUTPUT: if (isStatic()) { trackType = 'S'; // static } else { trackType = ' '; // normal } break; case TYPE_PATCH: trackType = 'P'; break; default: trackType = '?'; } if (isFastTrack()) { result.appendFormat("F%d %c %6d", mFastIndex, trackType, mId); } else { result.appendFormat(" %c %6d", trackType, mId); } char nowInUnderrun; switch (mObservedUnderruns.mBitFields.mMostRecent) { case UNDERRUN_FULL: nowInUnderrun = ' '; break; case UNDERRUN_PARTIAL: nowInUnderrun = '<'; break; case UNDERRUN_EMPTY: nowInUnderrun = '*'; break; default: nowInUnderrun = '?'; break; } char fillingStatus; switch (mFillingUpStatus) { case FS_INVALID: fillingStatus = 'I'; break; case FS_FILLING: fillingStatus = 'f'; break; case FS_FILLED: fillingStatus = 'F'; break; case FS_ACTIVE: fillingStatus = 'A'; break; default: fillingStatus = '?'; break; } // clip framesReadySafe to max representation in dump const size_t framesReadySafe = std::min(mAudioTrackServerProxy->framesReadySafe(), (size_t)99999999); // obtain volumes const gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR(); const std::pair vsVolume = mVolumeHandler->getLastVolume(); // Our effective frame count is obtained by ServerProxy::getBufferSizeInFrames() // as it may be reduced by the application. const size_t bufferSizeInFrames = (size_t)mAudioTrackServerProxy->getBufferSizeInFrames(); // Check whether the buffer size has been modified by the app. const char modifiedBufferChar = bufferSizeInFrames < mFrameCount ? 'r' /* buffer reduced */: bufferSizeInFrames > mFrameCount ? 'e' /* error */ : ' ' /* identical */; result.appendFormat("%7s %6u %7u %7u %2s 0x%03X " "%08X %08X %6u " "%2u %3x %2x " "%5.2g %5.2g %5.2g %5.2g%c " "%08X %6zu%c %6zu %c %9u%c %7u", active ? "yes" : "no", (mClient == 0) ? getpid() : mClient->pid(), mSessionId, mPortId, getTrackStateAsCodedString(), mCblk->mFlags, mFormat, mChannelMask, sampleRate(), mStreamType, mAttr.usage, mAttr.content_type, 20.0 * log10(mFinalVolume), 20.0 * log10(float_from_gain(gain_minifloat_unpack_left(vlr))), 20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))), 20.0 * log10(vsVolume.first), // VolumeShaper(s) total volume vsVolume.second ? 'A' : ' ', // if any VolumeShapers active mCblk->mServer, bufferSizeInFrames, modifiedBufferChar, framesReadySafe, fillingStatus, mAudioTrackServerProxy->getUnderrunFrames(), nowInUnderrun, (unsigned)mAudioTrackServerProxy->framesFlushed() % 10000000 ); if (isServerLatencySupported()) { double latencyMs; bool fromTrack; if (getTrackLatencyMs(&latencyMs, &fromTrack) == OK) { // Show latency in msec, followed by 't' if from track timestamp (the most accurate) // or 'k' if estimated from kernel because track frames haven't been presented yet. result.appendFormat(" %7.2lf %c", latencyMs, fromTrack ? 't' : 'k'); } else { result.appendFormat("%10s", mCblk->mServer != 0 ? "unavail" : "new"); } } result.append("\n"); } uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { return mAudioTrackServerProxy->getSampleRate(); } // AudioBufferProvider interface status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) { ServerProxy::Buffer buf; size_t desiredFrames = buffer->frameCount; buf.mFrameCount = desiredFrames; status_t status = mServerProxy->obtainBuffer(&buf); buffer->frameCount = buf.mFrameCount; buffer->raw = buf.mRaw; if (buf.mFrameCount == 0 && !isStopping() && !isStopped() && !isPaused()) { ALOGV("%s(%d): underrun, framesReady(%zu) < framesDesired(%zd), state: %d", __func__, mId, buf.mFrameCount, desiredFrames, mState); mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); } else { mAudioTrackServerProxy->tallyUnderrunFrames(0); } return status; } void AudioFlinger::PlaybackThread::Track::releaseBuffer(AudioBufferProvider::Buffer* buffer) { interceptBuffer(*buffer); TrackBase::releaseBuffer(buffer); } // TODO: compensate for time shift between HW modules. void AudioFlinger::PlaybackThread::Track::interceptBuffer( const AudioBufferProvider::Buffer& sourceBuffer) { auto start = std::chrono::steady_clock::now(); const size_t frameCount = sourceBuffer.frameCount; if (frameCount == 0) { return; // No audio to intercept. // Additionally PatchProxyBufferProvider::obtainBuffer (called by PathTrack::getNextBuffer) // does not allow 0 frame size request contrary to getNextBuffer } for (auto& teePatch : mTeePatches) { RecordThread::PatchRecord* patchRecord = teePatch.patchRecord.get(); const size_t framesWritten = patchRecord->writeFrames( sourceBuffer.i8, frameCount, mFrameSize); const size_t framesLeft = frameCount - framesWritten; ALOGW_IF(framesLeft != 0, "%s(%d) PatchRecord %d can not provide big enough " "buffer %zu/%zu, dropping %zu frames", __func__, mId, patchRecord->mId, framesWritten, frameCount, framesLeft); } auto spent = ceil(std::chrono::steady_clock::now() - start); using namespace std::chrono_literals; // Average is ~20us per track, this should virtually never be logged (Logging takes >200us) ALOGD_IF(spent > 500us, "%s: took %lldus to intercept %zu tracks", __func__, spent.count(), mTeePatches.size()); } // ExtendedAudioBufferProvider interface // framesReady() may return an approximation of the number of frames if called // from a different thread than the one calling Proxy->obtainBuffer() and // Proxy->releaseBuffer(). Also note there is no mutual exclusion in the // AudioTrackServerProxy so be especially careful calling with FastTracks. size_t AudioFlinger::PlaybackThread::Track::framesReady() const { if (mSharedBuffer != 0 && (isStopped() || isStopping())) { // Static tracks return zero frames immediately upon stopping (for FastTracks). // The remainder of the buffer is not drained. return 0; } return mAudioTrackServerProxy->framesReady(); } int64_t AudioFlinger::PlaybackThread::Track::framesReleased() const { return mAudioTrackServerProxy->framesReleased(); } void AudioFlinger::PlaybackThread::Track::onTimestamp(const ExtendedTimestamp ×tamp) { // This call comes from a FastTrack and should be kept lockless. // The server side frames are already translated to client frames. mAudioTrackServerProxy->setTimestamp(timestamp); // We do not set drained here, as FastTrack timestamp may not go to very last frame. // Compute latency. // TODO: Consider whether the server latency may be passed in by FastMixer // as a constant for all active FastTracks. const double latencyMs = timestamp.getOutputServerLatencyMs(sampleRate()); mServerLatencyFromTrack.store(true); mServerLatencyMs.store(latencyMs); } // 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 (isStopping()) { if (framesReady() > 0) { mFillingUpStatus = FS_FILLED; } return true; } size_t bufferSizeInFrames = mServerProxy->getBufferSizeInFrames(); size_t framesToBeReady = std::min(mFrameCountToBeReady, bufferSizeInFrames); if (framesReady() >= framesToBeReady || (mCblk->mFlags & CBLK_FORCEREADY)) { ALOGV("%s(%d): consider track ready with %zu/%zu, target was %zu)", __func__, mId, framesReady(), bufferSizeInFrames, framesToBeReady); mFillingUpStatus = FS_FILLED; android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); return true; } return false; } status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused, audio_session_t triggerSession __unused) { status_t status = NO_ERROR; ALOGV("%s(%d): calling pid %d session %d", __func__, mId, IPCThreadState::self()->getCallingPid(), mSessionId); sp thread = mThread.promote(); if (thread != 0) { if (isOffloaded()) { Mutex::Autolock _laf(thread->mAudioFlinger->mLock); Mutex::Autolock _lth(thread->mLock); sp ec = thread->getEffectChain_l(mSessionId); if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() || (ec != 0 && ec->isNonOffloadableEnabled())) { invalidate(); return PERMISSION_DENIED; } } Mutex::Autolock _lth(thread->mLock); track_state state = mState; // here the track could be either new, or restarted // in both cases "unstop" the track // initial state-stopping. next state-pausing. // What if resume is called ? if (state == PAUSED || state == PAUSING) { if (mResumeToStopping) { // happened we need to resume to STOPPING_1 mState = TrackBase::STOPPING_1; ALOGV("%s(%d): PAUSED => STOPPING_1 on thread %d", __func__, mId, (int)mThreadIoHandle); } else { mState = TrackBase::RESUMING; ALOGV("%s(%d): PAUSED => RESUMING on thread %d", __func__, mId, (int)mThreadIoHandle); } } else { mState = TrackBase::ACTIVE; ALOGV("%s(%d): ? => ACTIVE on thread %d", __func__, mId, (int)mThreadIoHandle); } // states to reset position info for non-offloaded/direct tracks if (!isOffloaded() && !isDirect() && (state == IDLE || state == STOPPED || state == FLUSHED)) { mFrameMap.reset(); } PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); if (isFastTrack()) { // refresh fast track underruns on start because that field is never cleared // by the fast mixer; furthermore, the same track can be recycled, i.e. start // after stop. mObservedUnderruns = playbackThread->getFastTrackUnderruns(mFastIndex); } status = playbackThread->addTrack_l(this); if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); // restore previous state if start was rejected by policy manager if (status == PERMISSION_DENIED) { mState = state; } } // Audio timing metrics are computed a few mix cycles after starting. { mLogStartCountdown = LOG_START_COUNTDOWN; mLogStartTimeNs = systemTime(); mLogStartFrames = mAudioTrackServerProxy->getTimestamp() .mPosition[ExtendedTimestamp::LOCATION_SERVER]; } if (status == NO_ERROR || status == ALREADY_EXISTS) { // for streaming tracks, remove the buffer read stop limit. mAudioTrackServerProxy->start(); } // track was already in the active list, not a problem if (status == ALREADY_EXISTS) { status = NO_ERROR; } else { // Acknowledge any pending flush(), so that subsequent new data isn't discarded. // It is usually unsafe to access the server proxy from a binder thread. // But in this case we know the mixer thread (whether normal mixer or fast mixer) // isn't looking at this track yet: we still hold the normal mixer thread lock, // and for fast tracks the track is not yet in the fast mixer thread's active set. // For static tracks, this is used to acknowledge change in position or loop. ServerProxy::Buffer buffer; buffer.mFrameCount = 1; (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/); } } else { status = BAD_VALUE; } if (status == NO_ERROR) { forEachTeePatchTrack([](auto patchTrack) { patchTrack->start(); }); } return status; } void AudioFlinger::PlaybackThread::Track::stop() { ALOGV("%s(%d): calling pid %d", __func__, mId, IPCThreadState::self()->getCallingPid()); sp 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() && !isOffloaded() && !isDirect()) { mState = STOPPED; } else { // For fast tracks prepareTracks_l() will set state to STOPPING_2 // presentation is complete // For an offloaded track this starts a drain and state will // move to STOPPING_2 when drain completes and then STOPPED mState = STOPPING_1; if (isOffloaded()) { mRetryCount = PlaybackThread::kMaxTrackStopRetriesOffload; } } playbackThread->broadcast_l(); ALOGV("%s(%d): not stopping/stopped => stopping/stopped on thread %d", __func__, mId, (int)mThreadIoHandle); } } forEachTeePatchTrack([](auto patchTrack) { patchTrack->stop(); }); } void AudioFlinger::PlaybackThread::Track::pause() { ALOGV("%s(%d): calling pid %d", __func__, mId, IPCThreadState::self()->getCallingPid()); sp thread = mThread.promote(); if (thread != 0) { Mutex::Autolock _l(thread->mLock); PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); switch (mState) { case STOPPING_1: case STOPPING_2: if (!isOffloaded()) { /* nothing to do if track is not offloaded */ break; } // Offloaded track was draining, we need to carry on draining when resumed mResumeToStopping = true; FALLTHROUGH_INTENDED; case ACTIVE: case RESUMING: mState = PAUSING; ALOGV("%s(%d): ACTIVE/RESUMING => PAUSING on thread %d", __func__, mId, (int)mThreadIoHandle); playbackThread->broadcast_l(); break; default: break; } } // Pausing the TeePatch to avoid a glitch on underrun, at the cost of buffered audio loss. forEachTeePatchTrack([](auto patchTrack) { patchTrack->pause(); }); } void AudioFlinger::PlaybackThread::Track::flush() { ALOGV("%s(%d)", __func__, mId); sp thread = mThread.promote(); if (thread != 0) { Mutex::Autolock _l(thread->mLock); PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); // Flush the ring buffer now if the track is not active in the PlaybackThread. // Otherwise the flush would not be done until the track is resumed. // Requires FastTrack removal be BLOCK_UNTIL_ACKED if (playbackThread->mActiveTracks.indexOf(this) < 0) { (void)mServerProxy->flushBufferIfNeeded(); } if (isOffloaded()) { // If offloaded we allow flush during any state except terminated // and keep the track active to avoid problems if user is seeking // rapidly and underlying hardware has a significant delay handling // a pause if (isTerminated()) { return; } ALOGV("%s(%d): offload flush", __func__, mId); reset(); if (mState == STOPPING_1 || mState == STOPPING_2) { ALOGV("%s(%d): flushed in STOPPING_1 or 2 state, change state to ACTIVE", __func__, mId); mState = ACTIVE; } mFlushHwPending = true; mResumeToStopping = false; } else { 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 if (isDirect()) { mFlushHwPending = true; } if (playbackThread->mActiveTracks.indexOf(this) < 0) { reset(); } } // Prevent flush being lost if the track is flushed and then resumed // before mixer thread can run. This is important when offloading // because the hardware buffer could hold a large amount of audio playbackThread->broadcast_l(); } // Flush the Tee to avoid on resume playing old data and glitching on the transition to new data forEachTeePatchTrack([](auto patchTrack) { patchTrack->flush(); }); } // must be called with thread lock held void AudioFlinger::PlaybackThread::Track::flushAck() { if (!isOffloaded() && !isDirect()) return; // Clear the client ring buffer so that the app can prime the buffer while paused. // Otherwise it might not get cleared until playback is resumed and obtainBuffer() is called. mServerProxy->flushBufferIfNeeded(); mFlushHwPending = false; } 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) { // Force underrun condition to avoid false underrun callback until first data is // written to buffer android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); mFillingUpStatus = FS_FILLING; mResetDone = true; if (mState == FLUSHED) { mState = IDLE; } } } status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) { sp thread = mThread.promote(); if (thread == 0) { ALOGE("%s(%d): thread is dead", __func__, mId); return FAILED_TRANSACTION; } else if ((thread->type() == ThreadBase::DIRECT) || (thread->type() == ThreadBase::OFFLOAD)) { return thread->setParameters(keyValuePairs); } else { return PERMISSION_DENIED; } } status_t AudioFlinger::PlaybackThread::Track::selectPresentation(int presentationId, int programId) { sp thread = mThread.promote(); if (thread == 0) { ALOGE("thread is dead"); return FAILED_TRANSACTION; } else if ((thread->type() == ThreadBase::DIRECT) || (thread->type() == ThreadBase::OFFLOAD)) { DirectOutputThread *directOutputThread = static_cast(thread.get()); return directOutputThread->selectPresentation(presentationId, programId); } return INVALID_OPERATION; } VolumeShaper::Status AudioFlinger::PlaybackThread::Track::applyVolumeShaper( const sp& configuration, const sp& operation) { sp newConfiguration; if (isOffloadedOrDirect()) { const VolumeShaper::Configuration::OptionFlag optionFlag = configuration->getOptionFlags(); if ((optionFlag & VolumeShaper::Configuration::OPTION_FLAG_CLOCK_TIME) == 0) { ALOGW("%s(%d): %s tracks do not support frame counted VolumeShaper," " using clock time instead", __func__, mId, isOffloaded() ? "Offload" : "Direct"); newConfiguration = new VolumeShaper::Configuration(*configuration); newConfiguration->setOptionFlags( VolumeShaper::Configuration::OptionFlag(optionFlag | VolumeShaper::Configuration::OPTION_FLAG_CLOCK_TIME)); } } VolumeShaper::Status status = mVolumeHandler->applyVolumeShaper( (newConfiguration.get() != nullptr ? newConfiguration : configuration), operation); if (isOffloadedOrDirect()) { // Signal thread to fetch new volume. sp thread = mThread.promote(); if (thread != 0) { Mutex::Autolock _l(thread->mLock); thread->broadcast_l(); } } return status; } sp AudioFlinger::PlaybackThread::Track::getVolumeShaperState(int id) { // Note: We don't check if Thread exists. // mVolumeHandler is thread safe. return mVolumeHandler->getVolumeShaperState(id); } void AudioFlinger::PlaybackThread::Track::setFinalVolume(float volume) { if (mFinalVolume != volume) { // Compare to an epsilon if too many meaningless updates mFinalVolume = volume; setMetadataHasChanged(); mTrackMetrics.logVolume(volume); } } void AudioFlinger::PlaybackThread::Track::copyMetadataTo(MetadataInserter& backInserter) const { *backInserter++ = { .usage = mAttr.usage, .content_type = mAttr.content_type, .gain = mFinalVolume, }; } void AudioFlinger::PlaybackThread::Track::setTeePatches(TeePatches teePatches) { forEachTeePatchTrack([](auto patchTrack) { patchTrack->destroy(); }); mTeePatches = std::move(teePatches); } status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp) { if (!isOffloaded() && !isDirect()) { return INVALID_OPERATION; // normal tracks handled through SSQ } sp thread = mThread.promote(); if (thread == 0) { return INVALID_OPERATION; } Mutex::Autolock _l(thread->mLock); PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); return playbackThread->getTimestamp_l(timestamp); } status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) { sp thread = mThread.promote(); if (thread == nullptr) { return DEAD_OBJECT; } sp dstThread = (PlaybackThread *)thread.get(); sp srcThread; // srcThread is initialized by call to moveAuxEffectToIo() sp af = mClient->audioFlinger(); status_t status = af->moveAuxEffectToIo(EffectId, dstThread, &srcThread); if (EffectId != 0 && status == NO_ERROR) { status = dstThread->attachAuxEffect(this, EffectId); if (status == NO_ERROR) { AudioSystem::moveEffectsToIo(std::vector(EffectId), dstThread->id()); } } if (status != NO_ERROR && srcThread != nullptr) { af->moveAuxEffectToIo(EffectId, srcThread, &dstThread); } return status; } void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) { mAuxEffectId = EffectId; mAuxBuffer = buffer; } bool AudioFlinger::PlaybackThread::Track::presentationComplete( int64_t framesWritten, size_t audioHalFrames) { // TODO: improve this based on FrameMap if it exists, to ensure full drain. // This assists in proper timestamp computation as well as wakelock management. // 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. // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used // to detect when all frames have been played. In this case framesWritten isn't // useful because it doesn't always reflect whether there is data in the h/w // buffers, particularly if a track has been paused and resumed during draining ALOGV("%s(%d): presentationComplete() mPresentationCompleteFrames %lld framesWritten %lld", __func__, mId, (long long)mPresentationCompleteFrames, (long long)framesWritten); if (mPresentationCompleteFrames == 0) { mPresentationCompleteFrames = framesWritten + audioHalFrames; ALOGV("%s(%d): presentationComplete() reset:" " mPresentationCompleteFrames %lld audioHalFrames %zu", __func__, mId, (long long)mPresentationCompleteFrames, audioHalFrames); } bool complete; if (isOffloaded()) { complete = true; } else if (isDirect() || isFastTrack()) { // these do not go through linear map complete = framesWritten >= (int64_t) mPresentationCompleteFrames; } else { // Normal tracks, OutputTracks, and PatchTracks complete = framesWritten >= (int64_t) mPresentationCompleteFrames && mAudioTrackServerProxy->isDrained(); } if (complete) { triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); mAudioTrackServerProxy->setStreamEndDone(); return true; } return false; } void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) { for (size_t i = 0; i < mSyncEvents.size();) { if (mSyncEvents[i]->type() == type) { mSyncEvents[i]->trigger(); mSyncEvents.removeAt(i); } else { ++i; } } } // implement VolumeBufferProvider interface gain_minifloat_packed_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)); gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR(); float vl = float_from_gain(gain_minifloat_unpack_left(vlr)); float vr = float_from_gain(gain_minifloat_unpack_right(vlr)); // track volumes come from shared memory, so can't be trusted and must be clamped if (vl > GAIN_FLOAT_UNITY) { vl = GAIN_FLOAT_UNITY; } if (vr > GAIN_FLOAT_UNITY) { vr = GAIN_FLOAT_UNITY; } // 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 packed minifloat vlr = gain_minifloat_pack(gain_from_float(vl), gain_from_float(vr)); // FIXME look at mute, pause, and stop flags return vlr; } status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp& event) { if (isTerminated() || mState == PAUSED || ((framesReady() == 0) && ((mSharedBuffer != 0) || (mState == STOPPED)))) { ALOGW("%s(%d): in invalid state %d on session %d %s mode, framesReady %zu", __func__, mId, mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); event->cancel(); return INVALID_OPERATION; } (void) TrackBase::setSyncEvent(event); return NO_ERROR; } void AudioFlinger::PlaybackThread::Track::invalidate() { TrackBase::invalidate(); signalClientFlag(CBLK_INVALID); } void AudioFlinger::PlaybackThread::Track::disable() { // TODO(b/142394888): the filling status should also be reset to filling signalClientFlag(CBLK_DISABLED); } void AudioFlinger::PlaybackThread::Track::signalClientFlag(int32_t flag) { // FIXME should use proxy, and needs work audio_track_cblk_t* cblk = mCblk; android_atomic_or(flag, &cblk->mFlags); android_atomic_release_store(0x40000000, &cblk->mFutex); // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX); } void AudioFlinger::PlaybackThread::Track::signal() { sp thread = mThread.promote(); if (thread != 0) { PlaybackThread *t = (PlaybackThread *)thread.get(); Mutex::Autolock _l(t->mLock); t->broadcast_l(); } } //To be called with thread lock held bool AudioFlinger::PlaybackThread::Track::isResumePending() { if (mState == RESUMING) return true; /* Resume is pending if track was stopping before pause was called */ if (mState == STOPPING_1 && mResumeToStopping) return true; return false; } //To be called with thread lock held void AudioFlinger::PlaybackThread::Track::resumeAck() { if (mState == RESUMING) mState = ACTIVE; // Other possibility of pending resume is stopping_1 state // Do not update the state from stopping as this prevents // drain being called. if (mState == STOPPING_1) { mResumeToStopping = false; } } //To be called with thread lock held void AudioFlinger::PlaybackThread::Track::updateTrackFrameInfo( int64_t trackFramesReleased, int64_t sinkFramesWritten, uint32_t halSampleRate, const ExtendedTimestamp &timeStamp) { // Make the kernel frametime available. const FrameTime ft{ timeStamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL], timeStamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]}; // ALOGD("FrameTime: %lld %lld", (long long)ft.frames, (long long)ft.timeNs); mKernelFrameTime.store(ft); if (!audio_is_linear_pcm(mFormat)) { return; } //update frame map mFrameMap.push(trackFramesReleased, sinkFramesWritten); // adjust server times and set drained state. // // Our timestamps are only updated when the track is on the Thread active list. // We need to ensure that tracks are not removed before full drain. ExtendedTimestamp local = timeStamp; bool drained = true; // default assume drained, if no server info found bool checked = false; for (int i = ExtendedTimestamp::LOCATION_MAX - 1; i >= ExtendedTimestamp::LOCATION_SERVER; --i) { // Lookup the track frame corresponding to the sink frame position. if (local.mTimeNs[i] > 0) { local.mPosition[i] = mFrameMap.findX(local.mPosition[i]); // check drain state from the latest stage in the pipeline. if (!checked && i <= ExtendedTimestamp::LOCATION_KERNEL) { drained = local.mPosition[i] >= mAudioTrackServerProxy->framesReleased(); checked = true; } } } mAudioTrackServerProxy->setDrained(drained); // Set correction for flushed frames that are not accounted for in released. local.mFlushed = mAudioTrackServerProxy->framesFlushed(); mServerProxy->setTimestamp(local); // Compute latency info. const bool useTrackTimestamp = !drained; const double latencyMs = useTrackTimestamp ? local.getOutputServerLatencyMs(sampleRate()) : timeStamp.getOutputServerLatencyMs(halSampleRate); mServerLatencyFromTrack.store(useTrackTimestamp); mServerLatencyMs.store(latencyMs); if (mLogStartCountdown > 0) { if (--mLogStartCountdown == 0) { // startup is the difference in times for the current timestamp and our start double startUpMs = (local.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] - mLogStartTimeNs) * 1e-6; // adjust for frames played. startUpMs -= (local.mPosition[ExtendedTimestamp::LOCATION_SERVER] - mLogStartFrames) * 1e3 / mSampleRate; ALOGV("%s: logging localTime:%lld, startTime:%lld" " localPosition:%lld, startPosition:%lld", __func__, (long long)local.mTimeNs[ExtendedTimestamp::LOCATION_SERVER], (long long)mLogStartTimeNs, (long long)local.mPosition[ExtendedTimestamp::LOCATION_SERVER], (long long)mLogStartFrames); mTrackMetrics.logLatencyAndStartup(latencyMs, startUpMs); } } } binder::Status AudioFlinger::PlaybackThread::Track::AudioVibrationController::mute( /*out*/ bool *ret) { *ret = false; sp thread = mTrack->mThread.promote(); if (thread != 0) { // Lock for updating mHapticPlaybackEnabled. Mutex::Autolock _l(thread->mLock); PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); if ((mTrack->channelMask() & AUDIO_CHANNEL_HAPTIC_ALL) != AUDIO_CHANNEL_NONE && playbackThread->mHapticChannelCount > 0) { mTrack->setHapticPlaybackEnabled(false); *ret = true; } } return binder::Status::ok(); } binder::Status AudioFlinger::PlaybackThread::Track::AudioVibrationController::unmute( /*out*/ bool *ret) { *ret = false; sp thread = mTrack->mThread.promote(); if (thread != 0) { // Lock for updating mHapticPlaybackEnabled. Mutex::Autolock _l(thread->mLock); PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); if ((mTrack->channelMask() & AUDIO_CHANNEL_HAPTIC_ALL) != AUDIO_CHANNEL_NONE && playbackThread->mHapticChannelCount > 0) { mTrack->setHapticPlaybackEnabled(true); *ret = true; } } return binder::Status::ok(); } // ---------------------------------------------------------------------------- #undef LOG_TAG #define LOG_TAG "AF::OutputTrack" AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( PlaybackThread *playbackThread, DuplicatingThread *sourceThread, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, uid_t uid) : Track(playbackThread, NULL, AUDIO_STREAM_PATCH, audio_attributes_t{} /* currently unused for output track */, sampleRate, format, channelMask, frameCount, nullptr /* buffer */, (size_t)0 /* bufferSize */, nullptr /* sharedBuffer */, AUDIO_SESSION_NONE, getpid(), uid, AUDIO_OUTPUT_FLAG_NONE, TYPE_OUTPUT), mActive(false), mSourceThread(sourceThread) { if (mCblk != NULL) { mOutBuffer.frameCount = 0; playbackThread->mTracks.add(this); ALOGV("%s(): mCblk %p, mBuffer %p, " "frameCount %zu, mChannelMask 0x%08x", __func__, mCblk, mBuffer, frameCount, mChannelMask); // since client and server are in the same process, // the buffer has the same virtual address on both sides mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, true /*clientInServer*/); mClientProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); mClientProxy->setSendLevel(0.0); mClientProxy->setSampleRate(sampleRate); } else { ALOGW("%s(%d): Error creating output track on thread %d", __func__, mId, (int)mThreadIoHandle); } } AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() { clearBufferQueue(); // superclass destructor will now delete the server proxy and shared memory both refer to } status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, audio_session_t 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; } ssize_t AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t frames) { Buffer *pInBuffer; Buffer inBuffer; bool outputBufferFull = false; inBuffer.frameCount = frames; inBuffer.raw = data; uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); if (!mActive && frames != 0) { (void) start(); } 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(); status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); if (status != NO_ERROR && status != NOT_ENOUGH_DATA) { ALOGV("%s(%d): thread %d no more output buffers; status %d", __func__, mId, (int)mThreadIoHandle, status); outputBufferFull = true; break; } uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); if (waitTimeLeftMs >= waitTimeMs) { waitTimeLeftMs -= waitTimeMs; } else { waitTimeLeftMs = 0; } if (status == NOT_ENOUGH_DATA) { restartIfDisabled(); continue; } } uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * mFrameSize); Proxy::Buffer buf; buf.mFrameCount = outFrames; buf.mRaw = NULL; mClientProxy->releaseBuffer(&buf); restartIfDisabled(); pInBuffer->frameCount -= outFrames; pInBuffer->raw = (int8_t *)pInBuffer->raw + outFrames * mFrameSize; mOutBuffer.frameCount -= outFrames; mOutBuffer.raw = (int8_t *)mOutBuffer.raw + outFrames * mFrameSize; if (pInBuffer->frameCount == 0) { if (mBufferQueue.size()) { mBufferQueue.removeAt(0); free(pInBuffer->mBuffer); if (pInBuffer != &inBuffer) { delete pInBuffer; } ALOGV("%s(%d): thread %d released overflow buffer %zu", __func__, mId, (int)mThreadIoHandle, mBufferQueue.size()); } else { break; } } } // If we could not write all frames, allocate a buffer and queue it for next time. if (inBuffer.frameCount) { sp thread = mThread.promote(); if (thread != 0 && !thread->standby()) { if (mBufferQueue.size() < kMaxOverFlowBuffers) { pInBuffer = new Buffer; pInBuffer->mBuffer = malloc(inBuffer.frameCount * mFrameSize); pInBuffer->frameCount = inBuffer.frameCount; pInBuffer->raw = pInBuffer->mBuffer; memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * mFrameSize); mBufferQueue.add(pInBuffer); ALOGV("%s(%d): thread %d adding overflow buffer %zu", __func__, mId, (int)mThreadIoHandle, mBufferQueue.size()); // audio data is consumed (stored locally); set frameCount to 0. inBuffer.frameCount = 0; } else { ALOGW("%s(%d): thread %d no more overflow buffers", __func__, mId, (int)mThreadIoHandle); // TODO: return error for this. } } } // Calling write() with a 0 length buffer means that no more data will be written: // We rely on stop() to set the appropriate flags to allow the remaining frames to play out. if (frames == 0 && mBufferQueue.size() == 0 && mActive) { stop(); } return frames - inBuffer.frameCount; // number of frames consumed. } void AudioFlinger::PlaybackThread::OutputTrack::copyMetadataTo(MetadataInserter& backInserter) const { std::lock_guard lock(mTrackMetadatasMutex); backInserter = std::copy(mTrackMetadatas.begin(), mTrackMetadatas.end(), backInserter); } void AudioFlinger::PlaybackThread::OutputTrack::setMetadatas(const SourceMetadatas& metadatas) { { std::lock_guard lock(mTrackMetadatasMutex); mTrackMetadatas = metadatas; } // No need to adjust metadata track volumes as OutputTrack volumes are always 0dBFS. setMetadataHasChanged(); } status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) { ClientProxy::Buffer buf; buf.mFrameCount = buffer->frameCount; struct timespec timeout; timeout.tv_sec = waitTimeMs / 1000; timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; status_t status = mClientProxy->obtainBuffer(&buf, &timeout); buffer->frameCount = buf.mFrameCount; buffer->raw = buf.mRaw; return status; } void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() { size_t size = mBufferQueue.size(); for (size_t i = 0; i < size; i++) { Buffer *pBuffer = mBufferQueue.itemAt(i); free(pBuffer->mBuffer); delete pBuffer; } mBufferQueue.clear(); } void AudioFlinger::PlaybackThread::OutputTrack::restartIfDisabled() { int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); if (mActive && (flags & CBLK_DISABLED)) { start(); } } // ---------------------------------------------------------------------------- #undef LOG_TAG #define LOG_TAG "AF::PatchTrack" AudioFlinger::PlaybackThread::PatchTrack::PatchTrack(PlaybackThread *playbackThread, audio_stream_type_t streamType, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format, size_t frameCount, void *buffer, size_t bufferSize, audio_output_flags_t flags, const Timeout& timeout, size_t frameCountToBeReady) : Track(playbackThread, NULL, streamType, audio_attributes_t{} /* currently unused for patch track */, sampleRate, format, channelMask, frameCount, buffer, bufferSize, nullptr /* sharedBuffer */, AUDIO_SESSION_NONE, getpid(), AID_AUDIOSERVER, flags, TYPE_PATCH, AUDIO_PORT_HANDLE_NONE, frameCountToBeReady), PatchTrackBase(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true), *playbackThread, timeout) { ALOGV("%s(%d): sampleRate %d mPeerTimeout %d.%03d sec", __func__, mId, sampleRate, (int)mPeerTimeout.tv_sec, (int)(mPeerTimeout.tv_nsec / 1000000)); } AudioFlinger::PlaybackThread::PatchTrack::~PatchTrack() { ALOGV("%s(%d)", __func__, mId); } size_t AudioFlinger::PlaybackThread::PatchTrack::framesReady() const { if (mPeerProxy && mPeerProxy->producesBufferOnDemand()) { return std::numeric_limits::max(); } else { return Track::framesReady(); } } status_t AudioFlinger::PlaybackThread::PatchTrack::start(AudioSystem::sync_event_t event, audio_session_t triggerSession) { status_t status = Track::start(event, triggerSession); if (status != NO_ERROR) { return status; } android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); return status; } // AudioBufferProvider interface status_t AudioFlinger::PlaybackThread::PatchTrack::getNextBuffer( AudioBufferProvider::Buffer* buffer) { ALOG_ASSERT(mPeerProxy != 0, "%s(%d): called without peer proxy", __func__, mId); Proxy::Buffer buf; buf.mFrameCount = buffer->frameCount; if (ATRACE_ENABLED()) { std::string traceName("PTnReq"); traceName += std::to_string(id()); ATRACE_INT(traceName.c_str(), buf.mFrameCount); } status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout); ALOGV_IF(status != NO_ERROR, "%s(%d): getNextBuffer status %d", __func__, mId, status); buffer->frameCount = buf.mFrameCount; if (ATRACE_ENABLED()) { std::string traceName("PTnObt"); traceName += std::to_string(id()); ATRACE_INT(traceName.c_str(), buf.mFrameCount); } if (buf.mFrameCount == 0) { return WOULD_BLOCK; } status = Track::getNextBuffer(buffer); return status; } void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(AudioBufferProvider::Buffer* buffer) { ALOG_ASSERT(mPeerProxy != 0, "%s(%d): called without peer proxy", __func__, mId); Proxy::Buffer buf; buf.mFrameCount = buffer->frameCount; buf.mRaw = buffer->raw; mPeerProxy->releaseBuffer(&buf); TrackBase::releaseBuffer(buffer); } status_t AudioFlinger::PlaybackThread::PatchTrack::obtainBuffer(Proxy::Buffer* buffer, const struct timespec *timeOut) { status_t status = NO_ERROR; static const int32_t kMaxTries = 5; int32_t tryCounter = kMaxTries; const size_t originalFrameCount = buffer->mFrameCount; do { if (status == NOT_ENOUGH_DATA) { restartIfDisabled(); buffer->mFrameCount = originalFrameCount; // cleared on error, must be restored. } status = mProxy->obtainBuffer(buffer, timeOut); } while ((status == NOT_ENOUGH_DATA) && (tryCounter-- > 0)); return status; } void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(Proxy::Buffer* buffer) { mProxy->releaseBuffer(buffer); restartIfDisabled(); } void AudioFlinger::PlaybackThread::PatchTrack::restartIfDisabled() { if (android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags) & CBLK_DISABLED) { ALOGW("%s(%d): disabled due to previous underrun, restarting", __func__, mId); start(); } } // ---------------------------------------------------------------------------- // Record // ---------------------------------------------------------------------------- // ---------------------------------------------------------------------------- // AppOp for audio recording // ------------------------------- #undef LOG_TAG #define LOG_TAG "AF::OpRecordAudioMonitor" // static sp AudioFlinger::RecordThread::OpRecordAudioMonitor::createIfNeeded( uid_t uid, const audio_attributes_t& attr, const String16& opPackageName) { if (isServiceUid(uid)) { ALOGV("not silencing record for service uid:%d pack:%s", uid, String8(opPackageName).string()); return nullptr; } // Capturing from FM TUNER output is not controlled by OP_RECORD_AUDIO // because it does not affect users privacy as does capturing from an actual microphone. if (attr.source == AUDIO_SOURCE_FM_TUNER) { ALOGV("not muting FM TUNER capture for uid %d", uid); return nullptr; } if (opPackageName.size() == 0) { Vector packages; // no package name, happens with SL ES clients // query package manager to find one PermissionController permissionController; permissionController.getPackagesForUid(uid, packages); if (packages.isEmpty()) { return nullptr; } else { ALOGV("using pack:%s for uid:%d", String8(packages[0]).string(), uid); return new OpRecordAudioMonitor(uid, packages[0]); } } return new OpRecordAudioMonitor(uid, opPackageName); } AudioFlinger::RecordThread::OpRecordAudioMonitor::OpRecordAudioMonitor( uid_t uid, const String16& opPackageName) : mHasOpRecordAudio(true), mUid(uid), mPackage(opPackageName) { } AudioFlinger::RecordThread::OpRecordAudioMonitor::~OpRecordAudioMonitor() { if (mOpCallback != 0) { mAppOpsManager.stopWatchingMode(mOpCallback); } mOpCallback.clear(); } void AudioFlinger::RecordThread::OpRecordAudioMonitor::onFirstRef() { checkRecordAudio(); mOpCallback = new RecordAudioOpCallback(this); ALOGV("start watching OP_RECORD_AUDIO for pack:%s", String8(mPackage).string()); mAppOpsManager.startWatchingMode(AppOpsManager::OP_RECORD_AUDIO, mPackage, mOpCallback); } bool AudioFlinger::RecordThread::OpRecordAudioMonitor::hasOpRecordAudio() const { return mHasOpRecordAudio.load(); } // Called by RecordAudioOpCallback when OP_RECORD_AUDIO is updated in AppOp callback // and in onFirstRef() // Note this method is never called (and never to be) for audio server / root track // due to the UID in createIfNeeded(). As a result for those record track, it's: // - not called from constructor, // - not called from RecordAudioOpCallback because the callback is not installed in this case void AudioFlinger::RecordThread::OpRecordAudioMonitor::checkRecordAudio() { const int32_t mode = mAppOpsManager.checkOp(AppOpsManager::OP_RECORD_AUDIO, mUid, mPackage); const bool hasIt = (mode == AppOpsManager::MODE_ALLOWED); // verbose logging only log when appOp changed ALOGI_IF(hasIt != mHasOpRecordAudio.load(), "OP_RECORD_AUDIO missing, %ssilencing record uid%d pack:%s", hasIt ? "un" : "", mUid, String8(mPackage).string()); mHasOpRecordAudio.store(hasIt); } AudioFlinger::RecordThread::OpRecordAudioMonitor::RecordAudioOpCallback::RecordAudioOpCallback( const wp& monitor) : mMonitor(monitor) { } void AudioFlinger::RecordThread::OpRecordAudioMonitor::RecordAudioOpCallback::opChanged(int32_t op, const String16& packageName) { UNUSED(packageName); if (op != AppOpsManager::OP_RECORD_AUDIO) { return; } sp monitor = mMonitor.promote(); if (monitor != NULL) { monitor->checkRecordAudio(); } } #undef LOG_TAG #define LOG_TAG "AF::RecordHandle" AudioFlinger::RecordHandle::RecordHandle( const sp& recordTrack) : BnAudioRecord(), mRecordTrack(recordTrack) { } AudioFlinger::RecordHandle::~RecordHandle() { stop_nonvirtual(); mRecordTrack->destroy(); } binder::Status AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, int /*audio_session_t*/ triggerSession) { ALOGV("%s()", __func__); return binder::Status::fromStatusT( mRecordTrack->start((AudioSystem::sync_event_t)event, (audio_session_t) triggerSession)); } binder::Status AudioFlinger::RecordHandle::stop() { stop_nonvirtual(); return binder::Status::ok(); } void AudioFlinger::RecordHandle::stop_nonvirtual() { ALOGV("%s()", __func__); mRecordTrack->stop(); } binder::Status AudioFlinger::RecordHandle::getActiveMicrophones( std::vector* activeMicrophones) { ALOGV("%s()", __func__); return binder::Status::fromStatusT( mRecordTrack->getActiveMicrophones(activeMicrophones)); } binder::Status AudioFlinger::RecordHandle::setPreferredMicrophoneDirection( int /*audio_microphone_direction_t*/ direction) { ALOGV("%s()", __func__); return binder::Status::fromStatusT(mRecordTrack->setPreferredMicrophoneDirection( static_cast(direction))); } binder::Status AudioFlinger::RecordHandle::setPreferredMicrophoneFieldDimension(float zoom) { ALOGV("%s()", __func__); return binder::Status::fromStatusT(mRecordTrack->setPreferredMicrophoneFieldDimension(zoom)); } // ---------------------------------------------------------------------------- #undef LOG_TAG #define LOG_TAG "AF::RecordTrack" // RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held AudioFlinger::RecordThread::RecordTrack::RecordTrack( RecordThread *thread, const sp& client, const audio_attributes_t& attr, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, size_t bufferSize, audio_session_t sessionId, pid_t creatorPid, uid_t uid, audio_input_flags_t flags, track_type type, const String16& opPackageName, audio_port_handle_t portId) : TrackBase(thread, client, attr, sampleRate, format, channelMask, frameCount, buffer, bufferSize, sessionId, creatorPid, uid, false /*isOut*/, (type == TYPE_DEFAULT) ? ((flags & AUDIO_INPUT_FLAG_FAST) ? ALLOC_PIPE : ALLOC_CBLK) : ((buffer == NULL) ? ALLOC_LOCAL : ALLOC_NONE), type, portId, std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_RECORD) + std::to_string(portId)), mOverflow(false), mFramesToDrop(0), mResamplerBufferProvider(NULL), // initialize in case of early constructor exit mRecordBufferConverter(NULL), mFlags(flags), mSilenced(false), mOpRecordAudioMonitor(OpRecordAudioMonitor::createIfNeeded(uid, attr, opPackageName)) { if (mCblk == NULL) { return; } if (!isDirect()) { mRecordBufferConverter = new RecordBufferConverter( thread->mChannelMask, thread->mFormat, thread->mSampleRate, channelMask, format, sampleRate); // Check if the RecordBufferConverter construction was successful. // If not, don't continue with construction. // // NOTE: It would be extremely rare that the record track cannot be created // for the current device, but a pending or future device change would make // the record track configuration valid. if (mRecordBufferConverter->initCheck() != NO_ERROR) { ALOGE("%s(%d): RecordTrack unable to create record buffer converter", __func__, mId); return; } } mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize, !isExternalTrack()); mResamplerBufferProvider = new ResamplerBufferProvider(this); if (flags & AUDIO_INPUT_FLAG_FAST) { ALOG_ASSERT(thread->mFastTrackAvail); thread->mFastTrackAvail = false; } else { // TODO: only Normal Record has timestamps (Fast Record does not). mServerLatencySupported = checkServerLatencySupported(mFormat, flags); } #ifdef TEE_SINK mTee.setId(std::string("_") + std::to_string(mThreadIoHandle) + "_" + std::to_string(mId) + "_R"); #endif // Once this item is logged by the server, the client can add properties. mTrackMetrics.logConstructor(creatorPid, uid); } AudioFlinger::RecordThread::RecordTrack::~RecordTrack() { ALOGV("%s()", __func__); delete mRecordBufferConverter; delete mResamplerBufferProvider; } status_t AudioFlinger::RecordThread::RecordTrack::initCheck() const { status_t status = TrackBase::initCheck(); if (status == NO_ERROR && mServerProxy == 0) { status = BAD_VALUE; } return status; } // AudioBufferProvider interface status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) { ServerProxy::Buffer buf; buf.mFrameCount = buffer->frameCount; status_t status = mServerProxy->obtainBuffer(&buf); buffer->frameCount = buf.mFrameCount; buffer->raw = buf.mRaw; if (buf.mFrameCount == 0) { // FIXME also wake futex so that overrun is noticed more quickly (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); } return status; } status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, audio_session_t triggerSession) { sp 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 thread = mThread.promote(); if (thread != 0) { RecordThread *recordThread = (RecordThread *)thread.get(); if (recordThread->stop(this) && isExternalTrack()) { AudioSystem::stopInput(mPortId); } } } void AudioFlinger::RecordThread::RecordTrack::destroy() { // see comments at AudioFlinger::PlaybackThread::Track::destroy() sp keep(this); { track_state priorState = mState; sp thread = mThread.promote(); if (thread != 0) { Mutex::Autolock _l(thread->mLock); RecordThread *recordThread = (RecordThread *) thread.get(); priorState = mState; recordThread->destroyTrack_l(this); // move mState to STOPPED, terminate } // APM portid/client management done outside of lock. // NOTE: if thread doesn't exist, the input descriptor probably doesn't either. if (isExternalTrack()) { switch (priorState) { case ACTIVE: // invalidated while still active case STARTING_2: // invalidated/start-aborted after startInput successfully called case PAUSING: // invalidated while in the middle of stop() pausing (still active) AudioSystem::stopInput(mPortId); break; case STARTING_1: // invalidated/start-aborted and startInput not successful case PAUSED: // OK, not active case IDLE: // OK, not active break; case STOPPED: // unexpected (destroyed) default: LOG_ALWAYS_FATAL("%s(%d): invalid prior state: %d", __func__, mId, priorState); } AudioSystem::releaseInput(mPortId); } } } void AudioFlinger::RecordThread::RecordTrack::invalidate() { TrackBase::invalidate(); // FIXME should use proxy, and needs work audio_track_cblk_t* cblk = mCblk; android_atomic_or(CBLK_INVALID, &cblk->mFlags); android_atomic_release_store(0x40000000, &cblk->mFutex); // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX); } void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) { result.appendFormat("Active Id Client Session Port Id S Flags " " Format Chn mask SRate Source " " Server FrmCnt FrmRdy Sil%s\n", isServerLatencySupported() ? " Latency" : ""); } void AudioFlinger::RecordThread::RecordTrack::appendDump(String8& result, bool active) { result.appendFormat("%c%5s %6d %6u %7u %7u %2s 0x%03X " "%08X %08X %6u %6X " "%08X %6zu %6zu %3c", isFastTrack() ? 'F' : ' ', active ? "yes" : "no", mId, (mClient == 0) ? getpid() : mClient->pid(), mSessionId, mPortId, getTrackStateAsCodedString(), mCblk->mFlags, mFormat, mChannelMask, mSampleRate, mAttr.source, mCblk->mServer, mFrameCount, mServerProxy->framesReadySafe(), isSilenced() ? 's' : 'n' ); if (isServerLatencySupported()) { double latencyMs; bool fromTrack; if (getTrackLatencyMs(&latencyMs, &fromTrack) == OK) { // Show latency in msec, followed by 't' if from track timestamp (the most accurate) // or 'k' if estimated from kernel (usually for debugging). result.appendFormat(" %7.2lf %c", latencyMs, fromTrack ? 't' : 'k'); } else { result.appendFormat("%10s", mCblk->mServer != 0 ? "unavail" : "new"); } } result.append("\n"); } void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp& event) { if (event == mSyncStartEvent) { ssize_t framesToDrop = 0; sp threadBase = mThread.promote(); if (threadBase != 0) { // TODO: use actual buffer filling status instead of 2 buffers when info is available // from audio HAL framesToDrop = threadBase->mFrameCount * 2; } mFramesToDrop = framesToDrop; } } void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent() { if (mSyncStartEvent != 0) { mSyncStartEvent->cancel(); mSyncStartEvent.clear(); } mFramesToDrop = 0; } void AudioFlinger::RecordThread::RecordTrack::updateTrackFrameInfo( int64_t trackFramesReleased, int64_t sourceFramesRead, uint32_t halSampleRate, const ExtendedTimestamp ×tamp) { // Make the kernel frametime available. const FrameTime ft{ timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL], timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]}; // ALOGD("FrameTime: %lld %lld", (long long)ft.frames, (long long)ft.timeNs); mKernelFrameTime.store(ft); if (!audio_is_linear_pcm(mFormat)) { return; } ExtendedTimestamp local = timestamp; // Convert HAL frames to server-side track frames at track sample rate. // We use trackFramesReleased and sourceFramesRead as an anchor point. for (int i = ExtendedTimestamp::LOCATION_SERVER; i < ExtendedTimestamp::LOCATION_MAX; ++i) { if (local.mTimeNs[i] != 0) { const int64_t relativeServerFrames = local.mPosition[i] - sourceFramesRead; const int64_t relativeTrackFrames = relativeServerFrames * mSampleRate / halSampleRate; // TODO: potential computation overflow local.mPosition[i] = relativeTrackFrames + trackFramesReleased; } } mServerProxy->setTimestamp(local); // Compute latency info. const bool useTrackTimestamp = true; // use track unless debugging. const double latencyMs = - (useTrackTimestamp ? local.getOutputServerLatencyMs(sampleRate()) : timestamp.getOutputServerLatencyMs(halSampleRate)); mServerLatencyFromTrack.store(useTrackTimestamp); mServerLatencyMs.store(latencyMs); } bool AudioFlinger::RecordThread::RecordTrack::isSilenced() const { if (mSilenced) { return true; } // The monitor is only created for record tracks that can be silenced. return mOpRecordAudioMonitor ? !mOpRecordAudioMonitor->hasOpRecordAudio() : false; } status_t AudioFlinger::RecordThread::RecordTrack::getActiveMicrophones( std::vector* activeMicrophones) { sp thread = mThread.promote(); if (thread != 0) { RecordThread *recordThread = (RecordThread *)thread.get(); return recordThread->getActiveMicrophones(activeMicrophones); } else { return BAD_VALUE; } } status_t AudioFlinger::RecordThread::RecordTrack::setPreferredMicrophoneDirection( audio_microphone_direction_t direction) { sp thread = mThread.promote(); if (thread != 0) { RecordThread *recordThread = (RecordThread *)thread.get(); return recordThread->setPreferredMicrophoneDirection(direction); } else { return BAD_VALUE; } } status_t AudioFlinger::RecordThread::RecordTrack::setPreferredMicrophoneFieldDimension(float zoom) { sp thread = mThread.promote(); if (thread != 0) { RecordThread *recordThread = (RecordThread *)thread.get(); return recordThread->setPreferredMicrophoneFieldDimension(zoom); } else { return BAD_VALUE; } } // ---------------------------------------------------------------------------- #undef LOG_TAG #define LOG_TAG "AF::PatchRecord" AudioFlinger::RecordThread::PatchRecord::PatchRecord(RecordThread *recordThread, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format, size_t frameCount, void *buffer, size_t bufferSize, audio_input_flags_t flags, const Timeout& timeout) : RecordTrack(recordThread, NULL, audio_attributes_t{} /* currently unused for patch track */, sampleRate, format, channelMask, frameCount, buffer, bufferSize, AUDIO_SESSION_NONE, getpid(), AID_AUDIOSERVER, flags, TYPE_PATCH, String16()), PatchTrackBase(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true), *recordThread, timeout) { ALOGV("%s(%d): sampleRate %d mPeerTimeout %d.%03d sec", __func__, mId, sampleRate, (int)mPeerTimeout.tv_sec, (int)(mPeerTimeout.tv_nsec / 1000000)); } AudioFlinger::RecordThread::PatchRecord::~PatchRecord() { ALOGV("%s(%d)", __func__, mId); } static size_t writeFramesHelper( AudioBufferProvider* dest, const void* src, size_t frameCount, size_t frameSize) { AudioBufferProvider::Buffer patchBuffer; patchBuffer.frameCount = frameCount; auto status = dest->getNextBuffer(&patchBuffer); if (status != NO_ERROR) { ALOGW("%s PathRecord getNextBuffer failed with error %d: %s", __func__, status, strerror(-status)); return 0; } ALOG_ASSERT(patchBuffer.frameCount <= frameCount); memcpy(patchBuffer.raw, src, patchBuffer.frameCount * frameSize); size_t framesWritten = patchBuffer.frameCount; dest->releaseBuffer(&patchBuffer); return framesWritten; } // static size_t AudioFlinger::RecordThread::PatchRecord::writeFrames( AudioBufferProvider* dest, const void* src, size_t frameCount, size_t frameSize) { size_t framesWritten = writeFramesHelper(dest, src, frameCount, frameSize); // On buffer wrap, the buffer frame count will be less than requested, // when this happens a second buffer needs to be used to write the leftover audio const size_t framesLeft = frameCount - framesWritten; if (framesWritten != 0 && framesLeft != 0) { framesWritten += writeFramesHelper(dest, (const char*)src + framesWritten * frameSize, framesLeft, frameSize); } return framesWritten; } // AudioBufferProvider interface status_t AudioFlinger::RecordThread::PatchRecord::getNextBuffer( AudioBufferProvider::Buffer* buffer) { ALOG_ASSERT(mPeerProxy != 0, "%s(%d): called without peer proxy", __func__, mId); Proxy::Buffer buf; buf.mFrameCount = buffer->frameCount; status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout); ALOGV_IF(status != NO_ERROR, "%s(%d): mPeerProxy->obtainBuffer status %d", __func__, mId, status); buffer->frameCount = buf.mFrameCount; if (ATRACE_ENABLED()) { std::string traceName("PRnObt"); traceName += std::to_string(id()); ATRACE_INT(traceName.c_str(), buf.mFrameCount); } if (buf.mFrameCount == 0) { return WOULD_BLOCK; } status = RecordTrack::getNextBuffer(buffer); return status; } void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(AudioBufferProvider::Buffer* buffer) { ALOG_ASSERT(mPeerProxy != 0, "%s(%d): called without peer proxy", __func__, mId); Proxy::Buffer buf; buf.mFrameCount = buffer->frameCount; buf.mRaw = buffer->raw; mPeerProxy->releaseBuffer(&buf); TrackBase::releaseBuffer(buffer); } status_t AudioFlinger::RecordThread::PatchRecord::obtainBuffer(Proxy::Buffer* buffer, const struct timespec *timeOut) { return mProxy->obtainBuffer(buffer, timeOut); } void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(Proxy::Buffer* buffer) { mProxy->releaseBuffer(buffer); } #undef LOG_TAG #define LOG_TAG "AF::PthrPatchRecord" static std::unique_ptr allocAligned(size_t alignment, size_t size) { void *ptr = nullptr; (void)posix_memalign(&ptr, alignment, size); return std::unique_ptr(ptr, free); } AudioFlinger::RecordThread::PassthruPatchRecord::PassthruPatchRecord( RecordThread *recordThread, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format, size_t frameCount, audio_input_flags_t flags) : PatchRecord(recordThread, sampleRate, channelMask, format, frameCount, nullptr /*buffer*/, 0 /*bufferSize*/, flags), mPatchRecordAudioBufferProvider(*this), mSinkBuffer(allocAligned(32, mFrameCount * mFrameSize)), mStubBuffer(allocAligned(32, mFrameCount * mFrameSize)) { memset(mStubBuffer.get(), 0, mFrameCount * mFrameSize); } sp AudioFlinger::RecordThread::PassthruPatchRecord::obtainStream( sp* thread) { *thread = mThread.promote(); if (!*thread) return nullptr; RecordThread *recordThread = static_cast((*thread).get()); Mutex::Autolock _l(recordThread->mLock); return recordThread->mInput ? recordThread->mInput->stream : nullptr; } // PatchProxyBufferProvider methods are called on DirectOutputThread status_t AudioFlinger::RecordThread::PassthruPatchRecord::obtainBuffer( Proxy::Buffer* buffer, const struct timespec* timeOut) { if (mUnconsumedFrames) { buffer->mFrameCount = std::min(buffer->mFrameCount, mUnconsumedFrames); // mUnconsumedFrames is decreased in releaseBuffer to use actual frame consumption figure. return PatchRecord::obtainBuffer(buffer, timeOut); } // Otherwise, execute a read from HAL and write into the buffer. nsecs_t startTimeNs = 0; if (timeOut && (timeOut->tv_sec != 0 || timeOut->tv_nsec != 0) && timeOut->tv_sec != INT_MAX) { // Will need to correct timeOut by elapsed time. startTimeNs = systemTime(); } const size_t framesToRead = std::min(buffer->mFrameCount, mFrameCount); buffer->mFrameCount = 0; buffer->mRaw = nullptr; sp thread; sp stream = obtainStream(&thread); if (!stream) return NO_INIT; // If there is no stream, RecordThread is not reading. status_t result = NO_ERROR; size_t bytesRead = 0; { ATRACE_NAME("read"); result = stream->read(mSinkBuffer.get(), framesToRead * mFrameSize, &bytesRead); if (result != NO_ERROR) goto stream_error; if (bytesRead == 0) return NO_ERROR; } { std::lock_guard lock(mReadLock); mReadBytes += bytesRead; mReadError = NO_ERROR; } mReadCV.notify_one(); // writeFrames handles wraparound and should write all the provided frames. // If it couldn't, there is something wrong with the client/server buffer of the software patch. buffer->mFrameCount = writeFrames( &mPatchRecordAudioBufferProvider, mSinkBuffer.get(), bytesRead / mFrameSize, mFrameSize); ALOGW_IF(buffer->mFrameCount < bytesRead / mFrameSize, "Lost %zu frames obtained from HAL", bytesRead / mFrameSize - buffer->mFrameCount); mUnconsumedFrames = buffer->mFrameCount; struct timespec newTimeOut; if (startTimeNs) { // Correct the timeout by elapsed time. nsecs_t newTimeOutNs = audio_utils_ns_from_timespec(timeOut) - (systemTime() - startTimeNs); if (newTimeOutNs < 0) newTimeOutNs = 0; newTimeOut.tv_sec = newTimeOutNs / NANOS_PER_SECOND; newTimeOut.tv_nsec = newTimeOutNs - newTimeOut.tv_sec * NANOS_PER_SECOND; timeOut = &newTimeOut; } return PatchRecord::obtainBuffer(buffer, timeOut); stream_error: stream->standby(); { std::lock_guard lock(mReadLock); mReadError = result; } mReadCV.notify_one(); return result; } void AudioFlinger::RecordThread::PassthruPatchRecord::releaseBuffer(Proxy::Buffer* buffer) { if (buffer->mFrameCount <= mUnconsumedFrames) { mUnconsumedFrames -= buffer->mFrameCount; } else { ALOGW("Write side has consumed more frames than we had: %zu > %zu", buffer->mFrameCount, mUnconsumedFrames); mUnconsumedFrames = 0; } PatchRecord::releaseBuffer(buffer); } // AudioBufferProvider and Source methods are called on RecordThread // 'read' emulates actual audio data with 0's. This is OK as 'getNextBuffer' // and 'releaseBuffer' are stubbed out and ignore their input. // It's not possible to retrieve actual data here w/o blocking 'obtainBuffer' // until we copy it. status_t AudioFlinger::RecordThread::PassthruPatchRecord::read( void* buffer, size_t bytes, size_t* read) { bytes = std::min(bytes, mFrameCount * mFrameSize); { std::unique_lock lock(mReadLock); mReadCV.wait(lock, [&]{ return mReadError != NO_ERROR || mReadBytes != 0; }); if (mReadError != NO_ERROR) { mLastReadFrames = 0; return mReadError; } *read = std::min(bytes, mReadBytes); mReadBytes -= *read; } mLastReadFrames = *read / mFrameSize; memset(buffer, 0, *read); return 0; } status_t AudioFlinger::RecordThread::PassthruPatchRecord::getCapturePosition( int64_t* frames, int64_t* time) { sp thread; sp stream = obtainStream(&thread); return stream ? stream->getCapturePosition(frames, time) : NO_INIT; } status_t AudioFlinger::RecordThread::PassthruPatchRecord::standby() { // RecordThread issues 'standby' command in two major cases: // 1. Error on read--this case is handled in 'obtainBuffer'. // 2. Track is stopping--as PassthruPatchRecord assumes continuous // output, this can only happen when the software patch // is being torn down. In this case, the RecordThread // will terminate and close the HAL stream. return 0; } // As the buffer gets filled in obtainBuffer, here we only simulate data consumption. status_t AudioFlinger::RecordThread::PassthruPatchRecord::getNextBuffer( AudioBufferProvider::Buffer* buffer) { buffer->frameCount = mLastReadFrames; buffer->raw = buffer->frameCount != 0 ? mStubBuffer.get() : nullptr; return NO_ERROR; } void AudioFlinger::RecordThread::PassthruPatchRecord::releaseBuffer( AudioBufferProvider::Buffer* buffer) { buffer->frameCount = 0; buffer->raw = nullptr; } // ---------------------------------------------------------------------------- #undef LOG_TAG #define LOG_TAG "AF::MmapTrack" AudioFlinger::MmapThread::MmapTrack::MmapTrack(ThreadBase *thread, const audio_attributes_t& attr, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, audio_session_t sessionId, bool isOut, uid_t uid, pid_t pid, pid_t creatorPid, audio_port_handle_t portId) : TrackBase(thread, NULL, attr, sampleRate, format, channelMask, (size_t)0 /* frameCount */, nullptr /* buffer */, (size_t)0 /* bufferSize */, sessionId, creatorPid, uid, isOut, ALLOC_NONE, TYPE_DEFAULT, portId, std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_MMAP) + std::to_string(portId)), mPid(pid), mSilenced(false), mSilencedNotified(false) { // Once this item is logged by the server, the client can add properties. mTrackMetrics.logConstructor(creatorPid, uid); } AudioFlinger::MmapThread::MmapTrack::~MmapTrack() { } status_t AudioFlinger::MmapThread::MmapTrack::initCheck() const { return NO_ERROR; } status_t AudioFlinger::MmapThread::MmapTrack::start(AudioSystem::sync_event_t event __unused, audio_session_t triggerSession __unused) { return NO_ERROR; } void AudioFlinger::MmapThread::MmapTrack::stop() { } // AudioBufferProvider interface status_t AudioFlinger::MmapThread::MmapTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) { buffer->frameCount = 0; buffer->raw = nullptr; return INVALID_OPERATION; } // ExtendedAudioBufferProvider interface size_t AudioFlinger::MmapThread::MmapTrack::framesReady() const { return 0; } int64_t AudioFlinger::MmapThread::MmapTrack::framesReleased() const { return 0; } void AudioFlinger::MmapThread::MmapTrack::onTimestamp(const ExtendedTimestamp ×tamp __unused) { } void AudioFlinger::MmapThread::MmapTrack::appendDumpHeader(String8& result) { result.appendFormat("Client Session Port Id Format Chn mask SRate Flags %s\n", isOut() ? "Usg CT": "Source"); } void AudioFlinger::MmapThread::MmapTrack::appendDump(String8& result, bool active __unused) { result.appendFormat("%6u %7u %7u %08X %08X %6u 0x%03X ", mPid, mSessionId, mPortId, mFormat, mChannelMask, mSampleRate, mAttr.flags); if (isOut()) { result.appendFormat("%3x %2x", mAttr.usage, mAttr.content_type); } else { result.appendFormat("%6x", mAttr.source); } result.append("\n"); } } // namespace android