/* * Copyright (C) 2010 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_NDEBUG 0 #define LOG_TAG "NuPlayerRenderer" #include #include "NuPlayerRenderer.h" #include #include #include #include #include #include #include #include #include #include #include #include #include #include "mediaplayerservice/AVNuExtensions.h" #include "stagefright/AVExtensions.h" namespace android { /* * Example of common configuration settings in shell script form #Turn offload audio off (use PCM for Play Music) -- AudioPolicyManager adb shell setprop audio.offload.disable 1 #Allow offload audio with video (requires offloading to be enabled) -- AudioPolicyManager adb shell setprop audio.offload.video 1 #Use audio callbacks for PCM data adb shell setprop media.stagefright.audio.cbk 1 #Use deep buffer for PCM data with video (it is generally enabled for audio-only) adb shell setprop media.stagefright.audio.deep 1 #Set size of buffers for pcm audio sink in msec (example: 1000 msec) adb shell setprop media.stagefright.audio.sink 1000 * These configurations take effect for the next track played (not the current track). */ static inline bool getUseAudioCallbackSetting() { return property_get_bool("media.stagefright.audio.cbk", false /* default_value */); } static inline int32_t getAudioSinkPcmMsSetting() { return property_get_int32( "media.stagefright.audio.sink", 500 /* default_value */); } // Maximum time in paused state when offloading audio decompression. When elapsed, the AudioSink // is closed to allow the audio DSP to power down. static const int64_t kOffloadPauseMaxUs = 10000000ll; // static const NuPlayer::Renderer::PcmInfo NuPlayer::Renderer::AUDIO_PCMINFO_INITIALIZER = { AUDIO_CHANNEL_NONE, AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID, 0, // mNumChannels 0 // mSampleRate }; // static const int64_t NuPlayer::Renderer::kMinPositionUpdateDelayUs = 100000ll; static bool sFrameAccurateAVsync = false; static void readProperties() { char value[PROPERTY_VALUE_MAX]; if (property_get("persist.sys.media.avsync", value, NULL)) { sFrameAccurateAVsync = !strcmp("1", value) || !strcasecmp("true", value); } } NuPlayer::Renderer::Renderer( const sp &sink, const sp ¬ify, uint32_t flags) : mAudioSink(sink), mNotify(notify), mFlags(flags), mNumFramesWritten(0), mDrainAudioQueuePending(false), mDrainVideoQueuePending(false), mAudioQueueGeneration(0), mVideoQueueGeneration(0), mAudioDrainGeneration(0), mVideoDrainGeneration(0), mPlaybackSettings(AUDIO_PLAYBACK_RATE_DEFAULT), mAudioFirstAnchorTimeMediaUs(-1), mAnchorTimeMediaUs(-1), mAnchorNumFramesWritten(-1), mVideoLateByUs(0ll), mHasAudio(false), mHasVideo(false), mFoundAudioEOS(false), mNotifyCompleteAudio(false), mNotifyCompleteVideo(false), mSyncQueues(false), mPaused(false), mPauseDrainAudioAllowedUs(0), mVideoSampleReceived(false), mVideoRenderingStarted(false), mVideoRenderingStartGeneration(0), mAudioRenderingStartGeneration(0), mRenderingDataDelivered(false), mAudioOffloadPauseTimeoutGeneration(0), mAudioTearingDown(false), mCurrentOffloadInfo(AUDIO_INFO_INITIALIZER), mCurrentPcmInfo(AUDIO_PCMINFO_INITIALIZER), mTotalBuffersQueued(0), mLastAudioBufferDrained(0), mUseAudioCallback(false), mWakeLock(new AWakeLock()) { mMediaClock = new MediaClock; mPlaybackRate = mPlaybackSettings.mSpeed; mMediaClock->setPlaybackRate(mPlaybackRate); readProperties(); } NuPlayer::Renderer::~Renderer() { if (offloadingAudio()) { mAudioSink->stop(); mAudioSink->flush(); mAudioSink->close(); } } void NuPlayer::Renderer::queueBuffer( bool audio, const sp &buffer, const sp ¬ifyConsumed) { sp msg = new AMessage(kWhatQueueBuffer, this); msg->setInt32("queueGeneration", getQueueGeneration(audio)); msg->setInt32("audio", static_cast(audio)); msg->setBuffer("buffer", buffer); msg->setMessage("notifyConsumed", notifyConsumed); msg->post(); } void NuPlayer::Renderer::queueEOS(bool audio, status_t finalResult) { CHECK_NE(finalResult, (status_t)OK); sp msg = new AMessage(kWhatQueueEOS, this); msg->setInt32("queueGeneration", getQueueGeneration(audio)); msg->setInt32("audio", static_cast(audio)); msg->setInt32("finalResult", finalResult); msg->post(); } status_t NuPlayer::Renderer::setPlaybackSettings(const AudioPlaybackRate &rate) { sp msg = new AMessage(kWhatConfigPlayback, this); writeToAMessage(msg, rate); sp response; status_t err = msg->postAndAwaitResponse(&response); if (err == OK && response != NULL) { CHECK(response->findInt32("err", &err)); } return err; } status_t NuPlayer::Renderer::onConfigPlayback(const AudioPlaybackRate &rate /* sanitized */) { if (rate.mSpeed == 0.f) { onPause(); // don't call audiosink's setPlaybackRate if pausing, as pitch does not // have to correspond to the any non-0 speed (e.g old speed). Keep // settings nonetheless, using the old speed, in case audiosink changes. AudioPlaybackRate newRate = rate; newRate.mSpeed = mPlaybackSettings.mSpeed; mPlaybackSettings = newRate; return OK; } if (mAudioSink != NULL && mAudioSink->ready()) { status_t err = mAudioSink->setPlaybackRate(rate); if (err != OK) { return err; } } mPlaybackSettings = rate; mPlaybackRate = rate.mSpeed; mMediaClock->setPlaybackRate(mPlaybackRate); return OK; } status_t NuPlayer::Renderer::getPlaybackSettings(AudioPlaybackRate *rate /* nonnull */) { sp msg = new AMessage(kWhatGetPlaybackSettings, this); sp response; status_t err = msg->postAndAwaitResponse(&response); if (err == OK && response != NULL) { CHECK(response->findInt32("err", &err)); if (err == OK) { readFromAMessage(response, rate); } } return err; } status_t NuPlayer::Renderer::onGetPlaybackSettings(AudioPlaybackRate *rate /* nonnull */) { if (mAudioSink != NULL && mAudioSink->ready()) { status_t err = mAudioSink->getPlaybackRate(rate); if (err == OK) { if (!isAudioPlaybackRateEqual(*rate, mPlaybackSettings)) { ALOGW("correcting mismatch in internal/external playback rate"); } // get playback settings used by audiosink, as it may be // slightly off due to audiosink not taking small changes. mPlaybackSettings = *rate; if (mPaused) { rate->mSpeed = 0.f; } } return err; } *rate = mPlaybackSettings; return OK; } status_t NuPlayer::Renderer::setSyncSettings(const AVSyncSettings &sync, float videoFpsHint) { sp msg = new AMessage(kWhatConfigSync, this); writeToAMessage(msg, sync, videoFpsHint); sp response; status_t err = msg->postAndAwaitResponse(&response); if (err == OK && response != NULL) { CHECK(response->findInt32("err", &err)); } return err; } status_t NuPlayer::Renderer::onConfigSync(const AVSyncSettings &sync, float videoFpsHint __unused) { if (sync.mSource != AVSYNC_SOURCE_DEFAULT) { return BAD_VALUE; } // TODO: support sync sources return INVALID_OPERATION; } status_t NuPlayer::Renderer::getSyncSettings(AVSyncSettings *sync, float *videoFps) { sp msg = new AMessage(kWhatGetSyncSettings, this); sp response; status_t err = msg->postAndAwaitResponse(&response); if (err == OK && response != NULL) { CHECK(response->findInt32("err", &err)); if (err == OK) { readFromAMessage(response, sync, videoFps); } } return err; } status_t NuPlayer::Renderer::onGetSyncSettings( AVSyncSettings *sync /* nonnull */, float *videoFps /* nonnull */) { *sync = mSyncSettings; *videoFps = -1.f; return OK; } void NuPlayer::Renderer::flush(bool audio, bool notifyComplete) { { Mutex::Autolock autoLock(mLock); if (audio) { mNotifyCompleteAudio |= notifyComplete; clearAudioFirstAnchorTime_l(); ++mAudioQueueGeneration; ++mAudioDrainGeneration; } else { mNotifyCompleteVideo |= notifyComplete; ++mVideoQueueGeneration; ++mVideoDrainGeneration; } clearAnchorTime_l(); mVideoLateByUs = 0; mSyncQueues = false; } sp msg = new AMessage(kWhatFlush, this); msg->setInt32("audio", static_cast(audio)); msg->post(); } void NuPlayer::Renderer::signalTimeDiscontinuity() { } void NuPlayer::Renderer::signalDisableOffloadAudio() { (new AMessage(kWhatDisableOffloadAudio, this))->post(); } void NuPlayer::Renderer::signalEnableOffloadAudio() { (new AMessage(kWhatEnableOffloadAudio, this))->post(); } void NuPlayer::Renderer::pause() { (new AMessage(kWhatPause, this))->post(); } void NuPlayer::Renderer::resume() { (new AMessage(kWhatResume, this))->post(); } void NuPlayer::Renderer::setVideoFrameRate(float fps) { sp msg = new AMessage(kWhatSetVideoFrameRate, this); msg->setFloat("frame-rate", fps); msg->post(); } // Called on any threads. status_t NuPlayer::Renderer::getCurrentPosition(int64_t *mediaUs) { return mMediaClock->getMediaTime( ALooper::GetNowUs(), mediaUs, (mHasAudio && mFoundAudioEOS)); } void NuPlayer::Renderer::clearAudioFirstAnchorTime_l() { mAudioFirstAnchorTimeMediaUs = -1; mMediaClock->setStartingTimeMedia(-1); } void NuPlayer::Renderer::setAudioFirstAnchorTimeIfNeeded_l(int64_t mediaUs) { if (mAudioFirstAnchorTimeMediaUs == -1) { mAudioFirstAnchorTimeMediaUs = mediaUs; mMediaClock->setStartingTimeMedia(mediaUs); } } void NuPlayer::Renderer::clearAnchorTime_l() { mMediaClock->clearAnchor(); mAnchorTimeMediaUs = -1; mAnchorNumFramesWritten = -1; } void NuPlayer::Renderer::setVideoLateByUs(int64_t lateUs) { Mutex::Autolock autoLock(mLock); mVideoLateByUs = lateUs; } int64_t NuPlayer::Renderer::getVideoLateByUs() { Mutex::Autolock autoLock(mLock); return mVideoLateByUs; } status_t NuPlayer::Renderer::openAudioSink( const sp &format, bool offloadOnly, bool hasVideo, uint32_t flags, bool *isOffloaded, bool isStreaming) { sp msg = new AMessage(kWhatOpenAudioSink, this); msg->setMessage("format", format); msg->setInt32("offload-only", offloadOnly); msg->setInt32("has-video", hasVideo); msg->setInt32("flags", flags); msg->setInt32("isStreaming", isStreaming); sp response; status_t postStatus = msg->postAndAwaitResponse(&response); int32_t err; if (postStatus != OK || !response->findInt32("err", &err)) { err = INVALID_OPERATION; } else if (err == OK && isOffloaded != NULL) { int32_t offload; CHECK(response->findInt32("offload", &offload)); *isOffloaded = (offload != 0); } return err; } void NuPlayer::Renderer::closeAudioSink() { sp msg = new AMessage(kWhatCloseAudioSink, this); sp response; msg->postAndAwaitResponse(&response); } void NuPlayer::Renderer::onMessageReceived(const sp &msg) { switch (msg->what()) { case kWhatOpenAudioSink: { sp format; CHECK(msg->findMessage("format", &format)); int32_t offloadOnly; CHECK(msg->findInt32("offload-only", &offloadOnly)); int32_t hasVideo; CHECK(msg->findInt32("has-video", &hasVideo)); uint32_t flags; CHECK(msg->findInt32("flags", (int32_t *)&flags)); uint32_t isStreaming; CHECK(msg->findInt32("isStreaming", (int32_t *)&isStreaming)); status_t err = onOpenAudioSink(format, offloadOnly, hasVideo, flags, isStreaming); sp response = new AMessage; response->setInt32("err", err); response->setInt32("offload", offloadingAudio()); sp replyID; CHECK(msg->senderAwaitsResponse(&replyID)); response->postReply(replyID); break; } case kWhatCloseAudioSink: { sp replyID; CHECK(msg->senderAwaitsResponse(&replyID)); onCloseAudioSink(); sp response = new AMessage; response->postReply(replyID); break; } case kWhatStopAudioSink: { mAudioSink->stop(); break; } case kWhatDrainAudioQueue: { mDrainAudioQueuePending = false; int32_t generation; CHECK(msg->findInt32("drainGeneration", &generation)); if (generation != getDrainGeneration(true /* audio */)) { break; } if (onDrainAudioQueue()) { uint32_t numFramesPlayed; if (mAudioSink->getPosition(&numFramesPlayed) != OK) { ALOGW("mAudioSink->getPosition failed"); break; } uint32_t numFramesPendingPlayout = mNumFramesWritten - numFramesPlayed; // This is how long the audio sink will have data to // play back. int64_t delayUs = mAudioSink->msecsPerFrame() * numFramesPendingPlayout * 1000ll; if (mPlaybackRate > 1.0f) { delayUs /= mPlaybackRate; } // Let's give it more data after about half that time // has elapsed. Mutex::Autolock autoLock(mLock); postDrainAudioQueue_l(delayUs / 2); } break; } case kWhatDrainVideoQueue: { int32_t generation; CHECK(msg->findInt32("drainGeneration", &generation)); if (generation != getDrainGeneration(false /* audio */)) { break; } mDrainVideoQueuePending = false; onDrainVideoQueue(); postDrainVideoQueue(); break; } case kWhatPostDrainVideoQueue: { int32_t generation; CHECK(msg->findInt32("drainGeneration", &generation)); if (generation != getDrainGeneration(false /* audio */)) { break; } mDrainVideoQueuePending = false; postDrainVideoQueue(); break; } case kWhatQueueBuffer: { onQueueBuffer(msg); break; } case kWhatQueueEOS: { onQueueEOS(msg); break; } case kWhatConfigPlayback: { sp replyID; CHECK(msg->senderAwaitsResponse(&replyID)); AudioPlaybackRate rate; readFromAMessage(msg, &rate); status_t err = onConfigPlayback(rate); sp response = new AMessage; response->setInt32("err", err); response->postReply(replyID); break; } case kWhatGetPlaybackSettings: { sp replyID; CHECK(msg->senderAwaitsResponse(&replyID)); AudioPlaybackRate rate = AUDIO_PLAYBACK_RATE_DEFAULT; status_t err = onGetPlaybackSettings(&rate); sp response = new AMessage; if (err == OK) { writeToAMessage(response, rate); } response->setInt32("err", err); response->postReply(replyID); break; } case kWhatConfigSync: { sp replyID; CHECK(msg->senderAwaitsResponse(&replyID)); AVSyncSettings sync; float videoFpsHint; readFromAMessage(msg, &sync, &videoFpsHint); status_t err = onConfigSync(sync, videoFpsHint); sp response = new AMessage; response->setInt32("err", err); response->postReply(replyID); break; } case kWhatGetSyncSettings: { sp replyID; CHECK(msg->senderAwaitsResponse(&replyID)); ALOGV("kWhatGetSyncSettings"); AVSyncSettings sync; float videoFps = -1.f; status_t err = onGetSyncSettings(&sync, &videoFps); sp response = new AMessage; if (err == OK) { writeToAMessage(response, sync, videoFps); } response->setInt32("err", err); response->postReply(replyID); break; } case kWhatFlush: { onFlush(msg); break; } case kWhatDisableOffloadAudio: { onDisableOffloadAudio(); break; } case kWhatEnableOffloadAudio: { onEnableOffloadAudio(); break; } case kWhatPause: { onPause(); break; } case kWhatResume: { onResume(); break; } case kWhatSetVideoFrameRate: { float fps; CHECK(msg->findFloat("frame-rate", &fps)); onSetVideoFrameRate(fps); break; } case kWhatAudioTearDown: { onAudioTearDown(kDueToError); break; } case kWhatAudioTearDownComplete: { onAudioTearDownComplete(); break; } case kWhatAudioOffloadPauseTimeout: { int32_t generation; CHECK(msg->findInt32("drainGeneration", &generation)); if (generation != mAudioOffloadPauseTimeoutGeneration) { break; } ALOGV("Audio Offload tear down due to pause timeout."); onAudioTearDown(kDueToTimeout); mWakeLock->release(); break; } default: TRESPASS(); break; } } void NuPlayer::Renderer::postDrainAudioQueue_l(int64_t delayUs) { if (mDrainAudioQueuePending || mSyncQueues || mUseAudioCallback) { return; } if (mAudioQueue.empty()) { return; } // FIXME: if paused, wait until AudioTrack stop() is complete before delivering data. if (mPaused) { const int64_t diffUs = mPauseDrainAudioAllowedUs - ALooper::GetNowUs(); if (diffUs > delayUs) { delayUs = diffUs; } } mDrainAudioQueuePending = true; sp msg = new AMessage(kWhatDrainAudioQueue, this); msg->setInt32("drainGeneration", mAudioDrainGeneration); msg->post(delayUs); } void NuPlayer::Renderer::prepareForMediaRenderingStart_l() { mAudioRenderingStartGeneration = mAudioDrainGeneration; mVideoRenderingStartGeneration = mVideoDrainGeneration; mRenderingDataDelivered = false; } void NuPlayer::Renderer::notifyIfMediaRenderingStarted_l() { if (mVideoRenderingStartGeneration == mVideoDrainGeneration && mAudioRenderingStartGeneration == mAudioDrainGeneration) { mRenderingDataDelivered = true; if (mPaused) { return; } mVideoRenderingStartGeneration = -1; mAudioRenderingStartGeneration = -1; sp notify = mNotify->dup(); notify->setInt32("what", kWhatMediaRenderingStart); notify->post(); } } // static size_t NuPlayer::Renderer::AudioSinkCallback( MediaPlayerBase::AudioSink * /* audioSink */, void *buffer, size_t size, void *cookie, MediaPlayerBase::AudioSink::cb_event_t event) { NuPlayer::Renderer *me = (NuPlayer::Renderer *)cookie; switch (event) { case MediaPlayerBase::AudioSink::CB_EVENT_FILL_BUFFER: { return me->fillAudioBuffer(buffer, size); break; } case MediaPlayerBase::AudioSink::CB_EVENT_STREAM_END: { ALOGV("AudioSink::CB_EVENT_STREAM_END"); me->notifyEOS(true /* audio */, ERROR_END_OF_STREAM); break; } case MediaPlayerBase::AudioSink::CB_EVENT_TEAR_DOWN: { ALOGV("AudioSink::CB_EVENT_TEAR_DOWN"); me->notifyAudioTearDown(); break; } } return 0; } size_t NuPlayer::Renderer::fillAudioBuffer(void *buffer, size_t size) { Mutex::Autolock autoLock(mLock); if (!mUseAudioCallback) { return 0; } bool hasEOS = false; size_t sizeCopied = 0; bool firstEntry = true; QueueEntry *entry; // will be valid after while loop if hasEOS is set. while (sizeCopied < size && !mAudioQueue.empty()) { entry = &*mAudioQueue.begin(); if (entry->mBuffer == NULL) { // EOS hasEOS = true; mAudioQueue.erase(mAudioQueue.begin()); break; } if (firstEntry && entry->mOffset == 0) { firstEntry = false; int64_t mediaTimeUs; CHECK(entry->mBuffer->meta()->findInt64("timeUs", &mediaTimeUs)); ALOGV("fillAudioBuffer: rendering audio at media time %.2f secs", mediaTimeUs / 1E6); setAudioFirstAnchorTimeIfNeeded_l(mediaTimeUs); } size_t copy = entry->mBuffer->size() - entry->mOffset; size_t sizeRemaining = size - sizeCopied; if (copy > sizeRemaining) { copy = sizeRemaining; } memcpy((char *)buffer + sizeCopied, entry->mBuffer->data() + entry->mOffset, copy); entry->mOffset += copy; if (entry->mOffset == entry->mBuffer->size()) { entry->mNotifyConsumed->post(); mAudioQueue.erase(mAudioQueue.begin()); entry = NULL; } sizeCopied += copy; notifyIfMediaRenderingStarted_l(); } if (mAudioFirstAnchorTimeMediaUs >= 0) { int64_t nowUs = ALooper::GetNowUs(); int64_t nowMediaUs = mAudioFirstAnchorTimeMediaUs + getPlayedOutAudioDurationUs(nowUs); // we don't know how much data we are queueing for offloaded tracks. mMediaClock->updateAnchor(nowMediaUs, nowUs, INT64_MAX); mAnchorTimeMediaUs = nowMediaUs; } // for non-offloaded audio, we need to compute the frames written because // there is no EVENT_STREAM_END notification. The frames written gives // an estimate on the pending played out duration. if (!offloadingAudio()) { mNumFramesWritten += sizeCopied / mAudioSink->frameSize(); } if (hasEOS) { (new AMessage(kWhatStopAudioSink, this))->post(); // As there is currently no EVENT_STREAM_END callback notification for // non-offloaded audio tracks, we need to post the EOS ourselves. if (!offloadingAudio()) { int64_t postEOSDelayUs = 0; if (mAudioSink->needsTrailingPadding()) { postEOSDelayUs = getPendingAudioPlayoutDurationUs(ALooper::GetNowUs()); } ALOGV("fillAudioBuffer: notifyEOS " "mNumFramesWritten:%u finalResult:%d postEOSDelay:%lld", mNumFramesWritten, entry->mFinalResult, (long long)postEOSDelayUs); notifyEOS(true /* audio */, entry->mFinalResult, postEOSDelayUs); } } return sizeCopied; } void NuPlayer::Renderer::drainAudioQueueUntilLastEOS() { List::iterator it = mAudioQueue.begin(), itEOS = it; bool foundEOS = false; while (it != mAudioQueue.end()) { int32_t eos; QueueEntry *entry = &*it++; if (entry->mBuffer == NULL || (entry->mNotifyConsumed->findInt32("eos", &eos) && eos != 0)) { itEOS = it; foundEOS = true; } } if (foundEOS) { // post all replies before EOS and drop the samples for (it = mAudioQueue.begin(); it != itEOS; it++) { if (it->mBuffer == NULL) { // delay doesn't matter as we don't even have an AudioTrack notifyEOS(true /* audio */, it->mFinalResult); } else { it->mNotifyConsumed->post(); } } mAudioQueue.erase(mAudioQueue.begin(), itEOS); } } bool NuPlayer::Renderer::onDrainAudioQueue() { // do not drain audio during teardown as queued buffers may be invalid. if (mAudioTearingDown) { return false; } // TODO: This call to getPosition checks if AudioTrack has been created // in AudioSink before draining audio. If AudioTrack doesn't exist, then // CHECKs on getPosition will fail. // We still need to figure out why AudioTrack is not created when // this function is called. One possible reason could be leftover // audio. Another possible place is to check whether decoder // has received INFO_FORMAT_CHANGED as the first buffer since // AudioSink is opened there, and possible interactions with flush // immediately after start. Investigate error message // "vorbis_dsp_synthesis returned -135", along with RTSP. uint32_t numFramesPlayed; if (mAudioSink->getPosition(&numFramesPlayed) != OK) { // When getPosition fails, renderer will not reschedule the draining // unless new samples are queued. // If we have pending EOS (or "eos" marker for discontinuities), we need // to post these now as NuPlayerDecoder might be waiting for it. drainAudioQueueUntilLastEOS(); ALOGW("onDrainAudioQueue(): audio sink is not ready"); return false; } #if 0 ssize_t numFramesAvailableToWrite = mAudioSink->frameCount() - (mNumFramesWritten - numFramesPlayed); if (numFramesAvailableToWrite == mAudioSink->frameCount()) { ALOGI("audio sink underrun"); } else { ALOGV("audio queue has %d frames left to play", mAudioSink->frameCount() - numFramesAvailableToWrite); } #endif uint32_t prevFramesWritten = mNumFramesWritten; while (!mAudioQueue.empty()) { QueueEntry *entry = &*mAudioQueue.begin(); mLastAudioBufferDrained = entry->mBufferOrdinal; if (entry->mBuffer == NULL) { // EOS int64_t postEOSDelayUs = 0; if (mAudioSink->needsTrailingPadding()) { postEOSDelayUs = getPendingAudioPlayoutDurationUs(ALooper::GetNowUs()); } notifyEOS(true /* audio */, entry->mFinalResult, postEOSDelayUs); mAudioQueue.erase(mAudioQueue.begin()); entry = NULL; if (mAudioSink->needsTrailingPadding()) { // If we're not in gapless playback (i.e. through setNextPlayer), we // need to stop the track here, because that will play out the last // little bit at the end of the file. Otherwise short files won't play. mAudioSink->stop(); mNumFramesWritten = 0; } return false; } // ignore 0-sized buffer which could be EOS marker with no data if (entry->mOffset == 0 && entry->mBuffer->size() > 0) { int64_t mediaTimeUs; CHECK(entry->mBuffer->meta()->findInt64("timeUs", &mediaTimeUs)); ALOGV("onDrainAudioQueue: rendering audio at media time %.2f secs", mediaTimeUs / 1E6); onNewAudioMediaTime(mediaTimeUs); } size_t copy = entry->mBuffer->size() - entry->mOffset; ssize_t written = mAudioSink->write(entry->mBuffer->data() + entry->mOffset, copy, false /* blocking */); if (written < 0) { // An error in AudioSink write. Perhaps the AudioSink was not properly opened. if (written == WOULD_BLOCK) { ALOGV("AudioSink write would block when writing %zu bytes", copy); } else { ALOGE("AudioSink write error(%zd) when writing %zu bytes", written, copy); // This can only happen when AudioSink was opened with doNotReconnect flag set to // true, in which case the NuPlayer will handle the reconnect. notifyAudioTearDown(); } break; } entry->mOffset += written; if (entry->mOffset == entry->mBuffer->size()) { entry->mNotifyConsumed->post(); mAudioQueue.erase(mAudioQueue.begin()); entry = NULL; } size_t copiedFrames = written / mAudioSink->frameSize(); mNumFramesWritten += copiedFrames; { Mutex::Autolock autoLock(mLock); int64_t maxTimeMedia; maxTimeMedia = mAnchorTimeMediaUs + (int64_t)(max((long long)mNumFramesWritten - mAnchorNumFramesWritten, 0LL) * 1000LL * mAudioSink->msecsPerFrame()); mMediaClock->updateMaxTimeMedia(maxTimeMedia); notifyIfMediaRenderingStarted_l(); } if (written != (ssize_t)copy) { // A short count was received from AudioSink::write() // // AudioSink write is called in non-blocking mode. // It may return with a short count when: // // 1) Size to be copied is not a multiple of the frame size. We consider this fatal. // 2) The data to be copied exceeds the available buffer in AudioSink. // 3) An error occurs and data has been partially copied to the buffer in AudioSink. // 4) AudioSink is an AudioCache for data retrieval, and the AudioCache is exceeded. // (Case 1) // Must be a multiple of the frame size. If it is not a multiple of a frame size, it // needs to fail, as we should not carry over fractional frames between calls. if (copy % mAudioSink->frameSize()) { // CHECK_EQ(copy % mAudioSink->frameSize(), 0); ALOGE("CHECK_EQ(copy % mAudioSink->frameSize(), 0) failed b/25372978"); ALOGE("mAudioSink->frameSize() %zu", mAudioSink->frameSize()); ALOGE("bytes to copy %zu", copy); ALOGE("entry size %zu, entry offset %zu", entry->mBuffer->size(), entry->mOffset - written); notifyEOS(true /*audio*/, UNKNOWN_ERROR); return false; } // (Case 2, 3, 4) // Return early to the caller. // Beware of calling immediately again as this may busy-loop if you are not careful. ALOGV("AudioSink write short frame count %zd < %zu", written, copy); break; } } // calculate whether we need to reschedule another write. bool reschedule = !mAudioQueue.empty() && (!mPaused || prevFramesWritten != mNumFramesWritten); // permit pause to fill buffers //ALOGD("reschedule:%d empty:%d mPaused:%d prevFramesWritten:%u mNumFramesWritten:%u", // reschedule, mAudioQueue.empty(), mPaused, prevFramesWritten, mNumFramesWritten); return reschedule; } int64_t NuPlayer::Renderer::getDurationUsIfPlayedAtSampleRate(uint32_t numFrames) { int32_t sampleRate = offloadingAudio() ? mCurrentOffloadInfo.sample_rate : mCurrentPcmInfo.mSampleRate; if (sampleRate == 0) { ALOGE("sampleRate is 0 in %s mode", offloadingAudio() ? "offload" : "non-offload"); return 0; } // TODO: remove the (int32_t) casting below as it may overflow at 12.4 hours. return (int64_t)((int32_t)numFrames * 1000000LL / sampleRate); } // Calculate duration of pending samples if played at normal rate (i.e., 1.0). int64_t NuPlayer::Renderer::getPendingAudioPlayoutDurationUs(int64_t nowUs) { int64_t writtenAudioDurationUs = getDurationUsIfPlayedAtSampleRate(mNumFramesWritten); return writtenAudioDurationUs - getPlayedOutAudioDurationUs(nowUs); } int64_t NuPlayer::Renderer::getRealTimeUs(int64_t mediaTimeUs, int64_t nowUs) { int64_t realUs; if (mMediaClock->getRealTimeFor(mediaTimeUs, &realUs) != OK) { // If failed to get current position, e.g. due to audio clock is // not ready, then just play out video immediately without delay. return nowUs; } return realUs; } void NuPlayer::Renderer::onNewAudioMediaTime(int64_t mediaTimeUs) { Mutex::Autolock autoLock(mLock); // TRICKY: vorbis decoder generates multiple frames with the same // timestamp, so only update on the first frame with a given timestamp if (mediaTimeUs == mAnchorTimeMediaUs) { return; } setAudioFirstAnchorTimeIfNeeded_l(mediaTimeUs); int64_t nowUs = ALooper::GetNowUs(); int64_t nowMediaUs = mediaTimeUs - getPendingAudioPlayoutDurationUs(nowUs); mMediaClock->updateAnchor(nowMediaUs, nowUs, mediaTimeUs); mAnchorNumFramesWritten = mNumFramesWritten; mAnchorTimeMediaUs = mediaTimeUs; } // Called without mLock acquired. void NuPlayer::Renderer::postDrainVideoQueue() { if (mDrainVideoQueuePending || getSyncQueues() || (mPaused && mVideoSampleReceived)) { return; } if (mVideoQueue.empty()) { return; } QueueEntry &entry = *mVideoQueue.begin(); sp msg = new AMessage(kWhatDrainVideoQueue, this); msg->setInt32("drainGeneration", getDrainGeneration(false /* audio */)); if (entry.mBuffer == NULL) { // EOS doesn't carry a timestamp. msg->post(); mDrainVideoQueuePending = true; return; } int64_t delayUs; int64_t nowUs = ALooper::GetNowUs(); int64_t realTimeUs; if (mFlags & FLAG_REAL_TIME) { int64_t mediaTimeUs; CHECK(entry.mBuffer->meta()->findInt64("timeUs", &mediaTimeUs)); realTimeUs = mediaTimeUs; } else { int64_t mediaTimeUs; CHECK(entry.mBuffer->meta()->findInt64("timeUs", &mediaTimeUs)); { Mutex::Autolock autoLock(mLock); if (mAnchorTimeMediaUs < 0) { mMediaClock->updateAnchor(mediaTimeUs, nowUs, mediaTimeUs); mAnchorTimeMediaUs = mediaTimeUs; realTimeUs = nowUs; } else if (!mVideoSampleReceived) { // Always render the first video frame. realTimeUs = nowUs; } else { realTimeUs = getRealTimeUs(mediaTimeUs, nowUs); } } if (!mHasAudio) { // smooth out videos >= 10fps mMediaClock->updateMaxTimeMedia(mediaTimeUs + 100000); } // Heuristics to handle situation when media time changed without a // discontinuity. If we have not drained an audio buffer that was // received after this buffer, repost in 10 msec. Otherwise repost // in 500 msec. delayUs = realTimeUs - nowUs; if (delayUs > 500000) { int64_t postDelayUs = 500000; if (mHasAudio && (mLastAudioBufferDrained - entry.mBufferOrdinal) <= 0) { postDelayUs = 10000; } msg->setWhat(kWhatPostDrainVideoQueue); msg->post(postDelayUs); mVideoScheduler->restart(); ALOGI("possible video time jump of %dms, retrying in %dms", (int)(delayUs / 1000), (int)(postDelayUs / 1000)); mDrainVideoQueuePending = true; return; } } realTimeUs = mVideoScheduler->schedule(realTimeUs * 1000) / 1000; int64_t twoVsyncsUs = 2 * (mVideoScheduler->getVsyncPeriod() / 1000); delayUs = realTimeUs - nowUs; ALOGW_IF(delayUs > 500000, "unusually high delayUs: %" PRId64, delayUs); // post 2 display refreshes before rendering is due // FIXME currently this increases power consumption, so unless frame-accurate // AV sync is requested, post closer to required render time (at 0.63 vsyncs) if (!sFrameAccurateAVsync) { twoVsyncsUs >>= 4; } msg->post(delayUs > twoVsyncsUs ? delayUs - twoVsyncsUs : 0); mDrainVideoQueuePending = true; } void NuPlayer::Renderer::onDrainVideoQueue() { if (mVideoQueue.empty()) { return; } QueueEntry *entry = &*mVideoQueue.begin(); if (entry->mBuffer == NULL) { // EOS notifyEOS(false /* audio */, entry->mFinalResult); mVideoQueue.erase(mVideoQueue.begin()); entry = NULL; setVideoLateByUs(0); return; } int64_t nowUs = ALooper::GetNowUs(); int64_t realTimeUs; if (mFlags & FLAG_REAL_TIME) { CHECK(entry->mBuffer->meta()->findInt64("timeUs", &realTimeUs)); } else { int64_t mediaTimeUs; CHECK(entry->mBuffer->meta()->findInt64("timeUs", &mediaTimeUs)); realTimeUs = getRealTimeUs(mediaTimeUs, nowUs); } bool tooLate = false; if (!mPaused) { setVideoLateByUs(nowUs - realTimeUs); tooLate = (mVideoLateByUs > 40000); if (tooLate) { ALOGV("video late by %lld us (%.2f secs)", (long long)mVideoLateByUs, mVideoLateByUs / 1E6); } else { int64_t mediaUs = 0; mMediaClock->getMediaTime(realTimeUs, &mediaUs); ALOGV("rendering video at media time %.2f secs", (mFlags & FLAG_REAL_TIME ? realTimeUs : mediaUs) / 1E6); } } else { setVideoLateByUs(0); if (!mVideoSampleReceived && !mHasAudio) { // This will ensure that the first frame after a flush won't be used as anchor // when renderer is in paused state, because resume can happen any time after seek. Mutex::Autolock autoLock(mLock); clearAnchorTime_l(); } } // Always render the first video frame while keeping stats on A/V sync. if (!mVideoSampleReceived) { realTimeUs = nowUs; tooLate = false; } entry->mNotifyConsumed->setInt64("timestampNs", realTimeUs * 1000ll); entry->mNotifyConsumed->setInt32("render", !tooLate); entry->mNotifyConsumed->post(); mVideoQueue.erase(mVideoQueue.begin()); entry = NULL; mVideoSampleReceived = true; if (!mPaused) { if (!mVideoRenderingStarted) { mVideoRenderingStarted = true; notifyVideoRenderingStart(); } Mutex::Autolock autoLock(mLock); notifyIfMediaRenderingStarted_l(); } } void NuPlayer::Renderer::notifyVideoRenderingStart() { sp notify = mNotify->dup(); notify->setInt32("what", kWhatVideoRenderingStart); notify->post(); } void NuPlayer::Renderer::notifyEOS(bool audio, status_t finalResult, int64_t delayUs) { if (audio) { mFoundAudioEOS = true; } sp notify = mNotify->dup(); notify->setInt32("what", kWhatEOS); notify->setInt32("audio", static_cast(audio)); notify->setInt32("finalResult", finalResult); notify->post(delayUs); } void NuPlayer::Renderer::notifyAudioTearDown() { (new AMessage(kWhatAudioTearDown, this))->post(); } void NuPlayer::Renderer::onQueueBuffer(const sp &msg) { int32_t audio; CHECK(msg->findInt32("audio", &audio)); if (dropBufferIfStale(audio, msg)) { return; } if (audio) { mHasAudio = true; } else { mHasVideo = true; } if (mHasVideo) { if (mVideoScheduler == NULL) { mVideoScheduler = new VideoFrameScheduler(); mVideoScheduler->init(); } } sp buffer; CHECK(msg->findBuffer("buffer", &buffer)); sp notifyConsumed; CHECK(msg->findMessage("notifyConsumed", ¬ifyConsumed)); QueueEntry entry; entry.mBuffer = buffer; entry.mNotifyConsumed = notifyConsumed; entry.mOffset = 0; entry.mFinalResult = OK; entry.mBufferOrdinal = ++mTotalBuffersQueued; if (audio) { Mutex::Autolock autoLock(mLock); #if 1 sp newBuffer; status_t err = AVNuUtils::get()->convertToSinkFormatIfNeeded( buffer, newBuffer, (offloadingAudio() ? mCurrentOffloadInfo.format : mCurrentPcmInfo.mFormat), offloadingAudio()); switch (err) { case NO_INIT: // passthru decoder pushes some buffers before the audio sink // is opened. Since the offload format is known only when the sink // is opened, pcm conversions cannot take place. So, retry. ALOGI("init pending, retrying in 10ms, this shouldn't happen"); msg->post(10000LL); return; case OK: break; default: ALOGW("error 0x%x in converting to sink format, drop buffer", err); notifyConsumed->post(); return; } CHECK(newBuffer != NULL); entry.mBuffer = newBuffer; #endif mAudioQueue.push_back(entry); postDrainAudioQueue_l(); } else { mVideoQueue.push_back(entry); postDrainVideoQueue(); } Mutex::Autolock autoLock(mLock); if (!mSyncQueues || mAudioQueue.empty() || mVideoQueue.empty()) { return; } sp firstAudioBuffer = (*mAudioQueue.begin()).mBuffer; sp firstVideoBuffer = (*mVideoQueue.begin()).mBuffer; if (firstAudioBuffer == NULL || firstVideoBuffer == NULL) { // EOS signalled on either queue. syncQueuesDone_l(); return; } int64_t firstAudioTimeUs; int64_t firstVideoTimeUs; CHECK(firstAudioBuffer->meta() ->findInt64("timeUs", &firstAudioTimeUs)); CHECK(firstVideoBuffer->meta() ->findInt64("timeUs", &firstVideoTimeUs)); int64_t diff = firstVideoTimeUs - firstAudioTimeUs; ALOGV("queueDiff = %.2f secs", diff / 1E6); if (diff > 100000ll) { // Audio data starts More than 0.1 secs before video. // Drop some audio. (*mAudioQueue.begin()).mNotifyConsumed->post(); mAudioQueue.erase(mAudioQueue.begin()); return; } syncQueuesDone_l(); } void NuPlayer::Renderer::syncQueuesDone_l() { if (!mSyncQueues) { return; } mSyncQueues = false; if (!mAudioQueue.empty()) { postDrainAudioQueue_l(); } if (!mVideoQueue.empty()) { mLock.unlock(); postDrainVideoQueue(); mLock.lock(); } } void NuPlayer::Renderer::onQueueEOS(const sp &msg) { int32_t audio; CHECK(msg->findInt32("audio", &audio)); if (dropBufferIfStale(audio, msg)) { return; } int32_t finalResult; CHECK(msg->findInt32("finalResult", &finalResult)); QueueEntry entry; entry.mOffset = 0; entry.mFinalResult = finalResult; if (audio) { Mutex::Autolock autoLock(mLock); if (mAudioQueue.empty() && mSyncQueues) { syncQueuesDone_l(); } mAudioQueue.push_back(entry); postDrainAudioQueue_l(); } else { if (mVideoQueue.empty() && getSyncQueues()) { Mutex::Autolock autoLock(mLock); syncQueuesDone_l(); } mVideoQueue.push_back(entry); postDrainVideoQueue(); } } void NuPlayer::Renderer::onFlush(const sp &msg) { int32_t audio, notifyComplete; CHECK(msg->findInt32("audio", &audio)); { Mutex::Autolock autoLock(mLock); if (audio) { notifyComplete = mNotifyCompleteAudio; mNotifyCompleteAudio = false; } else { notifyComplete = mNotifyCompleteVideo; mNotifyCompleteVideo = false; } // If we're currently syncing the queues, i.e. dropping audio while // aligning the first audio/video buffer times and only one of the // two queues has data, we may starve that queue by not requesting // more buffers from the decoder. If the other source then encounters // a discontinuity that leads to flushing, we'll never find the // corresponding discontinuity on the other queue. // Therefore we'll stop syncing the queues if at least one of them // is flushed. syncQueuesDone_l(); clearAnchorTime_l(); } ALOGV("flushing %s", audio ? "audio" : "video"); if (audio) { { Mutex::Autolock autoLock(mLock); flushQueue(&mAudioQueue); ++mAudioDrainGeneration; prepareForMediaRenderingStart_l(); // the frame count will be reset after flush. clearAudioFirstAnchorTime_l(); } mDrainAudioQueuePending = false; if (offloadingAudio()) { mAudioSink->pause(); mAudioSink->flush(); if (!mPaused) { mAudioSink->start(); } } else { mAudioSink->pause(); mAudioSink->flush(); // Call stop() to signal to the AudioSink to completely fill the // internal buffer before resuming playback. // FIXME: this is ignored after flush(). mAudioSink->stop(); if (mPaused) { // Race condition: if renderer is paused and audio sink is stopped, // we need to make sure that the audio track buffer fully drains // before delivering data. // FIXME: remove this if we can detect if stop() is complete. const int delayUs = 2 * 50 * 1000; // (2 full mixer thread cycles at 50ms) mPauseDrainAudioAllowedUs = ALooper::GetNowUs() + delayUs; } else { mAudioSink->start(); } mNumFramesWritten = 0; } } else { flushQueue(&mVideoQueue); mDrainVideoQueuePending = false; if (mVideoScheduler != NULL) { mVideoScheduler->restart(); } Mutex::Autolock autoLock(mLock); ++mVideoDrainGeneration; prepareForMediaRenderingStart_l(); } mVideoSampleReceived = false; if (notifyComplete) { notifyFlushComplete(audio); } } void NuPlayer::Renderer::flushQueue(List *queue) { while (!queue->empty()) { QueueEntry *entry = &*queue->begin(); if (entry->mBuffer != NULL) { entry->mNotifyConsumed->post(); } queue->erase(queue->begin()); entry = NULL; } } void NuPlayer::Renderer::notifyFlushComplete(bool audio) { sp notify = mNotify->dup(); notify->setInt32("what", kWhatFlushComplete); notify->setInt32("audio", static_cast(audio)); notify->post(); } bool NuPlayer::Renderer::dropBufferIfStale( bool audio, const sp &msg) { int32_t queueGeneration; CHECK(msg->findInt32("queueGeneration", &queueGeneration)); if (queueGeneration == getQueueGeneration(audio)) { return false; } sp notifyConsumed; if (msg->findMessage("notifyConsumed", ¬ifyConsumed)) { notifyConsumed->post(); } return true; } void NuPlayer::Renderer::onAudioSinkChanged() { if (offloadingAudio()) { return; } CHECK(!mDrainAudioQueuePending); mNumFramesWritten = 0; { Mutex::Autolock autoLock(mLock); mAnchorNumFramesWritten = -1; } uint32_t written; if (mAudioSink->getFramesWritten(&written) == OK) { mNumFramesWritten = written; } } void NuPlayer::Renderer::onDisableOffloadAudio() { Mutex::Autolock autoLock(mLock); mFlags &= ~FLAG_OFFLOAD_AUDIO; ++mAudioDrainGeneration; if (mAudioRenderingStartGeneration != -1) { prepareForMediaRenderingStart_l(); } } void NuPlayer::Renderer::onEnableOffloadAudio() { Mutex::Autolock autoLock(mLock); mFlags |= FLAG_OFFLOAD_AUDIO; ++mAudioDrainGeneration; if (mAudioRenderingStartGeneration != -1) { prepareForMediaRenderingStart_l(); } } void NuPlayer::Renderer::onPause() { if (mPaused) { return; } { Mutex::Autolock autoLock(mLock); // we do not increment audio drain generation so that we fill audio buffer during pause. ++mVideoDrainGeneration; prepareForMediaRenderingStart_l(); mPaused = true; mMediaClock->setPlaybackRate(0.0); } mDrainAudioQueuePending = false; mDrainVideoQueuePending = false; mVideoRenderingStarted = false; // force-notify NOTE_INFO MEDIA_INFO_RENDERING_START after resume if (mHasAudio) { mAudioSink->pause(); startAudioOffloadPauseTimeout(); } ALOGV("now paused audio queue has %zu entries, video has %zu entries", mAudioQueue.size(), mVideoQueue.size()); } void NuPlayer::Renderer::onResume() { readProperties(); if (!mPaused) { return; } if (mHasAudio) { status_t status = NO_ERROR; cancelAudioOffloadPauseTimeout(); status = mAudioSink->start(); if (offloadingAudio() && status != NO_ERROR && status != INVALID_OPERATION) { ALOGD("received error :%d on resume for offload track posting TEAR_DOWN event",status); notifyAudioTearDown(); } //Update anchor time after resuming playback. if (offloadingAudio() && status == NO_ERROR) { int64_t nowUs = ALooper::GetNowUs(); int64_t nowMediaUs = mAudioFirstAnchorTimeMediaUs + getPlayedOutAudioDurationUs(nowUs); mMediaClock->updateAnchor(nowMediaUs, nowUs, INT64_MAX); } } { Mutex::Autolock autoLock(mLock); mPaused = false; // rendering started message may have been delayed if we were paused. if (mRenderingDataDelivered) { notifyIfMediaRenderingStarted_l(); } // configure audiosink as we did not do it when pausing if (mAudioSink != NULL && mAudioSink->ready()) { mAudioSink->setPlaybackRate(mPlaybackSettings); } mMediaClock->setPlaybackRate(mPlaybackRate); if (!mAudioQueue.empty()) { postDrainAudioQueue_l(); } } if (!mVideoQueue.empty()) { postDrainVideoQueue(); } } void NuPlayer::Renderer::onSetVideoFrameRate(float fps) { if (mVideoScheduler == NULL) { mVideoScheduler = new VideoFrameScheduler(); } mVideoScheduler->init(fps); } int32_t NuPlayer::Renderer::getQueueGeneration(bool audio) { Mutex::Autolock autoLock(mLock); return (audio ? mAudioQueueGeneration : mVideoQueueGeneration); } int32_t NuPlayer::Renderer::getDrainGeneration(bool audio) { Mutex::Autolock autoLock(mLock); return (audio ? mAudioDrainGeneration : mVideoDrainGeneration); } bool NuPlayer::Renderer::getSyncQueues() { Mutex::Autolock autoLock(mLock); return mSyncQueues; } // TODO: Remove unnecessary calls to getPlayedOutAudioDurationUs() // as it acquires locks and may query the audio driver. // // Some calls could conceivably retrieve extrapolated data instead of // accessing getTimestamp() or getPosition() every time a data buffer with // a media time is received. // // Calculate duration of played samples if played at normal rate (i.e., 1.0). int64_t NuPlayer::Renderer::getPlayedOutAudioDurationUs(int64_t nowUs) { uint32_t numFramesPlayed; int64_t numFramesPlayedAt; AudioTimestamp ts; static const int64_t kStaleTimestamp100ms = 100000; int64_t durationUs; status_t res = mAudioSink->getTimestamp(ts); if (res == OK) { // case 1: mixing audio tracks and offloaded tracks. numFramesPlayed = ts.mPosition; numFramesPlayedAt = ts.mTime.tv_sec * 1000000LL + ts.mTime.tv_nsec / 1000; const int64_t timestampAge = nowUs - numFramesPlayedAt; if (timestampAge > kStaleTimestamp100ms) { // This is an audio FIXME. // getTimestamp returns a timestamp which may come from audio mixing threads. // After pausing, the MixerThread may go idle, thus the mTime estimate may // become stale. Assuming that the MixerThread runs 20ms, with FastMixer at 5ms, // the max latency should be about 25ms with an average around 12ms (to be verified). // For safety we use 100ms. ALOGV("getTimestamp: returned stale timestamp nowUs(%lld) numFramesPlayedAt(%lld)", (long long)nowUs, (long long)numFramesPlayedAt); numFramesPlayedAt = nowUs - kStaleTimestamp100ms; } //ALOGD("getTimestamp: OK %d %lld", numFramesPlayed, (long long)numFramesPlayedAt); } else if (res == WOULD_BLOCK) { // case 2: transitory state on start of a new track numFramesPlayed = 0; numFramesPlayedAt = nowUs; //ALOGD("getTimestamp: WOULD_BLOCK %d %lld", // numFramesPlayed, (long long)numFramesPlayedAt); } else { // case 3: transitory at new track or audio fast tracks. res = mAudioSink->getPosition(&numFramesPlayed); if (res != OK) { //query to getPosition fails, use media clock to simulate render position getCurrentPosition(&durationUs); durationUs = durationUs - mAnchorTimeMediaUs; return durationUs; } else { numFramesPlayedAt = nowUs; numFramesPlayedAt += 1000LL * mAudioSink->latency() / 2; /* XXX */ } //ALOGD("getPosition: %u %lld", numFramesPlayed, (long long)numFramesPlayedAt); } //CHECK_EQ(numFramesPlayed & (1 << 31), 0); // can't be negative until 12.4 hrs, test durationUs = getDurationUsIfPlayedAtSampleRate(numFramesPlayed) + nowUs - numFramesPlayedAt; if (durationUs < 0) { // Occurs when numFramesPlayed position is very small and the following: // (1) In case 1, the time nowUs is computed before getTimestamp() is called and // numFramesPlayedAt is greater than nowUs by time more than numFramesPlayed. // (2) In case 3, using getPosition and adding mAudioSink->latency() to // numFramesPlayedAt, by a time amount greater than numFramesPlayed. // // Both of these are transitory conditions. ALOGV("getPlayedOutAudioDurationUs: negative duration %lld set to zero", (long long)durationUs); durationUs = 0; } ALOGV("getPlayedOutAudioDurationUs(%lld) nowUs(%lld) frames(%u) framesAt(%lld)", (long long)durationUs, (long long)nowUs, numFramesPlayed, (long long)numFramesPlayedAt); return durationUs; } void NuPlayer::Renderer::onAudioTearDown(AudioTearDownReason reason) { if (mAudioTearingDown) { return; } mAudioTearingDown = true; int64_t currentPositionUs; sp notify = mNotify->dup(); if (getCurrentPosition(¤tPositionUs) == OK) { notify->setInt64("positionUs", currentPositionUs); } mAudioSink->stop(); mAudioSink->flush(); notify->setInt32("what", kWhatAudioTearDown); notify->setInt32("reason", reason); notify->post(); } void NuPlayer::Renderer::startAudioOffloadPauseTimeout() { if (offloadingAudio()) { mWakeLock->acquire(); sp msg = new AMessage(kWhatAudioOffloadPauseTimeout, this); msg->setInt32("drainGeneration", mAudioOffloadPauseTimeoutGeneration); msg->post(kOffloadPauseMaxUs); } } void NuPlayer::Renderer::cancelAudioOffloadPauseTimeout() { if (offloadingAudio()) { mWakeLock->release(true); ++mAudioOffloadPauseTimeoutGeneration; } } status_t NuPlayer::Renderer::onOpenAudioSink( const sp &format, bool offloadOnly, bool hasVideo, uint32_t flags, bool isStreaming) { ALOGV("openAudioSink: offloadOnly(%d) offloadingAudio(%d)", offloadOnly, offloadingAudio()); if (mAudioTearingDown) { ALOGW("openAudioSink: not opening now!, would happen after teardown"); return OK; } bool audioSinkChanged = false; int32_t numChannels; CHECK(format->findInt32("channel-count", &numChannels)); int32_t channelMask; if (!format->findInt32("channel-mask", &channelMask)) { // signal to the AudioSink to derive the mask from count. channelMask = CHANNEL_MASK_USE_CHANNEL_ORDER; } int32_t bitWidth = 16; format->findInt32("bits-per-sample", &bitWidth); int32_t sampleRate; CHECK(format->findInt32("sample-rate", &sampleRate)); AString mime; CHECK(format->findString("mime", &mime)); if (offloadingAudio()) { audio_format_t audioFormat = AUDIO_FORMAT_PCM_16_BIT; status_t err = mapMimeToAudioFormat(audioFormat, mime.c_str()); if (err != OK) { ALOGE("Couldn't map mime \"%s\" to a valid " "audio_format", mime.c_str()); onDisableOffloadAudio(); } else { audioFormat = AVUtils::get()->updateAudioFormat(audioFormat, format); bitWidth = AVUtils::get()->getAudioSampleBits(format); ALOGV("Mime \"%s\" mapped to audio_format 0x%x", mime.c_str(), audioFormat); int avgBitRate = -1; format->findInt32("bitrate", &avgBitRate); int32_t aacProfile = -1; if (audioFormat == AUDIO_FORMAT_AAC && format->findInt32("aac-profile", &aacProfile)) { // Redefine AAC format as per aac profile int32_t isADTSSupported; isADTSSupported = AVUtils::get()->mapAACProfileToAudioFormat(format, audioFormat, aacProfile); if (!isADTSSupported) { mapAACProfileToAudioFormat(audioFormat, aacProfile); } else { ALOGV("Format is AAC ADTS\n"); } } ALOGV("onOpenAudioSink: %s", format->debugString().c_str()); int32_t offloadBufferSize = AVUtils::get()->getAudioMaxInputBufferSize( audioFormat, format); audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER; offloadInfo.duration_us = -1; format->findInt64( "durationUs", &offloadInfo.duration_us); offloadInfo.sample_rate = sampleRate; offloadInfo.channel_mask = channelMask; offloadInfo.format = audioFormat; offloadInfo.stream_type = AUDIO_STREAM_MUSIC; offloadInfo.bit_rate = avgBitRate; offloadInfo.has_video = hasVideo; offloadInfo.is_streaming = isStreaming; offloadInfo.bit_width = bitWidth; offloadInfo.offload_buffer_size = offloadBufferSize; if (memcmp(&mCurrentOffloadInfo, &offloadInfo, sizeof(offloadInfo)) == 0) { ALOGV("openAudioSink: no change in offload mode"); // no change from previous configuration, everything ok. return OK; } mCurrentPcmInfo = AUDIO_PCMINFO_INITIALIZER; ALOGV("openAudioSink: try to open AudioSink in offload mode"); uint32_t offloadFlags = flags; offloadFlags |= AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD; offloadFlags &= ~AUDIO_OUTPUT_FLAG_DEEP_BUFFER; audioSinkChanged = true; mAudioSink->close(); err = mAudioSink->open( sampleRate, numChannels, (audio_channel_mask_t)channelMask, audioFormat, 0 /* bufferCount - unused */, &NuPlayer::Renderer::AudioSinkCallback, this, (audio_output_flags_t)offloadFlags, &offloadInfo); if (err == OK) { err = mAudioSink->setPlaybackRate(mPlaybackSettings); } if (err == OK) { // If the playback is offloaded to h/w, we pass // the HAL some metadata information. // We don't want to do this for PCM because it // will be going through the AudioFlinger mixer // before reaching the hardware. // TODO mCurrentOffloadInfo = offloadInfo; if (!mPaused) { // for preview mode, don't start if paused err = mAudioSink->start(); } ALOGV_IF(err == OK, "openAudioSink: offload succeeded"); } if (err != OK) { // Clean up, fall back to non offload mode. mAudioSink->close(); onDisableOffloadAudio(); mCurrentOffloadInfo = AUDIO_INFO_INITIALIZER; ALOGV("openAudioSink: offload failed"); } else { mUseAudioCallback = true; // offload mode transfers data through callback ++mAudioDrainGeneration; // discard pending kWhatDrainAudioQueue message. mFlags |= FLAG_OFFLOAD_AUDIO; } } } if (!offloadOnly && !offloadingAudio()) { ALOGV("openAudioSink: open AudioSink in NON-offload mode"); uint32_t pcmFlags = flags; pcmFlags &= ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD; const PcmInfo info = { (audio_channel_mask_t)channelMask, (audio_output_flags_t)pcmFlags, AVNuUtils::get()->getPCMFormat(format), numChannels, sampleRate }; if (memcmp(&mCurrentPcmInfo, &info, sizeof(info)) == 0) { ALOGV("openAudioSink: no change in pcm mode"); // no change from previous configuration, everything ok. return OK; } audioSinkChanged = true; mAudioSink->close(); mCurrentOffloadInfo = AUDIO_INFO_INITIALIZER; // Note: It is possible to set up the callback, but not use it to send audio data. // This requires a fix in AudioSink to explicitly specify the transfer mode. mUseAudioCallback = getUseAudioCallbackSetting(); if (mUseAudioCallback) { ++mAudioDrainGeneration; // discard pending kWhatDrainAudioQueue message. } // Compute the desired buffer size. // For callback mode, the amount of time before wakeup is about half the buffer size. const uint32_t frameCount = (unsigned long long)sampleRate * getAudioSinkPcmMsSetting() / 1000; // The doNotReconnect means AudioSink will signal back and let NuPlayer to re-construct // AudioSink. We don't want this when there's video because it will cause a video seek to // the previous I frame. But we do want this when there's only audio because it will give // NuPlayer a chance to switch from non-offload mode to offload mode. // So we only set doNotReconnect when there's no video. const bool doNotReconnect = !hasVideo; status_t err = mAudioSink->open( sampleRate, numChannels, (audio_channel_mask_t)channelMask, AVNuUtils::get()->getPCMFormat(format), 0 /* bufferCount - unused */, mUseAudioCallback ? &NuPlayer::Renderer::AudioSinkCallback : NULL, mUseAudioCallback ? this : NULL, (audio_output_flags_t)pcmFlags, NULL, doNotReconnect, frameCount); if (err == OK) { err = mAudioSink->setPlaybackRate(mPlaybackSettings); } if (err != OK) { ALOGW("openAudioSink: non offloaded open failed status: %d", err); mAudioSink->close(); mCurrentPcmInfo = AUDIO_PCMINFO_INITIALIZER; return err; } mCurrentPcmInfo = info; if (!mPaused) { // for preview mode, don't start if paused mAudioSink->start(); } } if (audioSinkChanged) { onAudioSinkChanged(); } return OK; } void NuPlayer::Renderer::onCloseAudioSink() { mAudioSink->close(); mCurrentOffloadInfo = AUDIO_INFO_INITIALIZER; mCurrentPcmInfo = AUDIO_PCMINFO_INITIALIZER; } void NuPlayer::Renderer::signalAudioTearDownComplete() { (new AMessage(kWhatAudioTearDownComplete, this))->post(); } void NuPlayer::Renderer::onAudioTearDownComplete() { mAudioTearingDown = false; } } // namespace android