/* * Copyright (C) 2011 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 "LibAAH_RTP" #include <utils/Log.h> #define __STDC_FORMAT_MACROS #include <inttypes.h> #include <netdb.h> #include <netinet/ip.h> #include <common_time/cc_helper.h> #include <media/IMediaPlayer.h> #include <media/stagefright/foundation/ADebug.h> #include <media/stagefright/foundation/AMessage.h> #include <media/stagefright/FileSource.h> #include <media/stagefright/MediaBuffer.h> #include <media/stagefright/MediaDefs.h> #include <media/stagefright/MetaData.h> #include <utils/Timers.h> #include "aah_tx_packet.h" #include "aah_tx_player.h" namespace android { static int64_t kLowWaterMarkUs = 2000000ll; // 2secs static int64_t kHighWaterMarkUs = 10000000ll; // 10secs static const size_t kLowWaterMarkBytes = 40000; static const size_t kHighWaterMarkBytes = 200000; // When we start up, how much lead time should we put on the first access unit? static const int64_t kAAHStartupLeadTimeUs = 300000LL; // How much time do we attempt to lead the clock by in steady state? static const int64_t kAAHBufferTimeUs = 1000000LL; // how long do we keep data in our retransmit buffer after sending it. const int64_t AAH_TXPlayer::kAAHRetryKeepAroundTimeNs = kAAHBufferTimeUs * 1100; sp<MediaPlayerBase> createAAH_TXPlayer() { sp<MediaPlayerBase> ret = new AAH_TXPlayer(); return ret; } template <typename T> static T clamp(T val, T min, T max) { if (val < min) { return min; } else if (val > max) { return max; } else { return val; } } struct AAH_TXEvent : public TimedEventQueue::Event { AAH_TXEvent(AAH_TXPlayer *player, void (AAH_TXPlayer::*method)()) : mPlayer(player) , mMethod(method) {} protected: virtual ~AAH_TXEvent() {} virtual void fire(TimedEventQueue *queue, int64_t /* now_us */) { (mPlayer->*mMethod)(); } private: AAH_TXPlayer *mPlayer; void (AAH_TXPlayer::*mMethod)(); AAH_TXEvent(const AAH_TXEvent &); AAH_TXEvent& operator=(const AAH_TXEvent &); }; AAH_TXPlayer::AAH_TXPlayer() : mQueueStarted(false) , mFlags(0) , mExtractorFlags(0) { DataSource::RegisterDefaultSniffers(); mBufferingEvent = new AAH_TXEvent(this, &AAH_TXPlayer::onBufferingUpdate); mBufferingEventPending = false; mPumpAudioEvent = new AAH_TXEvent(this, &AAH_TXPlayer::onPumpAudio); mPumpAudioEventPending = false; mAudioCodecData = NULL; reset_l(); } AAH_TXPlayer::~AAH_TXPlayer() { if (mQueueStarted) { mQueue.stop(); } reset_l(); } void AAH_TXPlayer::cancelPlayerEvents(bool keepBufferingGoing) { if (!keepBufferingGoing) { mQueue.cancelEvent(mBufferingEvent->eventID()); mBufferingEventPending = false; mQueue.cancelEvent(mPumpAudioEvent->eventID()); mPumpAudioEventPending = false; } } status_t AAH_TXPlayer::initCheck() { // Check for the presense of the common time service by attempting to query // for CommonTime's frequency. If we get an error back, we cannot talk to // the service at all and should abort now. status_t res; uint64_t freq; res = mCCHelper.getCommonFreq(&freq); if (OK != res) { ALOGE("Failed to connect to common time service! (res %d)", res); return res; } return OK; } status_t AAH_TXPlayer::setDataSource( const char *url, const KeyedVector<String8, String8> *headers) { Mutex::Autolock autoLock(mLock); return setDataSource_l(url, headers); } status_t AAH_TXPlayer::setDataSource_l( const char *url, const KeyedVector<String8, String8> *headers) { reset_l(); mUri.setTo(url); if (headers) { mUriHeaders = *headers; ssize_t index = mUriHeaders.indexOfKey(String8("x-hide-urls-from-log")); if (index >= 0) { // Browser is in "incognito" mode, suppress logging URLs. // This isn't something that should be passed to the server. mUriHeaders.removeItemsAt(index); mFlags |= INCOGNITO; } } // The URL may optionally contain a "#" character followed by a Skyjam // cookie. Ideally the cookie header should just be passed in the headers // argument, but the Java API for supplying headers is apparently not yet // exposed in the SDK used by application developers. const char kSkyjamCookieDelimiter = '#'; char* skyjamCookie = strrchr(mUri.string(), kSkyjamCookieDelimiter); if (skyjamCookie) { skyjamCookie++; mUriHeaders.add(String8("Cookie"), String8(skyjamCookie)); mUri = String8(mUri.string(), skyjamCookie - mUri.string()); } return OK; } status_t AAH_TXPlayer::setDataSource(int fd, int64_t offset, int64_t length) { Mutex::Autolock autoLock(mLock); reset_l(); sp<DataSource> dataSource = new FileSource(dup(fd), offset, length); status_t err = dataSource->initCheck(); if (err != OK) { return err; } mFileSource = dataSource; sp<MediaExtractor> extractor = MediaExtractor::Create(dataSource); if (extractor == NULL) { return UNKNOWN_ERROR; } return setDataSource_l(extractor); } status_t AAH_TXPlayer::setVideoSurface(const sp<Surface>& surface) { return OK; } status_t AAH_TXPlayer::setVideoSurfaceTexture( const sp<ISurfaceTexture>& surfaceTexture) { return OK; } status_t AAH_TXPlayer::prepare() { return INVALID_OPERATION; } status_t AAH_TXPlayer::prepareAsync() { Mutex::Autolock autoLock(mLock); return prepareAsync_l(); } status_t AAH_TXPlayer::prepareAsync_l() { if (mFlags & PREPARING) { return UNKNOWN_ERROR; // async prepare already pending } mAAH_Sender = AAH_TXSender::GetInstance(); if (mAAH_Sender == NULL) { return NO_MEMORY; } if (!mQueueStarted) { mQueue.start(); mQueueStarted = true; } mFlags |= PREPARING; mAsyncPrepareEvent = new AAH_TXEvent( this, &AAH_TXPlayer::onPrepareAsyncEvent); mQueue.postEvent(mAsyncPrepareEvent); return OK; } status_t AAH_TXPlayer::finishSetDataSource_l() { sp<DataSource> dataSource; if (!strncasecmp("http://", mUri.string(), 7) || !strncasecmp("https://", mUri.string(), 8)) { mConnectingDataSource = HTTPBase::Create( (mFlags & INCOGNITO) ? HTTPBase::kFlagIncognito : 0); mLock.unlock(); status_t err = mConnectingDataSource->connect(mUri, &mUriHeaders); mLock.lock(); if (err != OK) { mConnectingDataSource.clear(); ALOGI("mConnectingDataSource->connect() returned %d", err); return err; } mCachedSource = new NuCachedSource2(mConnectingDataSource); mConnectingDataSource.clear(); dataSource = mCachedSource; // We're going to prefill the cache before trying to instantiate // the extractor below, as the latter is an operation that otherwise // could block on the datasource for a significant amount of time. // During that time we'd be unable to abort the preparation phase // without this prefill. mLock.unlock(); for (;;) { status_t finalStatus; size_t cachedDataRemaining = mCachedSource->approxDataRemaining(&finalStatus); if (finalStatus != OK || cachedDataRemaining >= kHighWaterMarkBytes || (mFlags & PREPARE_CANCELLED)) { break; } usleep(200000); } mLock.lock(); if (mFlags & PREPARE_CANCELLED) { ALOGI("Prepare cancelled while waiting for initial cache fill."); return UNKNOWN_ERROR; } } else { dataSource = DataSource::CreateFromURI(mUri.string(), &mUriHeaders); } if (dataSource == NULL) { return UNKNOWN_ERROR; } sp<MediaExtractor> extractor = MediaExtractor::Create(dataSource); if (extractor == NULL) { return UNKNOWN_ERROR; } return setDataSource_l(extractor); } status_t AAH_TXPlayer::setDataSource_l(const sp<MediaExtractor> &extractor) { // Attempt to approximate overall stream bitrate by summing all // tracks' individual bitrates, if not all of them advertise bitrate, // we have to fail. int64_t totalBitRate = 0; for (size_t i = 0; i < extractor->countTracks(); ++i) { sp<MetaData> meta = extractor->getTrackMetaData(i); int32_t bitrate; if (!meta->findInt32(kKeyBitRate, &bitrate)) { totalBitRate = -1; break; } totalBitRate += bitrate; } mBitrate = totalBitRate; ALOGV("mBitrate = %lld bits/sec", mBitrate); bool haveAudio = false; for (size_t i = 0; i < extractor->countTracks(); ++i) { sp<MetaData> meta = extractor->getTrackMetaData(i); const char *mime; CHECK(meta->findCString(kKeyMIMEType, &mime)); if (!strncasecmp(mime, "audio/", 6)) { mAudioSource = extractor->getTrack(i); CHECK(mAudioSource != NULL); haveAudio = true; break; } } if (!haveAudio) { return UNKNOWN_ERROR; } mExtractorFlags = extractor->flags(); return OK; } void AAH_TXPlayer::abortPrepare(status_t err) { CHECK(err != OK); notifyListener_l(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, err); mPrepareResult = err; mFlags &= ~(PREPARING|PREPARE_CANCELLED|PREPARING_CONNECTED); mPreparedCondition.broadcast(); } void AAH_TXPlayer::onPrepareAsyncEvent() { Mutex::Autolock autoLock(mLock); if (mFlags & PREPARE_CANCELLED) { ALOGI("prepare was cancelled before doing anything"); abortPrepare(UNKNOWN_ERROR); return; } if (mUri.size() > 0) { status_t err = finishSetDataSource_l(); if (err != OK) { abortPrepare(err); return; } } mAudioFormat = mAudioSource->getFormat(); if (!mAudioFormat->findInt64(kKeyDuration, &mDurationUs)) mDurationUs = 1; const char* mime_type = NULL; if (!mAudioFormat->findCString(kKeyMIMEType, &mime_type)) { ALOGE("Failed to find audio substream MIME type during prepare."); abortPrepare(BAD_VALUE); return; } if (!strcmp(mime_type, MEDIA_MIMETYPE_AUDIO_MPEG)) { mAudioCodec = TRTPAudioPacket::kCodecMPEG1Audio; } else if (!strcmp(mime_type, MEDIA_MIMETYPE_AUDIO_AAC)) { mAudioCodec = TRTPAudioPacket::kCodecAACAudio; uint32_t type; int32_t sample_rate; int32_t channel_count; const void* esds_data; size_t esds_len; if (!mAudioFormat->findInt32(kKeySampleRate, &sample_rate)) { ALOGE("Failed to find sample rate for AAC substream."); abortPrepare(BAD_VALUE); return; } if (!mAudioFormat->findInt32(kKeyChannelCount, &channel_count)) { ALOGE("Failed to find channel count for AAC substream."); abortPrepare(BAD_VALUE); return; } if (!mAudioFormat->findData(kKeyESDS, &type, &esds_data, &esds_len)) { ALOGE("Failed to find codec init data for AAC substream."); abortPrepare(BAD_VALUE); return; } CHECK(NULL == mAudioCodecData); mAudioCodecDataSize = esds_len + sizeof(sample_rate) + sizeof(channel_count); mAudioCodecData = new uint8_t[mAudioCodecDataSize]; if (NULL == mAudioCodecData) { ALOGE("Failed to allocate %u bytes for AAC substream codec aux" " data.", mAudioCodecDataSize); mAudioCodecDataSize = 0; abortPrepare(BAD_VALUE); return; } uint8_t* tmp = mAudioCodecData; tmp[0] = static_cast<uint8_t>((sample_rate >> 24) & 0xFF); tmp[1] = static_cast<uint8_t>((sample_rate >> 16) & 0xFF); tmp[2] = static_cast<uint8_t>((sample_rate >> 8) & 0xFF); tmp[3] = static_cast<uint8_t>((sample_rate ) & 0xFF); tmp[4] = static_cast<uint8_t>((channel_count >> 24) & 0xFF); tmp[5] = static_cast<uint8_t>((channel_count >> 16) & 0xFF); tmp[6] = static_cast<uint8_t>((channel_count >> 8) & 0xFF); tmp[7] = static_cast<uint8_t>((channel_count ) & 0xFF); memcpy(tmp + 8, esds_data, esds_len); } else { ALOGE("Unsupported MIME type \"%s\" in audio substream", mime_type); abortPrepare(BAD_VALUE); return; } status_t err = mAudioSource->start(); if (err != OK) { ALOGI("failed to start audio source, err=%d", err); abortPrepare(err); return; } mFlags |= PREPARING_CONNECTED; if (mCachedSource != NULL) { postBufferingEvent_l(); } else { finishAsyncPrepare_l(); } } void AAH_TXPlayer::finishAsyncPrepare_l() { notifyListener_l(MEDIA_PREPARED); mPrepareResult = OK; mFlags &= ~(PREPARING|PREPARE_CANCELLED|PREPARING_CONNECTED); mFlags |= PREPARED; mPreparedCondition.broadcast(); } status_t AAH_TXPlayer::start() { Mutex::Autolock autoLock(mLock); mFlags &= ~CACHE_UNDERRUN; return play_l(); } status_t AAH_TXPlayer::play_l() { if (mFlags & PLAYING) { return OK; } if (!(mFlags & PREPARED)) { return INVALID_OPERATION; } { Mutex::Autolock lock(mEndpointLock); if (!mEndpointValid) { return INVALID_OPERATION; } if (!mEndpointRegistered) { mProgramID = mAAH_Sender->registerEndpoint(mEndpoint); mEndpointRegistered = true; } } mFlags |= PLAYING; updateClockTransform_l(false); postPumpAudioEvent_l(-1); return OK; } status_t AAH_TXPlayer::stop() { status_t ret = pause(); sendEOS_l(); return ret; } status_t AAH_TXPlayer::pause() { Mutex::Autolock autoLock(mLock); mFlags &= ~CACHE_UNDERRUN; return pause_l(); } status_t AAH_TXPlayer::pause_l(bool doClockUpdate) { if (!(mFlags & PLAYING)) { return OK; } cancelPlayerEvents(true /* keepBufferingGoing */); mFlags &= ~PLAYING; if (doClockUpdate) { updateClockTransform_l(true); } return OK; } void AAH_TXPlayer::updateClockTransform_l(bool pause) { // record the new pause status so that onPumpAudio knows what rate to apply // when it initializes the transform mPlayRateIsPaused = pause; // if we haven't yet established a valid clock transform, then we can't // do anything here if (!mCurrentClockTransformValid) { return; } // sample the current common time int64_t commonTimeNow; if (OK != mCCHelper.getCommonTime(&commonTimeNow)) { ALOGE("updateClockTransform_l get common time failed"); mCurrentClockTransformValid = false; return; } // convert the current common time to media time using the old // transform int64_t mediaTimeNow; if (!mCurrentClockTransform.doReverseTransform( commonTimeNow, &mediaTimeNow)) { ALOGE("updateClockTransform_l reverse transform failed"); mCurrentClockTransformValid = false; return; } // calculate a new transform that preserves the old transform's // result for the current time mCurrentClockTransform.a_zero = mediaTimeNow; mCurrentClockTransform.b_zero = commonTimeNow; mCurrentClockTransform.a_to_b_numer = 1; mCurrentClockTransform.a_to_b_denom = pause ? 0 : 1; // send a packet announcing the new transform sp<TRTPControlPacket> packet = new TRTPControlPacket(); packet->setClockTransform(mCurrentClockTransform); packet->setCommandID(TRTPControlPacket::kCommandNop); queuePacketToSender_l(packet); } void AAH_TXPlayer::sendEOS_l() { sp<TRTPControlPacket> packet = new TRTPControlPacket(); packet->setCommandID(TRTPControlPacket::kCommandEOS); queuePacketToSender_l(packet); } bool AAH_TXPlayer::isPlaying() { return (mFlags & PLAYING) || (mFlags & CACHE_UNDERRUN); } status_t AAH_TXPlayer::seekTo(int msec) { if (mExtractorFlags & MediaExtractor::CAN_SEEK) { Mutex::Autolock autoLock(mLock); return seekTo_l(static_cast<int64_t>(msec) * 1000); } notifyListener_l(MEDIA_SEEK_COMPLETE); return OK; } status_t AAH_TXPlayer::seekTo_l(int64_t timeUs) { mIsSeeking = true; mSeekTimeUs = timeUs; mCurrentClockTransformValid = false; mLastQueuedMediaTimePTSValid = false; // send a flush command packet sp<TRTPControlPacket> packet = new TRTPControlPacket(); packet->setCommandID(TRTPControlPacket::kCommandFlush); queuePacketToSender_l(packet); return OK; } status_t AAH_TXPlayer::getCurrentPosition(int *msec) { if (!msec) { return BAD_VALUE; } Mutex::Autolock lock(mLock); int position; if (mIsSeeking) { position = mSeekTimeUs / 1000; } else if (mCurrentClockTransformValid) { // sample the current common time int64_t commonTimeNow; if (OK != mCCHelper.getCommonTime(&commonTimeNow)) { ALOGE("getCurrentPosition get common time failed"); return INVALID_OPERATION; } int64_t mediaTimeNow; if (!mCurrentClockTransform.doReverseTransform(commonTimeNow, &mediaTimeNow)) { ALOGE("getCurrentPosition reverse transform failed"); return INVALID_OPERATION; } position = static_cast<int>(mediaTimeNow / 1000); } else { position = 0; } int duration; if (getDuration_l(&duration) == OK) { *msec = clamp(position, 0, duration); } else { *msec = (position >= 0) ? position : 0; } return OK; } status_t AAH_TXPlayer::getDuration(int* msec) { if (!msec) { return BAD_VALUE; } Mutex::Autolock lock(mLock); return getDuration_l(msec); } status_t AAH_TXPlayer::getDuration_l(int* msec) { if (mDurationUs < 0) { return UNKNOWN_ERROR; } *msec = (mDurationUs + 500) / 1000; return OK; } status_t AAH_TXPlayer::reset() { Mutex::Autolock autoLock(mLock); reset_l(); return OK; } void AAH_TXPlayer::reset_l() { if (mFlags & PREPARING) { mFlags |= PREPARE_CANCELLED; if (mConnectingDataSource != NULL) { ALOGI("interrupting the connection process"); mConnectingDataSource->disconnect(); } if (mFlags & PREPARING_CONNECTED) { // We are basically done preparing, we're just buffering // enough data to start playback, we can safely interrupt that. finishAsyncPrepare_l(); } } while (mFlags & PREPARING) { mPreparedCondition.wait(mLock); } cancelPlayerEvents(); sendEOS_l(); mCachedSource.clear(); if (mAudioSource != NULL) { mAudioSource->stop(); } mAudioSource.clear(); mAudioCodec = TRTPAudioPacket::kCodecInvalid; mAudioFormat = NULL; delete[] mAudioCodecData; mAudioCodecData = NULL; mAudioCodecDataSize = 0; mFlags = 0; mExtractorFlags = 0; mDurationUs = -1; mIsSeeking = false; mSeekTimeUs = 0; mUri.setTo(""); mUriHeaders.clear(); mFileSource.clear(); mBitrate = -1; { Mutex::Autolock lock(mEndpointLock); if (mAAH_Sender != NULL && mEndpointRegistered) { mAAH_Sender->unregisterEndpoint(mEndpoint); } mEndpointRegistered = false; mEndpointValid = false; } mProgramID = 0; mAAH_Sender.clear(); mLastQueuedMediaTimePTSValid = false; mCurrentClockTransformValid = false; mPlayRateIsPaused = false; mTRTPVolume = 255; } status_t AAH_TXPlayer::setLooping(int loop) { return OK; } player_type AAH_TXPlayer::playerType() { return AAH_TX_PLAYER; } status_t AAH_TXPlayer::setParameter(int key, const Parcel &request) { return ERROR_UNSUPPORTED; } status_t AAH_TXPlayer::getParameter(int key, Parcel *reply) { return ERROR_UNSUPPORTED; } status_t AAH_TXPlayer::invoke(const Parcel& request, Parcel *reply) { return INVALID_OPERATION; } status_t AAH_TXPlayer::getMetadata(const media::Metadata::Filter& ids, Parcel* records) { using media::Metadata; Metadata metadata(records); metadata.appendBool(Metadata::kPauseAvailable, true); metadata.appendBool(Metadata::kSeekBackwardAvailable, false); metadata.appendBool(Metadata::kSeekForwardAvailable, false); metadata.appendBool(Metadata::kSeekAvailable, false); return OK; } status_t AAH_TXPlayer::setVolume(float leftVolume, float rightVolume) { if (leftVolume != rightVolume) { ALOGE("%s does not support per channel volume: %f, %f", __PRETTY_FUNCTION__, leftVolume, rightVolume); } float volume = clamp(leftVolume, 0.0f, 1.0f); Mutex::Autolock lock(mLock); mTRTPVolume = static_cast<uint8_t>((leftVolume * 255.0) + 0.5); return OK; } status_t AAH_TXPlayer::setAudioStreamType(audio_stream_type_t streamType) { return OK; } status_t AAH_TXPlayer::setRetransmitEndpoint( const struct sockaddr_in* endpoint) { Mutex::Autolock lock(mLock); if (NULL == endpoint) return BAD_VALUE; // Once the endpoint has been registered, it may not be changed. if (mEndpointRegistered) return INVALID_OPERATION; mEndpoint.addr = endpoint->sin_addr.s_addr; mEndpoint.port = endpoint->sin_port; mEndpointValid = true; return OK; } void AAH_TXPlayer::notifyListener_l(int msg, int ext1, int ext2) { sendEvent(msg, ext1, ext2); } bool AAH_TXPlayer::getBitrate_l(int64_t *bitrate) { off64_t size; if (mDurationUs >= 0 && mCachedSource != NULL && mCachedSource->getSize(&size) == OK) { *bitrate = size * 8000000ll / mDurationUs; // in bits/sec return true; } if (mBitrate >= 0) { *bitrate = mBitrate; return true; } *bitrate = 0; return false; } // Returns true iff cached duration is available/applicable. bool AAH_TXPlayer::getCachedDuration_l(int64_t *durationUs, bool *eos) { int64_t bitrate; if (mCachedSource != NULL && getBitrate_l(&bitrate)) { status_t finalStatus; size_t cachedDataRemaining = mCachedSource->approxDataRemaining( &finalStatus); *durationUs = cachedDataRemaining * 8000000ll / bitrate; *eos = (finalStatus != OK); return true; } return false; } void AAH_TXPlayer::ensureCacheIsFetching_l() { if (mCachedSource != NULL) { mCachedSource->resumeFetchingIfNecessary(); } } void AAH_TXPlayer::postBufferingEvent_l() { if (mBufferingEventPending) { return; } mBufferingEventPending = true; mQueue.postEventWithDelay(mBufferingEvent, 1000000ll); } void AAH_TXPlayer::postPumpAudioEvent_l(int64_t delayUs) { if (mPumpAudioEventPending) { return; } mPumpAudioEventPending = true; mQueue.postEventWithDelay(mPumpAudioEvent, delayUs < 0 ? 10000 : delayUs); } void AAH_TXPlayer::onBufferingUpdate() { Mutex::Autolock autoLock(mLock); if (!mBufferingEventPending) { return; } mBufferingEventPending = false; if (mCachedSource != NULL) { status_t finalStatus; size_t cachedDataRemaining = mCachedSource->approxDataRemaining( &finalStatus); bool eos = (finalStatus != OK); if (eos) { if (finalStatus == ERROR_END_OF_STREAM) { notifyListener_l(MEDIA_BUFFERING_UPDATE, 100); } if (mFlags & PREPARING) { ALOGV("cache has reached EOS, prepare is done."); finishAsyncPrepare_l(); } } else { int64_t bitrate; if (getBitrate_l(&bitrate)) { size_t cachedSize = mCachedSource->cachedSize(); int64_t cachedDurationUs = cachedSize * 8000000ll / bitrate; int percentage = (100.0 * (double) cachedDurationUs) / mDurationUs; if (percentage > 100) { percentage = 100; } notifyListener_l(MEDIA_BUFFERING_UPDATE, percentage); } else { // We don't know the bitrate of the stream, use absolute size // limits to maintain the cache. if ((mFlags & PLAYING) && !eos && (cachedDataRemaining < kLowWaterMarkBytes)) { ALOGI("cache is running low (< %d) , pausing.", kLowWaterMarkBytes); mFlags |= CACHE_UNDERRUN; pause_l(); ensureCacheIsFetching_l(); notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_START); } else if (eos || cachedDataRemaining > kHighWaterMarkBytes) { if (mFlags & CACHE_UNDERRUN) { ALOGI("cache has filled up (> %d), resuming.", kHighWaterMarkBytes); mFlags &= ~CACHE_UNDERRUN; play_l(); notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_END); } else if (mFlags & PREPARING) { ALOGV("cache has filled up (> %d), prepare is done", kHighWaterMarkBytes); finishAsyncPrepare_l(); } } } } } int64_t cachedDurationUs; bool eos; if (getCachedDuration_l(&cachedDurationUs, &eos)) { ALOGV("cachedDurationUs = %.2f secs, eos=%d", cachedDurationUs / 1E6, eos); if ((mFlags & PLAYING) && !eos && (cachedDurationUs < kLowWaterMarkUs)) { ALOGI("cache is running low (%.2f secs) , pausing.", cachedDurationUs / 1E6); mFlags |= CACHE_UNDERRUN; pause_l(); ensureCacheIsFetching_l(); notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_START); } else if (eos || cachedDurationUs > kHighWaterMarkUs) { if (mFlags & CACHE_UNDERRUN) { ALOGI("cache has filled up (%.2f secs), resuming.", cachedDurationUs / 1E6); mFlags &= ~CACHE_UNDERRUN; play_l(); notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_END); } else if (mFlags & PREPARING) { ALOGV("cache has filled up (%.2f secs), prepare is done", cachedDurationUs / 1E6); finishAsyncPrepare_l(); } } } postBufferingEvent_l(); } void AAH_TXPlayer::onPumpAudio() { while (true) { Mutex::Autolock autoLock(mLock); // If this flag is clear, its because someone has externally canceled // this pump operation (probably because we a resetting/shutting down). // Get out immediately, do not reschedule ourselves. if (!mPumpAudioEventPending) { return; } // Start by checking if there is still work to be doing. If we have // never queued a payload (so we don't know what the last queued PTS is) // or we have never established a MediaTime->CommonTime transformation, // then we have work to do (one time through this loop should establish // both). Otherwise, we want to keep a fixed amt of presentation time // worth of data buffered. If we cannot get common time (service is // unavailable, or common time is undefined)) then we don't have a lot // of good options here. For now, signal an error up to the app level // and shut down the transmission pump. int64_t commonTimeNow; if (OK != mCCHelper.getCommonTime(&commonTimeNow)) { // Failed to get common time; either the service is down or common // time is not synced. Raise an error and shutdown the player. ALOGE("*** Cannot pump audio, unable to fetch common time." " Shutting down."); notifyListener_l(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, UNKNOWN_ERROR); mPumpAudioEventPending = false; break; } if (mCurrentClockTransformValid && mLastQueuedMediaTimePTSValid) { int64_t mediaTimeNow; bool conversionResult = mCurrentClockTransform.doReverseTransform( commonTimeNow, &mediaTimeNow); CHECK(conversionResult); if ((mediaTimeNow + kAAHBufferTimeUs - mLastQueuedMediaTimePTS) <= 0) { break; } } MediaSource::ReadOptions options; if (mIsSeeking) { options.setSeekTo(mSeekTimeUs); } MediaBuffer* mediaBuffer; status_t err = mAudioSource->read(&mediaBuffer, &options); if (err != NO_ERROR) { if (err == ERROR_END_OF_STREAM) { ALOGI("*** %s reached end of stream", __PRETTY_FUNCTION__); notifyListener_l(MEDIA_BUFFERING_UPDATE, 100); notifyListener_l(MEDIA_PLAYBACK_COMPLETE); pause_l(false); sendEOS_l(); } else { ALOGE("*** %s read failed err=%d", __PRETTY_FUNCTION__, err); } return; } if (mIsSeeking) { mIsSeeking = false; notifyListener_l(MEDIA_SEEK_COMPLETE); } uint8_t* data = (static_cast<uint8_t*>(mediaBuffer->data()) + mediaBuffer->range_offset()); ALOGV("*** %s got media buffer data=[%02hhx %02hhx %02hhx %02hhx]" " offset=%d length=%d", __PRETTY_FUNCTION__, data[0], data[1], data[2], data[3], mediaBuffer->range_offset(), mediaBuffer->range_length()); int64_t mediaTimeUs; CHECK(mediaBuffer->meta_data()->findInt64(kKeyTime, &mediaTimeUs)); ALOGV("*** timeUs=%lld", mediaTimeUs); if (!mCurrentClockTransformValid) { if (OK == mCCHelper.getCommonTime(&commonTimeNow)) { mCurrentClockTransform.a_zero = mediaTimeUs; mCurrentClockTransform.b_zero = commonTimeNow + kAAHStartupLeadTimeUs; mCurrentClockTransform.a_to_b_numer = 1; mCurrentClockTransform.a_to_b_denom = mPlayRateIsPaused ? 0 : 1; mCurrentClockTransformValid = true; } else { // Failed to get common time; either the service is down or // common time is not synced. Raise an error and shutdown the // player. ALOGE("*** Cannot begin transmission, unable to fetch common" " time. Dropping sample with pts=%lld", mediaTimeUs); notifyListener_l(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, UNKNOWN_ERROR); mPumpAudioEventPending = false; break; } } ALOGV("*** transmitting packet with pts=%lld", mediaTimeUs); sp<TRTPAudioPacket> packet = new TRTPAudioPacket(); packet->setPTS(mediaTimeUs); packet->setSubstreamID(1); packet->setCodecType(mAudioCodec); packet->setVolume(mTRTPVolume); // TODO : introduce a throttle for this so we can control the // frequency with which transforms get sent. packet->setClockTransform(mCurrentClockTransform); packet->setAccessUnitData(data, mediaBuffer->range_length()); // TODO : while its pretty much universally true that audio ES payloads // are all RAPs across all codecs, it might be a good idea to throttle // the frequency with which we send codec out of band data to the RXers. // If/when we do, we need to flag only those payloads which have // required out of band data attached to them as RAPs. packet->setRandomAccessPoint(true); if (mAudioCodecData && mAudioCodecDataSize) { packet->setAuxData(mAudioCodecData, mAudioCodecDataSize); } queuePacketToSender_l(packet); mediaBuffer->release(); mLastQueuedMediaTimePTSValid = true; mLastQueuedMediaTimePTS = mediaTimeUs; } { // Explicit scope for the autolock pattern. Mutex::Autolock autoLock(mLock); // If someone externally has cleared this flag, its because we should be // shutting down. Do not reschedule ourselves. if (!mPumpAudioEventPending) { return; } // Looks like no one canceled us explicitly. Clear our flag and post a // new event to ourselves. mPumpAudioEventPending = false; postPumpAudioEvent_l(10000); } } void AAH_TXPlayer::queuePacketToSender_l(const sp<TRTPPacket>& packet) { if (mAAH_Sender == NULL) { return; } sp<AMessage> message = new AMessage(AAH_TXSender::kWhatSendPacket, mAAH_Sender->handlerID()); { Mutex::Autolock lock(mEndpointLock); if (!mEndpointValid) { return; } message->setInt32(AAH_TXSender::kSendPacketIPAddr, mEndpoint.addr); message->setInt32(AAH_TXSender::kSendPacketPort, mEndpoint.port); } packet->setProgramID(mProgramID); packet->setExpireTime(systemTime() + kAAHRetryKeepAroundTimeNs); packet->pack(); message->setObject(AAH_TXSender::kSendPacketTRTPPacket, packet); message->post(); } } // namespace android