diff options
Diffstat (limited to 'media/libaah_rtp/aah_tx_player.cpp')
| -rw-r--r-- | media/libaah_rtp/aah_tx_player.cpp | 1177 |
1 files changed, 1177 insertions, 0 deletions
diff --git a/media/libaah_rtp/aah_tx_player.cpp b/media/libaah_rtp/aah_tx_player.cpp new file mode 100644 index 0000000..974805b --- /dev/null +++ b/media/libaah_rtp/aah_tx_player.cpp @@ -0,0 +1,1177 @@ +/* + * 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 |