| /* |
| * Copyright (C) 2012 The Android Open Source Project |
| * |
| * Licensed under the Apache License, Version 2.0 (the "License"); |
| * you may not use this file except in compliance with the License. |
| * You may obtain a copy of the License at |
| * |
| * http://www.apache.org/licenses/LICENSE-2.0 |
| * |
| * Unless required by applicable law or agreed to in writing, software |
| * distributed under the License is distributed on an "AS IS" BASIS, |
| * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| * See the License for the specific language governing permissions and |
| * limitations under the License. |
| */ |
| |
| //#define LOG_NDEBUG 0 |
| #define LOG_TAG "PlaylistFetcher" |
| #include <utils/Log.h> |
| |
| #include "PlaylistFetcher.h" |
| |
| #include "LiveDataSource.h" |
| #include "LiveSession.h" |
| #include "M3UParser.h" |
| |
| #include "include/avc_utils.h" |
| #include "include/HTTPBase.h" |
| #include "include/ID3.h" |
| #include "mpeg2ts/AnotherPacketSource.h" |
| |
| #include <media/IStreamSource.h> |
| #include <media/stagefright/foundation/ABitReader.h> |
| #include <media/stagefright/foundation/ABuffer.h> |
| #include <media/stagefright/foundation/ADebug.h> |
| #include <media/stagefright/foundation/hexdump.h> |
| #include <media/stagefright/FileSource.h> |
| #include <media/stagefright/MediaDefs.h> |
| #include <media/stagefright/MetaData.h> |
| #include <media/stagefright/Utils.h> |
| |
| #include <ctype.h> |
| #include <openssl/aes.h> |
| #include <openssl/md5.h> |
| |
| namespace android { |
| |
| // static |
| const int64_t PlaylistFetcher::kMinBufferedDurationUs = 10000000ll; |
| const int64_t PlaylistFetcher::kMaxMonitorDelayUs = 3000000ll; |
| const int32_t PlaylistFetcher::kNumSkipFrames = 10; |
| |
| PlaylistFetcher::PlaylistFetcher( |
| const sp<AMessage> ¬ify, |
| const sp<LiveSession> &session, |
| const char *uri) |
| : mNotify(notify), |
| mStartTimeUsNotify(notify->dup()), |
| mSession(session), |
| mURI(uri), |
| mStreamTypeMask(0), |
| mStartTimeUs(-1ll), |
| mMinStartTimeUs(0ll), |
| mStopParams(NULL), |
| mLastPlaylistFetchTimeUs(-1ll), |
| mSeqNumber(-1), |
| mNumRetries(0), |
| mStartup(true), |
| mPrepared(false), |
| mNextPTSTimeUs(-1ll), |
| mMonitorQueueGeneration(0), |
| mRefreshState(INITIAL_MINIMUM_RELOAD_DELAY), |
| mFirstPTSValid(false), |
| mAbsoluteTimeAnchorUs(0ll) { |
| memset(mPlaylistHash, 0, sizeof(mPlaylistHash)); |
| mStartTimeUsNotify->setInt32("what", kWhatStartedAt); |
| mStartTimeUsNotify->setInt32("streamMask", 0); |
| } |
| |
| PlaylistFetcher::~PlaylistFetcher() { |
| } |
| |
| int64_t PlaylistFetcher::getSegmentStartTimeUs(int32_t seqNumber) const { |
| CHECK(mPlaylist != NULL); |
| |
| int32_t firstSeqNumberInPlaylist; |
| if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32( |
| "media-sequence", &firstSeqNumberInPlaylist)) { |
| firstSeqNumberInPlaylist = 0; |
| } |
| |
| int32_t lastSeqNumberInPlaylist = |
| firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1; |
| |
| CHECK_GE(seqNumber, firstSeqNumberInPlaylist); |
| CHECK_LE(seqNumber, lastSeqNumberInPlaylist); |
| |
| int64_t segmentStartUs = 0ll; |
| for (int32_t index = 0; |
| index < seqNumber - firstSeqNumberInPlaylist; ++index) { |
| sp<AMessage> itemMeta; |
| CHECK(mPlaylist->itemAt( |
| index, NULL /* uri */, &itemMeta)); |
| |
| int64_t itemDurationUs; |
| CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); |
| |
| segmentStartUs += itemDurationUs; |
| } |
| |
| return segmentStartUs; |
| } |
| |
| int64_t PlaylistFetcher::delayUsToRefreshPlaylist() const { |
| int64_t nowUs = ALooper::GetNowUs(); |
| |
| if (mPlaylist == NULL || mLastPlaylistFetchTimeUs < 0ll) { |
| CHECK_EQ((int)mRefreshState, (int)INITIAL_MINIMUM_RELOAD_DELAY); |
| return 0ll; |
| } |
| |
| if (mPlaylist->isComplete()) { |
| return (~0llu >> 1); |
| } |
| |
| int32_t targetDurationSecs; |
| CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs)); |
| |
| int64_t targetDurationUs = targetDurationSecs * 1000000ll; |
| |
| int64_t minPlaylistAgeUs; |
| |
| switch (mRefreshState) { |
| case INITIAL_MINIMUM_RELOAD_DELAY: |
| { |
| size_t n = mPlaylist->size(); |
| if (n > 0) { |
| sp<AMessage> itemMeta; |
| CHECK(mPlaylist->itemAt(n - 1, NULL /* uri */, &itemMeta)); |
| |
| int64_t itemDurationUs; |
| CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); |
| |
| minPlaylistAgeUs = itemDurationUs; |
| break; |
| } |
| |
| // fall through |
| } |
| |
| case FIRST_UNCHANGED_RELOAD_ATTEMPT: |
| { |
| minPlaylistAgeUs = targetDurationUs / 2; |
| break; |
| } |
| |
| case SECOND_UNCHANGED_RELOAD_ATTEMPT: |
| { |
| minPlaylistAgeUs = (targetDurationUs * 3) / 2; |
| break; |
| } |
| |
| case THIRD_UNCHANGED_RELOAD_ATTEMPT: |
| { |
| minPlaylistAgeUs = targetDurationUs * 3; |
| break; |
| } |
| |
| default: |
| TRESPASS(); |
| break; |
| } |
| |
| int64_t delayUs = mLastPlaylistFetchTimeUs + minPlaylistAgeUs - nowUs; |
| return delayUs > 0ll ? delayUs : 0ll; |
| } |
| |
| status_t PlaylistFetcher::decryptBuffer( |
| size_t playlistIndex, const sp<ABuffer> &buffer, |
| bool first) { |
| sp<AMessage> itemMeta; |
| bool found = false; |
| AString method; |
| |
| for (ssize_t i = playlistIndex; i >= 0; --i) { |
| AString uri; |
| CHECK(mPlaylist->itemAt(i, &uri, &itemMeta)); |
| |
| if (itemMeta->findString("cipher-method", &method)) { |
| found = true; |
| break; |
| } |
| } |
| |
| if (!found) { |
| method = "NONE"; |
| } |
| buffer->meta()->setString("cipher-method", method.c_str()); |
| |
| if (method == "NONE") { |
| return OK; |
| } else if (!(method == "AES-128")) { |
| ALOGE("Unsupported cipher method '%s'", method.c_str()); |
| return ERROR_UNSUPPORTED; |
| } |
| |
| AString keyURI; |
| if (!itemMeta->findString("cipher-uri", &keyURI)) { |
| ALOGE("Missing key uri"); |
| return ERROR_MALFORMED; |
| } |
| |
| ssize_t index = mAESKeyForURI.indexOfKey(keyURI); |
| |
| sp<ABuffer> key; |
| if (index >= 0) { |
| key = mAESKeyForURI.valueAt(index); |
| } else { |
| status_t err = mSession->fetchFile(keyURI.c_str(), &key); |
| |
| if (err != OK) { |
| ALOGE("failed to fetch cipher key from '%s'.", keyURI.c_str()); |
| return ERROR_IO; |
| } else if (key->size() != 16) { |
| ALOGE("key file '%s' wasn't 16 bytes in size.", keyURI.c_str()); |
| return ERROR_MALFORMED; |
| } |
| |
| mAESKeyForURI.add(keyURI, key); |
| } |
| |
| AES_KEY aes_key; |
| if (AES_set_decrypt_key(key->data(), 128, &aes_key) != 0) { |
| ALOGE("failed to set AES decryption key."); |
| return UNKNOWN_ERROR; |
| } |
| |
| size_t n = buffer->size(); |
| if (!n) { |
| return OK; |
| } |
| CHECK(n % 16 == 0); |
| |
| if (first) { |
| // If decrypting the first block in a file, read the iv from the manifest |
| // or derive the iv from the file's sequence number. |
| |
| AString iv; |
| if (itemMeta->findString("cipher-iv", &iv)) { |
| if ((!iv.startsWith("0x") && !iv.startsWith("0X")) |
| || iv.size() != 16 * 2 + 2) { |
| ALOGE("malformed cipher IV '%s'.", iv.c_str()); |
| return ERROR_MALFORMED; |
| } |
| |
| memset(mAESInitVec, 0, sizeof(mAESInitVec)); |
| for (size_t i = 0; i < 16; ++i) { |
| char c1 = tolower(iv.c_str()[2 + 2 * i]); |
| char c2 = tolower(iv.c_str()[3 + 2 * i]); |
| if (!isxdigit(c1) || !isxdigit(c2)) { |
| ALOGE("malformed cipher IV '%s'.", iv.c_str()); |
| return ERROR_MALFORMED; |
| } |
| uint8_t nibble1 = isdigit(c1) ? c1 - '0' : c1 - 'a' + 10; |
| uint8_t nibble2 = isdigit(c2) ? c2 - '0' : c2 - 'a' + 10; |
| |
| mAESInitVec[i] = nibble1 << 4 | nibble2; |
| } |
| } else { |
| memset(mAESInitVec, 0, sizeof(mAESInitVec)); |
| mAESInitVec[15] = mSeqNumber & 0xff; |
| mAESInitVec[14] = (mSeqNumber >> 8) & 0xff; |
| mAESInitVec[13] = (mSeqNumber >> 16) & 0xff; |
| mAESInitVec[12] = (mSeqNumber >> 24) & 0xff; |
| } |
| } |
| |
| AES_cbc_encrypt( |
| buffer->data(), buffer->data(), buffer->size(), |
| &aes_key, mAESInitVec, AES_DECRYPT); |
| |
| return OK; |
| } |
| |
| status_t PlaylistFetcher::checkDecryptPadding(const sp<ABuffer> &buffer) { |
| status_t err; |
| AString method; |
| CHECK(buffer->meta()->findString("cipher-method", &method)); |
| if (method == "NONE") { |
| return OK; |
| } |
| |
| uint8_t padding = 0; |
| if (buffer->size() > 0) { |
| padding = buffer->data()[buffer->size() - 1]; |
| } |
| |
| if (padding > 16) { |
| return ERROR_MALFORMED; |
| } |
| |
| for (size_t i = buffer->size() - padding; i < padding; i++) { |
| if (buffer->data()[i] != padding) { |
| return ERROR_MALFORMED; |
| } |
| } |
| |
| buffer->setRange(buffer->offset(), buffer->size() - padding); |
| return OK; |
| } |
| |
| void PlaylistFetcher::postMonitorQueue(int64_t delayUs, int64_t minDelayUs) { |
| int64_t maxDelayUs = delayUsToRefreshPlaylist(); |
| if (maxDelayUs < minDelayUs) { |
| maxDelayUs = minDelayUs; |
| } |
| if (delayUs > maxDelayUs) { |
| ALOGV("Need to refresh playlist in %lld", maxDelayUs); |
| delayUs = maxDelayUs; |
| } |
| sp<AMessage> msg = new AMessage(kWhatMonitorQueue, id()); |
| msg->setInt32("generation", mMonitorQueueGeneration); |
| msg->post(delayUs); |
| } |
| |
| void PlaylistFetcher::cancelMonitorQueue() { |
| ++mMonitorQueueGeneration; |
| } |
| |
| void PlaylistFetcher::startAsync( |
| const sp<AnotherPacketSource> &audioSource, |
| const sp<AnotherPacketSource> &videoSource, |
| const sp<AnotherPacketSource> &subtitleSource, |
| int64_t startTimeUs, |
| int64_t minStartTimeUs, |
| int32_t startSeqNumberHint) { |
| sp<AMessage> msg = new AMessage(kWhatStart, id()); |
| |
| uint32_t streamTypeMask = 0ul; |
| |
| if (audioSource != NULL) { |
| msg->setPointer("audioSource", audioSource.get()); |
| streamTypeMask |= LiveSession::STREAMTYPE_AUDIO; |
| } |
| |
| if (videoSource != NULL) { |
| msg->setPointer("videoSource", videoSource.get()); |
| streamTypeMask |= LiveSession::STREAMTYPE_VIDEO; |
| } |
| |
| if (subtitleSource != NULL) { |
| msg->setPointer("subtitleSource", subtitleSource.get()); |
| streamTypeMask |= LiveSession::STREAMTYPE_SUBTITLES; |
| } |
| |
| msg->setInt32("streamTypeMask", streamTypeMask); |
| msg->setInt64("startTimeUs", startTimeUs); |
| msg->setInt64("minStartTimeUs", minStartTimeUs); |
| msg->setInt32("startSeqNumberHint", startSeqNumberHint); |
| msg->post(); |
| } |
| |
| void PlaylistFetcher::pauseAsync() { |
| (new AMessage(kWhatPause, id()))->post(); |
| } |
| |
| void PlaylistFetcher::stopAsync(bool selfTriggered) { |
| sp<AMessage> msg = new AMessage(kWhatStop, id()); |
| msg->setInt32("selfTriggered", selfTriggered); |
| msg->post(); |
| } |
| |
| void PlaylistFetcher::resumeUntilAsync(const sp<AMessage> ¶ms) { |
| AMessage* msg = new AMessage(kWhatResumeUntil, id()); |
| msg->setMessage("params", params); |
| msg->post(); |
| } |
| |
| void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) { |
| switch (msg->what()) { |
| case kWhatStart: |
| { |
| status_t err = onStart(msg); |
| |
| sp<AMessage> notify = mNotify->dup(); |
| notify->setInt32("what", kWhatStarted); |
| notify->setInt32("err", err); |
| notify->post(); |
| break; |
| } |
| |
| case kWhatPause: |
| { |
| onPause(); |
| |
| sp<AMessage> notify = mNotify->dup(); |
| notify->setInt32("what", kWhatPaused); |
| notify->post(); |
| break; |
| } |
| |
| case kWhatStop: |
| { |
| onStop(msg); |
| |
| sp<AMessage> notify = mNotify->dup(); |
| notify->setInt32("what", kWhatStopped); |
| notify->post(); |
| break; |
| } |
| |
| case kWhatMonitorQueue: |
| case kWhatDownloadNext: |
| { |
| int32_t generation; |
| CHECK(msg->findInt32("generation", &generation)); |
| |
| if (generation != mMonitorQueueGeneration) { |
| // Stale event |
| break; |
| } |
| |
| if (msg->what() == kWhatMonitorQueue) { |
| onMonitorQueue(); |
| } else { |
| onDownloadNext(); |
| } |
| break; |
| } |
| |
| case kWhatResumeUntil: |
| { |
| onResumeUntil(msg); |
| break; |
| } |
| |
| default: |
| TRESPASS(); |
| } |
| } |
| |
| status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) { |
| mPacketSources.clear(); |
| |
| uint32_t streamTypeMask; |
| CHECK(msg->findInt32("streamTypeMask", (int32_t *)&streamTypeMask)); |
| |
| int64_t startTimeUs; |
| int32_t startSeqNumberHint; |
| CHECK(msg->findInt64("startTimeUs", &startTimeUs)); |
| CHECK(msg->findInt64("minStartTimeUs", (int64_t *) &mMinStartTimeUs)); |
| CHECK(msg->findInt32("startSeqNumberHint", &startSeqNumberHint)); |
| |
| if (streamTypeMask & LiveSession::STREAMTYPE_AUDIO) { |
| void *ptr; |
| CHECK(msg->findPointer("audioSource", &ptr)); |
| |
| mPacketSources.add( |
| LiveSession::STREAMTYPE_AUDIO, |
| static_cast<AnotherPacketSource *>(ptr)); |
| } |
| |
| if (streamTypeMask & LiveSession::STREAMTYPE_VIDEO) { |
| void *ptr; |
| CHECK(msg->findPointer("videoSource", &ptr)); |
| |
| mPacketSources.add( |
| LiveSession::STREAMTYPE_VIDEO, |
| static_cast<AnotherPacketSource *>(ptr)); |
| } |
| |
| if (streamTypeMask & LiveSession::STREAMTYPE_SUBTITLES) { |
| void *ptr; |
| CHECK(msg->findPointer("subtitleSource", &ptr)); |
| |
| mPacketSources.add( |
| LiveSession::STREAMTYPE_SUBTITLES, |
| static_cast<AnotherPacketSource *>(ptr)); |
| } |
| |
| mStreamTypeMask = streamTypeMask; |
| mStartTimeUs = startTimeUs; |
| |
| if (mStartTimeUs >= 0ll) { |
| mSeqNumber = -1; |
| mStartup = true; |
| mPrepared = false; |
| } |
| |
| if (startSeqNumberHint >= 0) { |
| mSeqNumber = startSeqNumberHint; |
| } |
| |
| postMonitorQueue(); |
| |
| return OK; |
| } |
| |
| void PlaylistFetcher::onPause() { |
| cancelMonitorQueue(); |
| } |
| |
| void PlaylistFetcher::onStop(const sp<AMessage> &msg) { |
| cancelMonitorQueue(); |
| |
| int32_t selfTriggered; |
| CHECK(msg->findInt32("selfTriggered", &selfTriggered)); |
| if (!selfTriggered) { |
| // Self triggered stops only happen during switching, in which case we do not want |
| // to clear the discontinuities queued at the end of packet sources. |
| for (size_t i = 0; i < mPacketSources.size(); i++) { |
| sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); |
| packetSource->clear(); |
| } |
| } |
| |
| mPacketSources.clear(); |
| mStreamTypeMask = 0; |
| } |
| |
| // Resume until we have reached the boundary timestamps listed in `msg`; when |
| // the remaining time is too short (within a resume threshold) stop immediately |
| // instead. |
| status_t PlaylistFetcher::onResumeUntil(const sp<AMessage> &msg) { |
| sp<AMessage> params; |
| CHECK(msg->findMessage("params", ¶ms)); |
| |
| bool stop = false; |
| for (size_t i = 0; i < mPacketSources.size(); i++) { |
| sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); |
| |
| const char *stopKey; |
| int streamType = mPacketSources.keyAt(i); |
| switch (streamType) { |
| case LiveSession::STREAMTYPE_VIDEO: |
| stopKey = "timeUsVideo"; |
| break; |
| |
| case LiveSession::STREAMTYPE_AUDIO: |
| stopKey = "timeUsAudio"; |
| break; |
| |
| case LiveSession::STREAMTYPE_SUBTITLES: |
| stopKey = "timeUsSubtitle"; |
| break; |
| |
| default: |
| TRESPASS(); |
| } |
| |
| // Don't resume if we would stop within a resume threshold. |
| int64_t latestTimeUs = 0, stopTimeUs = 0; |
| sp<AMessage> latestMeta = packetSource->getLatestMeta(); |
| if (latestMeta != NULL |
| && (latestMeta->findInt64("timeUs", &latestTimeUs) |
| && params->findInt64(stopKey, &stopTimeUs))) { |
| int64_t diffUs = stopTimeUs - latestTimeUs; |
| if (diffUs < resumeThreshold(latestMeta)) { |
| stop = true; |
| } |
| } |
| } |
| |
| if (stop) { |
| for (size_t i = 0; i < mPacketSources.size(); i++) { |
| mPacketSources.valueAt(i)->queueAccessUnit(mSession->createFormatChangeBuffer()); |
| } |
| stopAsync(/* selfTriggered = */ true); |
| return OK; |
| } |
| |
| mStopParams = params; |
| postMonitorQueue(); |
| |
| return OK; |
| } |
| |
| void PlaylistFetcher::notifyError(status_t err) { |
| sp<AMessage> notify = mNotify->dup(); |
| notify->setInt32("what", kWhatError); |
| notify->setInt32("err", err); |
| notify->post(); |
| } |
| |
| void PlaylistFetcher::queueDiscontinuity( |
| ATSParser::DiscontinuityType type, const sp<AMessage> &extra) { |
| for (size_t i = 0; i < mPacketSources.size(); ++i) { |
| mPacketSources.valueAt(i)->queueDiscontinuity(type, extra); |
| } |
| } |
| |
| void PlaylistFetcher::onMonitorQueue() { |
| bool downloadMore = false; |
| refreshPlaylist(); |
| |
| int32_t targetDurationSecs; |
| int64_t targetDurationUs = kMinBufferedDurationUs; |
| if (mPlaylist != NULL) { |
| CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs)); |
| targetDurationUs = targetDurationSecs * 1000000ll; |
| } |
| |
| // buffer at least 3 times the target duration, or up to 10 seconds |
| int64_t durationToBufferUs = targetDurationUs * 3; |
| if (durationToBufferUs > kMinBufferedDurationUs) { |
| durationToBufferUs = kMinBufferedDurationUs; |
| } |
| |
| int64_t bufferedDurationUs = 0ll; |
| status_t finalResult = NOT_ENOUGH_DATA; |
| if (mStreamTypeMask == LiveSession::STREAMTYPE_SUBTITLES) { |
| sp<AnotherPacketSource> packetSource = |
| mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES); |
| |
| bufferedDurationUs = |
| packetSource->getBufferedDurationUs(&finalResult); |
| finalResult = OK; |
| } else { |
| // Use max stream duration to prevent us from waiting on a non-existent stream; |
| // when we cannot make out from the manifest what streams are included in a playlist |
| // we might assume extra streams. |
| for (size_t i = 0; i < mPacketSources.size(); ++i) { |
| if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0) { |
| continue; |
| } |
| |
| int64_t bufferedStreamDurationUs = |
| mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult); |
| ALOGV("buffered %lld for stream %d", |
| bufferedStreamDurationUs, mPacketSources.keyAt(i)); |
| if (bufferedStreamDurationUs > bufferedDurationUs) { |
| bufferedDurationUs = bufferedStreamDurationUs; |
| } |
| } |
| } |
| downloadMore = (bufferedDurationUs < durationToBufferUs); |
| |
| // signal start if buffered up at least the target size |
| if (!mPrepared && bufferedDurationUs > targetDurationUs && downloadMore) { |
| mPrepared = true; |
| |
| ALOGV("prepared, buffered=%lld > %lld", |
| bufferedDurationUs, targetDurationUs); |
| sp<AMessage> msg = mNotify->dup(); |
| msg->setInt32("what", kWhatTemporarilyDoneFetching); |
| msg->post(); |
| } |
| |
| if (finalResult == OK && downloadMore) { |
| ALOGV("monitoring, buffered=%lld < %lld", |
| bufferedDurationUs, durationToBufferUs); |
| // delay the next download slightly; hopefully this gives other concurrent fetchers |
| // a better chance to run. |
| // onDownloadNext(); |
| sp<AMessage> msg = new AMessage(kWhatDownloadNext, id()); |
| msg->setInt32("generation", mMonitorQueueGeneration); |
| msg->post(1000l); |
| } else { |
| // Nothing to do yet, try again in a second. |
| |
| sp<AMessage> msg = mNotify->dup(); |
| msg->setInt32("what", kWhatTemporarilyDoneFetching); |
| msg->post(); |
| |
| int64_t delayUs = mPrepared ? kMaxMonitorDelayUs : targetDurationUs / 2; |
| ALOGV("pausing for %lld, buffered=%lld > %lld", |
| delayUs, bufferedDurationUs, durationToBufferUs); |
| // :TRICKY: need to enforce minimum delay because the delay to |
| // refresh the playlist will become 0 |
| postMonitorQueue(delayUs, mPrepared ? targetDurationUs * 2 : 0); |
| } |
| } |
| |
| status_t PlaylistFetcher::refreshPlaylist() { |
| if (delayUsToRefreshPlaylist() <= 0) { |
| bool unchanged; |
| sp<M3UParser> playlist = mSession->fetchPlaylist( |
| mURI.c_str(), mPlaylistHash, &unchanged); |
| |
| if (playlist == NULL) { |
| if (unchanged) { |
| // We succeeded in fetching the playlist, but it was |
| // unchanged from the last time we tried. |
| |
| if (mRefreshState != THIRD_UNCHANGED_RELOAD_ATTEMPT) { |
| mRefreshState = (RefreshState)(mRefreshState + 1); |
| } |
| } else { |
| ALOGE("failed to load playlist at url '%s'", mURI.c_str()); |
| notifyError(ERROR_IO); |
| return ERROR_IO; |
| } |
| } else { |
| mRefreshState = INITIAL_MINIMUM_RELOAD_DELAY; |
| mPlaylist = playlist; |
| |
| if (mPlaylist->isComplete() || mPlaylist->isEvent()) { |
| updateDuration(); |
| } |
| } |
| |
| mLastPlaylistFetchTimeUs = ALooper::GetNowUs(); |
| } |
| return OK; |
| } |
| |
| void PlaylistFetcher::onDownloadNext() { |
| if (refreshPlaylist() != OK) { |
| return; |
| } |
| |
| int32_t firstSeqNumberInPlaylist; |
| if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32( |
| "media-sequence", &firstSeqNumberInPlaylist)) { |
| firstSeqNumberInPlaylist = 0; |
| } |
| |
| bool seekDiscontinuity = false; |
| bool explicitDiscontinuity = false; |
| |
| const int32_t lastSeqNumberInPlaylist = |
| firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1; |
| |
| if (mStartup && mSeqNumber >= 0 |
| && (mSeqNumber < firstSeqNumberInPlaylist || mSeqNumber > lastSeqNumberInPlaylist)) { |
| // in case we guessed wrong during reconfiguration, try fetching the latest content. |
| mSeqNumber = lastSeqNumberInPlaylist; |
| } |
| |
| if (mSeqNumber < 0) { |
| CHECK_GE(mStartTimeUs, 0ll); |
| |
| if (mPlaylist->isComplete() || mPlaylist->isEvent()) { |
| mSeqNumber = getSeqNumberForTime(mStartTimeUs); |
| ALOGV("Initial sequence number for time %lld is %ld from (%ld .. %ld)", |
| mStartTimeUs, mSeqNumber, firstSeqNumberInPlaylist, |
| lastSeqNumberInPlaylist); |
| } else { |
| // If this is a live session, start 3 segments from the end. |
| mSeqNumber = lastSeqNumberInPlaylist - 3; |
| if (mSeqNumber < firstSeqNumberInPlaylist) { |
| mSeqNumber = firstSeqNumberInPlaylist; |
| } |
| ALOGV("Initial sequence number for live event %ld from (%ld .. %ld)", |
| mSeqNumber, firstSeqNumberInPlaylist, |
| lastSeqNumberInPlaylist); |
| } |
| |
| mStartTimeUs = -1ll; |
| } |
| |
| if (mSeqNumber < firstSeqNumberInPlaylist |
| || mSeqNumber > lastSeqNumberInPlaylist) { |
| if (!mPlaylist->isComplete() && mNumRetries < kMaxNumRetries) { |
| ++mNumRetries; |
| |
| if (mSeqNumber > lastSeqNumberInPlaylist) { |
| // refresh in increasing fraction (1/2, 1/3, ...) of the |
| // playlist's target duration or 3 seconds, whichever is less |
| int32_t targetDurationSecs; |
| CHECK(mPlaylist->meta()->findInt32( |
| "target-duration", &targetDurationSecs)); |
| int64_t delayUs = mPlaylist->size() * targetDurationSecs * |
| 1000000ll / (1 + mNumRetries); |
| if (delayUs > kMaxMonitorDelayUs) { |
| delayUs = kMaxMonitorDelayUs; |
| } |
| ALOGV("sequence number high: %ld from (%ld .. %ld), monitor in %lld (retry=%d)", |
| mSeqNumber, firstSeqNumberInPlaylist, |
| lastSeqNumberInPlaylist, delayUs, mNumRetries); |
| postMonitorQueue(delayUs); |
| return; |
| } |
| |
| // we've missed the boat, let's start from the lowest sequence |
| // number available and signal a discontinuity. |
| |
| ALOGI("We've missed the boat, restarting playback." |
| " mStartup=%d, was looking for %d in %d-%d", |
| mStartup, mSeqNumber, firstSeqNumberInPlaylist, |
| lastSeqNumberInPlaylist); |
| mSeqNumber = lastSeqNumberInPlaylist - 3; |
| if (mSeqNumber < firstSeqNumberInPlaylist) { |
| mSeqNumber = firstSeqNumberInPlaylist; |
| } |
| explicitDiscontinuity = true; |
| |
| // fall through |
| } else { |
| ALOGE("Cannot find sequence number %d in playlist " |
| "(contains %d - %d)", |
| mSeqNumber, firstSeqNumberInPlaylist, |
| firstSeqNumberInPlaylist + mPlaylist->size() - 1); |
| |
| notifyError(ERROR_END_OF_STREAM); |
| return; |
| } |
| } |
| |
| mNumRetries = 0; |
| |
| AString uri; |
| sp<AMessage> itemMeta; |
| CHECK(mPlaylist->itemAt( |
| mSeqNumber - firstSeqNumberInPlaylist, |
| &uri, |
| &itemMeta)); |
| |
| int32_t val; |
| if (itemMeta->findInt32("discontinuity", &val) && val != 0) { |
| explicitDiscontinuity = true; |
| } |
| |
| int64_t range_offset, range_length; |
| if (!itemMeta->findInt64("range-offset", &range_offset) |
| || !itemMeta->findInt64("range-length", &range_length)) { |
| range_offset = 0; |
| range_length = -1; |
| } |
| |
| ALOGV("fetching segment %d from (%d .. %d)", |
| mSeqNumber, firstSeqNumberInPlaylist, lastSeqNumberInPlaylist); |
| |
| ALOGV("fetching '%s'", uri.c_str()); |
| |
| sp<ABuffer> buffer; |
| status_t err = mSession->fetchFile( |
| uri.c_str(), &buffer, range_offset, range_length); |
| |
| if (err != OK) { |
| ALOGE("failed to fetch .ts segment at url '%s'", uri.c_str()); |
| notifyError(err); |
| return; |
| } |
| |
| CHECK(buffer != NULL); |
| |
| err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, buffer); |
| if (err == OK) { |
| err = checkDecryptPadding(buffer); |
| } |
| |
| if (err != OK) { |
| ALOGE("decryptBuffer failed w/ error %d", err); |
| |
| notifyError(err); |
| return; |
| } |
| |
| if (mStartup || seekDiscontinuity || explicitDiscontinuity) { |
| // Signal discontinuity. |
| |
| if (mPlaylist->isComplete() || mPlaylist->isEvent()) { |
| // If this was a live event this made no sense since |
| // we don't have access to all the segment before the current |
| // one. |
| mNextPTSTimeUs = getSegmentStartTimeUs(mSeqNumber); |
| } |
| |
| if (seekDiscontinuity || explicitDiscontinuity) { |
| ALOGI("queueing discontinuity (seek=%d, explicit=%d)", |
| seekDiscontinuity, explicitDiscontinuity); |
| |
| queueDiscontinuity( |
| explicitDiscontinuity |
| ? ATSParser::DISCONTINUITY_FORMATCHANGE |
| : ATSParser::DISCONTINUITY_SEEK, |
| NULL /* extra */); |
| } |
| } |
| |
| err = extractAndQueueAccessUnits(buffer, itemMeta); |
| |
| if (err == -EAGAIN) { |
| // bad starting sequence number hint |
| postMonitorQueue(); |
| return; |
| } |
| |
| if (err == ERROR_OUT_OF_RANGE) { |
| // reached stopping point |
| stopAsync(/* selfTriggered = */ true); |
| return; |
| } |
| |
| if (err != OK) { |
| notifyError(err); |
| return; |
| } |
| |
| ++mSeqNumber; |
| |
| postMonitorQueue(); |
| |
| mStartup = false; |
| } |
| |
| int32_t PlaylistFetcher::getSeqNumberForTime(int64_t timeUs) const { |
| int32_t firstSeqNumberInPlaylist; |
| if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32( |
| "media-sequence", &firstSeqNumberInPlaylist)) { |
| firstSeqNumberInPlaylist = 0; |
| } |
| |
| size_t index = 0; |
| int64_t segmentStartUs = 0; |
| while (index < mPlaylist->size()) { |
| sp<AMessage> itemMeta; |
| CHECK(mPlaylist->itemAt( |
| index, NULL /* uri */, &itemMeta)); |
| |
| int64_t itemDurationUs; |
| CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); |
| |
| if (timeUs < segmentStartUs + itemDurationUs) { |
| break; |
| } |
| |
| segmentStartUs += itemDurationUs; |
| ++index; |
| } |
| |
| if (index >= mPlaylist->size()) { |
| index = mPlaylist->size() - 1; |
| } |
| |
| return firstSeqNumberInPlaylist + index; |
| } |
| |
| status_t PlaylistFetcher::extractAndQueueAccessUnits( |
| const sp<ABuffer> &buffer, const sp<AMessage> &itemMeta) { |
| if (buffer->size() > 0 && buffer->data()[0] == 0x47) { |
| // Let's assume this is an MPEG2 transport stream. |
| |
| if ((buffer->size() % 188) != 0) { |
| ALOGE("MPEG2 transport stream is not an even multiple of 188 " |
| "bytes in length."); |
| return ERROR_MALFORMED; |
| } |
| |
| if (mTSParser == NULL) { |
| // Use TS_TIMESTAMPS_ARE_ABSOLUTE so pts carry over between fetchers. |
| mTSParser = new ATSParser(ATSParser::TS_TIMESTAMPS_ARE_ABSOLUTE); |
| } |
| |
| if (mNextPTSTimeUs >= 0ll) { |
| sp<AMessage> extra = new AMessage; |
| // Since we are using absolute timestamps, signal an offset of 0 to prevent |
| // ATSParser from skewing the timestamps of access units. |
| extra->setInt64(IStreamListener::kKeyMediaTimeUs, 0); |
| |
| mTSParser->signalDiscontinuity( |
| ATSParser::DISCONTINUITY_SEEK, extra); |
| |
| mNextPTSTimeUs = -1ll; |
| } |
| |
| size_t offset = 0; |
| while (offset < buffer->size()) { |
| status_t err = mTSParser->feedTSPacket(buffer->data() + offset, 188); |
| |
| if (err != OK) { |
| return err; |
| } |
| |
| offset += 188; |
| } |
| |
| status_t err = OK; |
| for (size_t i = mPacketSources.size(); i-- > 0;) { |
| sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); |
| |
| const char *key; |
| ATSParser::SourceType type; |
| const LiveSession::StreamType stream = mPacketSources.keyAt(i); |
| switch (stream) { |
| |
| case LiveSession::STREAMTYPE_VIDEO: |
| type = ATSParser::VIDEO; |
| key = "timeUsVideo"; |
| break; |
| |
| case LiveSession::STREAMTYPE_AUDIO: |
| type = ATSParser::AUDIO; |
| key = "timeUsAudio"; |
| break; |
| |
| case LiveSession::STREAMTYPE_SUBTITLES: |
| { |
| ALOGE("MPEG2 Transport streams do not contain subtitles."); |
| return ERROR_MALFORMED; |
| break; |
| } |
| |
| default: |
| TRESPASS(); |
| } |
| |
| sp<AnotherPacketSource> source = |
| static_cast<AnotherPacketSource *>( |
| mTSParser->getSource(type).get()); |
| |
| if (source == NULL) { |
| ALOGW("MPEG2 Transport stream does not contain %s data.", |
| type == ATSParser::VIDEO ? "video" : "audio"); |
| |
| mStreamTypeMask &= ~mPacketSources.keyAt(i); |
| mPacketSources.removeItemsAt(i); |
| continue; |
| } |
| |
| int64_t timeUs; |
| sp<ABuffer> accessUnit; |
| status_t finalResult; |
| while (source->hasBufferAvailable(&finalResult) |
| && source->dequeueAccessUnit(&accessUnit) == OK) { |
| |
| CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs)); |
| if (mMinStartTimeUs > 0) { |
| if (timeUs < mMinStartTimeUs) { |
| // TODO untested path |
| // try a later ts |
| int32_t targetDuration; |
| mPlaylist->meta()->findInt32("target-duration", &targetDuration); |
| int32_t incr = (mMinStartTimeUs - timeUs) / 1000000 / targetDuration; |
| if (incr == 0) { |
| // increment mSeqNumber by at least one |
| incr = 1; |
| } |
| mSeqNumber += incr; |
| err = -EAGAIN; |
| break; |
| } else { |
| int64_t startTimeUs; |
| if (mStartTimeUsNotify != NULL |
| && !mStartTimeUsNotify->findInt64(key, &startTimeUs)) { |
| mStartTimeUsNotify->setInt64(key, timeUs); |
| |
| uint32_t streamMask = 0; |
| mStartTimeUsNotify->findInt32("streamMask", (int32_t *) &streamMask); |
| streamMask |= mPacketSources.keyAt(i); |
| mStartTimeUsNotify->setInt32("streamMask", streamMask); |
| |
| if (streamMask == mStreamTypeMask) { |
| mStartTimeUsNotify->post(); |
| mStartTimeUsNotify.clear(); |
| } |
| } |
| } |
| } |
| |
| if (mStopParams != NULL) { |
| // Queue discontinuity in original stream. |
| int64_t stopTimeUs; |
| if (!mStopParams->findInt64(key, &stopTimeUs) || timeUs >= stopTimeUs) { |
| packetSource->queueAccessUnit(mSession->createFormatChangeBuffer()); |
| mStreamTypeMask &= ~stream; |
| mPacketSources.removeItemsAt(i); |
| break; |
| } |
| } |
| |
| // Note that we do NOT dequeue any discontinuities except for format change. |
| |
| // for simplicity, store a reference to the format in each unit |
| sp<MetaData> format = source->getFormat(); |
| if (format != NULL) { |
| accessUnit->meta()->setObject("format", format); |
| } |
| |
| // Stash the sequence number so we can hint future fetchers where to start at. |
| accessUnit->meta()->setInt32("seq", mSeqNumber); |
| packetSource->queueAccessUnit(accessUnit); |
| } |
| |
| if (err != OK) { |
| break; |
| } |
| } |
| |
| if (err != OK) { |
| for (size_t i = mPacketSources.size(); i-- > 0;) { |
| sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i); |
| packetSource->clear(); |
| } |
| return err; |
| } |
| |
| if (!mStreamTypeMask) { |
| // Signal gap is filled between original and new stream. |
| ALOGV("ERROR OUT OF RANGE"); |
| return ERROR_OUT_OF_RANGE; |
| } |
| |
| return OK; |
| } else if (buffer->size() >= 7 && !memcmp("WEBVTT\n", buffer->data(), 7)) { |
| if (mStreamTypeMask != LiveSession::STREAMTYPE_SUBTITLES) { |
| ALOGE("This stream only contains subtitles."); |
| return ERROR_MALFORMED; |
| } |
| |
| const sp<AnotherPacketSource> packetSource = |
| mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES); |
| |
| int64_t durationUs; |
| CHECK(itemMeta->findInt64("durationUs", &durationUs)); |
| buffer->meta()->setInt64("timeUs", getSegmentStartTimeUs(mSeqNumber)); |
| buffer->meta()->setInt64("durationUs", durationUs); |
| buffer->meta()->setInt32("seq", mSeqNumber); |
| |
| packetSource->queueAccessUnit(buffer); |
| return OK; |
| } |
| |
| if (mNextPTSTimeUs >= 0ll) { |
| mFirstPTSValid = false; |
| mAbsoluteTimeAnchorUs = mNextPTSTimeUs; |
| mNextPTSTimeUs = -1ll; |
| } |
| |
| // This better be an ISO 13818-7 (AAC) or ISO 13818-1 (MPEG) audio |
| // stream prefixed by an ID3 tag. |
| |
| bool firstID3Tag = true; |
| uint64_t PTS = 0; |
| |
| for (;;) { |
| // Make sure to skip all ID3 tags preceding the audio data. |
| // At least one must be present to provide the PTS timestamp. |
| |
| ID3 id3(buffer->data(), buffer->size(), true /* ignoreV1 */); |
| if (!id3.isValid()) { |
| if (firstID3Tag) { |
| ALOGE("Unable to parse ID3 tag."); |
| return ERROR_MALFORMED; |
| } else { |
| break; |
| } |
| } |
| |
| if (firstID3Tag) { |
| bool found = false; |
| |
| ID3::Iterator it(id3, "PRIV"); |
| while (!it.done()) { |
| size_t length; |
| const uint8_t *data = it.getData(&length); |
| |
| static const char *kMatchName = |
| "com.apple.streaming.transportStreamTimestamp"; |
| static const size_t kMatchNameLen = strlen(kMatchName); |
| |
| if (length == kMatchNameLen + 1 + 8 |
| && !strncmp((const char *)data, kMatchName, kMatchNameLen)) { |
| found = true; |
| PTS = U64_AT(&data[kMatchNameLen + 1]); |
| } |
| |
| it.next(); |
| } |
| |
| if (!found) { |
| ALOGE("Unable to extract transportStreamTimestamp from ID3 tag."); |
| return ERROR_MALFORMED; |
| } |
| } |
| |
| // skip the ID3 tag |
| buffer->setRange( |
| buffer->offset() + id3.rawSize(), buffer->size() - id3.rawSize()); |
| |
| firstID3Tag = false; |
| } |
| |
| if (!mFirstPTSValid) { |
| mFirstPTSValid = true; |
| mFirstPTS = PTS; |
| } |
| PTS -= mFirstPTS; |
| |
| int64_t timeUs = (PTS * 100ll) / 9ll + mAbsoluteTimeAnchorUs; |
| |
| if (mStreamTypeMask != LiveSession::STREAMTYPE_AUDIO) { |
| ALOGW("This stream only contains audio data!"); |
| |
| mStreamTypeMask &= LiveSession::STREAMTYPE_AUDIO; |
| |
| if (mStreamTypeMask == 0) { |
| return OK; |
| } |
| } |
| |
| sp<AnotherPacketSource> packetSource = |
| mPacketSources.valueFor(LiveSession::STREAMTYPE_AUDIO); |
| |
| if (packetSource->getFormat() == NULL && buffer->size() >= 7) { |
| ABitReader bits(buffer->data(), buffer->size()); |
| |
| // adts_fixed_header |
| |
| CHECK_EQ(bits.getBits(12), 0xfffu); |
| bits.skipBits(3); // ID, layer |
| bool protection_absent = bits.getBits(1) != 0; |
| |
| unsigned profile = bits.getBits(2); |
| CHECK_NE(profile, 3u); |
| unsigned sampling_freq_index = bits.getBits(4); |
| bits.getBits(1); // private_bit |
| unsigned channel_configuration = bits.getBits(3); |
| CHECK_NE(channel_configuration, 0u); |
| bits.skipBits(2); // original_copy, home |
| |
| sp<MetaData> meta = MakeAACCodecSpecificData( |
| profile, sampling_freq_index, channel_configuration); |
| |
| meta->setInt32(kKeyIsADTS, true); |
| |
| packetSource->setFormat(meta); |
| } |
| |
| int64_t numSamples = 0ll; |
| int32_t sampleRate; |
| CHECK(packetSource->getFormat()->findInt32(kKeySampleRate, &sampleRate)); |
| |
| size_t offset = 0; |
| while (offset < buffer->size()) { |
| const uint8_t *adtsHeader = buffer->data() + offset; |
| CHECK_LT(offset + 5, buffer->size()); |
| |
| unsigned aac_frame_length = |
| ((adtsHeader[3] & 3) << 11) |
| | (adtsHeader[4] << 3) |
| | (adtsHeader[5] >> 5); |
| |
| CHECK_LE(offset + aac_frame_length, buffer->size()); |
| |
| sp<ABuffer> unit = new ABuffer(aac_frame_length); |
| memcpy(unit->data(), adtsHeader, aac_frame_length); |
| |
| int64_t unitTimeUs = timeUs + numSamples * 1000000ll / sampleRate; |
| unit->meta()->setInt64("timeUs", unitTimeUs); |
| |
| // Each AAC frame encodes 1024 samples. |
| numSamples += 1024; |
| |
| unit->meta()->setInt32("seq", mSeqNumber); |
| packetSource->queueAccessUnit(unit); |
| |
| offset += aac_frame_length; |
| } |
| |
| return OK; |
| } |
| |
| void PlaylistFetcher::updateDuration() { |
| int64_t durationUs = 0ll; |
| for (size_t index = 0; index < mPlaylist->size(); ++index) { |
| sp<AMessage> itemMeta; |
| CHECK(mPlaylist->itemAt( |
| index, NULL /* uri */, &itemMeta)); |
| |
| int64_t itemDurationUs; |
| CHECK(itemMeta->findInt64("durationUs", &itemDurationUs)); |
| |
| durationUs += itemDurationUs; |
| } |
| |
| sp<AMessage> msg = mNotify->dup(); |
| msg->setInt32("what", kWhatDurationUpdate); |
| msg->setInt64("durationUs", durationUs); |
| msg->post(); |
| } |
| |
| int64_t PlaylistFetcher::resumeThreshold(const sp<AMessage> &msg) { |
| int64_t durationUs, threshold; |
| if (msg->findInt64("durationUs", &durationUs)) { |
| return kNumSkipFrames * durationUs; |
| } |
| |
| sp<RefBase> obj; |
| msg->findObject("format", &obj); |
| MetaData *format = static_cast<MetaData *>(obj.get()); |
| |
| const char *mime; |
| CHECK(format->findCString(kKeyMIMEType, &mime)); |
| bool audio = !strncasecmp(mime, "audio/", 6); |
| if (audio) { |
| // Assumes 1000 samples per frame. |
| int32_t sampleRate; |
| CHECK(format->findInt32(kKeySampleRate, &sampleRate)); |
| return kNumSkipFrames /* frames */ * 1000 /* samples */ |
| * (1000000 / sampleRate) /* sample duration (us) */; |
| } else { |
| int32_t frameRate; |
| if (format->findInt32(kKeyFrameRate, &frameRate) && frameRate > 0) { |
| return kNumSkipFrames * (1000000 / frameRate); |
| } |
| } |
| |
| return 500000ll; |
| } |
| |
| } // namespace android |
