From 6708a45ba1b5eacf583d5e80b2075b68940fe0ab Mon Sep 17 00:00:00 2001 From: Lajos Molnar Date: Mon, 16 Dec 2013 10:16:32 -0800 Subject: stagefright: Fix bugs in playlist fetcher logic 1. Improve start time: start playback as soon as the target duration or 10s was buffered. 2. Select playlist monitor time based on target duration to avoid continuously missing the boat. 3. If "we miss the boat" we still must request a safe sequence number (last - 3) Change-Id: Ie99c360ac67b152ad9af19e9c6e520016f67e4e3 Signed-off-by: Lajos Molnar Bug: 12060952 --- media/libstagefright/httplive/PlaylistFetcher.cpp | 127 +++++++++++++++++----- media/libstagefright/httplive/PlaylistFetcher.h | 7 +- 2 files changed, 105 insertions(+), 29 deletions(-) (limited to 'media/libstagefright/httplive') diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp index 973b779..1754bf2 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.cpp +++ b/media/libstagefright/httplive/PlaylistFetcher.cpp @@ -47,6 +47,7 @@ namespace android { // static const int64_t PlaylistFetcher::kMinBufferedDurationUs = 10000000ll; +const int64_t PlaylistFetcher::kMaxMonitorDelayUs = 3000000ll; PlaylistFetcher::PlaylistFetcher( const sp ¬ify, @@ -61,6 +62,7 @@ PlaylistFetcher::PlaylistFetcher( mSeqNumber(-1), mNumRetries(0), mStartup(true), + mPrepared(false), mNextPTSTimeUs(-1ll), mMonitorQueueGeneration(0), mRefreshState(INITIAL_MINIMUM_RELOAD_DELAY), @@ -103,10 +105,16 @@ int64_t PlaylistFetcher::getSegmentStartTimeUs(int32_t seqNumber) const { return segmentStartUs; } -bool PlaylistFetcher::timeToRefreshPlaylist(int64_t nowUs) const { - if (mPlaylist == NULL) { +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 true; + return 0ll; + } + + if (mPlaylist->isComplete()) { + return (~0llu >> 1); } int32_t targetDurationSecs; @@ -157,7 +165,8 @@ bool PlaylistFetcher::timeToRefreshPlaylist(int64_t nowUs) const { break; } - return mLastPlaylistFetchTimeUs + minPlaylistAgeUs <= nowUs; + int64_t delayUs = mLastPlaylistFetchTimeUs + minPlaylistAgeUs - nowUs; + return delayUs > 0ll ? delayUs : 0ll; } status_t PlaylistFetcher::decryptBuffer( @@ -274,7 +283,15 @@ status_t PlaylistFetcher::decryptBuffer( return OK; } -void PlaylistFetcher::postMonitorQueue(int64_t delayUs) { +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 msg = new AMessage(kWhatMonitorQueue, id()); msg->setInt32("generation", mMonitorQueueGeneration); msg->post(delayUs); @@ -415,6 +432,7 @@ status_t PlaylistFetcher::onStart(const sp &msg) { if (mStartTimeUs >= 0ll) { mSeqNumber = -1; mStartup = true; + mPrepared = false; } postMonitorQueue(); @@ -456,40 +474,62 @@ void PlaylistFetcher::queueDiscontinuity( 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; + } - status_t finalResult; + // 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 packetSource = mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES); - int64_t bufferedDurationUs = + bufferedDurationUs = packetSource->getBufferedDurationUs(&finalResult); - - downloadMore = (bufferedDurationUs < kMinBufferedDurationUs); finalResult = OK; } else { bool first = true; - int64_t minBufferedDurationUs = 0ll; for (size_t i = 0; i < mPacketSources.size(); ++i) { if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0) { continue; } - int64_t bufferedDurationUs = + int64_t bufferedStreamDurationUs = mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult); - - if (first || bufferedDurationUs < minBufferedDurationUs) { - minBufferedDurationUs = bufferedDurationUs; + if (first || bufferedStreamDurationUs < bufferedDurationUs) { + bufferedDurationUs = bufferedStreamDurationUs; first = false; } } + } + downloadMore = (bufferedDurationUs < durationToBufferUs); + + // signal start if buffered up at least the target size + if (!mPrepared && bufferedDurationUs > targetDurationUs && downloadMore) { + mPrepared = true; - downloadMore = - !first && (minBufferedDurationUs < kMinBufferedDurationUs); + ALOGV("prepared, buffered=%lld > %lld", + bufferedDurationUs, targetDurationUs); + sp msg = mNotify->dup(); + msg->setInt32("what", kWhatTemporarilyDoneFetching); + msg->post(); } if (finalResult == OK && downloadMore) { + ALOGV("monitoring, buffered=%lld < %lld", + bufferedDurationUs, durationToBufferUs); onDownloadNext(); } else { // Nothing to do yet, try again in a second. @@ -498,15 +538,17 @@ void PlaylistFetcher::onMonitorQueue() { msg->setInt32("what", kWhatTemporarilyDoneFetching); msg->post(); - postMonitorQueue(1000000ll); + 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); } } -void PlaylistFetcher::onDownloadNext() { - int64_t nowUs = ALooper::GetNowUs(); - - if (mLastPlaylistFetchTimeUs < 0ll - || (!mPlaylist->isComplete() && timeToRefreshPlaylist(nowUs))) { +status_t PlaylistFetcher::refreshPlaylist() { + if (delayUsToRefreshPlaylist() <= 0) { bool unchanged; sp playlist = mSession->fetchPlaylist( mURI.c_str(), mPlaylistHash, &unchanged); @@ -522,7 +564,7 @@ void PlaylistFetcher::onDownloadNext() { } else { ALOGE("failed to load playlist at url '%s'", mURI.c_str()); notifyError(ERROR_IO); - return; + return ERROR_IO; } } else { mRefreshState = INITIAL_MINIMUM_RELOAD_DELAY; @@ -535,6 +577,13 @@ void PlaylistFetcher::onDownloadNext() { mLastPlaylistFetchTimeUs = ALooper::GetNowUs(); } + return OK; +} + +void PlaylistFetcher::onDownloadNext() { + if (refreshPlaylist() != OK) { + return; + } int32_t firstSeqNumberInPlaylist; if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32( @@ -553,12 +602,18 @@ void PlaylistFetcher::onDownloadNext() { 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; @@ -570,16 +625,34 @@ void PlaylistFetcher::onDownloadNext() { ++mNumRetries; if (mSeqNumber > lastSeqNumberInPlaylist) { - mLastPlaylistFetchTimeUs = -1; - postMonitorQueue(3000000ll); + // 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."); - mSeqNumber = lastSeqNumberInPlaylist; + 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 diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h index 1648e02..78dea20 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.h +++ b/media/libstagefright/httplive/PlaylistFetcher.h @@ -79,6 +79,7 @@ private: }; static const int64_t kMinBufferedDurationUs; + static const int64_t kMaxMonitorDelayUs; sp mNotify; sp mSession; @@ -97,6 +98,7 @@ private: int32_t mSeqNumber; int32_t mNumRetries; bool mStartup; + bool mPrepared; int64_t mNextPTSTimeUs; int32_t mMonitorQueueGeneration; @@ -120,10 +122,11 @@ private: status_t decryptBuffer( size_t playlistIndex, const sp &buffer); - void postMonitorQueue(int64_t delayUs = 0); + void postMonitorQueue(int64_t delayUs = 0, int64_t minDelayUs = 0); void cancelMonitorQueue(); - bool timeToRefreshPlaylist(int64_t nowUs) const; + int64_t delayUsToRefreshPlaylist() const; + status_t refreshPlaylist(); // Returns the media time in us of the segment specified by seqNumber. // This is computed by summing the durations of all segments before it. -- cgit v1.1 From e62d20268de87f63083896b5eef083d541078322 Mon Sep 17 00:00:00 2001 From: Lajos Molnar Date: Tue, 17 Dec 2013 14:10:46 -0800 Subject: stagefright: Fix issue with tracking media format in packet source Media format in AnotherPacketSource is now tracked across discontinuities. This fixes a bug where format was set on queueAccessUnit and cleared on dequeueAccessUnit, thereby allowing it to remain cleared. Change-Id: I20975a630443f4a223a2b4344e8244f34b9560b9 Signed-off-by: Lajos Molnar Bug: 12060952 --- media/libstagefright/httplive/PlaylistFetcher.cpp | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'media/libstagefright/httplive') diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp index 1754bf2..f095987 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.cpp +++ b/media/libstagefright/httplive/PlaylistFetcher.cpp @@ -861,12 +861,13 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( && source->dequeueAccessUnit(&accessUnit) == OK) { // Note that we do NOT dequeue any discontinuities. + // for simplicity, store a reference to the format in each unit + sp format = source->getFormat(); + if (format != NULL) { + accessUnit->meta()->setObject("format", format); + } packetSource->queueAccessUnit(accessUnit); } - - if (packetSource->getFormat() == NULL) { - packetSource->setFormat(source->getFormat()); - } } return OK; -- cgit v1.1 From f7e34cfcb9d649989c8e391f23dde782d4931849 Mon Sep 17 00:00:00 2001 From: Robert Shih Date: Thu, 23 Jan 2014 15:00:53 -0800 Subject: stagefright: use CODECS field in EXT-X_STREAM-INF as fallback Use the CODECS field in EXT-X-STREAM-INF as a viable fallback for identifying available streams in a variant playlist. Bug: 11854054 Change-Id: I5011809dc7c56220b023eb1a16f2d6a392ddd6f1 --- media/libstagefright/httplive/M3UParser.cpp | 160 ++++++++++++++++++++++++++-- media/libstagefright/httplive/M3UParser.h | 4 + 2 files changed, 157 insertions(+), 7 deletions(-) (limited to 'media/libstagefright/httplive') diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp index 243888c..ae19ffa 100644 --- a/media/libstagefright/httplive/M3UParser.cpp +++ b/media/libstagefright/httplive/M3UParser.cpp @@ -24,6 +24,7 @@ #include #include #include +#include #include namespace android { @@ -352,9 +353,27 @@ bool M3UParser::getTypeURI(size_t index, const char *key, AString *uri) const { if (!meta->findString(key, &groupID)) { *uri = mItems.itemAt(index).mURI; - // Assume media without any more specific attribute contains - // audio and video, but no subtitles. - return !strcmp("audio", key) || !strcmp("video", key); + AString codecs; + if (!meta->findString("codecs", &codecs)) { + // Assume media without any more specific attribute contains + // audio and video, but no subtitles. + return !strcmp("audio", key) || !strcmp("video", key); + } else { + // Split the comma separated list of codecs. + size_t offset = 0; + ssize_t commaPos = -1; + codecs.append(','); + while ((commaPos = codecs.find(",", offset)) >= 0) { + AString codec(codecs, offset, commaPos - offset); + // return true only if a codec of type `key` ("audio"/"video") + // is found. + if (codecIsType(codec, key)) { + return true; + } + offset = commaPos + 1; + } + return false; + } } sp group = mMediaGroups.valueFor(groupID); @@ -684,12 +703,22 @@ status_t M3UParser::parseStreamInf( *meta = new AMessage; } (*meta)->setInt32("bandwidth", x); + } else if (!strcasecmp("codecs", key.c_str())) { + if (!isQuotedString(val)) { + ALOGE("Expected quoted string for %s attribute, " + "got '%s' instead.", + key.c_str(), val.c_str());; + + return ERROR_MALFORMED; + } + + key.tolower(); + const AString &codecs = unquoteString(val); + (*meta)->setString(key.c_str(), codecs.c_str()); } else if (!strcasecmp("audio", key.c_str()) || !strcasecmp("video", key.c_str()) || !strcasecmp("subtitles", key.c_str())) { - if (val.size() < 2 - || val.c_str()[0] != '"' - || val.c_str()[val.size() - 1] != '"') { + if (!isQuotedString(val)) { ALOGE("Expected quoted string for %s attribute, " "got '%s' instead.", key.c_str(), val.c_str()); @@ -697,7 +726,7 @@ status_t M3UParser::parseStreamInf( return ERROR_MALFORMED; } - AString groupID(val, 1, val.size() - 2); + const AString &groupID = unquoteString(val); ssize_t groupIndex = mMediaGroups.indexOfKey(groupID); if (groupIndex < 0) { @@ -1086,4 +1115,121 @@ status_t M3UParser::ParseDouble(const char *s, double *x) { return OK; } +// static +bool M3UParser::isQuotedString(const AString &str) { + if (str.size() < 2 + || str.c_str()[0] != '"' + || str.c_str()[str.size() - 1] != '"') { + return false; + } + return true; +} + +// static +AString M3UParser::unquoteString(const AString &str) { + if (!isQuotedString(str)) { + return str; + } + return AString(str, 1, str.size() - 2); +} + +// static +bool M3UParser::codecIsType(const AString &codec, const char *type) { + if (codec.size() < 4) { + return false; + } + const char *c = codec.c_str(); + switch (FOURCC(c[0], c[1], c[2], c[3])) { + // List extracted from http://www.mp4ra.org/codecs.html + case 'ac-3': + case 'alac': + case 'dra1': + case 'dtsc': + case 'dtse': + case 'dtsh': + case 'dtsl': + case 'ec-3': + case 'enca': + case 'g719': + case 'g726': + case 'm4ae': + case 'mlpa': + case 'mp4a': + case 'raw ': + case 'samr': + case 'sawb': + case 'sawp': + case 'sevc': + case 'sqcp': + case 'ssmv': + case 'twos': + case 'agsm': + case 'alaw': + case 'dvi ': + case 'fl32': + case 'fl64': + case 'ima4': + case 'in24': + case 'in32': + case 'lpcm': + case 'Qclp': + case 'QDM2': + case 'QDMC': + case 'ulaw': + case 'vdva': + return !strcmp("audio", type); + + case 'avc1': + case 'avc2': + case 'avcp': + case 'drac': + case 'encv': + case 'mjp2': + case 'mp4v': + case 'mvc1': + case 'mvc2': + case 'resv': + case 's263': + case 'svc1': + case 'vc-1': + case 'CFHD': + case 'civd': + case 'DV10': + case 'dvh5': + case 'dvh6': + case 'dvhp': + case 'DVOO': + case 'DVOR': + case 'DVTV': + case 'DVVT': + case 'flic': + case 'gif ': + case 'h261': + case 'h263': + case 'HD10': + case 'jpeg': + case 'M105': + case 'mjpa': + case 'mjpb': + case 'png ': + case 'PNTG': + case 'rle ': + case 'rpza': + case 'Shr0': + case 'Shr1': + case 'Shr2': + case 'Shr3': + case 'Shr4': + case 'SVQ1': + case 'SVQ3': + case 'tga ': + case 'tiff': + case 'WRLE': + return !strcmp("video", type); + + default: + return false; + } +} + } // namespace android diff --git a/media/libstagefright/httplive/M3UParser.h b/media/libstagefright/httplive/M3UParser.h index 5248004..b93b0e5 100644 --- a/media/libstagefright/httplive/M3UParser.h +++ b/media/libstagefright/httplive/M3UParser.h @@ -100,6 +100,10 @@ private: static status_t ParseInt32(const char *s, int32_t *x); static status_t ParseDouble(const char *s, double *x); + static bool isQuotedString(const AString &str); + static AString unquoteString(const AString &str); + static bool codecIsType(const AString &codec, const char *type); + DISALLOW_EVIL_CONSTRUCTORS(M3UParser); }; -- cgit v1.1 From 822a489e595336be447f47f5c2a051e8fdd1cdff Mon Sep 17 00:00:00 2001 From: Robert Shih Date: Thu, 6 Feb 2014 14:01:30 -0800 Subject: LiveSession: Add support for block-by-block fetchFile. Bug: 11854054 Change-Id: I4025ba7fab8fab2e0c720f73894e908fd98a43d8 --- media/libstagefright/httplive/LiveSession.cpp | 97 +++++++++++++++++---------- media/libstagefright/httplive/LiveSession.h | 18 ++++- 2 files changed, 80 insertions(+), 35 deletions(-) (limited to 'media/libstagefright/httplive') diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp index bd12ddc..0333a2a 100644 --- a/media/libstagefright/httplive/LiveSession.cpp +++ b/media/libstagefright/httplive/LiveSession.cpp @@ -512,53 +512,80 @@ sp LiveSession::addFetcher(const char *uri) { return info.mFetcher; } +/* + * Illustration of parameters: + * + * 0 `range_offset` + * +------------+-------------------------------------------------------+--+--+ + * | | | next block to fetch | | | + * | | `source` handle => `out` buffer | | | | + * | `url` file |<--------- buffer size --------->|<--- `block_size` -->| | | + * | |<----------- `range_length` / buffer capacity ----------->| | + * |<------------------------------ file_size ------------------------------->| + * + * Special parameter values: + * - range_length == -1 means entire file + * - block_size == 0 means entire range + * + */ status_t LiveSession::fetchFile( const char *url, sp *out, - int64_t range_offset, int64_t range_length) { - *out = NULL; + int64_t range_offset, int64_t range_length, + uint32_t block_size, /* download block size */ + sp *source /* to return and reuse source */) { + off64_t size; + sp temp_source; + if (source == NULL) { + source = &temp_source; + } - sp source; + if (*source == NULL) { + if (!strncasecmp(url, "file://", 7)) { + *source = new FileSource(url + 7); + } else if (strncasecmp(url, "http://", 7) + && strncasecmp(url, "https://", 8)) { + return ERROR_UNSUPPORTED; + } else { + KeyedVector headers = mExtraHeaders; + if (range_offset > 0 || range_length >= 0) { + headers.add( + String8("Range"), + String8( + StringPrintf( + "bytes=%lld-%s", + range_offset, + range_length < 0 + ? "" : StringPrintf("%lld", + range_offset + range_length - 1).c_str()).c_str())); + } + status_t err = mHTTPDataSource->connect(url, &headers); - if (!strncasecmp(url, "file://", 7)) { - source = new FileSource(url + 7); - } else if (strncasecmp(url, "http://", 7) - && strncasecmp(url, "https://", 8)) { - return ERROR_UNSUPPORTED; - } else { - KeyedVector headers = mExtraHeaders; - if (range_offset > 0 || range_length >= 0) { - headers.add( - String8("Range"), - String8( - StringPrintf( - "bytes=%lld-%s", - range_offset, - range_length < 0 - ? "" : StringPrintf("%lld", range_offset + range_length - 1).c_str()).c_str())); - } - status_t err = mHTTPDataSource->connect(url, &headers); + if (err != OK) { + return err; + } - if (err != OK) { - return err; + *source = mHTTPDataSource; } - - source = mHTTPDataSource; } - off64_t size; - status_t err = source->getSize(&size); - - if (err != OK) { + status_t getSizeErr = (*source)->getSize(&size); + if (getSizeErr != OK) { size = 65536; } - sp buffer = new ABuffer(size); - buffer->setRange(0, 0); + sp buffer = *out != NULL ? *out : new ABuffer(size); + if (*out == NULL) { + buffer->setRange(0, 0); + } + // adjust range_length if only reading partial block + if (block_size > 0 && (range_length == -1 || buffer->size() + block_size < range_length)) { + range_length = buffer->size() + block_size; + } for (;;) { + // Only resize when we don't know the size. size_t bufferRemaining = buffer->capacity() - buffer->size(); - - if (bufferRemaining == 0) { + if (bufferRemaining == 0 && getSizeErr != OK) { bufferRemaining = 32768; ALOGV("increasing download buffer to %d bytes", @@ -583,7 +610,9 @@ status_t LiveSession::fetchFile( } } - ssize_t n = source->readAt( + // The DataSource is responsible for informing us of error (n < 0) or eof (n == 0) + // to help us break out of the loop. + ssize_t n = (*source)->readAt( buffer->size(), buffer->data() + buffer->size(), maxBytesToRead); diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h index 99b480a8..f42d7ae 100644 --- a/media/libstagefright/httplive/LiveSession.h +++ b/media/libstagefright/httplive/LiveSession.h @@ -145,9 +145,25 @@ private: status_t onSeek(const sp &msg); void onFinishDisconnect2(); + // If given a non-zero block_size (default 0), it is used to cap the number of + // bytes read in from the DataSource. If given a non-NULL buffer, new content + // is read into the end. + // + // The DataSource we read from is responsible for signaling error or EOF to help us + // break out of the read loop. The DataSource can be returned to the caller, so + // that the caller can reuse it for subsequent fetches (within the initially + // requested range). + // + // For reused HTTP sources, the caller must download a file sequentially without + // any overlaps or gaps to prevent reconnection. status_t fetchFile( const char *url, sp *out, - int64_t range_offset = 0, int64_t range_length = -1); + /* request/open a file starting at range_offset for range_length bytes */ + int64_t range_offset = 0, int64_t range_length = -1, + /* download block size */ + uint32_t block_size = 0, + /* reuse DataSource if doing partial fetch */ + sp *source = NULL); sp fetchPlaylist( const char *url, uint8_t *curPlaylistHash, bool *unchanged); -- cgit v1.1 From 2cd94583868b775a548233a4f7cd1d988fc6344f Mon Sep 17 00:00:00 2001 From: Robert Shih Date: Thu, 6 Feb 2014 14:25:25 -0800 Subject: PlaylistFetcher: Add support for block-by-block decryption. Bug: 11854054 Change-Id: Ifd3f3369275889e716b360087b5b60d01635b578 --- media/libstagefright/httplive/PlaylistFetcher.cpp | 95 ++++++++++++++--------- media/libstagefright/httplive/PlaylistFetcher.h | 17 +++- 2 files changed, 75 insertions(+), 37 deletions(-) (limited to 'media/libstagefright/httplive') diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp index f095987..9cb16fe 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.cpp +++ b/media/libstagefright/httplive/PlaylistFetcher.cpp @@ -170,7 +170,8 @@ int64_t PlaylistFetcher::delayUsToRefreshPlaylist() const { } status_t PlaylistFetcher::decryptBuffer( - size_t playlistIndex, const sp &buffer) { + size_t playlistIndex, const sp &buffer, + bool first) { sp itemMeta; bool found = false; AString method; @@ -188,6 +189,7 @@ status_t PlaylistFetcher::decryptBuffer( if (!found) { method = "NONE"; } + buffer->meta()->setString("cipher-method", method.c_str()); if (method == "NONE") { return OK; @@ -227,59 +229,77 @@ status_t PlaylistFetcher::decryptBuffer( return UNKNOWN_ERROR; } - unsigned char aes_ivec[16]; + size_t n = buffer->size(); + if (!n) { + return OK; + } + CHECK(n % 16 == 0); - 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; - } + 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. - memset(aes_ivec, 0, sizeof(aes_ivec)); - 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)) { + 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; } - uint8_t nibble1 = isdigit(c1) ? c1 - '0' : c1 - 'a' + 10; - uint8_t nibble2 = isdigit(c2) ? c2 - '0' : c2 - 'a' + 10; - aes_ivec[i] = nibble1 << 4 | nibble2; + 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; } - } else { - memset(aes_ivec, 0, sizeof(aes_ivec)); - aes_ivec[15] = mSeqNumber & 0xff; - aes_ivec[14] = (mSeqNumber >> 8) & 0xff; - aes_ivec[13] = (mSeqNumber >> 16) & 0xff; - aes_ivec[12] = (mSeqNumber >> 24) & 0xff; } AES_cbc_encrypt( buffer->data(), buffer->data(), buffer->size(), - &aes_key, aes_ivec, AES_DECRYPT); - - // hexdump(buffer->data(), buffer->size()); + &aes_key, mAESInitVec, AES_DECRYPT); - size_t n = buffer->size(); - CHECK_GT(n, 0u); + return OK; +} - size_t pad = buffer->data()[n - 1]; +status_t PlaylistFetcher::checkDecryptPadding(const sp &buffer) { + status_t err; + AString method; + CHECK(buffer->meta()->findString("cipher-method", &method)); + if (method == "NONE") { + return OK; + } - CHECK_GT(pad, 0u); - CHECK_LE(pad, 16u); - CHECK_GE((size_t)n, pad); - for (size_t i = 0; i < pad; ++i) { - CHECK_EQ((unsigned)buffer->data()[n - 1 - i], pad); + uint8_t padding = 0; + if (buffer->size() > 0) { + padding = buffer->data()[buffer->size() - 1]; } - n -= pad; + if (padding > 16) { + return ERROR_MALFORMED; + } - buffer->setRange(buffer->offset(), n); + 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; } @@ -706,6 +726,9 @@ void PlaylistFetcher::onDownloadNext() { CHECK(buffer != NULL); err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, buffer); + if (err == OK) { + err = checkDecryptPadding(buffer); + } if (err != OK) { ALOGE("decryptBuffer failed w/ error %d", err); diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h index 78dea20..ac04a77 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.h +++ b/media/libstagefright/httplive/PlaylistFetcher.h @@ -119,8 +119,23 @@ private: uint64_t mFirstPTS; int64_t mAbsoluteTimeAnchorUs; + // Stores the initialization vector to decrypt the next block of cipher text, which can + // either be derived from the sequence number, read from the manifest, or copied from + // the last block of cipher text (cipher-block chaining). + unsigned char mAESInitVec[16]; + + // Set first to true if decrypting the first segment of a playlist segment. When + // first is true, reset the initialization vector based on the available + // information in the manifest; otherwise, use the initialization vector as + // updated by the last call to AES_cbc_encrypt. + // + // For the input to decrypt correctly, decryptBuffer must be called on + // consecutive byte ranges on block boundaries, e.g. 0..15, 16..47, 48..63, + // and so on. status_t decryptBuffer( - size_t playlistIndex, const sp &buffer); + size_t playlistIndex, const sp &buffer, + bool first = true); + status_t checkDecryptPadding(const sp &buffer); void postMonitorQueue(int64_t delayUs = 0, int64_t minDelayUs = 0); void cancelMonitorQueue(); -- cgit v1.1 From 6c8495c8f1ccc35db972ee7ac0dbb8baf5843548 Mon Sep 17 00:00:00 2001 From: Robert Shih Date: Fri, 14 Feb 2014 12:21:50 -0800 Subject: LiveSession refactor Bug: 11854054 Change-Id: Ia91dce109835e042f72934376d4838b4cc72cb10 --- media/libstagefright/httplive/LiveSession.cpp | 186 ++++++++++---------------- media/libstagefright/httplive/LiveSession.h | 28 +++- media/libstagefright/httplive/M3UParser.cpp | 12 -- media/libstagefright/httplive/M3UParser.h | 6 +- 4 files changed, 92 insertions(+), 140 deletions(-) (limited to 'media/libstagefright/httplive') diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp index 0333a2a..79ad100 100644 --- a/media/libstagefright/httplive/LiveSession.cpp +++ b/media/libstagefright/httplive/LiveSession.cpp @@ -66,14 +66,13 @@ LiveSession::LiveSession( mHTTPDataSource->setUID(mUID); } - mPacketSources.add( - STREAMTYPE_AUDIO, new AnotherPacketSource(NULL /* meta */)); + mStreams[kAudioIndex] = StreamItem("audio"); + mStreams[kVideoIndex] = StreamItem("video"); + mStreams[kSubtitleIndex] = StreamItem("subtitle"); - mPacketSources.add( - STREAMTYPE_VIDEO, new AnotherPacketSource(NULL /* meta */)); - - mPacketSources.add( - STREAMTYPE_SUBTITLES, new AnotherPacketSource(NULL /* meta */)); + for (size_t i = 0; i < kMaxStreams; ++i) { + mPacketSources.add(indexToType(i), new AnotherPacketSource(NULL /* meta */)); + } } LiveSession::~LiveSession() { @@ -374,6 +373,12 @@ int LiveSession::SortByBandwidth(const BandwidthItem *a, const BandwidthItem *b) return 1; } +// static +LiveSession::StreamType LiveSession::indexToType(int idx) { + CHECK(idx >= 0 && idx < kMaxStreams); + return (StreamType)(1 << idx); +} + void LiveSession::onConnect(const sp &msg) { AString url; CHECK(msg->findString("url", &url)); @@ -858,19 +863,11 @@ void LiveSession::changeConfiguration( uint32_t streamMask = 0; - AString audioURI; - if (mPlaylist->getAudioURI(item.mPlaylistIndex, &audioURI)) { - streamMask |= STREAMTYPE_AUDIO; - } - - AString videoURI; - if (mPlaylist->getVideoURI(item.mPlaylistIndex, &videoURI)) { - streamMask |= STREAMTYPE_VIDEO; - } - - AString subtitleURI; - if (mPlaylist->getSubtitleURI(item.mPlaylistIndex, &subtitleURI)) { - streamMask |= STREAMTYPE_SUBTITLES; + AString URIs[kMaxStreams]; + for (size_t i = 0; i < kMaxStreams; ++i) { + if (mPlaylist->getTypeURI(item.mPlaylistIndex, mStreams[i].mType, &URIs[i])) { + streamMask |= indexToType(i); + } } // Step 1, stop and discard fetchers that are no longer needed. @@ -882,10 +879,10 @@ void LiveSession::changeConfiguration( // If we're seeking all current fetchers are discarded. if (timeUs < 0ll) { - if (((streamMask & STREAMTYPE_AUDIO) && uri == audioURI) - || ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI) - || ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI)) { - discardFetcher = false; + for (size_t j = 0; j < kMaxStreams; ++j) { + if ((streamMask & indexToType(j)) && uri == URIs[j]) { + discardFetcher = false; + } } } @@ -899,14 +896,10 @@ void LiveSession::changeConfiguration( sp msg = new AMessage(kWhatChangeConfiguration2, id()); msg->setInt32("streamMask", streamMask); msg->setInt64("timeUs", timeUs); - if (streamMask & STREAMTYPE_AUDIO) { - msg->setString("audioURI", audioURI.c_str()); - } - if (streamMask & STREAMTYPE_VIDEO) { - msg->setString("videoURI", videoURI.c_str()); - } - if (streamMask & STREAMTYPE_SUBTITLES) { - msg->setString("subtitleURI", subtitleURI.c_str()); + for (size_t i = 0; i < kMaxStreams; ++i) { + if (streamMask & indexToType(i)) { + msg->setString(mStreams[i].uriKey().c_str(), URIs[i].c_str()); + } } // Every time a fetcher acknowledges the stopAsync or pauseAsync request @@ -937,18 +930,13 @@ void LiveSession::onChangeConfiguration2(const sp &msg) { uint32_t streamMask; CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask)); - AString audioURI, videoURI, subtitleURI; - if (streamMask & STREAMTYPE_AUDIO) { - CHECK(msg->findString("audioURI", &audioURI)); - ALOGV("audioURI = '%s'", audioURI.c_str()); - } - if (streamMask & STREAMTYPE_VIDEO) { - CHECK(msg->findString("videoURI", &videoURI)); - ALOGV("videoURI = '%s'", videoURI.c_str()); - } - if (streamMask & STREAMTYPE_SUBTITLES) { - CHECK(msg->findString("subtitleURI", &subtitleURI)); - ALOGV("subtitleURI = '%s'", subtitleURI.c_str()); + AString URIs[kMaxStreams]; + for (size_t i = 0; i < kMaxStreams; ++i) { + if (streamMask & indexToType(i)) { + const AString &uriKey = mStreams[i].uriKey(); + CHECK(msg->findString(uriKey.c_str(), &URIs[i])); + ALOGV("%s = '%s'", uriKey.c_str(), URIs[i].c_str()); + } } // Determine which decoders to shutdown on the player side, @@ -958,15 +946,12 @@ void LiveSession::onChangeConfiguration2(const sp &msg) { // 2) its streamtype was already active and still is but the URI // has changed. uint32_t changedMask = 0; - if (((mStreamMask & streamMask & STREAMTYPE_AUDIO) - && !(audioURI == mAudioURI)) - || (mStreamMask & ~streamMask & STREAMTYPE_AUDIO)) { - changedMask |= STREAMTYPE_AUDIO; - } - if (((mStreamMask & streamMask & STREAMTYPE_VIDEO) - && !(videoURI == mVideoURI)) - || (mStreamMask & ~streamMask & STREAMTYPE_VIDEO)) { - changedMask |= STREAMTYPE_VIDEO; + for (size_t i = 0; i < kMaxStreams && i != kSubtitleIndex; ++i) { + if (((mStreamMask & streamMask & indexToType(i)) + && !(URIs[i] == mStreams[i].mUri)) + || (mStreamMask & ~streamMask & indexToType(i))) { + changedMask |= indexToType(i); + } } if (changedMask == 0) { @@ -998,15 +983,10 @@ void LiveSession::onChangeConfiguration3(const sp &msg) { uint32_t streamMask; CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask)); - AString audioURI, videoURI, subtitleURI; - if (streamMask & STREAMTYPE_AUDIO) { - CHECK(msg->findString("audioURI", &audioURI)); - } - if (streamMask & STREAMTYPE_VIDEO) { - CHECK(msg->findString("videoURI", &videoURI)); - } - if (streamMask & STREAMTYPE_SUBTITLES) { - CHECK(msg->findString("subtitleURI", &subtitleURI)); + for (size_t i = 0; i < kMaxStreams; ++i) { + if (streamMask & indexToType(i)) { + CHECK(msg->findString(mStreams[i].uriKey().c_str(), &mStreams[i].mUri)); + } } int64_t timeUs; @@ -1018,9 +998,6 @@ void LiveSession::onChangeConfiguration3(const sp &msg) { mRealTimeBaseUs = ALooper::GetNowUs() - timeUs; mStreamMask = streamMask; - mAudioURI = audioURI; - mVideoURI = videoURI; - mSubtitleURI = subtitleURI; // Resume all existing fetchers and assign them packet sources. for (size_t i = 0; i < mFetcherInfos.size(); ++i) { @@ -1028,22 +1005,12 @@ void LiveSession::onChangeConfiguration3(const sp &msg) { uint32_t resumeMask = 0; - sp audioSource; - if ((streamMask & STREAMTYPE_AUDIO) && uri == audioURI) { - audioSource = mPacketSources.valueFor(STREAMTYPE_AUDIO); - resumeMask |= STREAMTYPE_AUDIO; - } - - sp videoSource; - if ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI) { - videoSource = mPacketSources.valueFor(STREAMTYPE_VIDEO); - resumeMask |= STREAMTYPE_VIDEO; - } - - sp subtitleSource; - if ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI) { - subtitleSource = mPacketSources.valueFor(STREAMTYPE_SUBTITLES); - resumeMask |= STREAMTYPE_SUBTITLES; + sp sources[kMaxStreams]; + for (size_t j = 0; j < kMaxStreams; ++j) { + if ((streamMask & indexToType(j)) && uri == mStreams[j].mUri) { + sources[j] = mPacketSources.valueFor(indexToType(j)); + resumeMask |= indexToType(j); + } } CHECK_NE(resumeMask, 0u); @@ -1053,7 +1020,7 @@ void LiveSession::onChangeConfiguration3(const sp &msg) { streamMask &= ~resumeMask; mFetcherInfos.valueAt(i).mFetcher->startAsync( - audioSource, videoSource, subtitleSource); + sources[kAudioIndex], sources[kVideoIndex], sources[kSubtitleIndex]); } // streamMask now only contains the types that need a new fetcher created. @@ -1062,52 +1029,33 @@ void LiveSession::onChangeConfiguration3(const sp &msg) { ALOGV("creating new fetchers for mask 0x%08x", streamMask); } - while (streamMask != 0) { - StreamType streamType = (StreamType)(streamMask & ~(streamMask - 1)); + for (size_t i = 0; i < kMaxStreams; i++) { + if (!(indexToType(i) & streamMask)) { + continue; + } AString uri; - switch (streamType) { - case STREAMTYPE_AUDIO: - uri = audioURI; - break; - case STREAMTYPE_VIDEO: - uri = videoURI; - break; - case STREAMTYPE_SUBTITLES: - uri = subtitleURI; - break; - default: - TRESPASS(); - } + uri = mStreams[i].mUri; sp fetcher = addFetcher(uri.c_str()); CHECK(fetcher != NULL); - sp audioSource; - if ((streamMask & STREAMTYPE_AUDIO) && uri == audioURI) { - audioSource = mPacketSources.valueFor(STREAMTYPE_AUDIO); - audioSource->clear(); - - streamMask &= ~STREAMTYPE_AUDIO; - } - - sp videoSource; - if ((streamMask & STREAMTYPE_VIDEO) && uri == videoURI) { - videoSource = mPacketSources.valueFor(STREAMTYPE_VIDEO); - videoSource->clear(); + sp sources[kMaxStreams]; + // TRICKY: looping from i as earlier streams are already removed from streamMask + for (size_t j = i; j < kMaxStreams; ++j) { + if ((streamMask & indexToType(j)) && uri == mStreams[j].mUri) { + sources[j] = mPacketSources.valueFor(indexToType(j)); + sources[j]->clear(); - streamMask &= ~STREAMTYPE_VIDEO; - } - - sp subtitleSource; - if ((streamMask & STREAMTYPE_SUBTITLES) && uri == subtitleURI) { - subtitleSource = mPacketSources.valueFor(STREAMTYPE_SUBTITLES); - subtitleSource->clear(); - - streamMask &= ~STREAMTYPE_SUBTITLES; + streamMask &= ~indexToType(j); + } } - fetcher->startAsync(audioSource, videoSource, subtitleSource, timeUs); + fetcher->startAsync( + sources[kAudioIndex], + sources[kVideoIndex], + sources[kSubtitleIndex], + timeUs); } // All fetchers have now been started, the configuration change diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h index f42d7ae..b38c518 100644 --- a/media/libstagefright/httplive/LiveSession.h +++ b/media/libstagefright/httplive/LiveSession.h @@ -42,10 +42,17 @@ struct LiveSession : public AHandler { const sp ¬ify, uint32_t flags = 0, bool uidValid = false, uid_t uid = 0); + enum StreamIndex { + kAudioIndex = 0, + kVideoIndex = 1, + kSubtitleIndex = 2, + kMaxStreams = 3, + }; + enum StreamType { - STREAMTYPE_AUDIO = 1, - STREAMTYPE_VIDEO = 2, - STREAMTYPE_SUBTITLES = 4, + STREAMTYPE_AUDIO = 1 << kAudioIndex, + STREAMTYPE_VIDEO = 1 << kVideoIndex, + STREAMTYPE_SUBTITLES = 1 << kSubtitleIndex, }; status_t dequeueAccessUnit(StreamType stream, sp *accessUnit); @@ -105,6 +112,19 @@ private: bool mIsPrepared; }; + struct StreamItem { + const char *mType; + AString mUri; + StreamItem() : mType("") {} + StreamItem(const char *type) : mType(type) {} + AString uriKey() { + AString key(mType); + key.append("URI"); + return key; + } + }; + StreamItem mStreams[kMaxStreams]; + sp mNotify; uint32_t mFlags; bool mUIDValid; @@ -123,7 +143,6 @@ private: sp mPlaylist; KeyedVector mFetcherInfos; - AString mAudioURI, mVideoURI, mSubtitleURI; uint32_t mStreamMask; KeyedVector > mPacketSources; @@ -171,6 +190,7 @@ private: size_t getBandwidthIndex(); static int SortByBandwidth(const BandwidthItem *, const BandwidthItem *); + static StreamType indexToType(int idx); void changeConfiguration( int64_t timeUs, size_t bandwidthIndex, bool pickTrack = false); diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp index ae19ffa..a9184b5 100644 --- a/media/libstagefright/httplive/M3UParser.cpp +++ b/media/libstagefright/httplive/M3UParser.cpp @@ -388,18 +388,6 @@ bool M3UParser::getTypeURI(size_t index, const char *key, AString *uri) const { return true; } -bool M3UParser::getAudioURI(size_t index, AString *uri) const { - return getTypeURI(index, "audio", uri); -} - -bool M3UParser::getVideoURI(size_t index, AString *uri) const { - return getTypeURI(index, "video", uri); -} - -bool M3UParser::getSubtitleURI(size_t index, AString *uri) const { - return getTypeURI(index, "subtitles", uri); -} - static bool MakeURL(const char *baseURL, const char *url, AString *out) { out->clear(); diff --git a/media/libstagefright/httplive/M3UParser.h b/media/libstagefright/httplive/M3UParser.h index b93b0e5..ccd6556 100644 --- a/media/libstagefright/httplive/M3UParser.h +++ b/media/libstagefright/httplive/M3UParser.h @@ -45,9 +45,7 @@ struct M3UParser : public RefBase { status_t getTrackInfo(Parcel* reply) const; ssize_t getSelectedIndex() const; - bool getAudioURI(size_t index, AString *uri) const; - bool getVideoURI(size_t index, AString *uri) const; - bool getSubtitleURI(size_t index, AString *uri) const; + bool getTypeURI(size_t index, const char *key, AString *uri) const; protected: virtual ~M3UParser(); @@ -95,8 +93,6 @@ private: status_t parseMedia(const AString &line); - bool getTypeURI(size_t index, const char *key, AString *uri) const; - static status_t ParseInt32(const char *s, int32_t *x); static status_t ParseDouble(const char *s, double *x); -- cgit v1.1 From 777ee5ed736c8f6c3f7d196ea022f7432bfd23e1 Mon Sep 17 00:00:00 2001 From: Robert Shih Date: Thu, 20 Feb 2014 13:07:26 -0800 Subject: Initial HLS seamless switch implementation. Bug: 11854054 Change-Id: I75fc2a258111295039ac13cc37e407df25891dd2 --- media/libstagefright/httplive/LiveSession.cpp | 280 +++++++++++++++++++--- media/libstagefright/httplive/LiveSession.h | 36 +++ media/libstagefright/httplive/PlaylistFetcher.cpp | 251 +++++++++++++++++-- media/libstagefright/httplive/PlaylistFetcher.h | 22 +- 4 files changed, 541 insertions(+), 48 deletions(-) (limited to 'media/libstagefright/httplive') diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp index 79ad100..9e30ebd 100644 --- a/media/libstagefright/httplive/LiveSession.cpp +++ b/media/libstagefright/httplive/LiveSession.cpp @@ -37,6 +37,8 @@ #include #include +#include + #include #include #include @@ -57,10 +59,14 @@ LiveSession::LiveSession( : 0)), mPrevBandwidthIndex(-1), mStreamMask(0), + mNewStreamMask(0), + mSwapMask(0), mCheckBandwidthGeneration(0), + mSwitchGeneration(0), mLastDequeuedTimeUs(0ll), mRealTimeBaseUs(0ll), mReconfigurationInProgress(false), + mSwitchInProgress(false), mDisconnectReplyID(0) { if (mUIDValid) { mHTTPDataSource->setUID(mUID); @@ -72,16 +78,37 @@ LiveSession::LiveSession( for (size_t i = 0; i < kMaxStreams; ++i) { mPacketSources.add(indexToType(i), new AnotherPacketSource(NULL /* meta */)); + mPacketSources2.add(indexToType(i), new AnotherPacketSource(NULL /* meta */)); } } LiveSession::~LiveSession() { } +sp LiveSession::createFormatChangeBuffer(bool swap) { + ABuffer *discontinuity = new ABuffer(0); + discontinuity->meta()->setInt32("discontinuity", ATSParser::DISCONTINUITY_FORMATCHANGE); + discontinuity->meta()->setInt32("swapPacketSource", swap); + discontinuity->meta()->setInt32("switchGeneration", mSwitchGeneration); + discontinuity->meta()->setInt64("timeUs", -1); + return discontinuity; +} + +void LiveSession::swapPacketSource(StreamType stream) { + sp &aps = mPacketSources.editValueFor(stream); + sp &aps2 = mPacketSources2.editValueFor(stream); + sp tmp = aps; + aps = aps2; + aps2 = tmp; + aps2->clear(); +} + status_t LiveSession::dequeueAccessUnit( StreamType stream, sp *accessUnit) { if (!(mStreamMask & stream)) { - return UNKNOWN_ERROR; + // return -EWOULDBLOCK to avoid halting the decoder + // when switching between audio/video and audio only. + return -EWOULDBLOCK; } sp packetSource = mPacketSources.valueFor(stream); @@ -121,6 +148,25 @@ status_t LiveSession::dequeueAccessUnit( streamStr, type, extra == NULL ? "NULL" : extra->debugString().c_str()); + + int32_t swap; + if (type == ATSParser::DISCONTINUITY_FORMATCHANGE + && (*accessUnit)->meta()->findInt32("swapPacketSource", &swap) + && swap) { + + int32_t switchGeneration; + CHECK((*accessUnit)->meta()->findInt32("switchGeneration", &switchGeneration)); + { + Mutex::Autolock lock(mSwapMutex); + if (switchGeneration == mSwitchGeneration) { + swapPacketSource(stream); + sp msg = new AMessage(kWhatSwapped, id()); + msg->setInt32("stream", stream); + msg->setInt32("switchGeneration", switchGeneration); + msg->post(); + } + } + } } else if (err == OK) { if (stream == STREAMTYPE_AUDIO || stream == STREAMTYPE_VIDEO) { int64_t timeUs; @@ -142,6 +188,7 @@ status_t LiveSession::dequeueAccessUnit( } status_t LiveSession::getStreamFormat(StreamType stream, sp *format) { + // No swapPacketSource race condition; called from the same thread as dequeueAccessUnit. if (!(mStreamMask & stream)) { return UNKNOWN_ERROR; } @@ -238,7 +285,12 @@ void LiveSession::onMessageReceived(const sp &msg) { if (what == PlaylistFetcher::kWhatStopped) { AString uri; CHECK(msg->findString("uri", &uri)); - mFetcherInfos.removeItem(uri); + if (mFetcherInfos.removeItem(uri) < 0) { + // ignore duplicated kWhatStopped messages. + break; + } + + tryToFinishBandwidthSwitch(); } if (mContinuation != NULL) { @@ -274,6 +326,8 @@ void LiveSession::onMessageReceived(const sp &msg) { postPrepared(err); } + cancelBandwidthSwitch(); + mPacketSources.valueFor(STREAMTYPE_AUDIO)->signalEOS(err); mPacketSources.valueFor(STREAMTYPE_VIDEO)->signalEOS(err); @@ -312,6 +366,27 @@ void LiveSession::onMessageReceived(const sp &msg) { break; } + case PlaylistFetcher::kWhatStartedAt: + { + int32_t switchGeneration; + CHECK(msg->findInt32("switchGeneration", &switchGeneration)); + + if (switchGeneration != mSwitchGeneration) { + break; + } + + // Resume fetcher for the original variant; the resumed fetcher should + // continue until the timestamps found in msg, which is stored by the + // new fetcher to indicate where the new variant has started buffering. + for (size_t i = 0; i < mFetcherInfos.size(); i++) { + const FetcherInfo info = mFetcherInfos.valueAt(i); + if (info.mToBeRemoved) { + info.mFetcher->resumeUntilAsync(msg); + } + } + break; + } + default: TRESPASS(); } @@ -356,6 +431,11 @@ void LiveSession::onMessageReceived(const sp &msg) { break; } + case kWhatSwapped: + { + onSwapped(msg); + break; + } default: TRESPASS(); break; @@ -466,6 +546,10 @@ void LiveSession::finishDisconnect() { // during disconnection either. cancelCheckBandwidthEvent(); + // Protect mPacketSources from a swapPacketSource race condition through disconnect. + // (finishDisconnect, onFinishDisconnect2) + cancelBandwidthSwitch(); + for (size_t i = 0; i < mFetcherInfos.size(); ++i) { mFetcherInfos.valueAt(i).mFetcher->stopAsync(); } @@ -505,11 +589,13 @@ sp LiveSession::addFetcher(const char *uri) { sp notify = new AMessage(kWhatFetcherNotify, id()); notify->setString("uri", uri); + notify->setInt32("switchGeneration", mSwitchGeneration); FetcherInfo info; info.mFetcher = new PlaylistFetcher(notify, this, uri); info.mDurationUs = -1ll; info.mIsPrepared = false; + info.mToBeRemoved = false; looper()->registerHandler(info.mFetcher); mFetcherInfos.add(uri, info); @@ -844,8 +930,25 @@ status_t LiveSession::selectTrack(size_t index, bool select) { return err; } +bool LiveSession::canSwitchUp() { + // Allow upwards bandwidth switch when a stream has buffered at least 10 seconds. + status_t err = OK; + for (size_t i = 0; i < mPacketSources.size(); ++i) { + sp source = mPacketSources.valueAt(i); + int64_t dur = source->getBufferedDurationUs(&err); + if (err == OK && dur > 10000000) { + return true; + } + } + return false; +} + void LiveSession::changeConfiguration( int64_t timeUs, size_t bandwidthIndex, bool pickTrack) { + // Protect mPacketSources from a swapPacketSource race condition through reconfiguration. + // (changeConfiguration, onChangeConfiguration2, onChangeConfiguration3). + cancelBandwidthSwitch(); + CHECK(!mReconfigurationInProgress); mReconfigurationInProgress = true; @@ -861,7 +964,8 @@ void LiveSession::changeConfiguration( CHECK_LT(bandwidthIndex, mBandwidthItems.size()); const BandwidthItem &item = mBandwidthItems.itemAt(bandwidthIndex); - uint32_t streamMask = 0; + uint32_t streamMask = 0; // streams that should be fetched by the new fetcher + uint32_t resumeMask = 0; // streams that should be fetched by the original fetcher AString URIs[kMaxStreams]; for (size_t i = 0; i < kMaxStreams; ++i) { @@ -879,9 +983,14 @@ void LiveSession::changeConfiguration( // If we're seeking all current fetchers are discarded. if (timeUs < 0ll) { + // delay fetcher removal + discardFetcher = false; + for (size_t j = 0; j < kMaxStreams; ++j) { - if ((streamMask & indexToType(j)) && uri == URIs[j]) { - discardFetcher = false; + StreamType type = indexToType(j); + if ((streamMask & type) && uri == URIs[j]) { + resumeMask |= type; + streamMask &= ~type; } } } @@ -893,8 +1002,15 @@ void LiveSession::changeConfiguration( } } - sp msg = new AMessage(kWhatChangeConfiguration2, id()); + sp msg; + if (timeUs < 0ll) { + // skip onChangeConfiguration2 (decoder destruction) if switching. + msg = new AMessage(kWhatChangeConfiguration3, id()); + } else { + msg = new AMessage(kWhatChangeConfiguration2, id()); + } msg->setInt32("streamMask", streamMask); + msg->setInt32("resumeMask", resumeMask); msg->setInt64("timeUs", timeUs); for (size_t i = 0; i < kMaxStreams; ++i) { if (streamMask & indexToType(i)) { @@ -977,11 +1093,13 @@ void LiveSession::onChangeConfiguration2(const sp &msg) { } void LiveSession::onChangeConfiguration3(const sp &msg) { + mContinuation.clear(); // All remaining fetchers are still suspended, the player has shutdown // any decoders that needed it. - uint32_t streamMask; + uint32_t streamMask, resumeMask; CHECK(msg->findInt32("streamMask", (int32_t *)&streamMask)); + CHECK(msg->findInt32("resumeMask", (int32_t *)&resumeMask)); for (size_t i = 0; i < kMaxStreams; ++i) { if (streamMask & indexToType(i)) { @@ -990,37 +1108,39 @@ void LiveSession::onChangeConfiguration3(const sp &msg) { } int64_t timeUs; + bool switching = false; CHECK(msg->findInt64("timeUs", &timeUs)); if (timeUs < 0ll) { timeUs = mLastDequeuedTimeUs; + switching = true; } mRealTimeBaseUs = ALooper::GetNowUs() - timeUs; - mStreamMask = streamMask; + mNewStreamMask = streamMask; - // Resume all existing fetchers and assign them packet sources. + // Of all existing fetchers: + // * Resume fetchers that are still needed and assign them original packet sources. + // * Mark otherwise unneeded fetchers for removal. + ALOGV("resuming fetchers for mask 0x%08x", resumeMask); for (size_t i = 0; i < mFetcherInfos.size(); ++i) { const AString &uri = mFetcherInfos.keyAt(i); - uint32_t resumeMask = 0; - sp sources[kMaxStreams]; for (size_t j = 0; j < kMaxStreams; ++j) { - if ((streamMask & indexToType(j)) && uri == mStreams[j].mUri) { + if ((resumeMask & indexToType(j)) && uri == mStreams[j].mUri) { sources[j] = mPacketSources.valueFor(indexToType(j)); - resumeMask |= indexToType(j); } } - CHECK_NE(resumeMask, 0u); - - ALOGV("resuming fetchers for mask 0x%08x", resumeMask); - - streamMask &= ~resumeMask; - - mFetcherInfos.valueAt(i).mFetcher->startAsync( - sources[kAudioIndex], sources[kVideoIndex], sources[kSubtitleIndex]); + FetcherInfo &info = mFetcherInfos.editValueAt(i); + if (sources[kAudioIndex] != NULL || sources[kVideoIndex] != NULL + || sources[kSubtitleIndex] != NULL) { + info.mFetcher->startAsync( + sources[kAudioIndex], sources[kVideoIndex], sources[kSubtitleIndex]); + } else { + info.mToBeRemoved = true; + } } // streamMask now only contains the types that need a new fetcher created. @@ -1029,6 +1149,8 @@ void LiveSession::onChangeConfiguration3(const sp &msg) { ALOGV("creating new fetchers for mask 0x%08x", streamMask); } + // Find out when the original fetchers have buffered up to and start the new fetchers + // at a later timestamp. for (size_t i = 0; i < kMaxStreams; i++) { if (!(indexToType(i) & streamMask)) { continue; @@ -1040,12 +1162,40 @@ void LiveSession::onChangeConfiguration3(const sp &msg) { sp fetcher = addFetcher(uri.c_str()); CHECK(fetcher != NULL); + int32_t latestSeq = -1; + int64_t latestTimeUs = 0ll; sp sources[kMaxStreams]; + // TRICKY: looping from i as earlier streams are already removed from streamMask for (size_t j = i; j < kMaxStreams; ++j) { if ((streamMask & indexToType(j)) && uri == mStreams[j].mUri) { sources[j] = mPacketSources.valueFor(indexToType(j)); - sources[j]->clear(); + + if (!switching) { + sources[j]->clear(); + } else { + int32_t type, seq; + int64_t srcTimeUs; + sp meta = sources[j]->getLatestMeta(); + + if (meta != NULL && !meta->findInt32("discontinuity", &type)) { + CHECK(meta->findInt32("seq", &seq)); + if (seq > latestSeq) { + latestSeq = seq; + } + CHECK(meta->findInt64("timeUs", &srcTimeUs)); + if (srcTimeUs > latestTimeUs) { + latestTimeUs = srcTimeUs; + } + } + + sources[j] = mPacketSources2.valueFor(indexToType(j)); + sources[j]->clear(); + uint32_t extraStreams = mNewStreamMask & (~mStreamMask); + if (extraStreams & indexToType(j)) { + sources[j]->queueAccessUnit(createFormatChangeBuffer(/* swap = */ false)); + } + } streamMask &= ~indexToType(j); } @@ -1055,7 +1205,9 @@ void LiveSession::onChangeConfiguration3(const sp &msg) { sources[kAudioIndex], sources[kVideoIndex], sources[kSubtitleIndex], - timeUs); + timeUs, + latestTimeUs /* min start time(us) */, + latestSeq >= 0 ? latestSeq + 1 : -1 /* starting sequence number hint */ ); } // All fetchers have now been started, the configuration change @@ -1064,14 +1216,61 @@ void LiveSession::onChangeConfiguration3(const sp &msg) { scheduleCheckBandwidthEvent(); ALOGV("XXX configuration change completed."); - mReconfigurationInProgress = false; + if (switching) { + mSwitchInProgress = true; + } else { + mStreamMask = mNewStreamMask; + } if (mDisconnectReplyID != 0) { finishDisconnect(); } } +void LiveSession::onSwapped(const sp &msg) { + int32_t switchGeneration; + CHECK(msg->findInt32("switchGeneration", &switchGeneration)); + if (switchGeneration != mSwitchGeneration) { + return; + } + + int32_t stream; + CHECK(msg->findInt32("stream", &stream)); + mSwapMask |= stream; + if (mSwapMask != mStreamMask) { + return; + } + + // Check if new variant contains extra streams. + uint32_t extraStreams = mNewStreamMask & (~mStreamMask); + while (extraStreams) { + StreamType extraStream = (StreamType) (extraStreams & ~(extraStreams - 1)); + swapPacketSource(extraStream); + extraStreams &= ~extraStream; + } + + tryToFinishBandwidthSwitch(); +} + +// Mark switch done when: +// 1. all old buffers are swapped out, AND +// 2. all old fetchers are removed. +void LiveSession::tryToFinishBandwidthSwitch() { + bool needToRemoveFetchers = false; + for (size_t i = 0; i < mFetcherInfos.size(); ++i) { + if (mFetcherInfos.valueAt(i).mToBeRemoved) { + needToRemoveFetchers = true; + break; + } + } + if (!needToRemoveFetchers && mSwapMask == mStreamMask) { + mStreamMask = mNewStreamMask; + mSwitchInProgress = false; + mSwapMask = 0; + } +} + void LiveSession::scheduleCheckBandwidthEvent() { sp msg = new AMessage(kWhatCheckBandwidth, id()); msg->setInt32("generation", mCheckBandwidthGeneration); @@ -1082,16 +1281,37 @@ void LiveSession::cancelCheckBandwidthEvent() { ++mCheckBandwidthGeneration; } -void LiveSession::onCheckBandwidth() { - if (mReconfigurationInProgress) { - scheduleCheckBandwidthEvent(); - return; +void LiveSession::cancelBandwidthSwitch() { + Mutex::Autolock lock(mSwapMutex); + mSwitchGeneration++; + mSwitchInProgress = false; + mSwapMask = 0; +} + +bool LiveSession::canSwitchBandwidthTo(size_t bandwidthIndex) { + if (mReconfigurationInProgress || mSwitchInProgress) { + return false; + } + + if (mPrevBandwidthIndex < 0) { + return true; } + if (bandwidthIndex == (size_t)mPrevBandwidthIndex) { + return false; + } else if (bandwidthIndex > (size_t)mPrevBandwidthIndex) { + return canSwitchUp(); + } else { + return true; + } +} + +void LiveSession::onCheckBandwidth() { size_t bandwidthIndex = getBandwidthIndex(); - if (mPrevBandwidthIndex < 0 - || bandwidthIndex != (size_t)mPrevBandwidthIndex) { + if (canSwitchBandwidthTo(bandwidthIndex)) { changeConfiguration(-1ll /* timeUs */, bandwidthIndex); + } else { + scheduleCheckBandwidthEvent(); } // Handling the kWhatCheckBandwidth even here does _not_ automatically diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h index b38c518..e26d024 100644 --- a/media/libstagefright/httplive/LiveSession.h +++ b/media/libstagefright/httplive/LiveSession.h @@ -81,6 +81,11 @@ struct LiveSession : public AHandler { kWhatPreparationFailed, }; + // create a format-change discontinuity + // + // swap: + // whether is format-change discontinuity should trigger a buffer swap + sp createFormatChangeBuffer(bool swap = true); protected: virtual ~LiveSession(); @@ -99,6 +104,7 @@ private: kWhatChangeConfiguration2 = 'chC2', kWhatChangeConfiguration3 = 'chC3', kWhatFinishDisconnect2 = 'fin2', + kWhatSwapped = 'swap', }; struct BandwidthItem { @@ -110,6 +116,7 @@ private: sp mFetcher; int64_t mDurationUs; bool mIsPrepared; + bool mToBeRemoved; }; struct StreamItem { @@ -145,9 +152,26 @@ private: KeyedVector mFetcherInfos; uint32_t mStreamMask; + // Masks used during reconfiguration: + // mNewStreamMask: streams in the variant playlist we're switching to; + // we don't want to immediately overwrite the original value. + uint32_t mNewStreamMask; + + // mSwapMask: streams that have started to playback content in the new variant playlist; + // we use this to track reconfiguration progress. + uint32_t mSwapMask; + KeyedVector > mPacketSources; + // A second set of packet sources that buffer content for the variant we're switching to. + KeyedVector > mPacketSources2; + + // A mutex used to serialize two sets of events: + // * the swapping of packet sources in dequeueAccessUnit on the player thread, AND + // * a forced bandwidth switch termination in cancelSwitch on the live looper. + Mutex mSwapMutex; int32_t mCheckBandwidthGeneration; + int32_t mSwitchGeneration; size_t mContinuationCounter; sp mContinuation; @@ -156,6 +180,7 @@ private: int64_t mRealTimeBaseUs; bool mReconfigurationInProgress; + bool mSwitchInProgress; uint32_t mDisconnectReplyID; sp addFetcher(const char *uri); @@ -197,16 +222,27 @@ private: void onChangeConfiguration(const sp &msg); void onChangeConfiguration2(const sp &msg); void onChangeConfiguration3(const sp &msg); + void onSwapped(const sp &msg); + void tryToFinishBandwidthSwitch(); void scheduleCheckBandwidthEvent(); void cancelCheckBandwidthEvent(); + // cancelBandwidthSwitch is atomic wrt swapPacketSource; call it to prevent packet sources + // from being swapped out on stale discontinuities while manipulating + // mPacketSources/mPacketSources2. + void cancelBandwidthSwitch(); + + bool canSwitchBandwidthTo(size_t bandwidthIndex); void onCheckBandwidth(); void finishDisconnect(); void postPrepared(status_t err); + void swapPacketSource(StreamType stream); + bool canSwitchUp(); + DISALLOW_EVIL_CONSTRUCTORS(LiveSession); }; diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp index 9cb16fe..0eac8b3 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.cpp +++ b/media/libstagefright/httplive/PlaylistFetcher.cpp @@ -48,16 +48,20 @@ 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 ¬ify, const sp &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), @@ -69,6 +73,8 @@ PlaylistFetcher::PlaylistFetcher( mFirstPTSValid(false), mAbsoluteTimeAnchorUs(0ll) { memset(mPlaylistHash, 0, sizeof(mPlaylistHash)); + mStartTimeUsNotify->setInt32("what", kWhatStartedAt); + mStartTimeUsNotify->setInt32("streamMask", 0); } PlaylistFetcher::~PlaylistFetcher() { @@ -325,7 +331,9 @@ void PlaylistFetcher::startAsync( const sp &audioSource, const sp &videoSource, const sp &subtitleSource, - int64_t startTimeUs) { + int64_t startTimeUs, + int64_t minStartTimeUs, + int32_t startSeqNumberHint) { sp msg = new AMessage(kWhatStart, id()); uint32_t streamTypeMask = 0ul; @@ -347,6 +355,8 @@ void PlaylistFetcher::startAsync( msg->setInt32("streamTypeMask", streamTypeMask); msg->setInt64("startTimeUs", startTimeUs); + msg->setInt64("minStartTimeUs", minStartTimeUs); + msg->setInt32("startSeqNumberHint", startSeqNumberHint); msg->post(); } @@ -358,6 +368,12 @@ void PlaylistFetcher::stopAsync() { (new AMessage(kWhatStop, id()))->post(); } +void PlaylistFetcher::resumeUntilAsync(const sp ¶ms) { + AMessage* msg = new AMessage(kWhatResumeUntil, id()); + msg->setMessage("params", params); + msg->post(); +} + void PlaylistFetcher::onMessageReceived(const sp &msg) { switch (msg->what()) { case kWhatStart: @@ -392,6 +408,7 @@ void PlaylistFetcher::onMessageReceived(const sp &msg) { } case kWhatMonitorQueue: + case kWhatDownloadNext: { int32_t generation; CHECK(msg->findInt32("generation", &generation)); @@ -401,7 +418,17 @@ void PlaylistFetcher::onMessageReceived(const sp &msg) { break; } - onMonitorQueue(); + if (msg->what() == kWhatMonitorQueue) { + onMonitorQueue(); + } else { + onDownloadNext(); + } + break; + } + + case kWhatResumeUntil: + { + onResumeUntil(msg); break; } @@ -417,7 +444,10 @@ status_t PlaylistFetcher::onStart(const sp &msg) { 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; @@ -455,6 +485,10 @@ status_t PlaylistFetcher::onStart(const sp &msg) { mPrepared = false; } + if (startSeqNumberHint >= 0) { + mSeqNumber = startSeqNumberHint; + } + postMonitorQueue(); return OK; @@ -462,20 +496,70 @@ status_t PlaylistFetcher::onStart(const sp &msg) { void PlaylistFetcher::onPause() { cancelMonitorQueue(); +} + +void PlaylistFetcher::onStop() { + cancelMonitorQueue(); mPacketSources.clear(); mStreamTypeMask = 0; } -void PlaylistFetcher::onStop() { - cancelMonitorQueue(); +// 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 &msg) { + sp params; + CHECK(msg->findMessage("params", ¶ms)); + + bool stop = false; + for (size_t i = 0; i < mPacketSources.size(); i++) { + sp packetSource = mPacketSources.valueAt(i); + + const char *stopKey; + int streamType = mPacketSources.keyAt(i); + switch (streamType) { + case LiveSession::STREAMTYPE_VIDEO: + stopKey = "timeUsVideo"; + break; - for (size_t i = 0; i < mPacketSources.size(); ++i) { - mPacketSources.valueAt(i)->clear(); + 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 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; + } + } } - mPacketSources.clear(); - mStreamTypeMask = 0; + if (stop) { + for (size_t i = 0; i < mPacketSources.size(); i++) { + mPacketSources.valueAt(i)->queueAccessUnit(mSession->createFormatChangeBuffer()); + } + stopAsync(); + return OK; + } + + mStopParams = params; + postMonitorQueue(); + + return OK; } void PlaylistFetcher::notifyError(status_t err) { @@ -519,8 +603,9 @@ void PlaylistFetcher::onMonitorQueue() { packetSource->getBufferedDurationUs(&finalResult); finalResult = OK; } else { - bool first = true; - + // 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; @@ -528,9 +613,10 @@ void PlaylistFetcher::onMonitorQueue() { int64_t bufferedStreamDurationUs = mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult); - if (first || bufferedStreamDurationUs < bufferedDurationUs) { + ALOGV("buffered %lld for stream %d", + bufferedStreamDurationUs, mPacketSources.keyAt(i)); + if (bufferedStreamDurationUs > bufferedDurationUs) { bufferedDurationUs = bufferedStreamDurationUs; - first = false; } } } @@ -550,7 +636,12 @@ void PlaylistFetcher::onMonitorQueue() { if (finalResult == OK && downloadMore) { ALOGV("monitoring, buffered=%lld < %lld", bufferedDurationUs, durationToBufferUs); - onDownloadNext(); + // delay the next download slightly; hopefully this gives other concurrent fetchers + // a better chance to run. + // onDownloadNext(); + sp msg = new AMessage(kWhatDownloadNext, id()); + msg->setInt32("generation", mMonitorQueueGeneration); + msg->post(1000l); } else { // Nothing to do yet, try again in a second. @@ -617,6 +708,12 @@ void PlaylistFetcher::onDownloadNext() { 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); @@ -761,6 +858,18 @@ void PlaylistFetcher::onDownloadNext() { err = extractAndQueueAccessUnits(buffer, itemMeta); + if (err == -EAGAIN) { + // bad starting sequence number hint + postMonitorQueue(); + return; + } + + if (err == ERROR_OUT_OF_RANGE) { + // reached stopping point + stopAsync(); + return; + } + if (err != OK) { notifyError(err); return; @@ -817,12 +926,15 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( } if (mTSParser == NULL) { - mTSParser = new ATSParser; + // Use TS_TIMESTAMPS_ARE_ABSOLUTE so pts carry over between fetchers. + mTSParser = new ATSParser(ATSParser::TS_TIMESTAMPS_ARE_ABSOLUTE); } if (mNextPTSTimeUs >= 0ll) { sp extra = new AMessage; - extra->setInt64(IStreamListener::kKeyMediaTimeUs, mNextPTSTimeUs); + // 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); @@ -841,17 +953,23 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( offset += 188; } + status_t err = OK; for (size_t i = mPacketSources.size(); i-- > 0;) { sp packetSource = mPacketSources.valueAt(i); + const char *key; ATSParser::SourceType type; - switch (mPacketSources.keyAt(i)) { + 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: @@ -878,19 +996,87 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( continue; } + int64_t timeUs; sp accessUnit; status_t finalResult; while (source->hasBufferAvailable(&finalResult) && source->dequeueAccessUnit(&accessUnit) == OK) { - // Note that we do NOT dequeue any discontinuities. + + 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 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 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; @@ -907,6 +1093,7 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( 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; @@ -1043,6 +1230,7 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( // Each AAC frame encodes 1024 samples. numSamples += 1024; + unit->meta()->setInt32("seq", mSeqNumber); packetSource->queueAccessUnit(unit); offset += aac_frame_length; @@ -1070,4 +1258,33 @@ void PlaylistFetcher::updateDuration() { msg->post(); } +int64_t PlaylistFetcher::resumeThreshold(const sp &msg) { + int64_t durationUs, threshold; + if (msg->findInt64("durationUs", &durationUs)) { + return kNumSkipFrames * durationUs; + } + + sp obj; + msg->findObject("format", &obj); + MetaData *format = static_cast(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 diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h index ac04a77..2e0349f 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.h +++ b/media/libstagefright/httplive/PlaylistFetcher.h @@ -43,6 +43,7 @@ struct PlaylistFetcher : public AHandler { kWhatTemporarilyDoneFetching, kWhatPrepared, kWhatPreparationFailed, + kWhatStartedAt, }; PlaylistFetcher( @@ -56,12 +57,16 @@ struct PlaylistFetcher : public AHandler { const sp &audioSource, const sp &videoSource, const sp &subtitleSource, - int64_t startTimeUs = -1ll); + int64_t startTimeUs = -1ll, + int64_t minStartTimeUs = 0ll /* start after this timestamp */, + int32_t startSeqNumberHint = -1 /* try starting at this sequence number */); void pauseAsync(); void stopAsync(); + void resumeUntilAsync(const sp ¶ms); + protected: virtual ~PlaylistFetcher(); virtual void onMessageReceived(const sp &msg); @@ -76,17 +81,25 @@ private: kWhatPause = 'paus', kWhatStop = 'stop', kWhatMonitorQueue = 'moni', + kWhatResumeUntil = 'rsme', + kWhatDownloadNext = 'dlnx', }; static const int64_t kMinBufferedDurationUs; static const int64_t kMaxMonitorDelayUs; + static const int32_t kNumSkipFrames; + // notifications to mSession sp mNotify; + sp mStartTimeUsNotify; + sp mSession; AString mURI; uint32_t mStreamTypeMask; int64_t mStartTimeUs; + int64_t mMinStartTimeUs; // start fetching no earlier than this value + sp mStopParams; // message containing the latest timestamps we should fetch. KeyedVector > mPacketSources; @@ -153,6 +166,9 @@ private: void onMonitorQueue(); void onDownloadNext(); + // Resume a fetcher to continue until the stopping point stored in msg. + status_t onResumeUntil(const sp &msg); + status_t extractAndQueueAccessUnits( const sp &buffer, const sp &itemMeta); @@ -165,6 +181,10 @@ private: void updateDuration(); + // Before resuming a fetcher in onResume, check the remaining duration is longer than that + // returned by resumeThreshold. + int64_t resumeThreshold(const sp &msg); + DISALLOW_EVIL_CONSTRUCTORS(PlaylistFetcher); }; -- cgit v1.1 From 75d03185ac7be95c700f3f375080989e5de03ef6 Mon Sep 17 00:00:00 2001 From: Martin Storsjo Date: Fri, 22 Nov 2013 17:05:05 +0200 Subject: M3UParser: Skip query strings when looking for the last slash in a URL Bug: 13174301 Change-Id: I72d3a5e11fef9bbd75b291bc490c9cab1dce58da --- media/libstagefright/httplive/M3UParser.cpp | 34 +++++++++++++++++++---------- 1 file changed, 22 insertions(+), 12 deletions(-) (limited to 'media/libstagefright/httplive') diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp index a9184b5..8f530ee 100644 --- a/media/libstagefright/httplive/M3UParser.cpp +++ b/media/libstagefright/httplive/M3UParser.cpp @@ -423,22 +423,32 @@ static bool MakeURL(const char *baseURL, const char *url, AString *out) { } else { // URL is a relative path - size_t n = strlen(baseURL); - if (baseURL[n - 1] == '/') { - out->setTo(baseURL); - out->append(url); + // Check for a possible query string + const char *qsPos = strchr(baseURL, '?'); + size_t end; + if (qsPos != NULL) { + end = qsPos - baseURL; } else { - const char *slashPos = strrchr(baseURL, '/'); - - if (slashPos > &baseURL[6]) { - out->setTo(baseURL, slashPos - baseURL); - } else { - out->setTo(baseURL); + end = strlen(baseURL); + } + // Check for the last slash before a potential query string + for (ssize_t pos = end - 1; pos >= 0; pos--) { + if (baseURL[pos] == '/') { + end = pos; + break; } + } - out->append("/"); - out->append(url); + // Check whether the found slash actually is part of the path + // and not part of the "http://". + if (end > 6) { + out->setTo(baseURL, end); + } else { + out->setTo(baseURL); } + + out->append("/"); + out->append(url); } ALOGV("base:'%s', url:'%s' => '%s'", baseURL, url, out->c_str()); -- cgit v1.1 From a1076eb135b74a32e9bdc1aed17aee4374eb41af Mon Sep 17 00:00:00 2001 From: Martin Storsjo Date: Fri, 22 Nov 2013 17:57:31 +0200 Subject: LiveSession: Use the actual, possibly redirected url as base in the M3U This fixes playback of HLS streams where the M3U playlists use relative paths and the main playlist is opened via a redirect. Bug: 13174301 Change-Id: I787e1c050daddc566be4c8e3f84803d3f91eec46 --- media/libstagefright/httplive/LiveSession.cpp | 14 +++++++++++--- media/libstagefright/httplive/LiveSession.h | 3 ++- 2 files changed, 13 insertions(+), 4 deletions(-) (limited to 'media/libstagefright/httplive') diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp index 9e30ebd..687e871 100644 --- a/media/libstagefright/httplive/LiveSession.cpp +++ b/media/libstagefright/httplive/LiveSession.cpp @@ -623,7 +623,8 @@ status_t LiveSession::fetchFile( const char *url, sp *out, int64_t range_offset, int64_t range_length, uint32_t block_size, /* download block size */ - sp *source /* to return and reuse source */) { + sp *source, /* to return and reuse source */ + String8 *actualUrl) { off64_t size; sp temp_source; if (source == NULL) { @@ -719,6 +720,12 @@ status_t LiveSession::fetchFile( } *out = buffer; + if (actualUrl != NULL) { + *actualUrl = (*source)->getUri(); + if (actualUrl->isEmpty()) { + *actualUrl = url; + } + } return OK; } @@ -730,7 +737,8 @@ sp LiveSession::fetchPlaylist( *unchanged = false; sp buffer; - status_t err = fetchFile(url, &buffer); + String8 actualUrl; + status_t err = fetchFile(url, &buffer, 0, -1, 0, NULL, &actualUrl); if (err != OK) { return NULL; @@ -764,7 +772,7 @@ sp LiveSession::fetchPlaylist( #endif sp playlist = - new M3UParser(url, buffer->data(), buffer->size()); + new M3UParser(actualUrl.string(), buffer->data(), buffer->size()); if (playlist->initCheck() != OK) { ALOGE("failed to parse .m3u8 playlist"); diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h index e26d024..376d451 100644 --- a/media/libstagefright/httplive/LiveSession.h +++ b/media/libstagefright/httplive/LiveSession.h @@ -207,7 +207,8 @@ private: /* download block size */ uint32_t block_size = 0, /* reuse DataSource if doing partial fetch */ - sp *source = NULL); + sp *source = NULL, + String8 *actualUrl = NULL); sp fetchPlaylist( const char *url, uint8_t *curPlaylistHash, bool *unchanged); -- cgit v1.1 From ceeabe15f4e7bc73efdfcafed917202de9d515cb Mon Sep 17 00:00:00 2001 From: Robert Shih Date: Mon, 3 Mar 2014 15:08:27 -0800 Subject: httplive: clear access units before returning from seekTo. Bug: 12060952 Change-Id: I6a69a718c082501003ee9b78a948a2f8bbfbb14e --- media/libstagefright/httplive/LiveSession.cpp | 26 +++++++++++++++++------ media/libstagefright/httplive/LiveSession.h | 2 ++ media/libstagefright/httplive/PlaylistFetcher.cpp | 25 ++++++++++++++++------ media/libstagefright/httplive/PlaylistFetcher.h | 4 ++-- 4 files changed, 42 insertions(+), 15 deletions(-) (limited to 'media/libstagefright/httplive') diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp index 687e871..c1c3e2b 100644 --- a/media/libstagefright/httplive/LiveSession.cpp +++ b/media/libstagefright/httplive/LiveSession.cpp @@ -67,7 +67,8 @@ LiveSession::LiveSession( mRealTimeBaseUs(0ll), mReconfigurationInProgress(false), mSwitchInProgress(false), - mDisconnectReplyID(0) { + mDisconnectReplyID(0), + mSeekReplyID(0) { if (mUIDValid) { mHTTPDataSource->setUID(mUID); } @@ -234,6 +235,10 @@ status_t LiveSession::seekTo(int64_t timeUs) { sp response; status_t err = msg->postAndAwaitResponse(&response); + uint32_t replyID; + CHECK(response == mSeekReply && 0 != mSeekReplyID); + mSeekReply.clear(); + mSeekReplyID = 0; return err; } @@ -259,15 +264,12 @@ void LiveSession::onMessageReceived(const sp &msg) { case kWhatSeek: { - uint32_t replyID; - CHECK(msg->senderAwaitsResponse(&replyID)); + CHECK(msg->senderAwaitsResponse(&mSeekReplyID)); status_t err = onSeek(msg); - sp response = new AMessage; - response->setInt32("err", err); - - response->postReply(replyID); + mSeekReply = new AMessage; + mSeekReply->setInt32("err", err); break; } @@ -297,6 +299,11 @@ void LiveSession::onMessageReceived(const sp &msg) { CHECK_GT(mContinuationCounter, 0); if (--mContinuationCounter == 0) { mContinuation->post(); + + if (mSeekReplyID != 0) { + CHECK(mSeekReply != NULL); + mSeekReply->postReply(mSeekReplyID); + } } } break; @@ -1035,6 +1042,11 @@ void LiveSession::changeConfiguration( if (mContinuationCounter == 0) { msg->post(); + + if (mSeekReplyID != 0) { + CHECK(mSeekReply != NULL); + mSeekReply->postReply(mSeekReplyID); + } } } diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h index 376d451..680792d 100644 --- a/media/libstagefright/httplive/LiveSession.h +++ b/media/libstagefright/httplive/LiveSession.h @@ -175,6 +175,7 @@ private: size_t mContinuationCounter; sp mContinuation; + sp mSeekReply; int64_t mLastDequeuedTimeUs; int64_t mRealTimeBaseUs; @@ -182,6 +183,7 @@ private: bool mReconfigurationInProgress; bool mSwitchInProgress; uint32_t mDisconnectReplyID; + uint32_t mSeekReplyID; sp addFetcher(const char *uri); diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp index 0eac8b3..012a68b 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.cpp +++ b/media/libstagefright/httplive/PlaylistFetcher.cpp @@ -364,8 +364,10 @@ void PlaylistFetcher::pauseAsync() { (new AMessage(kWhatPause, id()))->post(); } -void PlaylistFetcher::stopAsync() { - (new AMessage(kWhatStop, id()))->post(); +void PlaylistFetcher::stopAsync(bool selfTriggered) { + sp msg = new AMessage(kWhatStop, id()); + msg->setInt32("selfTriggered", selfTriggered); + msg->post(); } void PlaylistFetcher::resumeUntilAsync(const sp ¶ms) { @@ -399,7 +401,7 @@ void PlaylistFetcher::onMessageReceived(const sp &msg) { case kWhatStop: { - onStop(); + onStop(msg); sp notify = mNotify->dup(); notify->setInt32("what", kWhatStopped); @@ -498,9 +500,20 @@ void PlaylistFetcher::onPause() { cancelMonitorQueue(); } -void PlaylistFetcher::onStop() { +void PlaylistFetcher::onStop(const sp &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 packetSource = mPacketSources.valueAt(i); + packetSource->clear(); + } + } + mPacketSources.clear(); mStreamTypeMask = 0; } @@ -552,7 +565,7 @@ status_t PlaylistFetcher::onResumeUntil(const sp &msg) { for (size_t i = 0; i < mPacketSources.size(); i++) { mPacketSources.valueAt(i)->queueAccessUnit(mSession->createFormatChangeBuffer()); } - stopAsync(); + stopAsync(/* selfTriggered = */ true); return OK; } @@ -866,7 +879,7 @@ void PlaylistFetcher::onDownloadNext() { if (err == ERROR_OUT_OF_RANGE) { // reached stopping point - stopAsync(); + stopAsync(/* selfTriggered = */ true); return; } diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h index 2e0349f..8404b8d 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.h +++ b/media/libstagefright/httplive/PlaylistFetcher.h @@ -63,7 +63,7 @@ struct PlaylistFetcher : public AHandler { void pauseAsync(); - void stopAsync(); + void stopAsync(bool selfTriggered = false); void resumeUntilAsync(const sp ¶ms); @@ -162,7 +162,7 @@ private: status_t onStart(const sp &msg); void onPause(); - void onStop(); + void onStop(const sp &msg); void onMonitorQueue(); void onDownloadNext(); -- cgit v1.1 From 9f434cfd021f60e26baf589dc34bf3839b832d4d Mon Sep 17 00:00:00 2001 From: Robert Shih Date: Mon, 10 Mar 2014 17:05:01 -0700 Subject: M3UParser: trim spaces when parsing comma separated codecs. Bug: 13402087 Change-Id: Idc92716bfefd6d1b0cb371d0d97d990d53288090 --- media/libstagefright/httplive/M3UParser.cpp | 1 + 1 file changed, 1 insertion(+) (limited to 'media/libstagefright/httplive') diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp index 8f530ee..f211bc8 100644 --- a/media/libstagefright/httplive/M3UParser.cpp +++ b/media/libstagefright/httplive/M3UParser.cpp @@ -365,6 +365,7 @@ bool M3UParser::getTypeURI(size_t index, const char *key, AString *uri) const { codecs.append(','); while ((commaPos = codecs.find(",", offset)) >= 0) { AString codec(codecs, offset, commaPos - offset); + codec.trim(); // return true only if a codec of type `key` ("audio"/"video") // is found. if (codecIsType(codec, key)) { -- cgit v1.1 From 7d3044d64294cca6fadd184648a57185e92cf5c6 Mon Sep 17 00:00:00 2001 From: Robert Shih Date: Mon, 10 Mar 2014 17:27:15 -0700 Subject: LiveSession: fix incorrect stream key (subtitle"s") Bug: 13402087 Change-Id: Ic46e3069c6e41f90ead47cae84cbe0123d11002a --- media/libstagefright/httplive/LiveSession.cpp | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'media/libstagefright/httplive') diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp index c1c3e2b..3e11759 100644 --- a/media/libstagefright/httplive/LiveSession.cpp +++ b/media/libstagefright/httplive/LiveSession.cpp @@ -75,7 +75,7 @@ LiveSession::LiveSession( mStreams[kAudioIndex] = StreamItem("audio"); mStreams[kVideoIndex] = StreamItem("video"); - mStreams[kSubtitleIndex] = StreamItem("subtitle"); + mStreams[kSubtitleIndex] = StreamItem("subtitles"); for (size_t i = 0; i < kMaxStreams; ++i) { mPacketSources.add(indexToType(i), new AnotherPacketSource(NULL /* meta */)); -- cgit v1.1 From 43ca783effd99bba0e6e2dd6fe177a8888578ef8 Mon Sep 17 00:00:00 2001 From: Robert Shih Date: Thu, 27 Feb 2014 12:33:24 -0800 Subject: httplive: block-by-block fetch, decrypt, and parse ts files. Bug: 12060952 Change-Id: I695345081fe23961b9d0ef6db264885f914703ec --- media/libstagefright/httplive/LiveSession.cpp | 10 +- media/libstagefright/httplive/LiveSession.h | 2 +- media/libstagefright/httplive/PlaylistFetcher.cpp | 451 +++++++++++++--------- media/libstagefright/httplive/PlaylistFetcher.h | 5 + 4 files changed, 282 insertions(+), 186 deletions(-) (limited to 'media/libstagefright/httplive') diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp index 3e11759..e53e35a 100644 --- a/media/libstagefright/httplive/LiveSession.cpp +++ b/media/libstagefright/httplive/LiveSession.cpp @@ -626,7 +626,7 @@ sp LiveSession::addFetcher(const char *uri) { * - block_size == 0 means entire range * */ -status_t LiveSession::fetchFile( +ssize_t LiveSession::fetchFile( const char *url, sp *out, int64_t range_offset, int64_t range_length, uint32_t block_size, /* download block size */ @@ -677,6 +677,7 @@ status_t LiveSession::fetchFile( buffer->setRange(0, 0); } + ssize_t bytesRead = 0; // adjust range_length if only reading partial block if (block_size > 0 && (range_length == -1 || buffer->size() + block_size < range_length)) { range_length = buffer->size() + block_size; @@ -724,6 +725,7 @@ status_t LiveSession::fetchFile( } buffer->setRange(0, buffer->size() + (size_t)n); + bytesRead += n; } *out = buffer; @@ -734,7 +736,7 @@ status_t LiveSession::fetchFile( } } - return OK; + return bytesRead; } sp LiveSession::fetchPlaylist( @@ -745,9 +747,9 @@ sp LiveSession::fetchPlaylist( sp buffer; String8 actualUrl; - status_t err = fetchFile(url, &buffer, 0, -1, 0, NULL, &actualUrl); + ssize_t err = fetchFile(url, &buffer, 0, -1, 0, NULL, &actualUrl); - if (err != OK) { + if (err <= 0) { return NULL; } diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h index 680792d..3f8fee5 100644 --- a/media/libstagefright/httplive/LiveSession.h +++ b/media/libstagefright/httplive/LiveSession.h @@ -202,7 +202,7 @@ private: // // For reused HTTP sources, the caller must download a file sequentially without // any overlaps or gaps to prevent reconnection. - status_t fetchFile( + ssize_t fetchFile( const char *url, sp *out, /* request/open a file starting at range_offset for range_length bytes */ int64_t range_offset = 0, int64_t range_length = -1, diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp index 012a68b..ada856d 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.cpp +++ b/media/libstagefright/httplive/PlaylistFetcher.cpp @@ -48,6 +48,7 @@ namespace android { // static const int64_t PlaylistFetcher::kMinBufferedDurationUs = 10000000ll; const int64_t PlaylistFetcher::kMaxMonitorDelayUs = 3000000ll; +const int32_t PlaylistFetcher::kDownloadBlockSize = 192; const int32_t PlaylistFetcher::kNumSkipFrames = 10; PlaylistFetcher::PlaylistFetcher( @@ -216,9 +217,9 @@ status_t PlaylistFetcher::decryptBuffer( if (index >= 0) { key = mAESKeyForURI.valueAt(index); } else { - status_t err = mSession->fetchFile(keyURI.c_str(), &key); + ssize_t err = mSession->fetchFile(keyURI.c_str(), &key); - if (err != OK) { + if (err < 0) { ALOGE("failed to fetch cipher key from '%s'.", keyURI.c_str()); return ERROR_IO; } else if (key->size() != 16) { @@ -704,6 +705,11 @@ status_t PlaylistFetcher::refreshPlaylist() { return OK; } +// static +bool PlaylistFetcher::bufferStartsWithTsSyncByte(const sp& buffer) { + return buffer->size() > 0 && buffer->data()[0] == 0x47; +} + void PlaylistFetcher::onDownloadNext() { if (refreshPlaylist() != OK) { return; @@ -823,66 +829,161 @@ void PlaylistFetcher::onDownloadNext() { ALOGV("fetching '%s'", uri.c_str()); - sp 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; + sp source; + sp buffer, tsBuffer; + // decrypt a junk buffer to prefetch key; since a session uses only one http connection, + // this avoids interleaved connections to the key and segment file. + { + sp junk = new ABuffer(16); + junk->setRange(0, 16); + status_t err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, junk, + true /* first */); + if (err != OK) { + notifyError(err); + return; + } } - CHECK(buffer != NULL); + // block-wise download + ssize_t bytesRead; + do { + bytesRead = mSession->fetchFile( + uri.c_str(), &buffer, range_offset, range_length, kDownloadBlockSize, &source); - err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, buffer); - if (err == OK) { - err = checkDecryptPadding(buffer); - } + if (bytesRead < 0) { + status_t err = bytesRead; + ALOGE("failed to fetch .ts segment at url '%s'", uri.c_str()); + notifyError(err); + return; + } - if (err != OK) { - ALOGE("decryptBuffer failed w/ error %d", err); + CHECK(buffer != NULL); - notifyError(err); - return; - } + size_t size = buffer->size(); + // Set decryption range. + buffer->setRange(size - bytesRead, bytesRead); + status_t err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, buffer, + buffer->offset() == 0 /* first */); + // Unset decryption range. + buffer->setRange(0, size); - if (mStartup || seekDiscontinuity || explicitDiscontinuity) { - // Signal discontinuity. + if (err != OK) { + ALOGE("decryptBuffer failed w/ error %d", err); - 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); + notifyError(err); + return; } - if (seekDiscontinuity || explicitDiscontinuity) { - ALOGI("queueing discontinuity (seek=%d, explicit=%d)", - seekDiscontinuity, explicitDiscontinuity); + 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 */); + queueDiscontinuity( + explicitDiscontinuity + ? ATSParser::DISCONTINUITY_FORMATCHANGE + : ATSParser::DISCONTINUITY_SEEK, + NULL /* extra */); + } } - } - err = extractAndQueueAccessUnits(buffer, itemMeta); + err = OK; + if (bufferStartsWithTsSyncByte(buffer)) { + // Incremental extraction is only supported for MPEG2 transport streams. + if (tsBuffer == NULL) { + tsBuffer = new ABuffer(buffer->data(), buffer->capacity()); + tsBuffer->setRange(0, 0); + } else if (tsBuffer->capacity() != buffer->capacity()) { + size_t tsOff = tsBuffer->offset(), tsSize = tsBuffer->size(); + tsBuffer = new ABuffer(buffer->data(), buffer->capacity()); + tsBuffer->setRange(tsOff, tsSize); + } + tsBuffer->setRange(tsBuffer->offset(), tsBuffer->size() + bytesRead); + + err = extractAndQueueAccessUnitsFromTs(tsBuffer); + } + + 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; + } + + mStartup = false; + } while (bytesRead != 0); + + if (bufferStartsWithTsSyncByte(buffer)) { + // If we still don't see a stream after fetching a full ts segment mark it as + // nonexistent. + const size_t kNumTypes = ATSParser::NUM_SOURCE_TYPES; + ATSParser::SourceType srcTypes[kNumTypes] = + { ATSParser::VIDEO, ATSParser::AUDIO }; + LiveSession::StreamType streamTypes[kNumTypes] = + { LiveSession::STREAMTYPE_VIDEO, LiveSession::STREAMTYPE_AUDIO }; + + for (size_t i = 0; i < kNumTypes; i++) { + ATSParser::SourceType srcType = srcTypes[i]; + LiveSession::StreamType streamType = streamTypes[i]; + + sp source = + static_cast( + mTSParser->getSource(srcType).get()); + + if (source == NULL) { + ALOGW("MPEG2 Transport stream does not contain %s data.", + srcType == ATSParser::VIDEO ? "video" : "audio"); + + mStreamTypeMask &= ~streamType; + mPacketSources.removeItem(streamType); + } + } - if (err == -EAGAIN) { - // bad starting sequence number hint - postMonitorQueue(); - return; } - if (err == ERROR_OUT_OF_RANGE) { - // reached stopping point - stopAsync(/* selfTriggered = */ true); + if (checkDecryptPadding(buffer) != OK) { + ALOGE("Incorrect padding bytes after decryption."); + notifyError(ERROR_MALFORMED); return; } + status_t err = OK; + if (tsBuffer != NULL) { + AString method; + CHECK(buffer->meta()->findString("cipher-method", &method)); + if ((tsBuffer->size() > 0 && method == "NONE") + || tsBuffer->size() > 16) { + ALOGE("MPEG2 transport stream is not an even multiple of 188 " + "bytes in length."); + notifyError(ERROR_MALFORMED); + return; + } + } + + // bulk extract non-ts files + if (tsBuffer == NULL) { + err = extractAndQueueAccessUnits(buffer, itemMeta); + } + if (err != OK) { notifyError(err); return; @@ -891,8 +992,6 @@ void PlaylistFetcher::onDownloadNext() { ++mSeqNumber; postMonitorQueue(); - - mStartup = false; } int32_t PlaylistFetcher::getSeqNumberForTime(int64_t timeUs) const { @@ -927,173 +1026,163 @@ int32_t PlaylistFetcher::getSeqNumberForTime(int64_t timeUs) const { return firstSeqNumberInPlaylist + index; } -status_t PlaylistFetcher::extractAndQueueAccessUnits( - const sp &buffer, const sp &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 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); +status_t PlaylistFetcher::extractAndQueueAccessUnitsFromTs(const sp &buffer) { + if (mTSParser == NULL) { + // Use TS_TIMESTAMPS_ARE_ABSOLUTE so pts carry over between fetchers. + mTSParser = new ATSParser(ATSParser::TS_TIMESTAMPS_ARE_ABSOLUTE); + } - mTSParser->signalDiscontinuity( - ATSParser::DISCONTINUITY_SEEK, extra); + if (mNextPTSTimeUs >= 0ll) { + sp 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); - mNextPTSTimeUs = -1ll; - } + mTSParser->signalDiscontinuity( + ATSParser::DISCONTINUITY_SEEK, extra); - size_t offset = 0; - while (offset < buffer->size()) { - status_t err = mTSParser->feedTSPacket(buffer->data() + offset, 188); + mNextPTSTimeUs = -1ll; + } - if (err != OK) { - return err; - } + size_t offset = 0; + while (offset + 188 <= buffer->size()) { + status_t err = mTSParser->feedTSPacket(buffer->data() + offset, 188); - offset += 188; + if (err != OK) { + return err; } - status_t err = OK; - for (size_t i = mPacketSources.size(); i-- > 0;) { - sp packetSource = mPacketSources.valueAt(i); - - const char *key; - ATSParser::SourceType type; - const LiveSession::StreamType stream = mPacketSources.keyAt(i); - switch (stream) { + offset += 188; + } + // setRange to indicate consumed bytes. + buffer->setRange(buffer->offset() + offset, buffer->size() - offset); - case LiveSession::STREAMTYPE_VIDEO: - type = ATSParser::VIDEO; - key = "timeUsVideo"; - break; + status_t err = OK; + for (size_t i = mPacketSources.size(); i-- > 0;) { + sp packetSource = mPacketSources.valueAt(i); - case LiveSession::STREAMTYPE_AUDIO: - type = ATSParser::AUDIO; - key = "timeUsAudio"; - break; + 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_SUBTITLES: - { - ALOGE("MPEG2 Transport streams do not contain subtitles."); - return ERROR_MALFORMED; - break; - } + case LiveSession::STREAMTYPE_AUDIO: + type = ATSParser::AUDIO; + key = "timeUsAudio"; + break; - default: - TRESPASS(); + case LiveSession::STREAMTYPE_SUBTITLES: + { + ALOGE("MPEG2 Transport streams do not contain subtitles."); + return ERROR_MALFORMED; + break; } - sp source = - static_cast( - mTSParser->getSource(type).get()); + default: + TRESPASS(); + } - if (source == NULL) { - ALOGW("MPEG2 Transport stream does not contain %s data.", - type == ATSParser::VIDEO ? "video" : "audio"); + sp source = + static_cast( + mTSParser->getSource(type).get()); - mStreamTypeMask &= ~mPacketSources.keyAt(i); - mPacketSources.removeItemsAt(i); - continue; - } + if (source == NULL) { + continue; + } - int64_t timeUs; - sp 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(); - } + int64_t timeUs; + sp 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; - } + 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 format = source->getFormat(); - if (format != NULL) { - accessUnit->meta()->setObject("format", format); - } + // Note that we do NOT dequeue any discontinuities except for format change. - // Stash the sequence number so we can hint future fetchers where to start at. - accessUnit->meta()->setInt32("seq", mSeqNumber); - packetSource->queueAccessUnit(accessUnit); + // for simplicity, store a reference to the format in each unit + sp format = source->getFormat(); + if (format != NULL) { + accessUnit->meta()->setObject("format", format); } - if (err != OK) { - break; - } + // Stash the sequence number so we can hint future playlist where to start at. + accessUnit->meta()->setInt32("seq", mSeqNumber); + packetSource->queueAccessUnit(accessUnit); } if (err != OK) { - for (size_t i = mPacketSources.size(); i-- > 0;) { - sp packetSource = mPacketSources.valueAt(i); - packetSource->clear(); - } - return err; + break; } + } - if (!mStreamTypeMask) { - // Signal gap is filled between original and new stream. - ALOGV("ERROR OUT OF RANGE"); - return ERROR_OUT_OF_RANGE; + if (err != OK) { + for (size_t i = mPacketSources.size(); i-- > 0;) { + sp packetSource = mPacketSources.valueAt(i); + packetSource->clear(); } + return err; + } - return OK; - } else if (buffer->size() >= 7 && !memcmp("WEBVTT\n", buffer->data(), 7)) { + if (!mStreamTypeMask) { + // Signal gap is filled between original and new stream. + ALOGV("ERROR OUT OF RANGE"); + return ERROR_OUT_OF_RANGE; + } + + return OK; +} + +status_t PlaylistFetcher::extractAndQueueAccessUnits( + const sp &buffer, const sp &itemMeta) { + if (buffer->size() >= 7 && !memcmp("WEBVTT\n", buffer->data(), 7)) { if (mStreamTypeMask != LiveSession::STREAMTYPE_SUBTITLES) { ALOGE("This stream only contains subtitles."); return ERROR_MALFORMED; diff --git a/media/libstagefright/httplive/PlaylistFetcher.h b/media/libstagefright/httplive/PlaylistFetcher.h index 8404b8d..7e21523 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.h +++ b/media/libstagefright/httplive/PlaylistFetcher.h @@ -87,8 +87,11 @@ private: static const int64_t kMinBufferedDurationUs; static const int64_t kMaxMonitorDelayUs; + static const int32_t kDownloadBlockSize; static const int32_t kNumSkipFrames; + static bool bufferStartsWithTsSyncByte(const sp& buffer); + // notifications to mSession sp mNotify; sp mStartTimeUsNotify; @@ -169,6 +172,8 @@ private: // Resume a fetcher to continue until the stopping point stored in msg. status_t onResumeUntil(const sp &msg); + status_t extractAndQueueAccessUnitsFromTs(const sp &buffer); + status_t extractAndQueueAccessUnits( const sp &buffer, const sp &itemMeta); -- cgit v1.1 From 65e20ffc984c541a8119420f917493dd7b703f77 Mon Sep 17 00:00:00 2001 From: Robert Shih Date: Fri, 7 Feb 2014 12:26:58 -0800 Subject: DO NOT MERGE: PlaylistFetcher: fix infinite loop when parsing ADTS. First check for embedded ID3 tag, then bail out if invalid. Bug: 12934795 Change-Id: I74acebed4bfb2c6ca44dfe936166fdba8510233f --- media/libstagefright/httplive/PlaylistFetcher.cpp | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'media/libstagefright/httplive') diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp index ada856d..668cbd4 100644 --- a/media/libstagefright/httplive/PlaylistFetcher.cpp +++ b/media/libstagefright/httplive/PlaylistFetcher.cpp @@ -1321,6 +1321,18 @@ status_t PlaylistFetcher::extractAndQueueAccessUnits( | (adtsHeader[4] << 3) | (adtsHeader[5] >> 5); + if (aac_frame_length == 0) { + const uint8_t *id3Header = adtsHeader; + if (!memcmp(id3Header, "ID3", 3)) { + ID3 id3(id3Header, buffer->size() - offset, true); + if (id3.isValid()) { + offset += id3.rawSize(); + continue; + }; + } + return ERROR_MALFORMED; + } + CHECK_LE(offset + aac_frame_length, buffer->size()); sp unit = new ABuffer(aac_frame_length); -- cgit v1.1