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/*
* Copyright (C) 2020 Apple Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS''
* AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
* THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
* PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS
* BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
* INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
* ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF
* THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "config.h"
#include "SourceBufferPrivate.h"
#if ENABLE(MEDIA_SOURCE)
#include "AudioTrackPrivate.h"
#include "Logging.h"
#include "MediaDescription.h"
#include "MediaSample.h"
#include "PlatformTimeRanges.h"
#include "SampleMap.h"
#include "SharedBuffer.h"
#include "SourceBufferPrivateClient.h"
#include "TimeRanges.h"
#include "VideoTrackPrivate.h"
#include <wtf/CheckedArithmetic.h>
#include <wtf/MediaTime.h>
#include <wtf/StringPrintStream.h>
namespace WebCore {
// Do not enqueue samples spanning a significant unbuffered gap.
// NOTE: one second is somewhat arbitrary. MediaSource::monitorSourceBuffers() is run
// on the playbackTimer, which is effectively every 350ms. Allowing > 350ms gap between
// enqueued samples allows for situations where we overrun the end of a buffered range
// but don't notice for 350ms of playback time, and the client can enqueue data for the
// new current time without triggering this early return.
// FIXME(135867): Make this gap detection logic less arbitrary.
static const MediaTime discontinuityTolerance = MediaTime(1, 1);
SourceBufferPrivate::TrackBuffer::TrackBuffer()
: lastDecodeTimestamp(MediaTime::invalidTime())
, greatestDecodeDuration(MediaTime::invalidTime())
, lastFrameDuration(MediaTime::invalidTime())
, highestPresentationTimestamp(MediaTime::invalidTime())
, highestEnqueuedPresentationTime(MediaTime::invalidTime())
, lastEnqueuedDecodeKey({MediaTime::invalidTime(), MediaTime::invalidTime()})
, enqueueDiscontinuityBoundary(discontinuityTolerance)
{
}
SourceBufferPrivate::SourceBufferPrivate()
: m_buffered(TimeRanges::create())
{
}
SourceBufferPrivate::~SourceBufferPrivate() = default;
void SourceBufferPrivate::resetTimestampOffsetInTrackBuffers()
{
for (auto& trackBuffer : m_trackBufferMap.values()) {
trackBuffer.get().lastFrameTimescale = 0;
trackBuffer.get().roundedTimestampOffset = MediaTime::invalidTime();
}
}
void SourceBufferPrivate::setBufferedDirty(bool flag)
{
if (m_client)
m_client->sourceBufferPrivateBufferedDirtyChanged(flag);
}
void SourceBufferPrivate::resetTrackBuffers()
{
for (auto& trackBufferPair : m_trackBufferMap.values()) {
trackBufferPair.get().lastDecodeTimestamp = MediaTime::invalidTime();
trackBufferPair.get().greatestDecodeDuration = MediaTime::invalidTime();
trackBufferPair.get().lastFrameDuration = MediaTime::invalidTime();
trackBufferPair.get().highestPresentationTimestamp = MediaTime::invalidTime();
trackBufferPair.get().needRandomAccessFlag = true;
}
}
void SourceBufferPrivate::updateHighestPresentationTimestamp()
{
MediaTime highestTime;
for (auto& trackBuffer : m_trackBufferMap.values()) {
auto lastSampleIter = trackBuffer.get().samples.presentationOrder().rbegin();
if (lastSampleIter == trackBuffer.get().samples.presentationOrder().rend())
continue;
highestTime = std::max(highestTime, lastSampleIter->first);
}
if (m_highestPresentationTimestamp == highestTime)
return;
m_highestPresentationTimestamp = highestTime;
if (m_client)
m_client->sourceBufferPrivateHighestPresentationTimestampChanged(m_highestPresentationTimestamp);
}
void SourceBufferPrivate::setBufferedRanges(const PlatformTimeRanges& timeRanges)
{
m_buffered->ranges() = timeRanges;
setBufferedDirty(true);
}
void SourceBufferPrivate::updateBufferedFromTrackBuffers(bool sourceIsEnded)
{
// 3.1 Attributes, buffered
// https://rawgit.com/w3c/media-source/45627646344eea0170dd1cbc5a3d508ca751abb8/media-source-respec.html#dom-sourcebuffer-buffered
// 2. Let highest end time be the largest track buffer ranges end time across all the track buffers managed by this SourceBuffer object.
MediaTime highestEndTime = MediaTime::negativeInfiniteTime();
for (auto& trackBuffer : m_trackBufferMap.values()) {
if (!trackBuffer.get().buffered.length())
continue;
highestEndTime = std::max(highestEndTime, trackBuffer.get().buffered.maximumBufferedTime());
}
// NOTE: Short circuit the following if none of the TrackBuffers have buffered ranges to avoid generating
// a single range of {0, 0}.
if (highestEndTime.isNegativeInfinite()) {
setBufferedRanges(PlatformTimeRanges());
return;
}
// 3. Let intersection ranges equal a TimeRange object containing a single range from 0 to highest end time.
PlatformTimeRanges intersectionRanges { MediaTime::zeroTime(), highestEndTime };
// 4. For each audio and video track buffer managed by this SourceBuffer, run the following steps:
for (auto& trackBuffer : m_trackBufferMap.values()) {
// 4.1 Let track ranges equal the track buffer ranges for the current track buffer.
PlatformTimeRanges trackRanges = trackBuffer.get().buffered;
if (!trackRanges.length())
continue;
// 4.2 If readyState is "ended", then set the end time on the last range in track ranges to highest end time.
if (sourceIsEnded)
trackRanges.add(trackRanges.maximumBufferedTime(), highestEndTime);
// 4.3 Let new intersection ranges equal the intersection between the intersection ranges and the track ranges.
// 4.4 Replace the ranges in intersection ranges with the new intersection ranges.
intersectionRanges.intersectWith(trackRanges);
}
// 5. If intersection ranges does not contain the exact same range information as the current value of this attribute,
// then update the current value of this attribute to intersection ranges.
setBufferedRanges(intersectionRanges);
}
void SourceBufferPrivate::appendCompleted(bool parsingSucceeded, bool isEnded)
{
DEBUG_LOG(LOGIDENTIFIER);
// Resolve the changes in TrackBuffers' buffered ranges
// into the SourceBuffer's buffered ranges
updateBufferedFromTrackBuffers(isEnded);
if (m_client) {
if (!m_didReceiveSampleErrored)
m_client->sourceBufferPrivateAppendComplete(parsingSucceeded ? SourceBufferPrivateClient::AppendResult::AppendSucceeded : SourceBufferPrivateClient::AppendResult::ParsingFailed);
m_client->sourceBufferPrivateReportExtraMemoryCost(totalTrackBufferSizeInBytes());
}
}
void SourceBufferPrivate::reenqueSamples(const AtomString& trackID)
{
if (!m_isAttached)
return;
auto* trackBuffer = m_trackBufferMap.get(trackID);
if (!trackBuffer)
return;
trackBuffer->needsReenqueueing = true;
reenqueueMediaForTime(*trackBuffer, trackID, currentMediaTime());
}
void SourceBufferPrivate::seekToTime(const MediaTime& time)
{
for (auto& trackBufferPair : m_trackBufferMap) {
TrackBuffer& trackBuffer = trackBufferPair.value;
const AtomString& trackID = trackBufferPair.key;
trackBuffer.needsReenqueueing = true;
reenqueueMediaForTime(trackBuffer, trackID, time);
}
}
void SourceBufferPrivate::clearTrackBuffers()
{
for (auto& trackBufferPair : m_trackBufferMap.values()) {
trackBufferPair.get().samples.clear();
trackBufferPair.get().decodeQueue.clear();
}
}
void SourceBufferPrivate::bufferedSamplesForTrackId(const AtomString& trackId, CompletionHandler<void(Vector<String>&&)>&& completionHandler)
{
auto* trackBuffer = m_trackBufferMap.get(trackId);
if (!trackBuffer)
completionHandler({ });
Vector<String> sampleDescriptions;
for (auto& pair : trackBuffer->samples.decodeOrder())
sampleDescriptions.append(toString(*pair.second));
completionHandler(WTFMove(sampleDescriptions));
}
void SourceBufferPrivate::enqueuedSamplesForTrackID(const AtomString&, CompletionHandler<void(Vector<String>&&)>&& completionHandler)
{
completionHandler({ });
}
MediaTime SourceBufferPrivate::fastSeekTimeForMediaTime(const MediaTime& targetTime, const MediaTime& negativeThreshold, const MediaTime& positiveThreshold)
{
if (!m_client)
return targetTime;
MediaTime seekTime = targetTime;
for (auto& trackBuffer : m_trackBufferMap.values()) {
// Find the sample which contains the target time time.
auto futureSyncSampleIterator = trackBuffer.get().samples.decodeOrder().findSyncSampleAfterPresentationTime(targetTime, positiveThreshold);
auto pastSyncSampleIterator = trackBuffer.get().samples.decodeOrder().findSyncSamplePriorToPresentationTime(targetTime, negativeThreshold);
auto upperBound = trackBuffer.get().samples.decodeOrder().end();
auto lowerBound = trackBuffer.get().samples.decodeOrder().rend();
if (futureSyncSampleIterator == upperBound && pastSyncSampleIterator == lowerBound)
continue;
MediaTime futureSeekTime = MediaTime::positiveInfiniteTime();
if (futureSyncSampleIterator != upperBound) {
RefPtr<MediaSample>& sample = futureSyncSampleIterator->second;
futureSeekTime = sample->presentationTime();
}
MediaTime pastSeekTime = MediaTime::negativeInfiniteTime();
if (pastSyncSampleIterator != lowerBound) {
RefPtr<MediaSample>& sample = pastSyncSampleIterator->second;
pastSeekTime = sample->presentationTime();
}
MediaTime trackSeekTime = abs(targetTime - futureSeekTime) < abs(targetTime - pastSeekTime) ? futureSeekTime : pastSeekTime;
if (abs(targetTime - trackSeekTime) > abs(targetTime - seekTime))
seekTime = trackSeekTime;
}
return seekTime;
}
void SourceBufferPrivate::updateMinimumUpcomingPresentationTime(TrackBuffer& trackBuffer, const AtomString& trackID)
{
if (!canSetMinimumUpcomingPresentationTime(trackID))
return;
if (trackBuffer.decodeQueue.empty()) {
trackBuffer.minimumEnqueuedPresentationTime = MediaTime::invalidTime();
clearMinimumUpcomingPresentationTime(trackID);
return;
}
auto minPts = std::min_element(trackBuffer.decodeQueue.begin(), trackBuffer.decodeQueue.end(), [](auto& left, auto& right) -> bool {
return left.second->presentationTime() < right.second->presentationTime();
});
if (minPts == trackBuffer.decodeQueue.end()) {
trackBuffer.minimumEnqueuedPresentationTime = MediaTime::invalidTime();
clearMinimumUpcomingPresentationTime(trackID);
return;
}
trackBuffer.minimumEnqueuedPresentationTime = minPts->second->presentationTime();
setMinimumUpcomingPresentationTime(trackID, trackBuffer.minimumEnqueuedPresentationTime);
}
void SourceBufferPrivate::setMediaSourceEnded(bool isEnded)
{
if (m_isMediaSourceEnded == isEnded)
return;
m_isMediaSourceEnded = isEnded;
if (m_isMediaSourceEnded) {
for (auto& trackBufferPair : m_trackBufferMap) {
TrackBuffer& trackBuffer = trackBufferPair.value;
const AtomString& trackID = trackBufferPair.key;
trySignalAllSamplesInTrackEnqueued(trackBuffer, trackID);
}
}
}
void SourceBufferPrivate::trySignalAllSamplesInTrackEnqueued(TrackBuffer& trackBuffer, const AtomString& trackID)
{
if (m_isMediaSourceEnded && trackBuffer.decodeQueue.empty()) {
DEBUG_LOG(LOGIDENTIFIER, "All samples in track \"", trackID, "\" enqueued.");
allSamplesInTrackEnqueued(trackID);
}
}
void SourceBufferPrivate::provideMediaData(const AtomString& trackID)
{
auto it = m_trackBufferMap.find(trackID);
if (it == m_trackBufferMap.end())
return;
provideMediaData(it->value, trackID);
}
void SourceBufferPrivate::provideMediaData(TrackBuffer& trackBuffer, const AtomString& trackID)
{
if (!m_isAttached || isSeeking())
return;
#if !RELEASE_LOG_DISABLED
unsigned enqueuedSamples = 0;
#endif
if (trackBuffer.needsMinimumUpcomingPresentationTimeUpdating && canSetMinimumUpcomingPresentationTime(trackID)) {
trackBuffer.minimumEnqueuedPresentationTime = MediaTime::invalidTime();
clearMinimumUpcomingPresentationTime(trackID);
}
while (!trackBuffer.decodeQueue.empty()) {
if (!isReadyForMoreSamples(trackID)) {
DEBUG_LOG(LOGIDENTIFIER, "bailing early, track id ", trackID, " is not ready for more data");
notifyClientWhenReadyForMoreSamples(trackID);
break;
}
// FIXME(rdar://problem/20635969): Remove this re-entrancy protection when the aforementioned radar is resolved; protecting
// against re-entrancy introduces a small inefficency when removing appended samples from the decode queue one at a time
// rather than when all samples have been enqueued.
auto sample = trackBuffer.decodeQueue.begin()->second;
if (sample->decodeTime() > trackBuffer.enqueueDiscontinuityBoundary) {
DEBUG_LOG(LOGIDENTIFIER, "bailing early because of unbuffered gap, new sample: ", sample->decodeTime(), " >= the current discontinuity boundary: ", trackBuffer.enqueueDiscontinuityBoundary);
break;
}
// Remove the sample from the decode queue now.
trackBuffer.decodeQueue.erase(trackBuffer.decodeQueue.begin());
MediaTime samplePresentationEnd = sample->presentationTime() + sample->duration();
if (trackBuffer.highestEnqueuedPresentationTime.isInvalid() || samplePresentationEnd > trackBuffer.highestEnqueuedPresentationTime)
trackBuffer.highestEnqueuedPresentationTime = samplePresentationEnd;
trackBuffer.lastEnqueuedDecodeKey = {sample->decodeTime(), sample->presentationTime()};
trackBuffer.enqueueDiscontinuityBoundary = sample->decodeTime() + sample->duration() + discontinuityTolerance;
enqueueSample(sample.releaseNonNull(), trackID);
#if !RELEASE_LOG_DISABLED
++enqueuedSamples;
#endif
}
updateMinimumUpcomingPresentationTime(trackBuffer, trackID);
#if !RELEASE_LOG_DISABLED
DEBUG_LOG(LOGIDENTIFIER, "enqueued ", enqueuedSamples, " samples, ", static_cast<uint64_t>(trackBuffer.decodeQueue.size()), " remaining");
#endif
trySignalAllSamplesInTrackEnqueued(trackBuffer, trackID);
}
void SourceBufferPrivate::reenqueueMediaForTime(TrackBuffer& trackBuffer, const AtomString& trackID, const MediaTime& time)
{
flush(trackID);
trackBuffer.decodeQueue.clear();
trackBuffer.highestEnqueuedPresentationTime = MediaTime::invalidTime();
trackBuffer.lastEnqueuedDecodeKey = {MediaTime::invalidTime(), MediaTime::invalidTime()};
trackBuffer.enqueueDiscontinuityBoundary = time + discontinuityTolerance;
// Find the sample which contains the current presentation time.
auto currentSamplePTSIterator = trackBuffer.samples.presentationOrder().findSampleContainingPresentationTime(time);
if (currentSamplePTSIterator == trackBuffer.samples.presentationOrder().end())
currentSamplePTSIterator = trackBuffer.samples.presentationOrder().findSampleStartingOnOrAfterPresentationTime(time);
if (currentSamplePTSIterator == trackBuffer.samples.presentationOrder().end()
|| (currentSamplePTSIterator->first - time) > timeFudgeFactor())
return;
// Seach backward for the previous sync sample.
DecodeOrderSampleMap::KeyType decodeKey(currentSamplePTSIterator->second->decodeTime(), currentSamplePTSIterator->second->presentationTime());
auto currentSampleDTSIterator = trackBuffer.samples.decodeOrder().findSampleWithDecodeKey(decodeKey);
ASSERT(currentSampleDTSIterator != trackBuffer.samples.decodeOrder().end());
auto reverseCurrentSampleIter = --DecodeOrderSampleMap::reverse_iterator(currentSampleDTSIterator);
auto reverseLastSyncSampleIter = trackBuffer.samples.decodeOrder().findSyncSamplePriorToDecodeIterator(reverseCurrentSampleIter);
if (reverseLastSyncSampleIter == trackBuffer.samples.decodeOrder().rend())
return;
// Fill the decode queue with the non-displaying samples.
for (auto iter = reverseLastSyncSampleIter; iter != reverseCurrentSampleIter; --iter) {
auto copy = iter->second->createNonDisplayingCopy();
DecodeOrderSampleMap::KeyType decodeKey(copy->decodeTime(), copy->presentationTime());
trackBuffer.decodeQueue.insert(DecodeOrderSampleMap::MapType::value_type(decodeKey, WTFMove(copy)));
}
// Fill the decode queue with the remaining samples.
for (auto iter = currentSampleDTSIterator; iter != trackBuffer.samples.decodeOrder().end(); ++iter)
trackBuffer.decodeQueue.insert(*iter);
provideMediaData(trackBuffer, trackID);
trackBuffer.needsReenqueueing = false;
}
void SourceBufferPrivate::reenqueueMediaIfNeeded(const MediaTime& currentTime)
{
for (auto& trackBufferPair : m_trackBufferMap) {
TrackBuffer& trackBuffer = trackBufferPair.value;
const AtomString& trackID = trackBufferPair.key;
if (trackBuffer.needsReenqueueing) {
DEBUG_LOG(LOGIDENTIFIER, "reenqueuing at time ", currentTime);
reenqueueMediaForTime(trackBuffer, trackID, currentTime);
} else
provideMediaData(trackBuffer, trackID);
}
}
static WARN_UNUSED_RETURN bool decodeTimeComparator(const PresentationOrderSampleMap::MapType::value_type& a, const PresentationOrderSampleMap::MapType::value_type& b)
{
return a.second->decodeTime() < b.second->decodeTime();
}
#if !RELEASE_LOG_DISABLED
static PlatformTimeRanges removeSamplesFromTrackBuffer(const DecodeOrderSampleMap::MapType& samples, SourceBufferPrivate::TrackBuffer& trackBuffer, const SourceBufferPrivate* sourceBufferPrivate, const char* logPrefix)
#else
static PlatformTimeRanges removeSamplesFromTrackBuffer(const DecodeOrderSampleMap::MapType& samples, SourceBufferPrivate::TrackBuffer& trackBuffer)
#endif
{
#if !RELEASE_LOG_DISABLED
MediaTime earliestSample = MediaTime::positiveInfiniteTime();
MediaTime latestSample = MediaTime::zeroTime();
uint64_t bytesRemoved = 0;
auto logIdentifier = Logger::LogSiteIdentifier(sourceBufferPrivate->logClassName(), logPrefix, sourceBufferPrivate->logIdentifier());
auto& logger = sourceBufferPrivate->logger();
auto willLog = logger.willLog(sourceBufferPrivate->logChannel(), WTFLogLevel::Debug);
#endif
PlatformTimeRanges erasedRanges;
for (const auto& sampleIt : samples) {
const DecodeOrderSampleMap::KeyType& decodeKey = sampleIt.first;
#if !RELEASE_LOG_DISABLED
uint64_t startBufferSize = trackBuffer.samples.sizeInBytes();
#endif
const RefPtr<MediaSample>& sample = sampleIt.second;
#if !RELEASE_LOG_DISABLED
if (willLog)
logger.debug(sourceBufferPrivate->logChannel(), logIdentifier, "removing sample ", *sampleIt.second);
#endif
// Remove the erased samples from the TrackBuffer sample map.
trackBuffer.samples.removeSample(sample.get());
// Also remove the erased samples from the TrackBuffer decodeQueue.
trackBuffer.decodeQueue.erase(decodeKey);
auto startTime = sample->presentationTime();
auto endTime = startTime + sample->duration();
erasedRanges.add(startTime, endTime);
#if !RELEASE_LOG_DISABLED
bytesRemoved += startBufferSize - trackBuffer.samples.sizeInBytes();
if (startTime < earliestSample)
earliestSample = startTime;
if (endTime > latestSample)
latestSample = endTime;
#endif
}
// Because we may have added artificial padding in the buffered ranges when adding samples, we may
// need to remove that padding when removing those same samples. Walk over the erased ranges looking
// for unbuffered areas and expand erasedRanges to encompass those areas.
PlatformTimeRanges additionalErasedRanges;
for (unsigned i = 0; i < erasedRanges.length(); ++i) {
auto erasedStart = erasedRanges.start(i);
auto erasedEnd = erasedRanges.end(i);
auto startIterator = trackBuffer.samples.presentationOrder().reverseFindSampleBeforePresentationTime(erasedStart);
if (startIterator == trackBuffer.samples.presentationOrder().rend())
additionalErasedRanges.add(MediaTime::zeroTime(), erasedStart);
else {
auto& previousSample = *startIterator->second;
if (previousSample.presentationTime() + previousSample.duration() < erasedStart)
additionalErasedRanges.add(previousSample.presentationTime() + previousSample.duration(), erasedStart);
}
auto endIterator = trackBuffer.samples.presentationOrder().findSampleStartingOnOrAfterPresentationTime(erasedEnd);
if (endIterator == trackBuffer.samples.presentationOrder().end())
additionalErasedRanges.add(erasedEnd, MediaTime::positiveInfiniteTime());
else {
auto& nextSample = *endIterator->second;
if (nextSample.presentationTime() > erasedEnd)
additionalErasedRanges.add(erasedEnd, nextSample.presentationTime());
}
}
if (additionalErasedRanges.length())
erasedRanges.unionWith(additionalErasedRanges);
#if !RELEASE_LOG_DISABLED
if (bytesRemoved && willLog)
logger.debug(sourceBufferPrivate->logChannel(), logIdentifier, "removed ", bytesRemoved, ", start = ", earliestSample, ", end = ", latestSample);
#endif
return erasedRanges;
}
void SourceBufferPrivate::removeCodedFrames(const MediaTime& start, const MediaTime& end, const MediaTime& currentTime, bool isEnded, CompletionHandler<void()>&& completionHandler)
{
ASSERT(start < end);
if (start >= end) {
completionHandler();
return;
}
// 3.5.9 Coded Frame Removal Algorithm
// https://dvcs.w3.org/hg/html-media/raw-file/tip/media-source/media-source.html#sourcebuffer-coded-frame-removal
// 1. Let start be the starting presentation timestamp for the removal range.
// 2. Let end be the end presentation timestamp for the removal range.
// 3. For each track buffer in this source buffer, run the following steps:
for (auto& trackBufferKeyValue : m_trackBufferMap) {
TrackBuffer& trackBuffer = trackBufferKeyValue.value;
AtomString trackID = trackBufferKeyValue.key;
// 3.1. Let remove end timestamp be the current value of duration
// 3.2 If this track buffer has a random access point timestamp that is greater than or equal to end, then update
// remove end timestamp to that random access point timestamp.
// NOTE: Step 3.2 will be incorrect for any random access point timestamp whose decode time is later than the sample at end,
// but whose presentation time is less than the sample at end. Skip this step until step 3.3 below.
// NOTE: To handle MediaSamples which may be an amalgamation of multiple shorter samples, find samples whose presentation
// interval straddles the start and end times, and divide them if possible:
auto divideSampleIfPossibleAtPresentationTime = [&] (const MediaTime& time) {
auto sampleIterator = trackBuffer.samples.presentationOrder().findSampleContainingPresentationTime(time);
if (sampleIterator == trackBuffer.samples.presentationOrder().end())
return;
RefPtr<MediaSample> sample = sampleIterator->second;
if (!sample->isDivisable())
return;
std::pair<RefPtr<MediaSample>, RefPtr<MediaSample>> replacementSamples = sample->divide(time);
if (!replacementSamples.first || !replacementSamples.second)
return;
DEBUG_LOG(LOGIDENTIFIER, "splitting sample ", *sample, " into ", *replacementSamples.first, " and ", *replacementSamples.second);
trackBuffer.samples.removeSample(sample.get());
trackBuffer.samples.addSample(*replacementSamples.first);
trackBuffer.samples.addSample(*replacementSamples.second);
};
divideSampleIfPossibleAtPresentationTime(start);
divideSampleIfPossibleAtPresentationTime(end);
auto removePresentationStart = trackBuffer.samples.presentationOrder().findSampleContainingOrAfterPresentationTime(start);
auto removePresentationEnd = trackBuffer.samples.presentationOrder().findSampleStartingOnOrAfterPresentationTime(end);
if (removePresentationStart == removePresentationEnd)
continue;
// 3.3 Remove all media data, from this track buffer, that contain starting timestamps greater than or equal to
// start and less than the remove end timestamp.
// NOTE: frames must be removed in decode order, so that all dependant frames between the frame to be removed
// and the next sync sample frame are removed. But we must start from the first sample in decode order, not
// presentation order.
auto minmaxDecodeTimeIterPair = std::minmax_element(removePresentationStart, removePresentationEnd, decodeTimeComparator);
auto& firstSample = *minmaxDecodeTimeIterPair.first->second;
auto& lastSample = *minmaxDecodeTimeIterPair.second->second;
auto removeDecodeStart = trackBuffer.samples.decodeOrder().findSampleWithDecodeKey({firstSample.decodeTime(), firstSample.presentationTime()});
auto removeDecodeLast = trackBuffer.samples.decodeOrder().findSampleWithDecodeKey({lastSample.decodeTime(), lastSample.presentationTime()});
auto removeDecodeEnd = trackBuffer.samples.decodeOrder().findSyncSampleAfterDecodeIterator(removeDecodeLast);
DecodeOrderSampleMap::MapType erasedSamples(removeDecodeStart, removeDecodeEnd);
#if !RELEASE_LOG_DISABLED
PlatformTimeRanges erasedRanges = removeSamplesFromTrackBuffer(erasedSamples, trackBuffer, this, "removeCodedFrames");
#else
PlatformTimeRanges erasedRanges = removeSamplesFromTrackBuffer(erasedSamples, trackBuffer);
#endif
// Only force the TrackBuffer to re-enqueue if the removed ranges overlap with enqueued and possibly
// not yet displayed samples.
if (trackBuffer.highestEnqueuedPresentationTime.isValid() && currentTime < trackBuffer.highestEnqueuedPresentationTime) {
PlatformTimeRanges possiblyEnqueuedRanges(currentTime, trackBuffer.highestEnqueuedPresentationTime);
possiblyEnqueuedRanges.intersectWith(erasedRanges);
if (possiblyEnqueuedRanges.length()) {
trackBuffer.needsReenqueueing = true;
DEBUG_LOG(LOGIDENTIFIER, "the range in removeCodedFrames() includes already enqueued samples, reenqueueing from ", currentTime);
reenqueueMediaForTime(trackBuffer, trackID, currentTime);
}
}
erasedRanges.invert();
trackBuffer.buffered.intersectWith(erasedRanges);
setBufferedDirty(true);
// 3.4 If this object is in activeSourceBuffers, the current playback position is greater than or equal to start
// and less than the remove end timestamp, and HTMLMediaElement.readyState is greater than HAVE_METADATA, then set
// the HTMLMediaElement.readyState attribute to HAVE_METADATA and stall playback.
if (isActive() && currentTime >= start && currentTime < end && readyState() > MediaPlayer::ReadyState::HaveMetadata)
setReadyState(MediaPlayer::ReadyState::HaveMetadata);
}
updateBufferedFromTrackBuffers(isEnded);
// 4. If buffer full flag equals true and this object is ready to accept more bytes, then set the buffer full flag to false.
// No-op
updateHighestPresentationTimestamp();
LOG(Media, "SourceBuffer::removeCodedFrames(%p) - buffered = %s", this, toString(m_buffered->ranges()).utf8().data());
completionHandler();
}
void SourceBufferPrivate::evictCodedFrames(uint64_t newDataSize, uint64_t maximumBufferSize, const MediaTime& currentTime, const MediaTime& duration, bool isEnded)
{
// 3.5.13 Coded Frame Eviction Algorithm
// http://www.w3.org/TR/media-source/#sourcebuffer-coded-frame-eviction
if (!m_isAttached)
return;
// This algorithm is run to free up space in this source buffer when new data is appended.
// 1. Let new data equal the data that is about to be appended to this SourceBuffer.
// 2. If the buffer full flag equals false, then abort these steps.
if (!isBufferFullFor(newDataSize, maximumBufferSize))
return;
// 3. Let removal ranges equal a list of presentation time ranges that can be evicted from
// the presentation to make room for the new data.
// NOTE: begin by removing data from the beginning of the buffered ranges, 30 seconds at
// a time, up to 30 seconds before currentTime.
MediaTime thirtySeconds = MediaTime(30, 1);
MediaTime maximumRangeEnd = currentTime - thirtySeconds;
#if !RELEASE_LOG_DISABLED
uint64_t initialBufferedSize = totalTrackBufferSizeInBytes();
DEBUG_LOG(LOGIDENTIFIER, "currentTime = ", currentTime, ", require ", initialBufferedSize + newDataSize, " bytes, maximum buffer size is ", maximumBufferSize);
#endif
MediaTime rangeStart = MediaTime::invalidTime();
for (auto& trackBuffer : m_trackBufferMap.values()) {
auto iter = trackBuffer.get().samples.presentationOrder().findSampleContainingOrAfterPresentationTime(MediaTime::zeroTime());
if (iter != trackBuffer.get().samples.presentationOrder().end()) {
MediaTime startTime = iter->first;
if (rangeStart.isInvalid() || startTime < rangeStart)
rangeStart = startTime;
}
}
if (rangeStart.isInvalid())
rangeStart = MediaTime::zeroTime();
MediaTime rangeEnd = rangeStart + thirtySeconds;
while (rangeStart < maximumRangeEnd) {
// 4. For each range in removal ranges, run the coded frame removal algorithm with start and
// end equal to the removal range start and end timestamp respectively.
removeCodedFrames(rangeStart, std::min(rangeEnd, maximumRangeEnd), currentTime, isEnded);
if (!isBufferFullFor(newDataSize, maximumBufferSize)) {
break;
}
rangeStart += thirtySeconds;
rangeEnd += thirtySeconds;
}
if (!isBufferFullFor(newDataSize, maximumBufferSize)) {
#if !RELEASE_LOG_DISABLED
DEBUG_LOG(LOGIDENTIFIER, "evicted ", initialBufferedSize - totalTrackBufferSizeInBytes());
#endif
return;
}
// If there still isn't enough free space and there buffers in time ranges after the current range (ie. there is a gap after
// the current buffered range), delete 30 seconds at a time from duration back to the current time range or 30 seconds after
// currenTime whichever we hit first.
auto buffered = m_buffered->ranges().copyWithEpsilon(timeFudgeFactor());
uint64_t currentTimeRange = buffered.findWithEpsilon(currentTime, timeFudgeFactor());
if (!buffered.length() || currentTimeRange == buffered.length() - 1) {
#if !RELEASE_LOG_DISABLED
ERROR_LOG(LOGIDENTIFIER, "FAILED to free enough after evicting ", initialBufferedSize - totalTrackBufferSizeInBytes());
#endif
return;
}
MediaTime minimumRangeStart =
currentTimeRange == notFound ? currentTime + thirtySeconds : std::max(currentTime + thirtySeconds, buffered.end(currentTimeRange));
rangeEnd = duration;
if (!rangeEnd.isFinite()) {
rangeEnd = buffered.maximumBufferedTime();
#if !RELEASE_LOG_DISABLED
DEBUG_LOG(LOGIDENTIFIER, "MediaSource duration is not a finite value, using maximum buffered time: ", rangeEnd);
#endif
}
rangeStart = rangeEnd - thirtySeconds;
while (rangeEnd > minimumRangeStart) {
// 4. For each range in removal ranges, run the coded frame removal algorithm with start and
// end equal to the removal range start and end timestamp respectively.
removeCodedFrames(std::max(minimumRangeStart, rangeStart), rangeEnd, currentTime, isEnded);
if (!isBufferFullFor(newDataSize, maximumBufferSize)) {
break;
}
rangeStart -= thirtySeconds;
rangeEnd -= thirtySeconds;
}
#if !RELEASE_LOG_DISABLED
if (isBufferFullFor(newDataSize, maximumBufferSize))
ERROR_LOG(LOGIDENTIFIER, "FAILED to free enough after evicting ", initialBufferedSize - totalTrackBufferSizeInBytes());
else
DEBUG_LOG(LOGIDENTIFIER, "evicted ", initialBufferedSize - totalTrackBufferSizeInBytes());
#endif
}
bool SourceBufferPrivate::isBufferFullFor(uint64_t requiredSize, uint64_t maximumBufferSize)
{
auto totalRequired = checkedSum<uint64_t>(totalTrackBufferSizeInBytes(), requiredSize);
if (totalRequired.hasOverflowed())
return true;
return totalRequired >= maximumBufferSize;
}
uint64_t SourceBufferPrivate::totalTrackBufferSizeInBytes() const
{
uint64_t totalSizeInBytes = 0;
for (auto& trackBuffer : m_trackBufferMap.values())
totalSizeInBytes += trackBuffer.get().samples.sizeInBytes();
return totalSizeInBytes;
}
void SourceBufferPrivate::addTrackBuffer(const AtomString& trackId, RefPtr<MediaDescription>&& description)
{
ASSERT(!m_trackBufferMap.contains(trackId));
auto& trackBuffer = m_trackBufferMap.add(trackId, makeUniqueRef<TrackBuffer>()).iterator->value;
// 5.2.9 Add the track description for this track to the track buffer.
trackBuffer.get().description = description;
m_hasAudio = m_hasAudio || trackBuffer.get().description->isAudio();
m_hasVideo = m_hasVideo || trackBuffer.get().description->isVideo();
}
void SourceBufferPrivate::updateTrackIds(Vector<std::pair<AtomString, AtomString>>&& trackIdPairs)
{
auto trackBufferMap = std::exchange(m_trackBufferMap, { });
for (auto& trackIdPair : trackIdPairs) {
auto oldId = trackIdPair.first;
auto newId = trackIdPair.second;
ASSERT(oldId != newId);
auto trackBuffer = trackBufferMap.take(oldId);
if (!trackBuffer)
continue;
m_trackBufferMap.add(newId, makeUniqueRefFromNonNullUniquePtr(WTFMove(trackBuffer)));
}
}
void SourceBufferPrivate::setAllTrackBuffersNeedRandomAccess()
{
for (auto& trackBuffer : m_trackBufferMap.values())
trackBuffer.get().needRandomAccessFlag = true;
}
void SourceBufferPrivate::didReceiveInitializationSegment(SourceBufferPrivateClient::InitializationSegment&& segment, CompletionHandler<void()>&& completionHandler)
{
if (!m_client) {
completionHandler();
return;
}
if (m_receivedFirstInitializationSegment && !validateInitializationSegment(segment)) {
m_client->sourceBufferPrivateAppendError(true);
return;
}
m_client->sourceBufferPrivateDidReceiveInitializationSegment(WTFMove(segment), WTFMove(completionHandler));
m_receivedFirstInitializationSegment = true;
m_pendingInitializationSegmentForChangeType = false;
}
bool SourceBufferPrivate::validateInitializationSegment(const SourceBufferPrivateClient::InitializationSegment& segment)
{
// * If more than one track for a single type are present (ie 2 audio tracks), then the Track
// IDs match the ones in the first initialization segment.
if (segment.audioTracks.size() >= 2) {
for (auto& audioTrackInfo : segment.audioTracks) {
if (!m_trackBufferMap.contains(audioTrackInfo.track->id()))
return false;
}
}
if (segment.videoTracks.size() >= 2) {
for (auto& videoTrackInfo : segment.videoTracks) {
if (!m_trackBufferMap.contains(videoTrackInfo.track->id()))
return false;
}
}
if (segment.textTracks.size() >= 2) {
for (auto& textTrackInfo : segment.videoTracks) {
if (!m_trackBufferMap.contains(textTrackInfo.track->id()))
return false;
}
}
return true;
}
void SourceBufferPrivate::didReceiveSample(Ref<MediaSample>&& originalSample)
{
if (!m_isAttached || m_didReceiveSampleErrored)
return;
// 3.5.1 Segment Parser Loop
// 6.1 If the first initialization segment received flag is false, (Note: Issue # 155 & changeType()
// algorithm) or the pending initialization segment for changeType flag is true, (End note)
// then run the append error algorithm
// with the decode error parameter set to true and abort this algorithm.
// Note: current design makes SourceBuffer somehow ignorant of append state - it's more a thing
// of SourceBufferPrivate. That's why this check can't really be done in appendInternal.
// unless we force some kind of design with state machine switching.
if ((!m_receivedFirstInitializationSegment || m_pendingInitializationSegmentForChangeType) && m_client) {
m_client->sourceBufferPrivateAppendError(true);
m_didReceiveSampleErrored = true;
return;
}
// 3.5.8 Coded Frame Processing
// http://www.w3.org/TR/media-source/#sourcebuffer-coded-frame-processing
// When complete coded frames have been parsed by the segment parser loop then the following steps
// are run:
// 1. For each coded frame in the media segment run the following steps:
// 1.1. Loop Top
Ref<MediaSample> sample = WTFMove(originalSample);
do {
MediaTime presentationTimestamp;
MediaTime decodeTimestamp;
// NOTE: this is out-of-order, but we need the timescale from the
// sample's duration for timestamp generation.
// 1.2 Let frame duration be a double precision floating point representation of the coded frame's
// duration in seconds.
MediaTime frameDuration = sample->duration();
if (m_shouldGenerateTimestamps) {
// ↳ If generate timestamps flag equals true:
// 1. Let presentation timestamp equal 0.
// NOTE: Use the duration timscale for the presentation timestamp, as this will eliminate
// timescale rounding when generating timestamps.
presentationTimestamp = { 0, frameDuration.timeScale() };
// 2. Let decode timestamp equal 0.
decodeTimestamp = { 0, frameDuration.timeScale() };
} else {
// ↳ Otherwise:
// 1. Let presentation timestamp be a double precision floating point representation of
// the coded frame's presentation timestamp in seconds.
presentationTimestamp = sample->presentationTime();
// 2. Let decode timestamp be a double precision floating point representation of the coded frame's
// decode timestamp in seconds.
decodeTimestamp = sample->decodeTime();
}
// 1.3 If mode equals "sequence" and group start timestamp is set, then run the following steps:
if (m_appendMode == SourceBufferAppendMode::Sequence && m_groupStartTimestamp.isValid()) {
// 1.3.1 Set timestampOffset equal to group start timestamp - presentation timestamp.
m_timestampOffset = m_groupStartTimestamp - presentationTimestamp;
for (auto& trackBuffer : m_trackBufferMap.values()) {
trackBuffer.get().lastFrameTimescale = 0;
trackBuffer.get().roundedTimestampOffset = MediaTime::invalidTime();
}
// 1.3.2 Set group end timestamp equal to group start timestamp.
m_groupEndTimestamp = m_groupStartTimestamp;
// 1.3.3 Set the need random access point flag on all track buffers to true.
for (auto& trackBuffer : m_trackBufferMap.values())
trackBuffer.get().needRandomAccessFlag = true;
// 1.3.4 Unset group start timestamp.
m_groupStartTimestamp = MediaTime::invalidTime();
}
// NOTE: this is out-of-order, but we need TrackBuffer to be able to cache the results of timestamp offset rounding
// 1.5 Let track buffer equal the track buffer that the coded frame will be added to.
AtomString trackID = sample->trackID();
auto it = m_trackBufferMap.find(trackID);
if (it == m_trackBufferMap.end()) {
// The client managed to append a sample with a trackID not present in the initialization
// segment. This would be a good place to post an message to the developer console.
m_client->sourceBufferPrivateDidDropSample();
return;
}
TrackBuffer& trackBuffer = it->value;
MediaTime microsecond(1, 1000000);
auto roundTowardsTimeScaleWithRoundingMargin = [] (const MediaTime& time, uint32_t timeScale, const MediaTime& roundingMargin) {
while (true) {
MediaTime roundedTime = time.toTimeScale(timeScale);
if (abs(roundedTime - time) < roundingMargin || timeScale >= MediaTime::MaximumTimeScale)
return roundedTime;
if (!WTF::safeMultiply(timeScale, 2, timeScale) || timeScale > MediaTime::MaximumTimeScale)
timeScale = MediaTime::MaximumTimeScale;
}
};
// 1.4 If timestampOffset is not 0, then run the following steps:
if (m_timestampOffset) {
if (!trackBuffer.roundedTimestampOffset.isValid() || presentationTimestamp.timeScale() != trackBuffer.lastFrameTimescale) {
trackBuffer.lastFrameTimescale = presentationTimestamp.timeScale();
trackBuffer.roundedTimestampOffset = roundTowardsTimeScaleWithRoundingMargin(m_timestampOffset, trackBuffer.lastFrameTimescale, microsecond);
}
// 1.4.1 Add timestampOffset to the presentation timestamp.
presentationTimestamp += trackBuffer.roundedTimestampOffset;
// 1.4.2 Add timestampOffset to the decode timestamp.
decodeTimestamp += trackBuffer.roundedTimestampOffset;
}
// 1.6 ↳ If last decode timestamp for track buffer is set and decode timestamp is less than last
// decode timestamp:
// OR
// ↳ If last decode timestamp for track buffer is set and the difference between decode timestamp and
// last decode timestamp is greater than 2 times last frame duration:
MediaTime decodeDurationToCheck = trackBuffer.greatestDecodeDuration;
if (decodeDurationToCheck.isValid() && trackBuffer.lastFrameDuration.isValid()
&& (trackBuffer.lastFrameDuration > decodeDurationToCheck))
decodeDurationToCheck = trackBuffer.lastFrameDuration;
if (trackBuffer.lastDecodeTimestamp.isValid() && (decodeTimestamp < trackBuffer.lastDecodeTimestamp
|| (decodeDurationToCheck.isValid() && abs(decodeTimestamp - trackBuffer.lastDecodeTimestamp) > (decodeDurationToCheck * 2)))) {
// 1.6.1:
if (m_appendMode == SourceBufferAppendMode::Segments) {
// ↳ If mode equals "segments":
// Set group end timestamp to presentation timestamp.
m_groupEndTimestamp = presentationTimestamp;
} else {
// ↳ If mode equals "sequence":
// Set group start timestamp equal to the group end timestamp.
m_groupStartTimestamp = m_groupEndTimestamp;
}
for (auto& trackBuffer : m_trackBufferMap.values()) {
// 1.6.2 Unset the last decode timestamp on all track buffers.
trackBuffer.get().lastDecodeTimestamp = MediaTime::invalidTime();
// 1.6.3 Unset the last frame duration on all track buffers.
trackBuffer.get().greatestDecodeDuration = MediaTime::invalidTime();
trackBuffer.get().lastFrameDuration = MediaTime::invalidTime();
// 1.6.4 Unset the highest presentation timestamp on all track buffers.
trackBuffer.get().highestPresentationTimestamp = MediaTime::invalidTime();
// 1.6.5 Set the need random access point flag on all track buffers to true.
trackBuffer.get().needRandomAccessFlag = true;
}
// 1.6.6 Jump to the Loop Top step above to restart processing of the current coded frame.
continue;
}
if (m_appendMode == SourceBufferAppendMode::Sequence) {
// Use the generated timestamps instead of the sample's timestamps.
sample->setTimestamps(presentationTimestamp, decodeTimestamp);
} else if (trackBuffer.roundedTimestampOffset) {
// Reflect the timestamp offset into the sample.
sample->offsetTimestampsBy(trackBuffer.roundedTimestampOffset);
}
DEBUG_LOG(LOGIDENTIFIER, sample.get());
// 1.7 Let frame end timestamp equal the sum of presentation timestamp and frame duration.
MediaTime frameEndTimestamp = presentationTimestamp + frameDuration;
// 1.8 If presentation timestamp is less than appendWindowStart, then set the need random access
// point flag to true, drop the coded frame, and jump to the top of the loop to start processing
// the next coded frame.
// 1.9 If frame end timestamp is greater than appendWindowEnd, then set the need random access
// point flag to true, drop the coded frame, and jump to the top of the loop to start processing
// the next coded frame.
if (presentationTimestamp < m_appendWindowStart || frameEndTimestamp > m_appendWindowEnd) {
// 1.8 Note.
// Some implementations MAY choose to collect some of these coded frames with presentation
// timestamp less than appendWindowStart and use them to generate a splice at the first coded
// frame that has a presentation timestamp greater than or equal to appendWindowStart even if
// that frame is not a random access point. Supporting this requires multiple decoders or
// faster than real-time decoding so for now this behavior will not be a normative
// requirement.
// 1.9 Note.
// Some implementations MAY choose to collect coded frames with presentation timestamp less
// than appendWindowEnd and frame end timestamp greater than appendWindowEnd and use them to
// generate a splice across the portion of the collected coded frames within the append
// window at time of collection, and the beginning portion of later processed frames which
// only partially overlap the end of the collected coded frames. Supporting this requires
// multiple decoders or faster than real-time decoding so for now this behavior will not be a
// normative requirement. In conjunction with collecting coded frames that span
// appendWindowStart, implementations MAY thus support gapless audio splicing.
// Audio MediaSamples are typically made of packed audio samples. Trim sample to make it fit within the appendWindow.
if (sample->isDivisable()) {
std::pair<RefPtr<MediaSample>, RefPtr<MediaSample>> replacementSamples = sample->divide(m_appendWindowStart);
if (replacementSamples.second) {
ASSERT(replacementSamples.second->presentationTime() >= m_appendWindowStart);
replacementSamples = replacementSamples.second->divide(m_appendWindowEnd, MediaSample::UseEndTime::Use);
if (replacementSamples.first) {
sample = replacementSamples.first.releaseNonNull();
ASSERT(sample->presentationTime() >= m_appendWindowStart && sample->presentationTime() + sample->duration() <= m_appendWindowEnd);
if (m_appendMode != SourceBufferAppendMode::Sequence && trackBuffer.roundedTimestampOffset)
sample->offsetTimestampsBy(-trackBuffer.roundedTimestampOffset);
continue;
}
}
}
trackBuffer.needRandomAccessFlag = true;
m_client->sourceBufferPrivateDidDropSample();
return;
}
// If the decode timestamp is less than the presentation start time, then run the end of stream
// algorithm with the error parameter set to "decode", and abort these steps.
// NOTE: Until <https://www.w3.org/Bugs/Public/show_bug.cgi?id=27487> is resolved, we will only check
// the presentation timestamp.
MediaTime presentationStartTime = MediaTime::zeroTime();
if (presentationTimestamp < presentationStartTime) {
ERROR_LOG(LOGIDENTIFIER, "failing because presentationTimestamp (", presentationTimestamp, ") < presentationStartTime (", presentationStartTime, ")");
m_client->sourceBufferPrivateStreamEndedWithDecodeError();
return;
}
// 1.10 If the need random access point flag on track buffer equals true, then run the following steps:
if (trackBuffer.needRandomAccessFlag) {
// 1.11.1 If the coded frame is not a random access point, then drop the coded frame and jump
// to the top of the loop to start processing the next coded frame.
if (!sample->isSync()) {
m_client->sourceBufferPrivateDidDropSample();
return;
}
// 1.11.2 Set the need random access point flag on track buffer to false.
trackBuffer.needRandomAccessFlag = false;
}
// 1.11 Let spliced audio frame be an unset variable for holding audio splice information
// 1.12 Let spliced timed text frame be an unset variable for holding timed text splice information
// FIXME: Add support for sample splicing.
SampleMap erasedSamples;
// 1.13 If last decode timestamp for track buffer is unset and presentation timestamp
// falls within the presentation interval of a coded frame in track buffer, then run the
// following steps:
if (trackBuffer.lastDecodeTimestamp.isInvalid()) {
auto iter = trackBuffer.samples.presentationOrder().findSampleContainingPresentationTime(presentationTimestamp);
if (iter != trackBuffer.samples.presentationOrder().end()) {
// 1.13.1 Let overlapped frame be the coded frame in track buffer that matches the condition above.
RefPtr<MediaSample> overlappedFrame = iter->second;
// 1.13.2 If track buffer contains audio coded frames:
// Run the audio splice frame algorithm and if a splice frame is returned, assign it to
// spliced audio frame.
// FIXME: Add support for sample splicing.
// If track buffer contains video coded frames:
if (trackBuffer.description && trackBuffer.description->isVideo()) {
// 1.13.2.1 Let overlapped frame presentation timestamp equal the presentation timestamp
// of overlapped frame.
MediaTime overlappedFramePresentationTimestamp = overlappedFrame->presentationTime();
// 1.13.2.2 Let remove window timestamp equal overlapped frame presentation timestamp
// plus 1 microsecond.
MediaTime removeWindowTimestamp = overlappedFramePresentationTimestamp + microsecond;
// 1.13.2.3 If the presentation timestamp is less than the remove window timestamp,
// then remove overlapped frame and any coded frames that depend on it from track buffer.
if (presentationTimestamp < removeWindowTimestamp)
erasedSamples.addSample(*iter->second);
}
// If track buffer contains timed text coded frames:
// Run the text splice frame algorithm and if a splice frame is returned, assign it to spliced timed text frame.
// FIXME: Add support for sample splicing.
}
}
// 1.14 Remove existing coded frames in track buffer:
// If highest presentation timestamp for track buffer is not set:
if (trackBuffer.highestPresentationTimestamp.isInvalid()) {
// Remove all coded frames from track buffer that have a presentation timestamp greater than or
// equal to presentation timestamp and less than frame end timestamp.
auto iterPair = trackBuffer.samples.presentationOrder().findSamplesBetweenPresentationTimes(presentationTimestamp, frameEndTimestamp);
if (iterPair.first != trackBuffer.samples.presentationOrder().end())
erasedSamples.addRange(iterPair.first, iterPair.second);
}
// When appending media containing B-frames (media whose samples' presentation timestamps
// do not increase monotonically, the prior erase steps could leave a sample in the trackBuffer
// which will be disconnected from its previous I-frame. If the incoming frame is an I-frame,
// remove all samples in decode order between the incoming I-frame's decode timestamp and the
// next I-frame. See <https://github.com/w3c/media-source/issues/187> for a discussion of what
// the how the MSE specification should handlie this secnario.
do {
if (!sample->isSync())
break;
DecodeOrderSampleMap::KeyType decodeKey(sample->decodeTime(), sample->presentationTime());
auto nextSampleInDecodeOrder = trackBuffer.samples.decodeOrder().findSampleAfterDecodeKey(decodeKey);
if (nextSampleInDecodeOrder == trackBuffer.samples.decodeOrder().end())
break;
if (nextSampleInDecodeOrder->second->isSync())
break;
auto nextSyncSample = trackBuffer.samples.decodeOrder().findSyncSampleAfterDecodeIterator(nextSampleInDecodeOrder);
INFO_LOG(LOGIDENTIFIER, "Discovered out-of-order frames, from: ", *nextSampleInDecodeOrder->second, " to: ", (nextSyncSample == trackBuffer.samples.decodeOrder().end() ? "[end]"_s : toString(*nextSyncSample->second)));
erasedSamples.addRange(nextSampleInDecodeOrder, nextSyncSample);
} while (false);
// There are many files out there where the frame times are not perfectly contiguous and may have small overlaps
// between the beginning of a frame and the end of the previous one; therefore a tolerance is needed whenever
// durations are considered.
// For instance, most WebM files are muxed rounded to the millisecond (the default TimecodeScale of the format)
// but their durations use a finer timescale (causing a sub-millisecond overlap). More rarely, there are also
// MP4 files with slightly off tfdt boxes, presenting a similar problem at the beginning of each fragment.
const MediaTime contiguousFrameTolerance = MediaTime(1, 1000);
// If highest presentation timestamp for track buffer is set and less than or equal to presentation timestamp
if (trackBuffer.highestPresentationTimestamp.isValid() && trackBuffer.highestPresentationTimestamp - contiguousFrameTolerance <= presentationTimestamp) {
// Remove all coded frames from track buffer that have a presentation timestamp greater than highest
// presentation timestamp and less than or equal to frame end timestamp.
do {
// NOTE: Searching from the end of the trackBuffer will be vastly more efficient if the search range is
// near the end of the buffered range. Use a linear-backwards search if the search range is within one
// frame duration of the end:
unsigned bufferedLength = trackBuffer.buffered.length();
if (!bufferedLength)
break;
MediaTime highestBufferedTime = trackBuffer.buffered.maximumBufferedTime();
MediaTime eraseBeginTime = trackBuffer.highestPresentationTimestamp - contiguousFrameTolerance;
MediaTime eraseEndTime = frameEndTimestamp - contiguousFrameTolerance;
PresentationOrderSampleMap::iterator_range range;
if (highestBufferedTime - trackBuffer.highestPresentationTimestamp < trackBuffer.lastFrameDuration) {
// If the new frame is at the end of the buffered ranges, perform a sequential scan from end (O(1)).
range = trackBuffer.samples.presentationOrder().findSamplesBetweenPresentationTimesFromEnd(eraseBeginTime, eraseEndTime);
} else {
// In any other case, perform a binary search (O(log(n)).
range = trackBuffer.samples.presentationOrder().findSamplesBetweenPresentationTimes(eraseBeginTime, eraseEndTime);
}
if (range.first != trackBuffer.samples.presentationOrder().end())
erasedSamples.addRange(range.first, range.second);
} while (false);
}
// 1.15 Remove decoding dependencies of the coded frames removed in the previous step:
DecodeOrderSampleMap::MapType dependentSamples;
if (!erasedSamples.empty()) {
// If detailed information about decoding dependencies is available:
// FIXME: Add support for detailed dependency information
// Otherwise: Remove all coded frames between the coded frames removed in the previous step
// and the next random access point after those removed frames.
auto firstDecodeIter = trackBuffer.samples.decodeOrder().findSampleWithDecodeKey(erasedSamples.decodeOrder().begin()->first);
auto lastDecodeIter = trackBuffer.samples.decodeOrder().findSampleWithDecodeKey(erasedSamples.decodeOrder().rbegin()->first);
auto nextSyncIter = trackBuffer.samples.decodeOrder().findSyncSampleAfterDecodeIterator(lastDecodeIter);
dependentSamples.insert(firstDecodeIter, nextSyncIter);
// NOTE: in the case of b-frames, the previous step may leave in place samples whose presentation
// timestamp < presentationTime, but whose decode timestamp >= decodeTime. These will eventually cause
// a decode error if left in place, so remove these samples as well.
DecodeOrderSampleMap::KeyType decodeKey(sample->decodeTime(), sample->presentationTime());
auto samplesWithHigherDecodeTimes = trackBuffer.samples.decodeOrder().findSamplesBetweenDecodeKeys(decodeKey, erasedSamples.decodeOrder().begin()->first);
if (samplesWithHigherDecodeTimes.first != samplesWithHigherDecodeTimes.second)
dependentSamples.insert(samplesWithHigherDecodeTimes.first, samplesWithHigherDecodeTimes.second);
#if !RELEASE_LOG_DISABLED
PlatformTimeRanges erasedRanges = removeSamplesFromTrackBuffer(dependentSamples, trackBuffer, this, "didReceiveSample");
#else
PlatformTimeRanges erasedRanges = removeSamplesFromTrackBuffer(dependentSamples, trackBuffer);
#endif
// Only force the TrackBuffer to re-enqueue if the removed ranges overlap with enqueued and possibly
// not yet displayed samples.
MediaTime currentTime = currentMediaTime();
if (trackBuffer.highestEnqueuedPresentationTime.isValid() && currentTime < trackBuffer.highestEnqueuedPresentationTime) {
PlatformTimeRanges possiblyEnqueuedRanges(currentTime, trackBuffer.highestEnqueuedPresentationTime);
possiblyEnqueuedRanges.intersectWith(erasedRanges);
if (possiblyEnqueuedRanges.length())
trackBuffer.needsReenqueueing = true;
}
erasedRanges.invert();
trackBuffer.buffered.intersectWith(erasedRanges);
setBufferedDirty(true);
}
// 1.16 If spliced audio frame is set:
// Add spliced audio frame to the track buffer.
// If spliced timed text frame is set:
// Add spliced timed text frame to the track buffer.
// FIXME: Add support for sample splicing.
// Otherwise:
// Add the coded frame with the presentation timestamp, decode timestamp, and frame duration to the track buffer.
trackBuffer.samples.addSample(sample);
// Note: The terminology here is confusing: "enqueuing" means providing a frame to the inner media framework.
// First, frames are inserted in the decode queue; later, at the end of the append some of the frames in the
// decode may be "enqueued" (sent to the inner media framework) in `provideMediaData()`.
//
// In order to check whether a frame should be added to the decode queue we check that it does not precede any
// frame already enqueued.
//
// Note that adding a frame to the decode queue is no guarantee that it will be actually enqueued at that point.
// If the frame is after the discontinuity boundary, the enqueueing algorithm will hold it there until samples
// with earlier timestamps are enqueued. The decode queue is not FIFO, but rather an ordered map.
DecodeOrderSampleMap::KeyType decodeKey(sample->decodeTime(), sample->presentationTime());
if (trackBuffer.lastEnqueuedDecodeKey.first.isInvalid() || decodeKey > trackBuffer.lastEnqueuedDecodeKey) {
trackBuffer.decodeQueue.insert(DecodeOrderSampleMap::MapType::value_type(decodeKey, &sample.get()));
if (trackBuffer.minimumEnqueuedPresentationTime.isValid() && sample->presentationTime() < trackBuffer.minimumEnqueuedPresentationTime)
trackBuffer.needsMinimumUpcomingPresentationTimeUpdating = true;
}
// NOTE: the spec considers "Coded Frame Duration" to be the presentation duration, but this is not necessarily equal
// to the decoded duration. When comparing deltas between decode timestamps, the decode duration, not the presentation.
if (trackBuffer.lastDecodeTimestamp.isValid()) {
MediaTime lastDecodeDuration = decodeTimestamp - trackBuffer.lastDecodeTimestamp;
if (!trackBuffer.greatestDecodeDuration.isValid() || lastDecodeDuration > trackBuffer.greatestDecodeDuration)
trackBuffer.greatestDecodeDuration = lastDecodeDuration;
}
// 1.17 Set last decode timestamp for track buffer to decode timestamp.
trackBuffer.lastDecodeTimestamp = decodeTimestamp;
// 1.18 Set last frame duration for track buffer to frame duration.
trackBuffer.lastFrameDuration = frameDuration;
// 1.19 If highest presentation timestamp for track buffer is unset or frame end timestamp is greater
// than highest presentation timestamp, then set highest presentation timestamp for track buffer
// to frame end timestamp.
if (trackBuffer.highestPresentationTimestamp.isInvalid() || frameEndTimestamp > trackBuffer.highestPresentationTimestamp)
trackBuffer.highestPresentationTimestamp = frameEndTimestamp;
// 1.20 If frame end timestamp is greater than group end timestamp, then set group end timestamp equal
// to frame end timestamp.
if (m_groupEndTimestamp.isInvalid() || frameEndTimestamp > m_groupEndTimestamp)
m_groupEndTimestamp = frameEndTimestamp;
// 1.21 If generate timestamps flag equals true, then set timestampOffset equal to frame end timestamp.
if (m_shouldGenerateTimestamps) {
m_timestampOffset = frameEndTimestamp;
for (auto& trackBuffer : m_trackBufferMap.values()) {
trackBuffer.get().lastFrameTimescale = 0;
trackBuffer.get().roundedTimestampOffset = MediaTime::invalidTime();
}
}
// Eliminate small gaps between buffered ranges by coalescing
// disjoint ranges separated by less than a "fudge factor".
auto presentationEndTime = presentationTimestamp + frameDuration;
auto nearestToPresentationStartTime = trackBuffer.buffered.nearest(presentationTimestamp);
if (nearestToPresentationStartTime.isValid() && (presentationTimestamp - nearestToPresentationStartTime).isBetween(MediaTime::zeroTime(), timeFudgeFactor()))
presentationTimestamp = nearestToPresentationStartTime;
auto nearestToPresentationEndTime = trackBuffer.buffered.nearest(presentationEndTime);
if (nearestToPresentationEndTime.isValid() && (nearestToPresentationEndTime - presentationEndTime).isBetween(MediaTime::zeroTime(), timeFudgeFactor()))
presentationEndTime = nearestToPresentationEndTime;
trackBuffer.buffered.add(presentationTimestamp, presentationEndTime);
m_client->sourceBufferPrivateDidParseSample(frameDuration.toDouble());
setBufferedDirty(true);
break;
} while (true);
// Steps 2-4 will be handled by MediaSource::monitorSourceBuffers()
// 5. If the media segment contains data beyond the current duration, then run the duration change algorithm with new
// duration set to the maximum of the current duration and the group end timestamp.
if (m_groupEndTimestamp > duration())
m_client->sourceBufferPrivateDurationChanged(m_groupEndTimestamp);
updateHighestPresentationTimestamp();
}
void SourceBufferPrivate::append(Ref<SharedBuffer>&& buffer)
{
append(buffer->extractData());
}
void SourceBufferPrivate::append(Vector<unsigned char>&&)
{
RELEASE_ASSERT_NOT_REACHED();
}
} // namespace WebCore
#endif