Attachment #439726 for bug #231019

View Details Raw Unified Return to Bug 231019

| Differences between



2021-09-27  Alicia Boya García  <aboya@igalia.com>

        [MSE][GStreamer] Honor MP4 edit lists
        https://bugs.webkit.org/show_bug.cgi?id=231019

        Reviewed by NOBODY (OOPS!).

        This patch takes into consideration the GstSegment attached to a
        sample to offset the PTS and DTS. This ensures accurate timestamps are
        obtained for MP4 files containing edit lists (commonly necessary for
        files containing video with B frames to have PTS starting at zero).

        Before this was implemented, a workaround was in place based on a
        heuristic (DTS = 0 && PTS > 0 && PTS < 0.1). The workaround is
        preserved for the sake of content without proper edit lists, but
        any edit list takes preference.

        The time fudge factor has been modified from 0.083 seconds up to
        0.100 seconds to accomodate the size of the empty edit in test.mp4
        used by Web Platform Tests.

        This test fixes improves expectation results and fixes two subtests in
        imported/w3c/web-platform-tests/media-source/mediasource-remove.html.

        * Modules/mediasource/MediaSource.cpp:
        (WebCore::MediaSource::currentTimeFudgeFactor):
        * platform/graphics/SourceBufferPrivate.h:
        (WebCore::SourceBufferPrivate::timeFudgeFactor const):
        * platform/graphics/gstreamer/MediaSampleGStreamer.cpp:
        (WebCore::MediaSampleGStreamer::extendToTheBeginning): Deleted.
        * platform/graphics/gstreamer/MediaSampleGStreamer.h:
        * platform/graphics/gstreamer/mse/AppendPipeline.cpp:
        (WebCore::bufferTimeToStreamTimeClamped):
        (WebCore::AppendPipeline::appsinkNewSample):

2021-09-27  Alicia Boya García  <aboya@igalia.com>

        [MSE][GStreamer] Don't create MediaSourceTrackGStreamer objects twice for the same track




const MediaTime& MediaSource::currentTimeFudgeFactor()
{
    // Allow hasCurrentTime() to be off by as much as 100ms.
    static NeverDestroyed<MediaTime> fudgeFactor(1, 10);
    return fudgeFactor;
}


Lines 150-156 public: a/Source/WebCore/platform/graphics/SourceBufferPrivate.h_sec1
- a/Source/WebCore/platform/graphics/SourceBufferPrivate.h -1 / +1 lines
150	protected:	150	protected:
151	// The following method should never be called directly and be overridden instead.	151	// The following method should never be called directly and be overridden instead.
152	WEBCORE_EXPORT virtual void append(Vector<unsigned char>&&);	152	WEBCORE_EXPORT virtual void append(Vector<unsigned char>&&);
153	virtual MediaTime timeFudgeFactor() const { return {2002, 24000}; }	153	virtual MediaTime timeFudgeFactor() const { return {1, 10}; }
154	virtual bool isActive() const { return false; }	154	virtual bool isActive() const { return false; }
155	virtual bool isSeeking() const { return false; }	155	virtual bool isSeeking() const { return false; }
156	virtual MediaTime currentMediaTime() const { return { }; }	156	virtual MediaTime currentMediaTime() const { return { }; }



    return JSC::Uint8ClampedArray::tryCreate(WTFMove(bufferStorage), 0, byteLength);
}

void MediaSampleGStreamer::extendToTheBeginning()
{
    // Only to be used with the first sample, as a hack for lack of support for edit lists.
    // See AppendPipeline::appsinkNewSample()
    ASSERT(m_dts == MediaTime::zeroTime());
    m_duration += m_pts;
    m_pts = MediaTime::zeroTime();
}

void MediaSampleGStreamer::setTimestamps(const MediaTime& presentationTime, const MediaTime& decodeTime)
{
    m_pts = presentationTime;

Lines 42-48 public: a/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.h_sec1
- a/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.h -1 lines
42	static Ref<MediaSampleGStreamer> createFakeSample(GstCaps*, MediaTime pts, MediaTime dts, MediaTime duration, const FloatSize& presentationSize, const AtomString& trackId);	42	static Ref<MediaSampleGStreamer> createFakeSample(GstCaps*, MediaTime pts, MediaTime dts, MediaTime duration, const FloatSize& presentationSize, const AtomString& trackId);
43	static Ref<MediaSampleGStreamer> createImageSample(PixelBuffer&&, const IntSize& destinationSize = { }, double frameRate = 1);	43	static Ref<MediaSampleGStreamer> createImageSample(PixelBuffer&&, const IntSize& destinationSize = { }, double frameRate = 1);
44		44
45	void extendToTheBeginning();
46	MediaTime presentationTime() const override { return m_pts; }	45	MediaTime presentationTime() const override { return m_pts; }
47	MediaTime decodeTime() const override { return m_dts; }	46	MediaTime decodeTime() const override { return m_dts; }
48	MediaTime duration() const override { return m_duration; }	47	MediaTime duration() const override { return m_duration; }



    sourceBufferPrivate().didReceiveAllPendingSamples();
}

static GstClockTime bufferTimeToStreamTimeClamped(const GstSegment* segment, GstClockTime bufferTime)
{
    guint64 streamTime;
    int result = gst_segment_to_stream_time_full(segment, GST_FORMAT_TIME, bufferTime, &streamTime);
    if (!result) {
        GST_ERROR("Couldn't map buffer time %" GST_TIME_FORMAT " to segment %" GST_PTR_FORMAT, GST_TIME_ARGS(bufferTime), segment);
        return bufferTime;
    }
    if (result < 0)
        return 0; // Clamp negative timestamps down to zero.
    return streamTime;
}

void AppendPipeline::appsinkNewSample(const Track& track, GRefPtr<GstSample>&& sample)
{
    ASSERT(isMainThread());

    {
        GstBuffer* buffer = gst_sample_get_buffer(sample.get());
        if (UNLIKELY(!buffer)) {
            GST_WARNING("Received sample without buffer from appsink.");
            return;
        }

        if (!GST_BUFFER_PTS_IS_VALID(buffer)) {
            // When demuxing Vorbis, matroskademux creates several PTS-less frames with header information. We don't need those.
            GST_DEBUG("Ignoring sample without PTS: %" GST_PTR_FORMAT, gst_sample_get_buffer(sample.get()));
            return;
        }

        GstSegment* segment = gst_sample_get_segment(sample.get());
        bool hasMappedTime = false;
        GstClockTime pts = GST_BUFFER_PTS(buffer);
        GstClockTime dts = GST_BUFFER_DTS(buffer);
        GstClockTime duration = GST_BUFFER_DURATION(buffer);
        if (segment && (segment->time || segment->start)) {
            // MP4 has the concept of edit lists, where some buffer time needs to be offsetted, often very slightly,
            // to get exact timestamps.
            pts = bufferTimeToStreamTimeClamped(segment, GST_BUFFER_PTS(buffer));
            dts = bufferTimeToStreamTimeClamped(segment, GST_BUFFER_DTS(buffer));
            GST_TRACE_OBJECT(track.appsinkPad.get(), "Mapped buffer to segment, PTS %" GST_TIME_FORMAT " -> %" GST_TIME_FORMAT " DTS %" GST_TIME_FORMAT " -> %" GST_TIME_FORMAT,
                GST_TIME_ARGS(GST_BUFFER_PTS(buffer)), GST_TIME_ARGS(pts), GST_TIME_ARGS(GST_BUFFER_DTS(buffer)), GST_TIME_ARGS(dts));
            hasMappedTime = true;
        } else if (!dts && pts > 0 && pts <= 100'000'000) {
            // Because a track presentation time starting at some close to zero, but not exactly zero time can cause unexpected
            // results for applications, we extend the duration of this first sample to the left so that it starts at zero.
            // This is relevant for files that should have an edit list but don't, or when using GStreamer < 1.16, where
            // edit lists are not parsed in push-mode.

            GST_DEBUG("Extending first sample of track '%s' to make it start at PTS=0 %" GST_PTR_FORMAT, track.trackId.string().utf8().data(), buffer);
            duration += pts;
            pts = 0;
            hasMappedTime = true;
        }

        if (hasMappedTime) {
            sample = adoptGRef(gst_sample_make_writable(sample.leakRef()));
            GRefPtr<GstBuffer> newBuffer = gst_sample_get_buffer(sample.get());
            gst_sample_set_buffer(sample.get(), nullptr);
            newBuffer = adoptGRef(gst_buffer_make_writable(newBuffer.leakRef()));
            GST_BUFFER_PTS(newBuffer.get()) = pts;
            GST_BUFFER_DTS(newBuffer.get()) = dts;
            GST_BUFFER_DURATION(newBuffer.get()) = duration;
            gst_sample_set_buffer(sample.get(), newBuffer.leakRef());
        }
    }

    auto mediaSample = MediaSampleGStreamer::create(WTFMove(sample), track.presentationSize, track.trackId);

        mediaSample->duration().toString().utf8().data(),
        mediaSample->presentationSize().width(), mediaSample->presentationSize().height());

    // Hack, rework when GStreamer >= 1.16 becomes a requirement:
    // We're not applying edit lists. GStreamer < 1.16 doesn't emit the correct segments to do so.
    // GStreamer fix in https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/commit/c2a0da8096009f0f99943f78dc18066965be60f9
    // Also, in order to apply them we would need to convert the timestamps to stream time, which we're not currently
    // doing for consistency between GStreamer versions.
    //
    // In consequence, the timestamps we're handling here are unedited track time. In track time, the first sample is
    // guaranteed to have DTS == 0, but in the case of streams with B-frames, often PTS > 0. Edit lists fix this by
    // offsetting all timestamps by that amount in movie time, but we can't do that if we don't have access to them.
    // (We could assume the track PTS of the sample with track DTS = 0 is the offset, but we don't have any guarantee
    // we will get appended that sample first, or ever).
    //
    // Because a track presentation time starting at some close to zero, but not exactly zero time can cause unexpected
    // results for applications, we extend the duration of this first sample to the left so that it starts at zero.
    if (mediaSample->decodeTime() == MediaTime::zeroTime() && mediaSample->presentationTime() > MediaTime::zeroTime() && mediaSample->presentationTime() <= MediaTime(1, 10)) {
        GST_DEBUG("Extending first sample to make it start at PTS=0");
        mediaSample->extendToTheBeginning();
    }

    m_sourceBufferPrivate.didReceiveSample(mediaSample.get());
}




2021-09-27  Alicia Boya García  <aboya@igalia.com>

        [MSE][GStreamer] Honor MP4 edit lists
        https://bugs.webkit.org/show_bug.cgi?id=231019

        Reviewed by NOBODY (OOPS!).

        Update expectations for mediasource-remove.html in the GStreamer
        ports, as a couple subtests get fixed.

        * platform/glib/imported/w3c/web-platform-tests/media-source/mediasource-remove-expected.txt:

2021-09-22  Jean-Yves Avenard  <jya@apple.com>

        Make SharedBuffer inherit from ThreadSafeRefCounted



PASS Test aborting a remove operation.
PASS Test remove with a start at the duration.
PASS Test remove transitioning readyState from 'ended' to 'open'.
PASS Test removing all appended data.
PASS Test removing beginning of appended data.
FAIL Test removing the middle of appended data. assert_equals: Buffered ranges after remove(). expected "{ [0.095, 0.997) [3.298, 6.548) }" but got "{ [0.095, 0.975) [3.298, 6.548) }"
FAIL Test removing the end of appended data. assert_equals: Buffered ranges after remove(). expected "{ [0.095, 1.022) }" but got "{ [0.095, 0.995) }"


Return to Bug 231019

Lines 322-329 ExceptionOr<void> MediaSource::clearLiveSeekableRange() a/Source/WebCore/Modules/mediasource/MediaSource.cpp_sec1
- a/Source/WebCore/Modules/mediasource/MediaSource.cpp -2 / +2 lines
322		322
323	const MediaTime& MediaSource::currentTimeFudgeFactor()	323	const MediaTime& MediaSource::currentTimeFudgeFactor()
324	{	324	{
325	// Allow hasCurrentTime() to be off by as much as the length of two 24fps video frames	325	// Allow hasCurrentTime() to be off by as much as 100ms.
326	static NeverDestroyed<MediaTime> fudgeFactor(2002, 24000);	326	static NeverDestroyed<MediaTime> fudgeFactor(1, 10);
327	return fudgeFactor;	327	return fudgeFactor;
328	}	328	}
329		329

Lines 183-197 RefPtr<JSC::Uint8ClampedArray> MediaSampleGStreamer::getRGBAImageData() const a/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.cpp_sec1
- a/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.cpp -9 lines
183	return JSC::Uint8ClampedArray::tryCreate(WTFMove(bufferStorage), 0, byteLength);	183	return JSC::Uint8ClampedArray::tryCreate(WTFMove(bufferStorage), 0, byteLength);
184	}	184	}
185		185
186	void MediaSampleGStreamer::extendToTheBeginning()
187	{
188	// Only to be used with the first sample, as a hack for lack of support for edit lists.
189	// See AppendPipeline::appsinkNewSample()
190	ASSERT(m_dts == MediaTime::zeroTime());
191	m_duration += m_pts;
192	m_pts = MediaTime::zeroTime();
193	}
194
195	void MediaSampleGStreamer::setTimestamps(const MediaTime& presentationTime, const MediaTime& decodeTime)	186	void MediaSampleGStreamer::setTimestamps(const MediaTime& presentationTime, const MediaTime& decodeTime)
196	{	187	{
197	m_pts = presentationTime;	188	m_pts = presentationTime;

Lines 375-393 void AppendPipeline::handleEndOfAppend() a/Source/WebCore/platform/graphics/gstreamer/mse/AppendPipeline.cpp_sec1
- a/Source/WebCore/platform/graphics/gstreamer/mse/AppendPipeline.cpp -27 / +60 lines
375	sourceBufferPrivate().didReceiveAllPendingSamples();	375	sourceBufferPrivate().didReceiveAllPendingSamples();
376	}	376	}
377		377
		378	static GstClockTime bufferTimeToStreamTimeClamped(const GstSegment* segment, GstClockTime bufferTime)
		379	{
		380	guint64 streamTime;
		381	int result = gst_segment_to_stream_time_full(segment, GST_FORMAT_TIME, bufferTime, &streamTime);
		382	if (!result) {
		383	GST_ERROR("Couldn't map buffer time %" GST_TIME_FORMAT " to segment %" GST_PTR_FORMAT, GST_TIME_ARGS(bufferTime), segment);
		384	return bufferTime;
		385	}
		386	if (result < 0)
		387	return 0; // Clamp negative timestamps down to zero.
		388	return streamTime;
		389	}
		390
378	void AppendPipeline::appsinkNewSample(const Track& track, GRefPtr<GstSample>&& sample)	391	void AppendPipeline::appsinkNewSample(const Track& track, GRefPtr<GstSample>&& sample)
379	{	392	{
380	ASSERT(isMainThread());	393	ASSERT(isMainThread());
381		394
382	if (UNLIKELY(!gst_sample_get_buffer(sample.get()))) {	395	{
383	GST_WARNING("Received sample without buffer from appsink.");	396	GstBuffer* buffer = gst_sample_get_buffer(sample.get());
384	return;	397	if (UNLIKELY(!buffer)) {
385	}	398	GST_WARNING("Received sample without buffer from appsink.");
		399	return;
		400	}
386		401
387	if (!GST_BUFFER_PTS_IS_VALID(gst_sample_get_buffer(sample.get()))) {	402	if (!GST_BUFFER_PTS_IS_VALID(buffer)) {
388	// When demuxing Vorbis, matroskademux creates several PTS-less frames with header information. We don't need those.	403	// When demuxing Vorbis, matroskademux creates several PTS-less frames with header information. We don't need those.
389	GST_DEBUG("Ignoring sample without PTS: %" GST_PTR_FORMAT, gst_sample_get_buffer(sample.get()));	404	GST_DEBUG("Ignoring sample without PTS: %" GST_PTR_FORMAT, gst_sample_get_buffer(sample.get()));
390	return;	405	return;
		406	}
		407
		408	GstSegment* segment = gst_sample_get_segment(sample.get());
		409	bool hasMappedTime = false;
		410	GstClockTime pts = GST_BUFFER_PTS(buffer);
		411	GstClockTime dts = GST_BUFFER_DTS(buffer);
		412	GstClockTime duration = GST_BUFFER_DURATION(buffer);
		413	if (segment && (segment->time \|\| segment->start)) {
		414	// MP4 has the concept of edit lists, where some buffer time needs to be offsetted, often very slightly,
		415	// to get exact timestamps.
		416	pts = bufferTimeToStreamTimeClamped(segment, GST_BUFFER_PTS(buffer));
		417	dts = bufferTimeToStreamTimeClamped(segment, GST_BUFFER_DTS(buffer));
		418	GST_TRACE_OBJECT(track.appsinkPad.get(), "Mapped buffer to segment, PTS %" GST_TIME_FORMAT " -> %" GST_TIME_FORMAT " DTS %" GST_TIME_FORMAT " -> %" GST_TIME_FORMAT,
		419	GST_TIME_ARGS(GST_BUFFER_PTS(buffer)), GST_TIME_ARGS(pts), GST_TIME_ARGS(GST_BUFFER_DTS(buffer)), GST_TIME_ARGS(dts));
		420	hasMappedTime = true;
		421	} else if (!dts && pts > 0 && pts <= 100'000'000) {
		422	// Because a track presentation time starting at some close to zero, but not exactly zero time can cause unexpected
		423	// results for applications, we extend the duration of this first sample to the left so that it starts at zero.
		424	// This is relevant for files that should have an edit list but don't, or when using GStreamer < 1.16, where
		425	// edit lists are not parsed in push-mode.
		426
		427	GST_DEBUG("Extending first sample of track '%s' to make it start at PTS=0 %" GST_PTR_FORMAT, track.trackId.string().utf8().data(), buffer);
		428	duration += pts;
		429	pts = 0;
		430	hasMappedTime = true;
		431	}
		432
		433	if (hasMappedTime) {
		434	sample = adoptGRef(gst_sample_make_writable(sample.leakRef()));
		435	GRefPtr<GstBuffer> newBuffer = gst_sample_get_buffer(sample.get());
		436	gst_sample_set_buffer(sample.get(), nullptr);
		437	newBuffer = adoptGRef(gst_buffer_make_writable(newBuffer.leakRef()));
		438	GST_BUFFER_PTS(newBuffer.get()) = pts;
		439	GST_BUFFER_DTS(newBuffer.get()) = dts;
		440	GST_BUFFER_DURATION(newBuffer.get()) = duration;
		441	gst_sample_set_buffer(sample.get(), newBuffer.leakRef());
		442	}
391	}	443	}
392		444
393	auto mediaSample = MediaSampleGStreamer::create(WTFMove(sample), track.presentationSize, track.trackId);	445	auto mediaSample = MediaSampleGStreamer::create(WTFMove(sample), track.presentationSize, track.trackId);
Lines 399-423 void AppendPipeline::appsinkNewSample(const Track& track, GRefPtr<GstSample>&& s a/Source/WebCore/platform/graphics/gstreamer/mse/AppendPipeline.cpp_sec2
399	mediaSample->duration().toString().utf8().data(),	451	mediaSample->duration().toString().utf8().data(),
400	mediaSample->presentationSize().width(), mediaSample->presentationSize().height());	452	mediaSample->presentationSize().width(), mediaSample->presentationSize().height());
401		453
402	// Hack, rework when GStreamer >= 1.16 becomes a requirement:
403	// We're not applying edit lists. GStreamer < 1.16 doesn't emit the correct segments to do so.
404	// GStreamer fix in https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/commit/c2a0da8096009f0f99943f78dc18066965be60f9
405	// Also, in order to apply them we would need to convert the timestamps to stream time, which we're not currently
406	// doing for consistency between GStreamer versions.
407	//
408	// In consequence, the timestamps we're handling here are unedited track time. In track time, the first sample is
409	// guaranteed to have DTS == 0, but in the case of streams with B-frames, often PTS > 0. Edit lists fix this by
410	// offsetting all timestamps by that amount in movie time, but we can't do that if we don't have access to them.
411	// (We could assume the track PTS of the sample with track DTS = 0 is the offset, but we don't have any guarantee
412	// we will get appended that sample first, or ever).
413	//
414	// Because a track presentation time starting at some close to zero, but not exactly zero time can cause unexpected
415	// results for applications, we extend the duration of this first sample to the left so that it starts at zero.
416	if (mediaSample->decodeTime() == MediaTime::zeroTime() && mediaSample->presentationTime() > MediaTime::zeroTime() && mediaSample->presentationTime() <= MediaTime(1, 10)) {
417	GST_DEBUG("Extending first sample to make it start at PTS=0");
418	mediaSample->extendToTheBeginning();
419	}
420
421	m_sourceBufferPrivate.didReceiveSample(mediaSample.get());	454	m_sourceBufferPrivate.didReceiveSample(mediaSample.get());
422	}	455	}
423		456

Lines 1-3 a/Source/WebCore/ChangeLog_sec1
- a/Source/WebCore/ChangeLog +35 lines
		1	2021-09-27 Alicia Boya García <aboya@igalia.com>
		2
		3	[MSE][GStreamer] Honor MP4 edit lists
		4	https://bugs.webkit.org/show_bug.cgi?id=231019
		5
		6	Reviewed by NOBODY (OOPS!).
		7
		8	This patch takes into consideration the GstSegment attached to a
		9	sample to offset the PTS and DTS. This ensures accurate timestamps are
		10	obtained for MP4 files containing edit lists (commonly necessary for
		11	files containing video with B frames to have PTS starting at zero).
		12
		13	Before this was implemented, a workaround was in place based on a
		14	heuristic (DTS = 0 && PTS > 0 && PTS < 0.1). The workaround is
		15	preserved for the sake of content without proper edit lists, but
		16	any edit list takes preference.
		17
		18	The time fudge factor has been modified from 0.083 seconds up to
		19	0.100 seconds to accomodate the size of the empty edit in test.mp4
		20	used by Web Platform Tests.
		21
		22	This test fixes improves expectation results and fixes two subtests in
		23	imported/w3c/web-platform-tests/media-source/mediasource-remove.html.
		24
		25	* Modules/mediasource/MediaSource.cpp:
		26	(WebCore::MediaSource::currentTimeFudgeFactor):
		27	* platform/graphics/SourceBufferPrivate.h:
		28	(WebCore::SourceBufferPrivate::timeFudgeFactor const):
		29	* platform/graphics/gstreamer/MediaSampleGStreamer.cpp:
		30	(WebCore::MediaSampleGStreamer::extendToTheBeginning): Deleted.
		31	* platform/graphics/gstreamer/MediaSampleGStreamer.h:
		32	* platform/graphics/gstreamer/mse/AppendPipeline.cpp:
		33	(WebCore::bufferTimeToStreamTimeClamped):
		34	(WebCore::AppendPipeline::appsinkNewSample):
		35
1	2021-09-27 Alicia Boya García <aboya@igalia.com>	36	2021-09-27 Alicia Boya García <aboya@igalia.com>
2		37
3	[MSE][GStreamer] Don't create MediaSourceTrackGStreamer objects twice for the same track	38	[MSE][GStreamer] Don't create MediaSourceTrackGStreamer objects twice for the same track

Lines 1-3 a/LayoutTests/ChangeLog_sec1
- a/LayoutTests/ChangeLog +12 lines
		1	2021-09-27 Alicia Boya García <aboya@igalia.com>
		2
		3	[MSE][GStreamer] Honor MP4 edit lists
		4	https://bugs.webkit.org/show_bug.cgi?id=231019
		5
		6	Reviewed by NOBODY (OOPS!).
		7
		8	Update expectations for mediasource-remove.html in the GStreamer
		9	ports, as a couple subtests get fixed.
		10
		11	* platform/glib/imported/w3c/web-platform-tests/media-source/mediasource-remove-expected.txt:
		12
1	2021-09-22 Jean-Yves Avenard <jya@apple.com>	13	2021-09-22 Jean-Yves Avenard <jya@apple.com>
2		14
3	Make SharedBuffer inherit from ThreadSafeRefCounted	15	Make SharedBuffer inherit from ThreadSafeRefCounted

Lines 11-18 PASS Test remove while update pending. a/LayoutTests/platform/glib/imported/w3c/web-platform-tests/media-source/mediasource-remove-expected.txt_sec1
- a/LayoutTests/platform/glib/imported/w3c/web-platform-tests/media-source/mediasource-remove-expected.txt -4 / +4 lines
11	PASS Test aborting a remove operation.	11	PASS Test aborting a remove operation.
12	PASS Test remove with a start at the duration.	12	PASS Test remove with a start at the duration.
13	PASS Test remove transitioning readyState from 'ended' to 'open'.	13	PASS Test remove transitioning readyState from 'ended' to 'open'.
14	FAIL Test removing all appended data. assert_equals: Initial buffered range. expected "{ [0.095, 6.548) }" but got "{ [0.000, 6.548) }"	14	PASS Test removing all appended data.
15	FAIL Test removing beginning of appended data. assert_equals: Initial buffered range. expected "{ [0.095, 6.548) }" but got "{ [0.000, 6.548) }"	15	PASS Test removing beginning of appended data.
16	FAIL Test removing the middle of appended data. assert_equals: Initial buffered range. expected "{ [0.095, 6.548) }" but got "{ [0.000, 6.548) }"	16	FAIL Test removing the middle of appended data. assert_equals: Buffered ranges after remove(). expected "{ [0.095, 0.997) [3.298, 6.548) }" but got "{ [0.095, 0.975) [3.298, 6.548) }"
17	FAIL Test removing the end of appended data. assert_equals: Initial buffered range. expected "{ [0.095, 6.548) }" but got "{ [0.000, 6.548) }"	17	FAIL Test removing the end of appended data. assert_equals: Buffered ranges after remove(). expected "{ [0.095, 1.022) }" but got "{ [0.095, 0.995) }"
18		18