Diff
Modified: trunk/LayoutTests/ChangeLog (284086 => 284087)
--- trunk/LayoutTests/ChangeLog 2021-10-13 10:00:32 UTC (rev 284086)
+++ trunk/LayoutTests/ChangeLog 2021-10-13 10:21:50 UTC (rev 284087)
@@ -1,3 +1,16 @@
+2021-10-13 Alicia Boya García <[email protected]>
+
+ Unreviewed, reverting r283609
+
+ Build failure with GStreamer 1.14 and regression causing frame
+ corruption.
+
+ Reverted changeset:
+
+ "[MSE][GStreamer] Honor MP4 edit lists"
+ https://bugs.webkit.org/show_bug.cgi?id=231019
+ https://commits.webkit.org/r283609
+
2021-10-13 Youenn Fablet <[email protected]>
Add support for WebRTC media capabilities
Modified: trunk/LayoutTests/platform/glib/imported/w3c/web-platform-tests/media-source/mediasource-remove-expected.txt (284086 => 284087)
--- trunk/LayoutTests/platform/glib/imported/w3c/web-platform-tests/media-source/mediasource-remove-expected.txt 2021-10-13 10:00:32 UTC (rev 284086)
+++ trunk/LayoutTests/platform/glib/imported/w3c/web-platform-tests/media-source/mediasource-remove-expected.txt 2021-10-13 10:21:50 UTC (rev 284087)
@@ -11,8 +11,8 @@
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) }"
+FAIL Test removing all appended data. assert_equals: Initial buffered range. expected "{ [0.095, 6.548) }" but got "{ [0.000, 6.548) }"
+FAIL Test removing beginning of appended data. assert_equals: Initial buffered range. expected "{ [0.095, 6.548) }" but got "{ [0.000, 6.548) }"
+FAIL Test removing the middle of appended data. assert_equals: Initial buffered range. expected "{ [0.095, 6.548) }" but got "{ [0.000, 6.548) }"
+FAIL Test removing the end of appended data. assert_equals: Initial buffered range. expected "{ [0.095, 6.548) }" but got "{ [0.000, 6.548) }"
Modified: trunk/Source/WebCore/ChangeLog (284086 => 284087)
--- trunk/Source/WebCore/ChangeLog 2021-10-13 10:00:32 UTC (rev 284086)
+++ trunk/Source/WebCore/ChangeLog 2021-10-13 10:21:50 UTC (rev 284087)
@@ -1,3 +1,16 @@
+2021-10-13 Alicia Boya García <[email protected]>
+
+ Unreviewed, reverting r283609
+
+ Build failure with GStreamer 1.14 and regression causing frame
+ corruption.
+
+ Reverted changeset:
+
+ "[MSE][GStreamer] Honor MP4 edit lists"
+ https://bugs.webkit.org/show_bug.cgi?id=231019
+ https://commits.webkit.org/r283609
+
2021-10-13 Carlos Garcia Campos <[email protected]>
Unreviewed. Fix GTK build with ATSPI enabled after r283851
Modified: trunk/Source/WebCore/Modules/mediasource/MediaSource.cpp (284086 => 284087)
--- trunk/Source/WebCore/Modules/mediasource/MediaSource.cpp 2021-10-13 10:00:32 UTC (rev 284086)
+++ trunk/Source/WebCore/Modules/mediasource/MediaSource.cpp 2021-10-13 10:21:50 UTC (rev 284087)
@@ -323,8 +323,8 @@
const MediaTime& MediaSource::currentTimeFudgeFactor()
{
- // Allow hasCurrentTime() to be off by as much as 100ms.
- static NeverDestroyed<MediaTime> fudgeFactor(1, 10);
+ // Allow hasCurrentTime() to be off by as much as the length of two 24fps video frames
+ static NeverDestroyed<MediaTime> fudgeFactor(2002, 24000);
return fudgeFactor;
}
Modified: trunk/Source/WebCore/platform/graphics/SourceBufferPrivate.h (284086 => 284087)
--- trunk/Source/WebCore/platform/graphics/SourceBufferPrivate.h 2021-10-13 10:00:32 UTC (rev 284086)
+++ trunk/Source/WebCore/platform/graphics/SourceBufferPrivate.h 2021-10-13 10:21:50 UTC (rev 284087)
@@ -150,7 +150,7 @@
protected:
// The following method should never be called directly and be overridden instead.
WEBCORE_EXPORT virtual void append(Vector<unsigned char>&&);
- virtual MediaTime timeFudgeFactor() const { return {1, 10}; }
+ virtual MediaTime timeFudgeFactor() const { return {2002, 24000}; }
virtual bool isActive() const { return false; }
virtual bool isSeeking() const { return false; }
virtual MediaTime currentMediaTime() const { return { }; }
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.cpp (284086 => 284087)
--- trunk/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.cpp 2021-10-13 10:00:32 UTC (rev 284086)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.cpp 2021-10-13 10:21:50 UTC (rev 284087)
@@ -183,6 +183,15 @@
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;
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.h (284086 => 284087)
--- trunk/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.h 2021-10-13 10:00:32 UTC (rev 284086)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.h 2021-10-13 10:21:50 UTC (rev 284087)
@@ -42,6 +42,7 @@
static Ref<MediaSampleGStreamer> createFakeSample(GstCaps*, MediaTime pts, MediaTime dts, MediaTime duration, const FloatSize& presentationSize, const AtomString& trackId);
static Ref<MediaSampleGStreamer> createImageSample(PixelBuffer&&, const IntSize& destinationSize = { }, double frameRate = 1);
+ void extendToTheBeginning();
MediaTime presentationTime() const override { return m_pts; }
MediaTime decodeTime() const override { return m_dts; }
MediaTime duration() const override { return m_duration; }
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/mse/AppendPipeline.cpp (284086 => 284087)
--- trunk/Source/WebCore/platform/graphics/gstreamer/mse/AppendPipeline.cpp 2021-10-13 10:00:32 UTC (rev 284086)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/mse/AppendPipeline.cpp 2021-10-13 10:21:50 UTC (rev 284087)
@@ -375,72 +375,21 @@
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)) {
+ if (UNLIKELY(!gst_sample_get_buffer(sample.get()))) {
GST_WARNING("Received sample without buffer from appsink.");
return;
}
- if (!GST_BUFFER_PTS_IS_VALID(buffer)) {
+ if (!GST_BUFFER_PTS_IS_VALID(gst_sample_get_buffer(sample.get()))) {
// 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());
- // Unset the buffer temporarily to ensure the buffer has refcount of 1 if possible when gst_buffer_make_writable is called, therefore avoiding a copy.
- 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.get());
- }
-
auto mediaSample = MediaSampleGStreamer::create(WTFMove(sample), track.presentationSize, track.trackId);
GST_TRACE("append: trackId=%s PTS=%s DTS=%s DUR=%s presentationSize=%.0fx%.0f",
@@ -450,6 +399,25 @@
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());
}
