Diff
Modified: trunk/LayoutTests/ChangeLog (287179 => 287180)
--- trunk/LayoutTests/ChangeLog 2021-12-17 12:41:14 UTC (rev 287179)
+++ trunk/LayoutTests/ChangeLog 2021-12-17 12:47:56 UTC (rev 287180)
@@ -1,3 +1,14 @@
+2021-12-17 Philippe Normand <[email protected]>
+
+ [GStreamer][WebRTC] Huge memory leak
+ https://bugs.webkit.org/show_bug.cgi?id=234134
+
+ Reviewed by Xabier Rodriguez-Calvar.
+
+ * platform/glib/TestExpectations: Update test expectations, #233740 was mostly due to video
+ decoder initialization wrongly reported as failed.
+ webrtc/captureCanvas-webrtc-software-h264-baseline.html no longer times out.
+
2021-12-16 Tyler Wilcock <[email protected]>
AX: Make aria-multiline.html, clipped-text-under-element.html, mixed-checkbox.html, and selection-states.html pass in isolated tree mode
Modified: trunk/LayoutTests/platform/glib/TestExpectations (287179 => 287180)
--- trunk/LayoutTests/platform/glib/TestExpectations 2021-12-17 12:41:14 UTC (rev 287179)
+++ trunk/LayoutTests/platform/glib/TestExpectations 2021-12-17 12:47:56 UTC (rev 287180)
@@ -1349,8 +1349,7 @@
webkit.org/b/208125 webrtc/peerconnection-new-candidate-page-cache.html [ Pass Crash ]
-# Was Slow Failure before libwebrtc bumped to M96 in r285577
-webkit.org/b/216538 webrtc/captureCanvas-webrtc-software-h264-baseline.html [ Timeout ]
+webkit.org/b/216538 webrtc/captureCanvas-webrtc-software-h264-baseline.html [ Slow Failure ]
webkit.org/b/216763 webrtc/captureCanvas-webrtc-software-h264-high.html [ Crash Failure ]
@@ -1370,10 +1369,6 @@
webkit.org/b/229055 http/wpt/webrtc/sframe-transform-error.html [ Failure ]
-webkit.org/b/233740 http/wpt/webrtc/video-script-transform-keyframe-only.html [ Failure ]
-webkit.org/b/233740 webrtc/vp9-svc.html [ Timeout ]
-webkit.org/b/233740 webrtc/vp9.html [ Timeout ]
-
webkit.org/b/233879 fast/mediastream/get-display-media-settings.html [ Failure ]
#////////////////////////////////////////////////////////////////////////////////////////
Modified: trunk/Source/WebCore/ChangeLog (287179 => 287180)
--- trunk/Source/WebCore/ChangeLog 2021-12-17 12:41:14 UTC (rev 287179)
+++ trunk/Source/WebCore/ChangeLog 2021-12-17 12:47:56 UTC (rev 287180)
@@ -1,3 +1,41 @@
+2021-12-17 Philippe Normand <[email protected]>
+
+ [GStreamer][WebRTC] Huge memory leak
+ https://bugs.webkit.org/show_bug.cgi?id=234134
+
+ Reviewed by Xabier Rodriguez-Calvar.
+
+ The main issue was RealtimeOutgoingVideoSourceLibWebRTC leaking GstSamples. Fixing this lead
+ me to further clean-ups in the GstSample<->LibWebRTCVideoFrame handling. Native frames
+ should not consume GstSamples, but reference them, otherwise there are crashes where the
+ dangling GstSamples would be passed to the libwebrtc video decoder...
+
+ Also the video decoders were not reporting their initialization status correctly,
+ WEBRTC_VIDEO_CODEC_OK is 0 hence false...
+
+ * platform/graphics/gstreamer/MediaSampleGStreamer.cpp:
+ (WebCore::MediaSampleGStreamer::MediaSampleGStreamer):
+ (WebCore::MediaSampleGStreamer::initializeFromBuffer):
+ * platform/graphics/gstreamer/MediaSampleGStreamer.h:
+ (WebCore::MediaSampleGStreamer::createWrappedSample):
+ * platform/mediastream/libwebrtc/gstreamer/GStreamerVideoDecoderFactory.cpp:
+ (WebCore::GStreamerVideoDecoder::pullSample):
+ * platform/mediastream/libwebrtc/gstreamer/GStreamerVideoEncoderFactory.cpp:
+ * platform/mediastream/libwebrtc/gstreamer/GStreamerVideoFrameLibWebRTC.cpp:
+ (WebCore::convertLibWebRTCVideoFrameToGStreamerSample):
+ (WebCore::convertGStreamerSampleToLibWebRTCVideoFrame):
+ (WebCore::GStreamerVideoFrameLibWebRTC::create):
+ (WebCore::GStreamerSampleFromLibWebRTCVideoFrame): Deleted.
+ (WebCore::LibWebRTCVideoFrameFromGStreamerSample): Deleted.
+ * platform/mediastream/libwebrtc/gstreamer/GStreamerVideoFrameLibWebRTC.h:
+ (WebCore::GStreamerVideoFrameLibWebRTC::GStreamerVideoFrameLibWebRTC):
+ (WebCore::GStreamerVideoFrameLibWebRTC::getSample const):
+ (WebCore::GStreamerVideoFrameLibWebRTC::takeSample): Deleted.
+ * platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.cpp:
+ (WebCore::RealtimeIncomingVideoSourceLibWebRTC::OnFrame):
+ * platform/mediastream/libwebrtc/gstreamer/RealtimeOutgoingVideoSourceLibWebRTC.cpp:
+ (WebCore::RealtimeOutgoingVideoSourceLibWebRTC::videoSampleAvailable):
+
2021-12-17 Antoine Quint <[email protected]>
ActiveDOMObject::suspendIfNeeded() should not be called within constructors
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.cpp (287179 => 287180)
--- trunk/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.cpp 2021-12-17 12:41:14 UTC (rev 287179)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.cpp 2021-12-17 12:47:56 UTC (rev 287180)
@@ -42,7 +42,6 @@
, m_videoRotation(videoRotation)
, m_videoMirrored(videoMirrored)
{
- const GstClockTime minimumDuration = 1000; // 1 us
ASSERT(sample);
GstBuffer* buffer = gst_sample_get_buffer(sample.get());
RELEASE_ASSERT(buffer);
@@ -50,35 +49,10 @@
if (metadata)
buffer = webkitGstBufferSetVideoSampleMetadata(buffer, WTFMove(metadata));
- if (GST_BUFFER_PTS_IS_VALID(buffer))
- m_pts = fromGstClockTime(GST_BUFFER_PTS(buffer));
- if (GST_BUFFER_DTS_IS_VALID(buffer) || GST_BUFFER_PTS_IS_VALID(buffer))
- m_dts = fromGstClockTime(GST_BUFFER_DTS_OR_PTS(buffer));
- if (GST_BUFFER_DURATION_IS_VALID(buffer)) {
- // Sometimes (albeit rarely, so far seen only at the end of a track)
- // frames have very small durations, so small that may be under the
- // precision we are working with and be truncated to zero.
- // SourceBuffer algorithms are not expecting frames with zero-duration,
- // so let's use something very small instead in those fringe cases.
- m_duration = fromGstClockTime(std::max(GST_BUFFER_DURATION(buffer), minimumDuration));
- } else {
- // Unfortunately, sometimes samples don't provide a duration. This can never happen in MP4 because of the way
- // the format is laid out, but it's pretty common in WebM.
- // The good part is that durations don't matter for playback, just for buffered ranges and coded frame deletion.
- // We want to pick something small enough to not cause unwanted frame deletion, but big enough to never be
- // mistaken for a rounding artifact.
- m_duration = fromGstClockTime(16666667); // 1/60 seconds
- }
-
- m_size = gst_buffer_get_size(buffer);
m_sample = adoptGRef(gst_sample_new(buffer, gst_sample_get_caps(sample.get()), nullptr,
gst_sample_get_info(sample.get()) ? gst_structure_copy(gst_sample_get_info(sample.get())) : nullptr));
- if (GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DELTA_UNIT))
- m_flags = MediaSample::None;
-
- if (GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DECODE_ONLY))
- m_flags = static_cast<MediaSample::SampleFlags>(m_flags | MediaSample::IsNonDisplaying);
+ initializeFromBuffer();
}
MediaSampleGStreamer::MediaSampleGStreamer(const FloatSize& presentationSize, const AtomString& trackId)
@@ -90,6 +64,13 @@
{
}
+MediaSampleGStreamer::MediaSampleGStreamer(const GRefPtr<GstSample>& sample, VideoRotation videoRotation)
+ : m_sample(sample)
+ , m_videoRotation(videoRotation)
+{
+ initializeFromBuffer();
+}
+
Ref<MediaSampleGStreamer> MediaSampleGStreamer::createFakeSample(GstCaps*, MediaTime pts, MediaTime dts, MediaTime duration, const FloatSize& presentationSize, const AtomString& trackId)
{
MediaSampleGStreamer* gstreamerMediaSample = new MediaSampleGStreamer(presentationSize, trackId);
@@ -154,6 +135,41 @@
return create(WTFMove(sample), FloatSize(width, height), { }, videoRotation, videoMirrored);
}
+void MediaSampleGStreamer::initializeFromBuffer()
+{
+ const GstClockTime minimumDuration = 1000; // 1 us
+ auto* buffer = gst_sample_get_buffer(m_sample.get());
+ RELEASE_ASSERT(buffer);
+
+ if (GST_BUFFER_PTS_IS_VALID(buffer))
+ m_pts = fromGstClockTime(GST_BUFFER_PTS(buffer));
+ if (GST_BUFFER_DTS_IS_VALID(buffer) || GST_BUFFER_PTS_IS_VALID(buffer))
+ m_dts = fromGstClockTime(GST_BUFFER_DTS_OR_PTS(buffer));
+ if (GST_BUFFER_DURATION_IS_VALID(buffer)) {
+ // Sometimes (albeit rarely, so far seen only at the end of a track)
+ // frames have very small durations, so small that may be under the
+ // precision we are working with and be truncated to zero.
+ // SourceBuffer algorithms are not expecting frames with zero-duration,
+ // so let's use something very small instead in those fringe cases.
+ m_duration = fromGstClockTime(std::max(GST_BUFFER_DURATION(buffer), minimumDuration));
+ } else {
+ // Unfortunately, sometimes samples don't provide a duration. This can never happen in MP4 because of the way
+ // the format is laid out, but it's pretty common in WebM.
+ // The good part is that durations don't matter for playback, just for buffered ranges and coded frame deletion.
+ // We want to pick something small enough to not cause unwanted frame deletion, but big enough to never be
+ // mistaken for a rounding artifact.
+ m_duration = fromGstClockTime(16666667); // 1/60 seconds
+ }
+
+ m_size = gst_buffer_get_size(buffer);
+
+ if (GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DELTA_UNIT))
+ m_flags = MediaSample::None;
+
+ if (GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DECODE_ONLY))
+ m_flags = static_cast<MediaSample::SampleFlags>(m_flags | MediaSample::IsNonDisplaying);
+}
+
RefPtr<JSC::Uint8ClampedArray> MediaSampleGStreamer::getRGBAImageData() const
{
auto* caps = gst_sample_get_caps(m_sample.get());
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.h (287179 => 287180)
--- trunk/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.h 2021-12-17 12:41:14 UTC (rev 287179)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.h 2021-12-17 12:47:56 UTC (rev 287180)
@@ -40,6 +40,11 @@
return adoptRef(*new MediaSampleGStreamer(WTFMove(sample), presentationSize, trackId, videoRotation, videoMirrored, WTFMove(metadata)));
}
+ static Ref<MediaSampleGStreamer> createWrappedSample(const GRefPtr<GstSample>& sample, VideoRotation videoRotation = VideoRotation::None)
+ {
+ return adoptRef(*new MediaSampleGStreamer(sample, videoRotation));
+ }
+
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, VideoRotation videoRotation = VideoRotation::None, bool videoMirrored = false, std::optional<VideoSampleMetadata>&& metadata = std::nullopt);
@@ -66,11 +71,14 @@
protected:
MediaSampleGStreamer(GRefPtr<GstSample>&&, const FloatSize& presentationSize, const AtomString& trackId, VideoRotation = VideoRotation::None, bool videoMirrored = false, std::optional<VideoSampleMetadata>&& = std::nullopt);
+ MediaSampleGStreamer(const GRefPtr<GstSample>&, VideoRotation = VideoRotation::None);
virtual ~MediaSampleGStreamer() = default;
private:
MediaSampleGStreamer(const FloatSize& presentationSize, const AtomString& trackId);
+ void initializeFromBuffer();
+
MediaTime m_pts;
MediaTime m_dts;
MediaTime m_duration;
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoDecoderFactory.cpp (287179 => 287180)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoDecoderFactory.cpp 2021-12-17 12:41:14 UTC (rev 287179)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoDecoderFactory.cpp 2021-12-17 12:47:56 UTC (rev 287180)
@@ -150,7 +150,7 @@
return WEBRTC_VIDEO_CODEC_ERROR;
}
- return WEBRTC_VIDEO_CODEC_OK;
+ return true;
}
int32_t RegisterDecodeCompleteCallback(webrtc::DecodedImageCallback* callback) override
@@ -240,7 +240,7 @@
auto timestamps = m_dtsPtsMap[GST_BUFFER_PTS(buffer)];
m_dtsPtsMap.erase(GST_BUFFER_PTS(buffer));
- auto frame(LibWebRTCVideoFrameFromGStreamerSample(WTFMove(sample), webrtc::kVideoRotation_0,
+ auto frame(convertGStreamerSampleToLibWebRTCVideoFrame(sample, webrtc::kVideoRotation_0,
timestamps.timestamp, timestamps.renderTimeMs));
GST_BUFFER_DTS(buffer) = GST_CLOCK_TIME_NONE;
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoEncoderFactory.cpp (287179 => 287180)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoEncoderFactory.cpp 2021-12-17 12:41:14 UTC (rev 287179)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoEncoderFactory.cpp 2021-12-17 12:47:56 UTC (rev 287180)
@@ -228,7 +228,7 @@
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
- auto sample = GStreamerSampleFromLibWebRTCVideoFrame(frame);
+ auto sample = convertLibWebRTCVideoFrameToGStreamerSample(frame);
auto buffer = gst_sample_get_buffer(sample.get());
if (!GST_CLOCK_TIME_IS_VALID(m_firstFramePts)) {
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoFrameLibWebRTC.cpp (287179 => 287180)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoFrameLibWebRTC.cpp 2021-12-17 12:41:14 UTC (rev 287179)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoFrameLibWebRTC.cpp 2021-12-17 12:47:56 UTC (rev 287180)
@@ -27,13 +27,9 @@
namespace WebCore {
-GRefPtr<GstSample> GStreamerSampleFromLibWebRTCVideoFrame(const webrtc::VideoFrame& frame)
+GRefPtr<GstSample> convertLibWebRTCVideoFrameToGStreamerSample(const webrtc::VideoFrame& frame)
{
- if (frame.video_frame_buffer()->type() == webrtc::VideoFrameBuffer::Type::kNative) {
- auto* framebuffer = static_cast<GStreamerVideoFrameLibWebRTC*>(frame.video_frame_buffer().get());
- return framebuffer->takeSample();
- }
-
+ RELEASE_ASSERT(frame.video_frame_buffer()->type() != webrtc::VideoFrameBuffer::Type::kNative);
auto* i420Buffer = frame.video_frame_buffer()->ToI420().release();
int height = i420Buffer->height();
int strides[3] = {
@@ -60,23 +56,22 @@
return sample;
}
-rtc::scoped_refptr<webrtc::VideoFrameBuffer> GStreamerVideoFrameLibWebRTC::create(GRefPtr<GstSample>&& sample)
+std::unique_ptr<webrtc::VideoFrame> convertGStreamerSampleToLibWebRTCVideoFrame(GRefPtr<GstSample>& sample, webrtc::VideoRotation rotation, int64_t timestamp, int64_t renderTimeMs)
{
+ auto frameBuffer(GStreamerVideoFrameLibWebRTC::create(sample));
+ return std::unique_ptr<webrtc::VideoFrame>(new webrtc::VideoFrame(frameBuffer, timestamp, renderTimeMs, rotation));
+}
+
+rtc::scoped_refptr<webrtc::VideoFrameBuffer> GStreamerVideoFrameLibWebRTC::create(const GRefPtr<GstSample>& sample)
+{
GstVideoInfo info;
if (!gst_video_info_from_caps(&info, gst_sample_get_caps(sample.get())))
ASSERT_NOT_REACHED();
- return rtc::scoped_refptr<webrtc::VideoFrameBuffer>(new GStreamerVideoFrameLibWebRTC(WTFMove(sample), info));
+ return rtc::scoped_refptr<webrtc::VideoFrameBuffer>(new GStreamerVideoFrameLibWebRTC(sample, info));
}
-std::unique_ptr<webrtc::VideoFrame> LibWebRTCVideoFrameFromGStreamerSample(GRefPtr<GstSample>&& sample, webrtc::VideoRotation rotation,
- int64_t timestamp, int64_t renderTimeMs)
-{
- auto frameBuffer(GStreamerVideoFrameLibWebRTC::create(WTFMove(sample)));
- return std::unique_ptr<webrtc::VideoFrame>(new webrtc::VideoFrame(frameBuffer, timestamp, renderTimeMs, rotation));
-}
-
rtc::scoped_refptr<webrtc::I420BufferInterface> GStreamerVideoFrameLibWebRTC::ToI420()
{
GstMappedFrame inFrame(m_sample, GST_MAP_READ);
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoFrameLibWebRTC.h (287179 => 287180)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoFrameLibWebRTC.h 2021-12-17 12:41:14 UTC (rev 287179)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoFrameLibWebRTC.h 2021-12-17 12:47:56 UTC (rev 287180)
@@ -31,19 +31,19 @@
namespace WebCore {
-WARN_UNUSED_RETURN GRefPtr<GstSample> GStreamerSampleFromLibWebRTCVideoFrame(const webrtc::VideoFrame&);
+WARN_UNUSED_RETURN GRefPtr<GstSample> convertLibWebRTCVideoFrameToGStreamerSample(const webrtc::VideoFrame&);
-std::unique_ptr<webrtc::VideoFrame> LibWebRTCVideoFrameFromGStreamerSample(GRefPtr<GstSample>&&, webrtc::VideoRotation, int64_t timestamp, int64_t renderTimeMs);
+std::unique_ptr<webrtc::VideoFrame> convertGStreamerSampleToLibWebRTCVideoFrame(GRefPtr<GstSample>&, webrtc::VideoRotation, int64_t timestamp, int64_t renderTimeMs);
class GStreamerVideoFrameLibWebRTC : public rtc::RefCountedObject<webrtc::VideoFrameBuffer> {
public:
- GStreamerVideoFrameLibWebRTC(GRefPtr<GstSample>&& sample, GstVideoInfo info)
- : m_sample(WTFMove(sample))
+ GStreamerVideoFrameLibWebRTC(const GRefPtr<GstSample>& sample, GstVideoInfo info)
+ : m_sample(sample)
, m_info(info) { }
- static rtc::scoped_refptr<webrtc::VideoFrameBuffer> create(GRefPtr<GstSample>&&);
+ static rtc::scoped_refptr<webrtc::VideoFrameBuffer> create(const GRefPtr<GstSample>&);
- GRefPtr<GstSample>&& takeSample() { return WTFMove(m_sample); }
+ GstSample* getSample() const { return m_sample.get(); }
rtc::scoped_refptr<webrtc::I420BufferInterface> ToI420() final;
int width() const final { return GST_VIDEO_INFO_WIDTH(&m_info); }
@@ -55,6 +55,7 @@
GRefPtr<GstSample> m_sample;
GstVideoInfo m_info;
};
+
}
#endif // USE(GSTREAMER) && USE(LIBWEBRTC)
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.cpp (287179 => 287180)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.cpp 2021-12-17 12:41:14 UTC (rev 287179)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.cpp 2021-12-17 12:47:56 UTC (rev 287180)
@@ -58,12 +58,14 @@
if (!isProducingData())
return;
- callOnMainThread([protectedThis = Ref { *this }, frame] {
- auto gstSample = GStreamerSampleFromLibWebRTCVideoFrame(frame);
+ if (frame.video_frame_buffer()->type() == webrtc::VideoFrameBuffer::Type::kNative) {
+ auto* framebuffer = static_cast<GStreamerVideoFrameLibWebRTC*>(frame.video_frame_buffer().get());
+ videoSampleAvailable(MediaSampleGStreamer::createWrappedSample(framebuffer->getSample(), static_cast<MediaSample::VideoRotation>(frame.rotation())), { });
+ } else {
+ auto gstSample = convertLibWebRTCVideoFrameToGStreamerSample(frame);
auto metadata = std::make_optional(metadataFromVideoFrame(frame));
- auto sample = MediaSampleGStreamer::create(WTFMove(gstSample), { }, { }, static_cast<MediaSample::VideoRotation>(frame.rotation()), false, WTFMove(metadata));
- protectedThis->videoSampleAvailable(sample.get(), { });
- });
+ videoSampleAvailable(MediaSampleGStreamer::create(WTFMove(gstSample), { }, { }, static_cast<MediaSample::VideoRotation>(frame.rotation()), false, WTFMove(metadata)), { });
+ }
}
} // namespace WebCore
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeOutgoingVideoSourceLibWebRTC.cpp (287179 => 287180)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeOutgoingVideoSourceLibWebRTC.cpp 2021-12-17 12:41:14 UTC (rev 287179)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeOutgoingVideoSourceLibWebRTC.cpp 2021-12-17 12:47:56 UTC (rev 287180)
@@ -69,7 +69,7 @@
ASSERT(sample.platformSample().type == PlatformSample::GStreamerSampleType);
auto& mediaSample = static_cast<MediaSampleGStreamer&>(sample);
- auto frameBuffer = GStreamerVideoFrameLibWebRTC::create(gst_sample_ref(mediaSample.platformSample().sample.gstSample));
+ auto frameBuffer = GStreamerVideoFrameLibWebRTC::create(mediaSample.platformSample().sample.gstSample);
sendFrame(WTFMove(frameBuffer));
}
