[GStreamer][LibWebRTC] Timestamp handling improvements in video decod…

…er factory

https://bugs.webkit.org/show_bug.cgi?id=273757

Reviewed by Xabier Rodriguez-Calvar.

The buffers injected into the parser pipeline are now timestamped by appsrc. The RTP timestamps are
required by the LibWebRTC generic decoder in order to match parsed frames with input buffers, so
they are attached on each buffers using a reference timestamp meta.

* Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoDecoderFactory.cpp:
(WebCore::GStreamerWebRTCVideoDecoder::GStreamerWebRTCVideoDecoder):
(WebCore::GStreamerWebRTCVideoDecoder::pullSample):
* Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoFrameLibWebRTC.cpp:
(WebCore::ensureDebugCategoryIsRegistered):
(WebCore::convertGStreamerSampleToLibWebRTCVideoFrame):
(WebCore::GStreamerVideoFrameLibWebRTC::ToI420):
* Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoFrameLibWebRTC.h:

Canonical link: https://commits.webkit.org/278447@main

Source/WebCore/platform/mediastream/libwebrtc/gstreamer/GStreamerVideoDecoderFactory.cpp

@@ -227,7 +227,8 @@ class GStreamerWebRTCVideoDecoder : public webrtc::VideoDecoder {
         gst_buffer_add_reference_timestamp_meta(buffer.get(), m_rtpTimestampCaps.get(), inputImage.RtpTimestamp(), GST_CLOCK_TIME_NONE);
 
         auto sample = adoptGRef(gst_sample_new(buffer.get(), m_caps.get(), nullptr, nullptr));
-        switch (gst_app_src_push_sample(GST_APP_SRC(m_src), sample.get())) {
+
+	switch (gst_app_src_push_sample(GST_APP_SRC(m_src), sample.get())) {
         case GST_FLOW_OK:
             break;
         case GST_FLOW_FLUSHING: