Attachment 346273 Details for Bug 188210 – Patch

[patch] Patch

bug-188210-20180801084110.patch (text/plain), 3.56 KB, created by Thibault Saunier on 2018-08-01 05:41:11 PDT

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Creator: Thibault Saunier

Created: 2018-08-01 05:41:11 PDT

Size: 3.56 KB

patch

obsolete

>Subversion Revision: 234422
>diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
>index db75c8ed54de943e879f0d2a724b838c2a485957..5c013cb1ad14f0bbcc1c2ef7a1b1f9e6aec56005 100644
>--- a/Source/WebCore/ChangeLog
>+++ b/Source/WebCore/ChangeLog
>@@ -1,3 +1,20 @@
>+2018-07-31  Thibault Saunier  <tsaunier@igalia.com>
>+
>+        [GStreamer] Make sure that first buffer running time is 0 in GStreamerMediaStreamSource
>+        https://bugs.webkit.org/show_bug.cgi?id=188210
>+
>+        This is a live source and the first frame is the beginning of the stream,
>+        but that doesn't mean that the incoming stream from the camera didn't start
>+        before. We need to set a pad offset on each srcpad of the source to compensate
>+        for that. This is the exact same logic as in webrtc GStreamerVideo/Enc/Dec/oder.
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        * platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp:
>+        (WebCore::webkit_media_stream_src_init):
>+        (WebCore::webkitMediaStreamSrcPushVideoSample):
>+        (WebCore::webkitMediaStreamSrcPushAudioSample):
>+
> 2018-07-25  Thibault Saunier  <tsaunier@igalia.com>
> 
>         [GStreamer] Make codecparsers optionnal
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp b/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp
>index 9536076cb228e1369a50ef9d0e0ca2c016fb2e30..ffdae4eb508ee625d282ca22a9987c5cc2fe30d6 100644
>--- a/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp
>@@ -149,7 +149,9 @@ struct _WebKitMediaStreamSrc {
>     gchar* uri;
> 
>     GstElement* audioSrc;
>+    GstClockTime firstAudioBufferPts;
>     GstElement* videoSrc;
>+    GstClockTime firstFramePts;
> 
>     std::unique_ptr<WebKitMediaStreamTrackObserver> observer;
>     String videoTrackID;
>@@ -320,6 +322,8 @@ static void webkit_media_stream_src_init(WebKitMediaStreamSrc* self)
> {
>     self->observer = std::make_unique<WebKitMediaStreamTrackObserver>(self);
>     self->flowCombiner = gst_flow_combiner_new();
>+    self->firstAudioBufferPts = GST_CLOCK_TIME_NONE;
>+    self->firstFramePts = GST_CLOCK_TIME_NONE;
> }
> 
> typedef struct {
>@@ -485,14 +489,31 @@ gboolean webkitMediaStreamSrcSetStream(WebKitMediaStreamSrc* self, MediaStreamPr
> 
> static void webkitMediaStreamSrcPushVideoSample(WebKitMediaStreamSrc* self, GstSample* gstsample)
> {
>-    if (self->videoSrc)
>+    if (self->videoSrc) {
>+        if (!GST_CLOCK_TIME_IS_VALID(self->firstFramePts)) {
>+            auto buffer = gst_sample_get_buffer(gstsample);
>+
>+            self->firstFramePts = GST_BUFFER_PTS(buffer);
>+            auto pad = adoptGRef(gst_element_get_static_pad(self->videoSrc, "src"));
>+            gst_pad_set_offset(pad.get(), -self->firstFramePts);
>+        }
>+
>         gst_app_src_push_sample(GST_APP_SRC(self->videoSrc), gstsample);
>+    }
> }
> 
> static void webkitMediaStreamSrcPushAudioSample(WebKitMediaStreamSrc* self, GstSample* gstsample)
> {
>-    if (self->audioSrc)
>+    if (self->audioSrc) {
>+        if (!GST_CLOCK_TIME_IS_VALID(self->firstAudioBufferPts)) {
>+            auto buffer = gst_sample_get_buffer(gstsample);
>+
>+            self->firstAudioBufferPts = GST_BUFFER_PTS(buffer);
>+            auto pad = adoptGRef(gst_element_get_static_pad(self->audioSrc, "src"));
>+            gst_pad_set_offset(pad.get(), -self->firstAudioBufferPts);
>+        }
>         gst_app_src_push_sample(GST_APP_SRC(self->audioSrc), gstsample);
>+    }
> }
> 
> static void webkitMediaStreamSrcTrackEnded(WebKitMediaStreamSrc* self,

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Attachments on bug 188210: 346222 | 346273