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<dt>Revision</dt> <dd><a href="http://trac.webkit.org/projects/webkit/changeset/206204">206204</a></dd>
<dt>Author</dt> <dd>philn@webkit.org</dd>
<dt>Date</dt> <dd>2016-09-21 02:44:06 -0700 (Wed, 21 Sep 2016)</dd>
</dl>
<h3>Log Message</h3>
<pre>[OpenWebRTC] Miscellaneous fixes
https://bugs.webkit.org/show_bug.cgi?id=162332
Reviewed by Alejandro G. Castro.
* platform/graphics/gstreamer/MediaPlayerPrivateGStreamerOwr.cpp:
(WebCore::MediaPlayerPrivateGStreamerOwr::currentTime): Improved logging.
(WebCore::MediaPlayerPrivateGStreamerOwr::load): Ditto.
(WebCore::MediaPlayerPrivateGStreamerOwr::loadingFailed): Ditto.
(WebCore::MediaPlayerPrivateGStreamerOwr::createGSTAudioSinkBin):
Pre-roll the autoaudiosink, fetch the underlying platform audio
sink and pass it to the OpenWebRTC renderer.
(WebCore::MediaPlayerPrivateGStreamerOwr::maybeHandleChangeMutedState): Improved logging.
(WebCore::MediaPlayerPrivateGStreamerOwr::setSize): Don't configure invalid video renderer.
* platform/mediastream/openwebrtc/RealtimeMediaSourceCenterOwr.cpp:
(WebCore::RealtimeMediaSourceCenterOwr::createMediaStream): Fix copy-paste error.</pre>
<h3>Modified Paths</h3>
<ul>
<li><a href="#trunkSourceWebCoreChangeLog">trunk/Source/WebCore/ChangeLog</a></li>
<li><a href="#trunkSourceWebCoreplatformgraphicsgstreamerMediaPlayerPrivateGStreamerOwrcpp">trunk/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamerOwr.cpp</a></li>
<li><a href="#trunkSourceWebCoreplatformmediastreamopenwebrtcRealtimeMediaSourceCenterOwrcpp">trunk/Source/WebCore/platform/mediastream/openwebrtc/RealtimeMediaSourceCenterOwr.cpp</a></li>
</ul>
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<h3>Diff</h3>
<a id="trunkSourceWebCoreChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/ChangeLog (206203 => 206204)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/ChangeLog 2016-09-21 08:43:23 UTC (rev 206203)
+++ trunk/Source/WebCore/ChangeLog 2016-09-21 09:44:06 UTC (rev 206204)
</span><span class="lines">@@ -1,3 +1,22 @@
</span><ins>+2016-09-21 Philippe Normand <pnormand@igalia.com>
+
+ [OpenWebRTC] Miscellaneous fixes
+ https://bugs.webkit.org/show_bug.cgi?id=162332
+
+ Reviewed by Alejandro G. Castro.
+
+ * platform/graphics/gstreamer/MediaPlayerPrivateGStreamerOwr.cpp:
+ (WebCore::MediaPlayerPrivateGStreamerOwr::currentTime): Improved logging.
+ (WebCore::MediaPlayerPrivateGStreamerOwr::load): Ditto.
+ (WebCore::MediaPlayerPrivateGStreamerOwr::loadingFailed): Ditto.
+ (WebCore::MediaPlayerPrivateGStreamerOwr::createGSTAudioSinkBin):
+ Pre-roll the autoaudiosink, fetch the underlying platform audio
+ sink and pass it to the OpenWebRTC renderer.
+ (WebCore::MediaPlayerPrivateGStreamerOwr::maybeHandleChangeMutedState): Improved logging.
+ (WebCore::MediaPlayerPrivateGStreamerOwr::setSize): Don't configure invalid video renderer.
+ * platform/mediastream/openwebrtc/RealtimeMediaSourceCenterOwr.cpp:
+ (WebCore::RealtimeMediaSourceCenterOwr::createMediaStream): Fix copy-paste error.
+
</ins><span class="cx"> 2016-09-21 Youenn Fablet <youenn@apple.com>
</span><span class="cx">
</span><span class="cx"> Refactor CachedResourceLoader::canRequest
</span></span></pre></div>
<a id="trunkSourceWebCoreplatformgraphicsgstreamerMediaPlayerPrivateGStreamerOwrcpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamerOwr.cpp (206203 => 206204)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamerOwr.cpp 2016-09-21 08:43:23 UTC (rev 206203)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamerOwr.cpp 2016-09-21 09:44:06 UTC (rev 206204)
</span><span class="lines">@@ -115,7 +115,7 @@
</span><span class="cx"> if (static_cast<GstClockTime>(position) != GST_CLOCK_TIME_NONE)
</span><span class="cx"> result = static_cast<double>(position) / GST_SECOND;
</span><span class="cx">
</span><del>- GST_DEBUG("Position %" GST_TIME_FORMAT, GST_TIME_ARGS(position));
</del><ins>+ GST_LOG("Position %" GST_TIME_FORMAT, GST_TIME_ARGS(position));
</ins><span class="cx"> gst_query_unref(query);
</span><span class="cx">
</span><span class="cx"> return result;
</span><span class="lines">@@ -148,7 +148,7 @@
</span><span class="cx"> if (streamPrivate.hasAudio() && !m_audioSink)
</span><span class="cx"> createGSTAudioSinkBin();
</span><span class="cx">
</span><del>- GST_DEBUG("Loading MediaStreamPrivate %p", &streamPrivate);
</del><ins>+ GST_DEBUG("Loading MediaStreamPrivate %p video: %s, audio: %s", &streamPrivate, streamPrivate.hasVideo() ? "yes":"no", streamPrivate.hasAudio() ? "yes":"no");
</ins><span class="cx">
</span><span class="cx"> m_streamPrivate = &streamPrivate;
</span><span class="cx"> if (!m_streamPrivate->active()) {
</span><span class="lines">@@ -188,6 +188,7 @@
</span><span class="cx"> void MediaPlayerPrivateGStreamerOwr::loadingFailed(MediaPlayer::NetworkState error)
</span><span class="cx"> {
</span><span class="cx"> if (m_networkState != error) {
</span><ins>+ GST_WARNING("Loading failed, error: %d", error);
</ins><span class="cx"> m_networkState = error;
</span><span class="cx"> m_player->networkStateChanged();
</span><span class="cx"> }
</span><span class="lines">@@ -259,11 +260,19 @@
</span><span class="cx"> GST_DEBUG("Creating audio sink");
</span><span class="cx"> // FIXME: volume/mute support: https://webkit.org/b/153828.
</span><span class="cx">
</span><del>- GRefPtr<GstElement> sink = gst_element_factory_make("autoaudiosink", 0);
</del><ins>+ // Pre-roll an autoaudiosink so that the platform audio sink is created and
+ // can be retrieved from the autoaudiosink bin.
+ GRefPtr<GstElement> sink = gst_element_factory_make("autoaudiosink", nullptr);
</ins><span class="cx"> GstChildProxy* childProxy = GST_CHILD_PROXY(sink.get());
</span><del>- m_audioSink = adoptGRef(GST_ELEMENT(gst_child_proxy_get_child_by_index(childProxy, 0)));
</del><ins>+ gst_element_set_state(sink.get(), GST_STATE_READY);
+ GRefPtr<GstElement> platformSink = adoptGRef(GST_ELEMENT(gst_child_proxy_get_child_by_index(childProxy, 0)));
+ GstElementFactory* factory = gst_element_get_factory(platformSink.get());
+
+ // Dispose now un-needed autoaudiosink.
</ins><span class="cx"> gst_element_set_state(sink.get(), GST_STATE_NULL);
</span><span class="cx">
</span><ins>+ // Create a fresh new audio sink compatible with the platform.
+ m_audioSink = gst_element_factory_create(factory, nullptr);
</ins><span class="cx"> m_audioRenderer = adoptGRef(owr_gst_audio_renderer_new(m_audioSink.get()));
</span><span class="cx"> }
</span><span class="cx">
</span><span class="lines">@@ -294,6 +303,7 @@
</span><span class="cx"> auto realTimeMediaSource = reinterpret_cast<RealtimeMediaSourceOwr*>(&track.source());
</span><span class="cx"> auto mediaSource = OWR_MEDIA_SOURCE(realTimeMediaSource->mediaSource());
</span><span class="cx">
</span><ins>+ GST_DEBUG("%s track now %s", track.type() == RealtimeMediaSource::Audio ? "audio":"video", realTimeMediaSource->muted() ? "muted":"un-muted");
</ins><span class="cx"> switch (track.type()) {
</span><span class="cx"> case RealtimeMediaSource::Audio:
</span><span class="cx"> if (!realTimeMediaSource->muted()) {
</span><span class="lines">@@ -356,7 +366,8 @@
</span><span class="cx"> return;
</span><span class="cx">
</span><span class="cx"> MediaPlayerPrivateGStreamerBase::setSize(size);
</span><del>- g_object_set(m_videoRenderer.get(), "width", size.width(), "height", size.height(), nullptr);
</del><ins>+ if (m_videoRenderer)
+ g_object_set(m_videoRenderer.get(), "width", size.width(), "height", size.height(), nullptr);
</ins><span class="cx"> }
</span><span class="cx">
</span><span class="cx"> } // namespace WebCore
</span></span></pre></div>
<a id="trunkSourceWebCoreplatformmediastreamopenwebrtcRealtimeMediaSourceCenterOwrcpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/platform/mediastream/openwebrtc/RealtimeMediaSourceCenterOwr.cpp (206203 => 206204)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/platform/mediastream/openwebrtc/RealtimeMediaSourceCenterOwr.cpp 2016-09-21 08:43:23 UTC (rev 206203)
+++ trunk/Source/WebCore/platform/mediastream/openwebrtc/RealtimeMediaSourceCenterOwr.cpp 2016-09-21 09:44:06 UTC (rev 206204)
</span><span class="lines">@@ -146,7 +146,7 @@
</span><span class="cx"> if (sourceIterator != m_sourceMap.end()) {
</span><span class="cx"> RefPtr<RealtimeMediaSource> source = sourceIterator->value;
</span><span class="cx"> if (source->type() == RealtimeMediaSource::Video)
</span><del>- audioSources.append(source.release());
</del><ins>+ videoSources.append(source.release());
</ins><span class="cx"> }
</span><span class="cx"> }
</span><span class="cx">
</span></span></pre>
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