Changeset 206204 in webkit
- Timestamp:
- Sep 21, 2016 2:44:06 AM (8 years ago)
- Location:
- trunk/Source/WebCore
- Files:
-
- 3 edited
Legend:
- Unmodified
- Added
- Removed
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trunk/Source/WebCore/ChangeLog
r206203 r206204 1 2016-09-21 Philippe Normand <pnormand@igalia.com> 2 3 [OpenWebRTC] Miscellaneous fixes 4 https://bugs.webkit.org/show_bug.cgi?id=162332 5 6 Reviewed by Alejandro G. Castro. 7 8 * platform/graphics/gstreamer/MediaPlayerPrivateGStreamerOwr.cpp: 9 (WebCore::MediaPlayerPrivateGStreamerOwr::currentTime): Improved logging. 10 (WebCore::MediaPlayerPrivateGStreamerOwr::load): Ditto. 11 (WebCore::MediaPlayerPrivateGStreamerOwr::loadingFailed): Ditto. 12 (WebCore::MediaPlayerPrivateGStreamerOwr::createGSTAudioSinkBin): 13 Pre-roll the autoaudiosink, fetch the underlying platform audio 14 sink and pass it to the OpenWebRTC renderer. 15 (WebCore::MediaPlayerPrivateGStreamerOwr::maybeHandleChangeMutedState): Improved logging. 16 (WebCore::MediaPlayerPrivateGStreamerOwr::setSize): Don't configure invalid video renderer. 17 * platform/mediastream/openwebrtc/RealtimeMediaSourceCenterOwr.cpp: 18 (WebCore::RealtimeMediaSourceCenterOwr::createMediaStream): Fix copy-paste error. 19 1 20 2016-09-21 Youenn Fablet <youenn@apple.com> 2 21 -
trunk/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamerOwr.cpp
r204410 r206204 116 116 result = static_cast<double>(position) / GST_SECOND; 117 117 118 GST_ DEBUG("Position %" GST_TIME_FORMAT, GST_TIME_ARGS(position));118 GST_LOG("Position %" GST_TIME_FORMAT, GST_TIME_ARGS(position)); 119 119 gst_query_unref(query); 120 120 … … 149 149 createGSTAudioSinkBin(); 150 150 151 GST_DEBUG("Loading MediaStreamPrivate %p ", &streamPrivate);151 GST_DEBUG("Loading MediaStreamPrivate %p video: %s, audio: %s", &streamPrivate, streamPrivate.hasVideo() ? "yes":"no", streamPrivate.hasAudio() ? "yes":"no"); 152 152 153 153 m_streamPrivate = &streamPrivate; … … 189 189 { 190 190 if (m_networkState != error) { 191 GST_WARNING("Loading failed, error: %d", error); 191 192 m_networkState = error; 192 193 m_player->networkStateChanged(); … … 260 261 // FIXME: volume/mute support: https://webkit.org/b/153828. 261 262 262 GRefPtr<GstElement> sink = gst_element_factory_make("autoaudiosink", 0); 263 // Pre-roll an autoaudiosink so that the platform audio sink is created and 264 // can be retrieved from the autoaudiosink bin. 265 GRefPtr<GstElement> sink = gst_element_factory_make("autoaudiosink", nullptr); 263 266 GstChildProxy* childProxy = GST_CHILD_PROXY(sink.get()); 264 m_audioSink = adoptGRef(GST_ELEMENT(gst_child_proxy_get_child_by_index(childProxy, 0))); 267 gst_element_set_state(sink.get(), GST_STATE_READY); 268 GRefPtr<GstElement> platformSink = adoptGRef(GST_ELEMENT(gst_child_proxy_get_child_by_index(childProxy, 0))); 269 GstElementFactory* factory = gst_element_get_factory(platformSink.get()); 270 271 // Dispose now un-needed autoaudiosink. 265 272 gst_element_set_state(sink.get(), GST_STATE_NULL); 266 273 274 // Create a fresh new audio sink compatible with the platform. 275 m_audioSink = gst_element_factory_create(factory, nullptr); 267 276 m_audioRenderer = adoptGRef(owr_gst_audio_renderer_new(m_audioSink.get())); 268 277 } … … 295 304 auto mediaSource = OWR_MEDIA_SOURCE(realTimeMediaSource->mediaSource()); 296 305 306 GST_DEBUG("%s track now %s", track.type() == RealtimeMediaSource::Audio ? "audio":"video", realTimeMediaSource->muted() ? "muted":"un-muted"); 297 307 switch (track.type()) { 298 308 case RealtimeMediaSource::Audio: … … 357 367 358 368 MediaPlayerPrivateGStreamerBase::setSize(size); 359 g_object_set(m_videoRenderer.get(), "width", size.width(), "height", size.height(), nullptr); 369 if (m_videoRenderer) 370 g_object_set(m_videoRenderer.get(), "width", size.width(), "height", size.height(), nullptr); 360 371 } 361 372 -
trunk/Source/WebCore/platform/mediastream/openwebrtc/RealtimeMediaSourceCenterOwr.cpp
r205929 r206204 147 147 RefPtr<RealtimeMediaSource> source = sourceIterator->value; 148 148 if (source->type() == RealtimeMediaSource::Video) 149 audioSources.append(source.release());149 videoSources.append(source.release()); 150 150 } 151 151 }
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