Changeset 207665 in webkit
- Timestamp:
- Oct 21, 2016, 3:20:23 AM (9 years ago)
- Location:
- trunk
- Files:
-
- 3 added
- 5 edited
Legend:
- Unmodified
- Added
- Removed
-
trunk/ChangeLog
r207658 r207665 1 2016-10-21 Adam Bergkvist <adam.bergkvist@ericsson.com> 2 3 WebRTC: [GTK] Add MediaEndpointOwr - an OpenWebRTC WebRTC backend 4 https://bugs.webkit.org/show_bug.cgi?id=163327 5 6 Reviewed by Philippe Normand. 7 8 Add manual WebRTC test. Test features: 9 - Two RTCPeerConnection instances communicate in a single browser tab. 10 - Supports setting up bidirectional media with a single SDP dialog, as 11 well as one direction at a time. 12 - Strips vendor prefixes (runs in Chrome and Firefox as well) 13 - Supports modern as well as legacy APIs (mainly to make the test run 14 in Chrome) 15 16 * ManualTests/webrtc-one-tab-p2p.html: Added. 17 1 18 2016-10-20 Carlos Garcia Campos <cgarcia@igalia.com> 2 19 -
trunk/Source/WebCore/CMakeLists.txt
r207617 r207665 2311 2311 2312 2312 platform/mediastream/MediaConstraints.cpp 2313 platform/mediastream/MediaEndpoint.cpp2314 2313 platform/mediastream/MediaEndpointConfiguration.cpp 2315 2314 platform/mediastream/MediaStreamPrivate.cpp -
trunk/Source/WebCore/ChangeLog
r207663 r207665 1 2016-10-21 Adam Bergkvist <adam.bergkvist@ericsson.com> 2 3 WebRTC: [GTK] Add MediaEndpointOwr - an OpenWebRTC WebRTC backend 4 https://bugs.webkit.org/show_bug.cgi?id=163327 5 6 Reviewed by Philippe Normand. 7 8 Add MediaEndpointOwr which is a MediaEndpoint implementation (WebRTC backend) based on 9 OpenWebRTC [1]. The WebRTC backend can be tested with a manual test. Automatic testing 10 is still done with MockMediaEndpoint. 11 12 [1] http://www.openwebrtc.org/ 13 14 Testing: Added manual test (webrtc-one-tab-p2p.html) 15 16 * CMakeLists.txt: 17 * platform/GStreamer.cmake: 18 * platform/mediastream/openwebrtc/MediaEndpointOwr.cpp: Added. 19 (WebCore::createMediaEndpointOwr): 20 (WebCore::MediaEndpointOwr::MediaEndpointOwr): 21 (WebCore::MediaEndpointOwr::~MediaEndpointOwr): 22 (WebCore::MediaEndpointOwr::setConfiguration): 23 (WebCore::cryptoDataCallback): 24 (WebCore::MediaEndpointOwr::generateDtlsInfo): 25 (WebCore::MediaEndpointOwr::getDefaultAudioPayloads): 26 (WebCore::MediaEndpointOwr::getDefaultVideoPayloads): 27 (WebCore::payloadsContainType): 28 (WebCore::MediaEndpointOwr::filterPayloads): 29 (WebCore::MediaEndpointOwr::updateReceiveConfiguration): 30 (WebCore::findRtxPayload): 31 (WebCore::MediaEndpointOwr::updateSendConfiguration): 32 (WebCore::MediaEndpointOwr::addRemoteCandidate): 33 (WebCore::MediaEndpointOwr::replaceMutedRemoteSourceMid): 34 (WebCore::MediaEndpointOwr::createMutedRemoteSource): 35 (WebCore::MediaEndpointOwr::replaceSendSource): 36 (WebCore::MediaEndpointOwr::stop): 37 (WebCore::MediaEndpointOwr::transceiverIndexForSession): 38 (WebCore::MediaEndpointOwr::sessionMid): 39 (WebCore::MediaEndpointOwr::matchTransceiverByMid): 40 (WebCore::MediaEndpointOwr::dispatchNewIceCandidate): 41 (WebCore::MediaEndpointOwr::dispatchGatheringDone): 42 (WebCore::MediaEndpointOwr::processIceTransportStateChange): 43 (WebCore::MediaEndpointOwr::dispatchDtlsFingerprint): 44 (WebCore::MediaEndpointOwr::unmuteRemoteSource): 45 (WebCore::MediaEndpointOwr::prepareSession): 46 (WebCore::MediaEndpointOwr::prepareMediaSession): 47 (WebCore::parseHelperServerUrl): 48 (WebCore::MediaEndpointOwr::ensureTransportAgentAndTransceivers): 49 (WebCore::MediaEndpointOwr::internalAddRemoteCandidate): 50 (WebCore::gotCandidate): 51 (WebCore::candidateGatheringDone): 52 (WebCore::iceConnectionStateChange): 53 (WebCore::gotIncomingSource): 54 * platform/mediastream/openwebrtc/MediaEndpointOwr.h: Added. 55 (WebCore::OwrTransceiver::create): 56 (WebCore::OwrTransceiver::~OwrTransceiver): 57 (WebCore::OwrTransceiver::mid): 58 (WebCore::OwrTransceiver::session): 59 (WebCore::OwrTransceiver::owrIceState): 60 (WebCore::OwrTransceiver::setOwrIceState): 61 (WebCore::OwrTransceiver::gotEndOfRemoteCandidates): 62 (WebCore::OwrTransceiver::markGotEndOfRemoteCandidates): 63 (WebCore::OwrTransceiver::OwrTransceiver): 64 * platform/mediastream/openwebrtc/RealtimeMediaSourceOwr.h: 65 (WebCore::RealtimeMediaSourceOwr::RealtimeMediaSourceOwr): 66 (WebCore::RealtimeMediaSourceOwr::swapOutShallowSource): 67 Add support for an initially muted source. This is used for early 68 creation of remote sources. 69 1 70 2016-10-21 Javier Fernandez <jfernandez@igalia.com> 2 71 -
trunk/Source/WebCore/platform/GStreamer.cmake
r206862 r207665 15 15 platform/graphics/gstreamer/MediaPlayerPrivateGStreamerOwr.cpp 16 16 17 platform/mediastream/openwebrtc/MediaEndpointOwr.cpp 17 18 platform/mediastream/openwebrtc/OpenWebRTCUtilities.cpp 18 19 platform/mediastream/openwebrtc/RealtimeMediaSourceCenterOwr.cpp -
trunk/Source/WebCore/platform/mediastream/openwebrtc/RealtimeMediaSourceOwr.h
r205961 r207665 55 55 , m_mediaSource(mediaSource) 56 56 { 57 if (!mediaSource) 58 m_muted = true; 57 59 } 58 60 … … 64 66 65 67 virtual ~RealtimeMediaSourceOwr() { } 68 69 void swapOutShallowSource(OwrMediaSource& realSource) 70 { 71 m_mediaSource = &realSource; 72 setMuted(false); 73 } 66 74 67 75 virtual RefPtr<RealtimeMediaSourceCapabilities> capabilities() { return m_capabilities; }
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