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<dl class="meta">
<dt>Revision</dt> <dd><a href="http://trac.webkit.org/projects/webkit/changeset/213736">213736</a></dd>
<dt>Author</dt> <dd>commit-queue@webkit.org</dd>
<dt>Date</dt> <dd>2017-03-10 16:46:44 -0800 (Fri, 10 Mar 2017)</dd>
</dl>
<h3>Log Message</h3>
<pre>Move libwebrtc backend to using tracks
https://bugs.webkit.org/show_bug.cgi?id=169472
Patch by Youenn Fablet <youenn@apple.com> on 2017-03-10
Reviewed by Alex Christensen.
Source/ThirdParty/libwebrtc:
* Source/webrtc/pc/rtcstatscollector.cc: Moving from using media stream to tracks.
Source/WebCore:
Covered by current test sets.
This change allows to move away from streams and use more tracks.
This allows future better alignment with the spec and better implementation of sender/receover/transceiver.
Small refactoring to use more Ref<> in WebRTC track-based code.
Added a notifyAddedTrack to notify the backend that a track was added.
For libwebrtc, this allows calling AddTrack before making the offer.
Updated mock libwebrtc backend accordingly.
* Modules/mediastream/MediaEndpointPeerConnection.cpp:
(WebCore::MediaEndpointPeerConnection::replaceTrack):
(WebCore::MediaEndpointPeerConnection::replaceTrackTask):
* Modules/mediastream/MediaEndpointPeerConnection.h:
* Modules/mediastream/PeerConnectionBackend.h:
(WebCore::PeerConnectionBackend::notifyAddedTrack):
* Modules/mediastream/RTCPeerConnection.cpp:
(WebCore::RTCPeerConnection::addTrack):
(WebCore::RTCPeerConnection::replaceTrack):
* Modules/mediastream/RTCPeerConnection.h:
* Modules/mediastream/RTCRtpSender.cpp:
(WebCore::RTCRtpSender::create):
(WebCore::RTCRtpSender::RTCRtpSender):
(WebCore::RTCRtpSender::setTrack):
* Modules/mediastream/RTCRtpSender.h:
* Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
(WebCore::LibWebRTCMediaEndpoint::addTrack):
(WebCore::LibWebRTCMediaEndpoint::doCreateOffer):
(WebCore::LibWebRTCMediaEndpoint::doCreateAnswer):
(WebCore::LibWebRTCMediaEndpoint::getStats):
* Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
* Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp:
(WebCore::LibWebRTCPeerConnectionBackend::doAddIceCandidate):
(WebCore::LibWebRTCPeerConnectionBackend::notifyAddedTrack):
* Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h:
* testing/MockLibWebRTCPeerConnection.cpp:
(WebCore::MockLibWebRTCPeerConnection::AddTrack):
(WebCore::MockLibWebRTCPeerConnection::RemoveTrack):
(WebCore::MockLibWebRTCPeerConnection::CreateOffer):
(WebCore::MockLibWebRTCPeerConnection::CreateAnswer):
* testing/MockLibWebRTCPeerConnection.h:
(WebCore::MockRtpSender::MockRtpSender):
(WebCore::MockRtpSender::ssrc):
(WebCore::MockRtpSender::media_type):
(WebCore::MockRtpSender::id):
(WebCore::MockRtpSender::stream_ids):
(WebCore::MockRtpSender::GetParameters):
(WebCore::MockRtpSender::SetParameters):
(WebCore::MockRtpSender::GetDtmfSender):
LayoutTests:
* TestExpectations:</pre>
<h3>Modified Paths</h3>
<ul>
<li><a href="#trunkLayoutTestsChangeLog">trunk/LayoutTests/ChangeLog</a></li>
<li><a href="#trunkLayoutTestsTestExpectations">trunk/LayoutTests/TestExpectations</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcChangeLog">trunk/Source/ThirdParty/libwebrtc/ChangeLog</a></li>
<li><a href="#trunkSourceThirdPartylibwebrtcSourcewebrtcpcrtcstatscollectorcc">trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc</a></li>
<li><a href="#trunkSourceWebCoreChangeLog">trunk/Source/WebCore/ChangeLog</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamMediaEndpointPeerConnectioncpp">trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.cpp</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamMediaEndpointPeerConnectionh">trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.h</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamPeerConnectionBackendh">trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamRTCPeerConnectioncpp">trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamRTCPeerConnectionh">trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.h</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamRTCRtpSendercpp">trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamRTCRtpSenderh">trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.h</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamlibwebrtcLibWebRTCMediaEndpointcpp">trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamlibwebrtcLibWebRTCMediaEndpointh">trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamlibwebrtcLibWebRTCPeerConnectionBackendcpp">trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp</a></li>
<li><a href="#trunkSourceWebCoreModulesmediastreamlibwebrtcLibWebRTCPeerConnectionBackendh">trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h</a></li>
<li><a href="#trunkSourceWebCoretestingMockLibWebRTCPeerConnectioncpp">trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp</a></li>
<li><a href="#trunkSourceWebCoretestingMockLibWebRTCPeerConnectionh">trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.h</a></li>
</ul>
</div>
<div id="patch">
<h3>Diff</h3>
<a id="trunkLayoutTestsChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/LayoutTests/ChangeLog (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/ChangeLog 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/LayoutTests/ChangeLog 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -1,3 +1,12 @@
</span><ins>+2017-03-10 Youenn Fablet <youenn@apple.com>
+
+ Move libwebrtc backend to using tracks
+ https://bugs.webkit.org/show_bug.cgi?id=169472
+
+ Reviewed by Alex Christensen.
+
+ * TestExpectations:
+
</ins><span class="cx"> 2017-03-10 Ryan Haddad <ryanhaddad@apple.com>
</span><span class="cx">
</span><span class="cx"> Mark compositing/regions/transformed-layer-inside-transformed-layer.html as failing on ios-simulator.
</span></span></pre></div>
<a id="trunkLayoutTestsTestExpectations"></a>
<div class="modfile"><h4>Modified: trunk/LayoutTests/TestExpectations (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/TestExpectations 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/LayoutTests/TestExpectations 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -715,8 +715,6 @@
</span><span class="cx">
</span><span class="cx"> # See webkit.org/b/168736
</span><span class="cx"> fast/mediastream/RTCPeerConnection-addIceCandidate.html [ Failure ]
</span><del>-fast/mediastream/RTCPeerConnection-addTrack-reuse-sender.html [ Failure ]
-fast/mediastream/RTCPeerConnection-legacy-stream-based-api.html [ Failure ]
</del><span class="cx"> imported/w3c/web-platform-tests/webrtc/datachannel-emptystring.html [ Failure ]
</span><span class="cx"> imported/w3c/web-platform-tests/webrtc/no-media-call.html [ Failure ]
</span><span class="cx"> imported/w3c/web-platform-tests/webrtc/promises-call.html [ Failure ]
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/ChangeLog (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/ChangeLog 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/Source/ThirdParty/libwebrtc/ChangeLog 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -1,3 +1,12 @@
</span><ins>+2017-03-10 Youenn Fablet <youenn@apple.com>
+
+ Move libwebrtc backend to using tracks
+ https://bugs.webkit.org/show_bug.cgi?id=169472
+
+ Reviewed by Alex Christensen.
+
+ * Source/webrtc/pc/rtcstatscollector.cc: Moving from using media stream to tracks.
+
</ins><span class="cx"> 2017-03-08 Youenn Fablet <youenn@apple.com>
</span><span class="cx">
</span><span class="cx"> Use H264 hardware encoder for Mac libwebrtc
</span></span></pre></div>
<a id="trunkSourceThirdPartylibwebrtcSourcewebrtcpcrtcstatscollectorcc"></a>
<div class="modfile"><h4>Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -1198,6 +1198,20 @@
</span><span class="cx"> RTCStatsCollector::PrepareTrackToID_s() const {
</span><span class="cx"> RTC_DCHECK(signaling_thread_->IsCurrent());
</span><span class="cx"> std::map<MediaStreamTrackInterface*, std::string> track_to_id;
</span><ins>+#if defined(WEBRTC_WEBKIT_BUILD)
+ for (auto& sender : pc_->GetSenders()) {
+ auto track = sender->track();
+ if (!track)
+ continue;
+ track_to_id[track.get()] = track->id();
+ }
+ for (auto& receiver : pc_->GetReceivers()) {
+ auto track = receiver->track();
+ if (!track)
+ continue;
+ track_to_id[track.get()] = track->id();
+ }
+#else
</ins><span class="cx"> StreamCollectionInterface* local_and_remote_streams[] =
</span><span class="cx"> { pc_->local_streams().get(), pc_->remote_streams().get() };
</span><span class="cx"> for (auto& streams : local_and_remote_streams) {
</span><span class="lines">@@ -1215,6 +1229,7 @@
</span><span class="cx"> }
</span><span class="cx"> }
</span><span class="cx"> }
</span><ins>+#endif
</ins><span class="cx"> return track_to_id;
</span><span class="cx"> }
</span><span class="cx">
</span></span></pre></div>
<a id="trunkSourceWebCoreChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/ChangeLog (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/ChangeLog 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/Source/WebCore/ChangeLog 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -1,3 +1,60 @@
</span><ins>+2017-03-10 Youenn Fablet <youenn@apple.com>
+
+ Move libwebrtc backend to using tracks
+ https://bugs.webkit.org/show_bug.cgi?id=169472
+
+ Reviewed by Alex Christensen.
+
+ Covered by current test sets.
+
+ This change allows to move away from streams and use more tracks.
+ This allows future better alignment with the spec and better implementation of sender/receover/transceiver.
+
+ Small refactoring to use more Ref<> in WebRTC track-based code.
+ Added a notifyAddedTrack to notify the backend that a track was added.
+ For libwebrtc, this allows calling AddTrack before making the offer.
+ Updated mock libwebrtc backend accordingly.
+
+ * Modules/mediastream/MediaEndpointPeerConnection.cpp:
+ (WebCore::MediaEndpointPeerConnection::replaceTrack):
+ (WebCore::MediaEndpointPeerConnection::replaceTrackTask):
+ * Modules/mediastream/MediaEndpointPeerConnection.h:
+ * Modules/mediastream/PeerConnectionBackend.h:
+ (WebCore::PeerConnectionBackend::notifyAddedTrack):
+ * Modules/mediastream/RTCPeerConnection.cpp:
+ (WebCore::RTCPeerConnection::addTrack):
+ (WebCore::RTCPeerConnection::replaceTrack):
+ * Modules/mediastream/RTCPeerConnection.h:
+ * Modules/mediastream/RTCRtpSender.cpp:
+ (WebCore::RTCRtpSender::create):
+ (WebCore::RTCRtpSender::RTCRtpSender):
+ (WebCore::RTCRtpSender::setTrack):
+ * Modules/mediastream/RTCRtpSender.h:
+ * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
+ (WebCore::LibWebRTCMediaEndpoint::addTrack):
+ (WebCore::LibWebRTCMediaEndpoint::doCreateOffer):
+ (WebCore::LibWebRTCMediaEndpoint::doCreateAnswer):
+ (WebCore::LibWebRTCMediaEndpoint::getStats):
+ * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
+ * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp:
+ (WebCore::LibWebRTCPeerConnectionBackend::doAddIceCandidate):
+ (WebCore::LibWebRTCPeerConnectionBackend::notifyAddedTrack):
+ * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h:
+ * testing/MockLibWebRTCPeerConnection.cpp:
+ (WebCore::MockLibWebRTCPeerConnection::AddTrack):
+ (WebCore::MockLibWebRTCPeerConnection::RemoveTrack):
+ (WebCore::MockLibWebRTCPeerConnection::CreateOffer):
+ (WebCore::MockLibWebRTCPeerConnection::CreateAnswer):
+ * testing/MockLibWebRTCPeerConnection.h:
+ (WebCore::MockRtpSender::MockRtpSender):
+ (WebCore::MockRtpSender::ssrc):
+ (WebCore::MockRtpSender::media_type):
+ (WebCore::MockRtpSender::id):
+ (WebCore::MockRtpSender::stream_ids):
+ (WebCore::MockRtpSender::GetParameters):
+ (WebCore::MockRtpSender::SetParameters):
+ (WebCore::MockRtpSender::GetDtmfSender):
+
</ins><span class="cx"> 2017-03-10 Zalan Bujtas <zalan@apple.com>
</span><span class="cx">
</span><span class="cx"> Simple line layout: Check how many orphans needed on the current page before breaking.
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamMediaEndpointPeerConnectioncpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.cpp (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.cpp 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.cpp 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -693,7 +693,7 @@
</span><span class="cx"> return m_mediaEndpoint->createDataChannelHandler(label, options);
</span><span class="cx"> }
</span><span class="cx">
</span><del>-void MediaEndpointPeerConnection::replaceTrack(RTCRtpSender& sender, RefPtr<MediaStreamTrack>&& withTrack, DOMPromise<void>&& promise)
</del><ins>+void MediaEndpointPeerConnection::replaceTrack(RTCRtpSender& sender, Ref<MediaStreamTrack>&& withTrack, DOMPromise<void>&& promise)
</ins><span class="cx"> {
</span><span class="cx"> RTCRtpTransceiver* transceiver = matchTransceiver(m_peerConnection.getTransceivers(), [&sender] (RTCRtpTransceiver& current) {
</span><span class="cx"> return &current.sender() == &sender;
</span><span class="lines">@@ -713,7 +713,7 @@
</span><span class="cx"> });
</span><span class="cx"> }
</span><span class="cx">
</span><del>-void MediaEndpointPeerConnection::replaceTrackTask(RTCRtpSender& sender, const String& mid, RefPtr<MediaStreamTrack>&& withTrack, DOMPromise<void>& promise)
</del><ins>+void MediaEndpointPeerConnection::replaceTrackTask(RTCRtpSender& sender, const String& mid, Ref<MediaStreamTrack>&& withTrack, DOMPromise<void>& promise)
</ins><span class="cx"> {
</span><span class="cx"> if (m_peerConnection.internalSignalingState() == SignalingState::Closed)
</span><span class="cx"> return;
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamMediaEndpointPeerConnectionh"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.h (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.h 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.h 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -62,7 +62,7 @@
</span><span class="cx"> Vector<RefPtr<MediaStream>> getRemoteStreams() const final;
</span><span class="cx">
</span><span class="cx"> Ref<RTCRtpReceiver> createReceiver(const String& transceiverMid, const String& trackKind, const String& trackId) final;
</span><del>- void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromise<void>&&) final;
</del><ins>+ void replaceTrack(RTCRtpSender&, Ref<MediaStreamTrack>&&, DOMPromise<void>&&) final;
</ins><span class="cx">
</span><span class="cx"> void emulatePlatformEvent(const String& action) final;
</span><span class="cx">
</span><span class="lines">@@ -84,7 +84,7 @@
</span><span class="cx">
</span><span class="cx"> void addIceCandidateTask(RTCIceCandidate&);
</span><span class="cx">
</span><del>- void replaceTrackTask(RTCRtpSender&, const String& mid, RefPtr<MediaStreamTrack>&&, DOMPromise<void>&);
</del><ins>+ void replaceTrackTask(RTCRtpSender&, const String& mid, Ref<MediaStreamTrack>&&, DOMPromise<void>&);
</ins><span class="cx">
</span><span class="cx"> bool localDescriptionTypeValidForState(RTCSessionDescription::SdpType) const;
</span><span class="cx"> bool remoteDescriptionTypeValidForState(RTCSessionDescription::SdpType) const;
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamPeerConnectionBackendh"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -93,7 +93,8 @@
</span><span class="cx"> virtual Vector<RefPtr<MediaStream>> getRemoteStreams() const = 0;
</span><span class="cx">
</span><span class="cx"> virtual Ref<RTCRtpReceiver> createReceiver(const String& transceiverMid, const String& trackKind, const String& trackId) = 0;
</span><del>- virtual void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromise<void>&&) = 0;
</del><ins>+ virtual void replaceTrack(RTCRtpSender&, Ref<MediaStreamTrack>&&, DOMPromise<void>&&) = 0;
+ virtual void notifyAddedTrack(RTCRtpSender&) { }
</ins><span class="cx">
</span><span class="cx"> void markAsNeedingNegotiation();
</span><span class="cx"> bool isNegotiationNeeded() const { return m_negotiationNeeded; };
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamRTCPeerConnectioncpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -123,6 +123,7 @@
</span><span class="cx"> existingSender.setMediaStreamIds(WTFMove(mediaStreamIds));
</span><span class="cx"> transceiver->enableSendingDirection();
</span><span class="cx"> sender = &existingSender;
</span><ins>+
</ins><span class="cx"> break;
</span><span class="cx"> }
</span><span class="cx"> }
</span><span class="lines">@@ -146,6 +147,7 @@
</span><span class="cx">
</span><span class="cx"> m_backend->markAsNeedingNegotiation();
</span><span class="cx">
</span><ins>+ m_backend->notifyAddedTrack(*sender);
</ins><span class="cx"> return Ref<RTCRtpSender> { *sender };
</span><span class="cx"> }
</span><span class="cx">
</span><span class="lines">@@ -510,7 +512,7 @@
</span><span class="cx"> dispatchEvent(event);
</span><span class="cx"> }
</span><span class="cx">
</span><del>-void RTCPeerConnection::replaceTrack(RTCRtpSender& sender, RefPtr<MediaStreamTrack>&& withTrack, DOMPromise<void>&& promise)
</del><ins>+void RTCPeerConnection::replaceTrack(RTCRtpSender& sender, Ref<MediaStreamTrack>&& withTrack, DOMPromise<void>&& promise)
</ins><span class="cx"> {
</span><span class="cx"> m_backend->replaceTrack(sender, WTFMove(withTrack), WTFMove(promise));
</span><span class="cx"> }
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamRTCPeerConnectionh"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.h (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.h 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.h 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -159,7 +159,7 @@
</span><span class="cx"> bool canSuspendForDocumentSuspension() const final;
</span><span class="cx">
</span><span class="cx"> // RTCRtpSenderClient
</span><del>- void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromise<void>&&) final;
</del><ins>+ void replaceTrack(RTCRtpSender&, Ref<MediaStreamTrack>&&, DOMPromise<void>&&) final;
</ins><span class="cx">
</span><span class="cx"> PeerConnectionStates::SignalingState m_signalingState { PeerConnectionStates::SignalingState::Stable };
</span><span class="cx"> PeerConnectionStates::IceGatheringState m_iceGatheringState { PeerConnectionStates::IceGatheringState::New };
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamRTCRtpSendercpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -39,31 +39,29 @@
</span><span class="cx">
</span><span class="cx"> Ref<RTCRtpSender> RTCRtpSender::create(Ref<MediaStreamTrack>&& track, Vector<String>&& mediaStreamIds, RTCRtpSenderClient& client)
</span><span class="cx"> {
</span><del>- const String& trackKind = track->kind();
- return adoptRef(*new RTCRtpSender(WTFMove(track), trackKind, WTFMove(mediaStreamIds), client));
</del><ins>+ auto sender = adoptRef(*new RTCRtpSender(track->kind(), WTFMove(mediaStreamIds), client));
+ sender->setTrack(WTFMove(track));
+ return sender;
</ins><span class="cx"> }
</span><span class="cx">
</span><span class="cx"> Ref<RTCRtpSender> RTCRtpSender::create(const String& trackKind, Vector<String>&& mediaStreamIds, RTCRtpSenderClient& client)
</span><span class="cx"> {
</span><del>- return adoptRef(*new RTCRtpSender(nullptr, trackKind, WTFMove(mediaStreamIds), client));
</del><ins>+ return adoptRef(*new RTCRtpSender(trackKind, WTFMove(mediaStreamIds), client));
</ins><span class="cx"> }
</span><span class="cx">
</span><del>-RTCRtpSender::RTCRtpSender(RefPtr<MediaStreamTrack>&& track, const String& trackKind, Vector<String>&& mediaStreamIds, RTCRtpSenderClient& client)
</del><ins>+RTCRtpSender::RTCRtpSender(const String& trackKind, Vector<String>&& mediaStreamIds, RTCRtpSenderClient& client)
</ins><span class="cx"> : RTCRtpSenderReceiverBase()
</span><span class="cx"> , m_trackKind(trackKind)
</span><span class="cx"> , m_mediaStreamIds(WTFMove(mediaStreamIds))
</span><span class="cx"> , m_client(&client)
</span><span class="cx"> {
</span><del>- setTrack(WTFMove(track));
</del><span class="cx"> }
</span><span class="cx">
</span><del>-void RTCRtpSender::setTrack(RefPtr<MediaStreamTrack>&& track)
</del><ins>+void RTCRtpSender::setTrack(Ref<MediaStreamTrack>&& track)
</ins><span class="cx"> {
</span><del>- // Save the id from the first non-null track set. That id will be used to negotiate the sender
- // even if the track is replaced.
- if (!m_track && track)
- m_trackId = track->id();
-
</del><ins>+ ASSERT(!isStopped());
+ ASSERT(!m_track);
+ m_trackId = track->id();
</ins><span class="cx"> m_track = WTFMove(track);
</span><span class="cx"> }
</span><span class="cx">
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamRTCRtpSenderh"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.h (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.h 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.h 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -39,7 +39,7 @@
</span><span class="cx">
</span><span class="cx"> class RTCRtpSenderClient {
</span><span class="cx"> public:
</span><del>- virtual void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromise<void>&&) = 0;
</del><ins>+ virtual void replaceTrack(RTCRtpSender&, Ref<MediaStreamTrack>&&, DOMPromise<void>&&) = 0;
</ins><span class="cx">
</span><span class="cx"> virtual ~RTCRtpSenderClient() { }
</span><span class="cx"> };
</span><span class="lines">@@ -57,12 +57,12 @@
</span><span class="cx">
</span><span class="cx"> bool isStopped() const { return !m_client; }
</span><span class="cx"> void stop() { m_client = nullptr; }
</span><del>- void setTrack(RefPtr<MediaStreamTrack>&&);
</del><ins>+ void setTrack(Ref<MediaStreamTrack>&&);
</ins><span class="cx">
</span><span class="cx"> ExceptionOr<void> replaceTrack(Ref<MediaStreamTrack>&&, DOMPromise<void>&&);
</span><span class="cx">
</span><span class="cx"> private:
</span><del>- RTCRtpSender(RefPtr<MediaStreamTrack>&&, const String& trackKind, Vector<String>&& mediaStreamIds, RTCRtpSenderClient&);
</del><ins>+ RTCRtpSender(const String& trackKind, Vector<String>&& mediaStreamIds, RTCRtpSenderClient&);
</ins><span class="cx">
</span><span class="cx"> String m_trackId;
</span><span class="cx"> String m_trackKind;
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamlibwebrtcLibWebRTCMediaEndpointcpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -158,19 +158,34 @@
</span><span class="cx"> m_backend->SetRemoteDescription(&m_setRemoteSessionDescriptionObserver, sessionDescription.release());
</span><span class="cx"> }
</span><span class="cx">
</span><del>-static inline std::string streamId(RTCPeerConnection& connection)
</del><ins>+void LibWebRTCMediaEndpoint::addTrack(MediaStreamTrack& track, const Vector<String>& mediaStreamIds)
</ins><span class="cx"> {
</span><del>- auto& senders = connection.getSenders();
- if (senders.size()) {
- for (RTCRtpSender& sender : senders) {
- auto* track = sender.track();
- if (track) {
- ASSERT(sender.mediaStreamIds().size() == 1);
- return std::string(sender.mediaStreamIds().first().utf8().data());
- }
- }
</del><ins>+ if (!LibWebRTCProvider::factory())
+ return;
+
+ std::vector<webrtc::MediaStreamInterface*> mediaStreams;
+ rtc::scoped_refptr<webrtc::MediaStreamInterface> mediaStream = nullptr;
+ if (mediaStreamIds.size()) {
+ // libwebrtc is only using the first one if any.
+ mediaStream = LibWebRTCProvider::factory()->CreateLocalMediaStream(mediaStreamIds[0].utf8().data());
+ mediaStreams.push_back(mediaStream.get());
</ins><span class="cx"> }
</span><del>- return "av_label";
</del><ins>+
+ auto& source = track.source();
+ ASSERT(source.type() != RealtimeMediaSource::None);
+
+ if (source.type() == RealtimeMediaSource::Audio) {
+ auto trackSource = RealtimeOutgoingAudioSource::create(source);
+ auto audioTrack = LibWebRTCProvider::factory()->CreateAudioTrack(track.id().utf8().data(), trackSource.ptr());
+ trackSource->setTrack(rtc::scoped_refptr<webrtc::AudioTrackInterface>(audioTrack));
+ m_peerConnectionBackend.addAudioSource(WTFMove(trackSource));
+ m_backend->AddTrack(audioTrack.get(), WTFMove(mediaStreams));
+ } else {
+ auto videoSource = RealtimeOutgoingVideoSource::create(source);
+ auto videoTrack = LibWebRTCProvider::factory()->CreateVideoTrack(track.id().utf8().data(), videoSource.ptr());
+ m_peerConnectionBackend.addVideoSource(WTFMove(videoSource));
+ m_backend->AddTrack(videoTrack.get(), WTFMove(mediaStreams));
+ }
</ins><span class="cx"> }
</span><span class="cx">
</span><span class="cx"> void LibWebRTCMediaEndpoint::doCreateOffer()
</span><span class="lines">@@ -181,31 +196,6 @@
</span><span class="cx"> }
</span><span class="cx">
</span><span class="cx"> m_isInitiator = true;
</span><del>- auto& senders = m_peerConnectionBackend.connection().getSenders();
- if (senders.size()) {
- // FIXME: We only support one stream for the moment.
- auto stream = LibWebRTCProvider::factory()->CreateLocalMediaStream(streamId(m_peerConnectionBackend.connection()));
- for (RTCRtpSender& sender : senders) {
- auto* track = sender.track();
- if (track) {
- ASSERT(sender.mediaStreamIds().size() == 1);
- auto& source = track->source();
- if (source.type() == RealtimeMediaSource::Audio) {
- auto trackSource = RealtimeOutgoingAudioSource::create(source);
- auto rtcTrack = LibWebRTCProvider::factory()->CreateAudioTrack(track->id().utf8().data(), trackSource.ptr());
- trackSource->setTrack(rtc::scoped_refptr<webrtc::AudioTrackInterface>(rtcTrack));
- m_peerConnectionBackend.addAudioSource(WTFMove(trackSource));
- stream->AddTrack(WTFMove(rtcTrack));
- } else {
- auto videoSource = RealtimeOutgoingVideoSource::create(source);
- auto videoTrack = LibWebRTCProvider::factory()->CreateVideoTrack(track->id().utf8().data(), videoSource.ptr());
- m_peerConnectionBackend.addVideoSource(WTFMove(videoSource));
- stream->AddTrack(WTFMove(videoTrack));
- }
- }
- }
- m_backend->AddStream(stream);
- }
</del><span class="cx"> m_backend->CreateOffer(&m_createSessionDescriptionObserver, nullptr);
</span><span class="cx"> }
</span><span class="cx">
</span><span class="lines">@@ -217,37 +207,13 @@
</span><span class="cx"> }
</span><span class="cx">
</span><span class="cx"> m_isInitiator = false;
</span><del>-
- auto& senders = m_peerConnectionBackend.connection().getSenders();
- if (senders.size()) {
- // FIXME: We only support one stream for the moment.
- auto stream = LibWebRTCProvider::factory()->CreateLocalMediaStream(streamId(m_peerConnectionBackend.connection()));
- for (RTCRtpSender& sender : senders) {
- auto* track = sender.track();
- if (track) {
- ASSERT(sender.mediaStreamIds().size() == 1);
- auto& source = track->source();
- if (source.type() == RealtimeMediaSource::Audio) {
- auto trackSource = RealtimeOutgoingAudioSource::create(source);
- auto rtcTrack = LibWebRTCProvider::factory()->CreateAudioTrack(track->id().utf8().data(), trackSource.ptr());
- trackSource->setTrack(rtc::scoped_refptr<webrtc::AudioTrackInterface>(rtcTrack));
- m_peerConnectionBackend.addAudioSource(WTFMove(trackSource));
- stream->AddTrack(WTFMove(rtcTrack));
- } else {
- auto videoSource = RealtimeOutgoingVideoSource::create(source);
- auto videoTrack = LibWebRTCProvider::factory()->CreateVideoTrack(track->id().utf8().data(), videoSource.ptr());
- m_peerConnectionBackend.addVideoSource(WTFMove(videoSource));
- stream->AddTrack(WTFMove(videoTrack));
- }
- }
- }
- m_backend->AddStream(stream);
- }
</del><span class="cx"> m_backend->CreateAnswer(&m_createSessionDescriptionObserver, nullptr);
</span><span class="cx"> }
</span><span class="cx">
</span><span class="cx"> void LibWebRTCMediaEndpoint::getStats(MediaStreamTrack* track, const DeferredPromise& promise)
</span><span class="cx"> {
</span><ins>+ UNUSED_PARAM(track);
+ UNUSED_PARAM(promise);
</ins><span class="cx"> m_backend->GetStats(StatsCollector::create(*this, promise, track).get());
</span><span class="cx"> }
</span><span class="cx">
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamlibwebrtcLibWebRTCMediaEndpointh"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -79,6 +79,8 @@
</span><span class="cx"> RefPtr<RTCSessionDescription> pendingLocalDescription() const;
</span><span class="cx"> RefPtr<RTCSessionDescription> pendingRemoteDescription() const;
</span><span class="cx">
</span><ins>+ void addTrack(MediaStreamTrack&, const Vector<String>&);
+
</ins><span class="cx"> private:
</span><span class="cx"> LibWebRTCMediaEndpoint(LibWebRTCPeerConnectionBackend&, LibWebRTCProvider&);
</span><span class="cx">
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamlibwebrtcLibWebRTCPeerConnectionBackendcpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -169,11 +169,6 @@
</span><span class="cx">
</span><span class="cx"> void LibWebRTCPeerConnectionBackend::doAddIceCandidate(RTCIceCandidate& candidate)
</span><span class="cx"> {
</span><del>- if (!m_isRemoteDescriptionSet) {
- addIceCandidateFailed(Exception { INVALID_STATE_ERR, "No remote description set" });
- return;
- }
-
</del><span class="cx"> webrtc::SdpParseError error;
</span><span class="cx"> int sdpMLineIndex = candidate.sdpMLineIndex() ? candidate.sdpMLineIndex().value() : 0;
</span><span class="cx"> std::unique_ptr<webrtc::IceCandidateInterface> rtcCandidate(webrtc::CreateIceCandidate(candidate.sdpMid().utf8().data(), sdpMLineIndex, candidate.candidate().utf8().data(), &error));
</span><span class="lines">@@ -250,6 +245,12 @@
</span><span class="cx"> return m_endpoint->remoteDescription();
</span><span class="cx"> }
</span><span class="cx">
</span><ins>+void LibWebRTCPeerConnectionBackend::notifyAddedTrack(RTCRtpSender& sender)
+{
+ ASSERT(sender.track());
+ m_endpoint->addTrack(*sender.track(), sender.mediaStreamIds());
+}
+
</ins><span class="cx"> } // namespace WebCore
</span><span class="cx">
</span><span class="cx"> #endif // USE(LIBWEBRTC)
</span></span></pre></div>
<a id="trunkSourceWebCoreModulesmediastreamlibwebrtcLibWebRTCPeerConnectionBackendh"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -67,9 +67,10 @@
</span><span class="cx"> RefPtr<RTCSessionDescription> currentRemoteDescription() const final;
</span><span class="cx"> RefPtr<RTCSessionDescription> pendingRemoteDescription() const final;
</span><span class="cx">
</span><ins>+ void notifyAddedTrack(RTCRtpSender&) final;
</ins><span class="cx"> // FIXME: API to implement for real
</span><span class="cx"> Vector<RefPtr<MediaStream>> getRemoteStreams() const final { return { }; }
</span><del>- void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromise<void>&&) final { }
</del><ins>+ void replaceTrack(RTCRtpSender&, Ref<MediaStreamTrack>&&, DOMPromise<void>&&) final { }
</ins><span class="cx">
</span><span class="cx"> void emulatePlatformEvent(const String&) final { }
</span><span class="cx">
</span></span></pre></div>
<a id="trunkSourceWebCoretestingMockLibWebRTCPeerConnectioncpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -226,21 +226,31 @@
</span><span class="cx"> return new rtc::RefCountedObject<MockLibWebRTCDataChannel>(std::string(label), parameters.ordered, parameters.reliable, parameters.id);
</span><span class="cx"> }
</span><span class="cx">
</span><del>-bool MockLibWebRTCPeerConnection::AddStream(webrtc::MediaStreamInterface* stream)
</del><ins>+rtc::scoped_refptr<webrtc::RtpSenderInterface> MockLibWebRTCPeerConnection::AddTrack(webrtc::MediaStreamTrackInterface* track, std::vector<webrtc::MediaStreamInterface*> streams)
</ins><span class="cx"> {
</span><del>- m_stream = stream;
</del><span class="cx"> LibWebRTCProvider::callOnWebRTCSignalingThread([observer = &m_observer] {
</span><span class="cx"> observer->OnRenegotiationNeeded();
</span><span class="cx"> });
</span><del>- return true;
</del><ins>+
+ if (streams.size())
+ m_streamLabel = streams.front()->label();
+
+ m_senders.append(new rtc::RefCountedObject<MockRtpSender>(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>(track)));
+ return m_senders.last().get();
</ins><span class="cx"> }
</span><span class="cx">
</span><del>-void MockLibWebRTCPeerConnection::RemoveStream(webrtc::MediaStreamInterface*)
</del><ins>+bool MockLibWebRTCPeerConnection::RemoveTrack(webrtc::RtpSenderInterface* sender)
</ins><span class="cx"> {
</span><span class="cx"> LibWebRTCProvider::callOnWebRTCSignalingThread([observer = &m_observer] {
</span><span class="cx"> observer->OnRenegotiationNeeded();
</span><span class="cx"> });
</span><del>- m_stream = nullptr;
</del><ins>+ bool isRemoved = false;
+ return m_senders.removeFirstMatching([&](auto& item) {
+ if (item.get() != sender)
+ return false;
+ isRemoved = true;
+ return true;
+ });
</ins><span class="cx"> }
</span><span class="cx">
</span><span class="cx"> void MockLibWebRTCPeerConnection::CreateOffer(webrtc::CreateSessionDescriptionObserver* observer, const webrtc::MediaConstraintsInterface*)
</span><span class="lines">@@ -252,10 +262,13 @@
</span><span class="cx"> "o=- 5667094644266930845 " << m_counter++ << " IN IP4 127.0.0.1\r\n"
</span><span class="cx"> "s=-\r\n"
</span><span class="cx"> "t=0 0\r\n";
</span><del>- if (m_stream) {
</del><ins>+ if (m_senders.size()) {
</ins><span class="cx"> unsigned partCounter = 1;
</span><del>- sdp << "a=msid-semantic:WMS " << m_stream->label() << "\r\n";
- for (auto& audioTrack : m_stream->GetAudioTracks()) {
</del><ins>+ sdp << "a=msid-semantic:WMS " << m_streamLabel << "\r\n";
+ for (auto& sender : m_senders) {
+ auto track = sender->track();
+ if (track->kind() != "audio")
+ continue;
</ins><span class="cx"> sdp <<
</span><span class="cx"> "m=audio 9 UDP/TLS/RTP/SAVPF 111 8 0\r\n"
</span><span class="cx"> "c=IN IP4 0.0.0.0\r\n"
</span><span class="lines">@@ -266,13 +279,16 @@
</span><span class="cx"> "a=rtpmap:8 PCMA/8000\r\n"
</span><span class="cx"> "a=rtpmap:0 PCMU/8000\r\n"
</span><span class="cx"> "a=ssrc:3409173717 cname:/chKzCS9K6KOgL0n\r\n"
</span><del>- "a=msid:" << m_stream->label() << " " << audioTrack->id() << "\r\n"
</del><ins>+ "a=msid:" << m_streamLabel << " " << track->id() << "\r\n"
</ins><span class="cx"> "a=ice-ufrag:e/B1\r\n"
</span><span class="cx"> "a=ice-pwd:Yotk3Im3mnyi+1Q38p51MDub\r\n"
</span><span class="cx"> "a=fingerprint:sha-256 8B:87:09:8A:5D:C2:F3:33:EF:C5:B1:F6:84:3A:3D:D6:A3:E2:9C:17:4C:E7:46:3B:1B:CE:84:98:DD:8E:AF:7B\r\n"
</span><span class="cx"> "a=setup:actpass\r\n";
</span><span class="cx"> }
</span><del>- for (auto& videoTrack : m_stream->GetVideoTracks()) {
</del><ins>+ for (auto& sender : m_senders) {
+ auto track = sender->track();
+ if (track->kind() != "video")
+ continue;
</ins><span class="cx"> sdp <<
</span><span class="cx"> "m=video 9 UDP/TLS/RTP/SAVPF 103 100 120\r\n"
</span><span class="cx"> "c=IN IP4 0.0.0.0\r\n"
</span><span class="lines">@@ -290,7 +306,7 @@
</span><span class="cx"> "a=rtcp-fb:103 ccm fir\r\n"
</span><span class="cx"> "a=rtcp-fb:100 ccm fir\r\n"
</span><span class="cx"> "a=ssrc:3409173718 cname:/chKzCS9K6KOgL0n\r\n"
</span><del>- "a=msid:" << m_stream->label() << " " << videoTrack->id() << "\r\n"
</del><ins>+ "a=msid:" << m_streamLabel << " " << track->id() << "\r\n"
</ins><span class="cx"> "a=ice-ufrag:e/B1\r\n"
</span><span class="cx"> "a=ice-pwd:Yotk3Im3mnyi+1Q38p51MDub\r\n"
</span><span class="cx"> "a=fingerprint:sha-256 8B:87:09:8A:5D:C2:F3:33:EF:C5:B1:F6:84:3A:3D:D6:A3:E2:9C:17:4C:E7:46:3B:1B:CE:84:98:DD:8E:AF:7B\r\n"
</span><span class="lines">@@ -311,9 +327,11 @@
</span><span class="cx"> "o=- 5667094644266930846 " << m_counter++ << " IN IP4 127.0.0.1\r\n"
</span><span class="cx"> "s=-\r\n"
</span><span class="cx"> "t=0 0\r\n";
</span><del>- if (m_stream) {
- for (auto& audioTrack : m_stream->GetAudioTracks()) {
- ASSERT_UNUSED(audioTrack, !!audioTrack);
</del><ins>+ if (m_senders.size()) {
+ for (auto& sender : m_senders) {
+ auto track = sender->track();
+ if (track->kind() != "audio")
+ continue;
</ins><span class="cx"> sdp <<
</span><span class="cx"> "m=audio 9 UDP/TLS/RTP/SAVPF 111 8 0\r\n"
</span><span class="cx"> "c=IN IP4 0.0.0.0\r\n"
</span><span class="lines">@@ -329,8 +347,10 @@
</span><span class="cx"> "a=fingerprint:sha-256 8B:87:09:8A:5D:C2:F3:33:EF:C5:B1:F6:84:3A:3D:D6:A3:E2:9C:17:4C:E7:46:3B:1B:CE:84:98:DD:8E:AF:7B\r\n"
</span><span class="cx"> "a=setup:active\r\n";
</span><span class="cx"> }
</span><del>- for (auto& videoTrack : m_stream->GetVideoTracks()) {
- ASSERT_UNUSED(videoTrack, !!videoTrack);
</del><ins>+ for (auto& sender : m_senders) {
+ auto track = sender->track();
+ if (track->kind() != "video")
+ continue;
</ins><span class="cx"> sdp <<
</span><span class="cx"> "m=video 9 UDP/TLS/RTP/SAVPF 103 100 120\r\n"
</span><span class="cx"> "c=IN IP4 0.0.0.0\r\n"
</span></span></pre></div>
<a id="trunkSourceWebCoretestingMockLibWebRTCPeerConnectionh"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.h (213735 => 213736)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.h 2017-03-11 00:46:00 UTC (rev 213735)
+++ trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.h 2017-03-11 00:46:44 UTC (rev 213736)
</span><span class="lines">@@ -34,6 +34,7 @@
</span><span class="cx"> namespace WebCore {
</span><span class="cx">
</span><span class="cx"> class LibWebRTCProvider;
</span><ins>+class MockRtpSender;
</ins><span class="cx">
</span><span class="cx"> void useMockRTCPeerConnectionFactory(LibWebRTCProvider*, const String&);
</span><span class="cx">
</span><span class="lines">@@ -58,12 +59,15 @@
</span><span class="cx"> void StopRtcEventLog() override { }
</span><span class="cx"> void Close() override { }
</span><span class="cx">
</span><ins>+ bool AddStream(webrtc::MediaStreamInterface*) final { return false; }
+ void RemoveStream(webrtc::MediaStreamInterface*) final { }
+
</ins><span class="cx"> protected:
</span><span class="cx"> void SetRemoteDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) final;
</span><span class="cx"> void CreateAnswer(webrtc::CreateSessionDescriptionObserver*, const webrtc::MediaConstraintsInterface*) final;
</span><span class="cx"> rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel(const std::string&, const webrtc::DataChannelInit*) final;
</span><del>- bool AddStream(webrtc::MediaStreamInterface*) final;
- void RemoveStream(webrtc::MediaStreamInterface*) final;
</del><ins>+ rtc::scoped_refptr<webrtc::RtpSenderInterface> AddTrack(webrtc::MediaStreamTrackInterface*, std::vector<webrtc::MediaStreamInterface*> streams) final;
+ bool RemoveTrack(webrtc::RtpSenderInterface*) final;
</ins><span class="cx">
</span><span class="cx"> void SetLocalDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) override;
</span><span class="cx"> bool GetStats(webrtc::StatsObserver*, webrtc::MediaStreamTrackInterface*, StatsOutputLevel) override { return false; }
</span><span class="lines">@@ -73,10 +77,11 @@
</span><span class="cx">
</span><span class="cx"> webrtc::PeerConnectionObserver& m_observer;
</span><span class="cx"> unsigned m_counter { 0 };
</span><del>- rtc::scoped_refptr<webrtc::MediaStreamInterface> m_stream;
</del><ins>+ Vector<rtc::scoped_refptr<MockRtpSender>> m_senders;
</ins><span class="cx"> bool m_isInitiator { true };
</span><span class="cx"> bool m_isReceivingAudio { false };
</span><span class="cx"> bool m_isReceivingVideo { false };
</span><ins>+ std::string m_streamLabel;
</ins><span class="cx"> };
</span><span class="cx">
</span><span class="cx"> class MockLibWebRTCSessionDescription: public webrtc::SessionDescriptionInterface {
</span><span class="lines">@@ -199,6 +204,25 @@
</span><span class="cx"> int m_id { -1 };
</span><span class="cx"> };
</span><span class="cx">
</span><ins>+class MockRtpSender : public webrtc::RtpSenderInterface {
+public:
+ MockRtpSender(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>&& track) : m_track(WTFMove(track)) { }
+
+ bool SetTrack(webrtc::MediaStreamTrackInterface*) final { return false; }
+ rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track() const final { return m_track; }
+
+ uint32_t ssrc() const { return 0; }
+ cricket::MediaType media_type() const { return cricket::MEDIA_TYPE_VIDEO; }
+ std::string id() const { return ""; }
+ std::vector<std::string> stream_ids() const { return { }; }
+ webrtc::RtpParameters GetParameters() const { return { }; }
+ bool SetParameters(const webrtc::RtpParameters&) { return false; }
+ rtc::scoped_refptr<webrtc::DtmfSenderInterface> GetDtmfSender() const { return nullptr; }
+
+private:
+ rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> m_track;
+};
+
</ins><span class="cx"> class MockLibWebRTCPeerConnectionFactory : public webrtc::PeerConnectionFactoryInterface {
</span><span class="cx"> public:
</span><span class="cx"> static rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> create(String&& testCase) { return new rtc::RefCountedObject<MockLibWebRTCPeerConnectionFactory>(WTFMove(testCase)); }
</span></span></pre>
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