Updated the MockWebRtcPeerConnectionHandler used in blink Layouttests.

content/shell/renderer/test_runner/mock_webrtc_peer_connection_handler.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/shell/renderer/test_runner/mock_webrtc_peer_connection_handler.h"
 
 #include "content/shell/renderer/test_runner/WebTestDelegate.h"
 #include "content/shell/renderer/test_runner/mock_constraints.h"
 #include "content/shell/renderer/test_runner/mock_webrtc_data_channel_handler.h"
 #include "content/shell/renderer/test_runner/mock_webrtc_dtmf_sender_handler.h"
 #include "content/shell/renderer/test_runner/test_interfaces.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/WebKit/public/platform/WebMediaStream.h"
+#include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h"
 #include "third_party/WebKit/public/platform/WebRTCPeerConnectionHandlerClient.h"
 #include "third_party/WebKit/public/platform/WebRTCStatsResponse.h"
 #include "third_party/WebKit/public/platform/WebRTCVoidRequest.h"
 #include "third_party/WebKit/public/platform/WebString.h"
 #include "third_party/WebKit/public/platform/WebVector.h"
 
 using namespace blink;
 
@@ skipping 113 matching lines @@
     client_->didAddRemoteDataChannel(remote_data_channel);
   }
 
  private:
   WebRTCPeerConnectionHandlerClient* client_;
   WebTestDelegate* delegate_;
 };
 
 /////////////////////
 
-MockWebRTCPeerConnectionHandler::MockWebRTCPeerConnectionHandler() {
+MockWebRTCPeerConnectionHandler::~MockWebRTCPeerConnectionHandler() {
 }
 
 MockWebRTCPeerConnectionHandler::MockWebRTCPeerConnectionHandler(
     WebRTCPeerConnectionHandlerClient* client,
     TestInterfaces* interfaces)
     : client_(client),
       stopped_(false),
       stream_count_(0),
       interfaces_(interfaces) {
 }
@@ skipping 58 matching lines @@
     interfaces_->GetDelegate()->postTask(
         new RTCVoidRequestTask(this, request, true));
   } else
     interfaces_->GetDelegate()->postTask(
         new RTCVoidRequestTask(this, request, false));
 }
 
 void MockWebRTCPeerConnectionHandler::setRemoteDescription(
     const WebRTCVoidRequest& request,
     const WebRTCSessionDescription& remote_description) {
+
   if (!remote_description.isNull() && remote_description.sdp() == "remote") {
+    UpdateRemoteStreams();
     remote_description_ = remote_description;
     interfaces_->GetDelegate()->postTask(
         new RTCVoidRequestTask(this, request, true));
   } else
     interfaces_->GetDelegate()->postTask(
         new RTCVoidRequestTask(this, request, false));
 }
 
+void MockWebRTCPeerConnectionHandler::UpdateRemoteStreams() {
+  // Find all removed streams.
+  // Set the readyState of the remote tracks to ended, remove them from the
+  // stream and notify the client.
+  StreamMap::iterator removed_it = remote_streams_.begin();
+  while(removed_it != remote_streams_.end()) {
+    if (local_streams_.find(removed_it->first) != local_streams_.end()) {
+      removed_it++;
+      continue;
+    }
+
+    // The stream have been removed. Loop through all tracks and set the
+    // source as ended and remove them from the stream.
+    blink::WebMediaStream stream = removed_it->second;
+    blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
+    stream.audioTracks(audio_tracks);
+    for (size_t i = 0; i < audio_tracks.size(); ++i) {
+      audio_tracks[i].source().setReadyState(
+          blink::WebMediaStreamSource::ReadyStateEnded);
+      stream.removeTrack(audio_tracks[i]);
+    }
+
+    blink::WebVector<blink::WebMediaStreamTrack> video_tracks;
+    stream.videoTracks(video_tracks);
+    for (size_t i = 0; i < video_tracks.size(); ++i) {
+      video_tracks[i].source().setReadyState(
+          blink::WebMediaStreamSource::ReadyStateEnded);
+      stream.removeTrack(video_tracks[i]);
+    }
+    client_->didRemoveRemoteStream(stream);
+    remote_streams_.erase(removed_it++);
+  }
+
+  // Find all new streams;
+  // Create new sources and tracks and notify the client about the new stream.
+  StreamMap::iterator added_it = local_streams_.begin();
+  while(added_it != local_streams_.end()) {
+    if (remote_streams_.find(added_it->first) != remote_streams_.end()) {
+      added_it++;
+      continue;
+    }
+
+    const blink::WebMediaStream& stream = added_it->second;
+
+    blink::WebVector<blink::WebMediaStreamTrack> local_audio_tracks;
+    stream.audioTracks(local_audio_tracks);
+    blink::WebVector<blink::WebMediaStreamTrack>
+        remote_audio_tracks(local_audio_tracks.size());
+
+    for (size_t i = 0; i < local_audio_tracks.size(); ++i) {
+      blink::WebMediaStreamSource webkit_source;
+      webkit_source.initialize(local_audio_tracks[i].id(),
+                               blink::WebMediaStreamSource::TypeAudio,
+                               local_audio_tracks[i].id());

[hta - Chromium, 2014/09/30 09:43:49]

Shouldn't you be using the new flags here? this seems to be calling the
deprecated initialize function (but does permit the CL to be landed
independently of the other one).

[perkj_chrome, 2014/09/30 18:38:30]

Yes, but I would prefer to do that when I change all other calls in Chromium. I
would like to land this first to add the test for removing remote remote tracks.

+      remote_audio_tracks[i].initialize(webkit_source);
+    }
+
+    blink::WebVector<blink::WebMediaStreamTrack> local_video_tracks;
+    stream.videoTracks(local_video_tracks);
+    blink::WebVector<blink::WebMediaStreamTrack>
+        remote_video_tracks(local_video_tracks.size());
+    for (size_t i = 0; i < local_video_tracks.size(); ++i) {
+      blink::WebMediaStreamSource webkit_source;
+      webkit_source.initialize(local_video_tracks[i].id(),
+                               blink::WebMediaStreamSource::TypeVideo,
+                               local_audio_tracks[i].id());

[hta - Chromium, 2014/09/30 09:43:49]

The occurence of "audio" in this line looks strange...

[perkj_chrome, 2014/09/30 18:38:30]

Yes, wrong. 

+      remote_video_tracks[i].initialize(webkit_source);
+    }
+
+    blink::WebMediaStream new_remote_stream;
+    new_remote_stream.initialize(remote_audio_tracks,
+                                 remote_video_tracks);
+    remote_streams_[added_it->first] = new_remote_stream;
+    client_->didAddRemoteStream(new_remote_stream);
+    ++added_it;
+  }
+}
+
 WebRTCSessionDescription MockWebRTCPeerConnectionHandler::localDescription() {
   return local_description_;
 }
 
 WebRTCSessionDescription MockWebRTCPeerConnectionHandler::remoteDescription() {
   return remote_description_;
 }
 
 bool MockWebRTCPeerConnectionHandler::updateICE(
     const WebRTCConfiguration& configuration,
@@ skipping 13 matching lines @@
   interfaces_->GetDelegate()->postTask(
       new RTCVoidRequestTask(this, request, true));
   return true;
 }
 
 bool MockWebRTCPeerConnectionHandler::addStream(
     const WebMediaStream& stream,
     const WebMediaConstraints& constraints) {
   ++stream_count_;
   client_->negotiationNeeded();
+  DCHECK(local_streams_.find(stream.id().utf8()) == local_streams_.end());

[hta - Chromium, 2014/09/30 09:43:49]

This is actually not according to spec. Spec says "If stream is already in
connection's local streams set, then abort these steps". (It doesn't say that an
error is returned, so it seems to be ignorable.)

[perkj_chrome, 2014/09/30 18:38:30]

ok- this was to help debugging tests. It looks like if I return false here, a
syntax error exception will be thrown. RTCPeerConnection.cpp.

+  local_streams_[stream.id().utf8()] = stream;
   return true;
 }
 
 void MockWebRTCPeerConnectionHandler::removeStream(
     const WebMediaStream& stream) {
   --stream_count_;
+  local_streams_.erase(stream.id().utf8());
   client_->negotiationNeeded();
 }
 
 void MockWebRTCPeerConnectionHandler::getStats(
     const WebRTCStatsRequest& request) {
   WebRTCStatsResponse response = request.createResponse();
   double current_date =
       interfaces_->GetDelegate()->getCurrentTimeInMillisecond();
   if (request.hasSelector()) {
     // FIXME: There is no check that the fetched values are valid.
@@ skipping 27 matching lines @@
     const WebMediaStreamTrack& track) {
   return new MockWebRTCDTMFSenderHandler(track, interfaces_->GetDelegate());
 }
 
 void MockWebRTCPeerConnectionHandler::stop() {
   stopped_ = true;
   task_list_.RevokeAll();
 }
 
 }  // namespace content