Wire up FlexFEC in VideoEngine2.

webrtc/media/engine/fakewebrtccall.cc

Index: webrtc/media/engine/fakewebrtccall.cc
diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc
index c07a2ebadd1d079e240072e74bf88e3cedd3b796..347d77ccc1a7e901a7451e7dd060acec5b8908e6 100644
--- a/webrtc/media/engine/fakewebrtccall.cc
+++ b/webrtc/media/engine/fakewebrtccall.cc
@@ -296,6 +296,28 @@ void FakeVideoReceiveStream::EnableEncodedFrameRecording(rtc::PlatformFile file,
   rtc::ClosePlatformFile(file);
 }
 
+FakeFlexfecReceiveStream::FakeFlexfecReceiveStream(
+    const webrtc::FlexfecReceiveStream::Config& config)
+    : config_(config), receiving_(false) {}
+
+const webrtc::FlexfecReceiveStream::Config&
+FakeFlexfecReceiveStream::GetConfig() {
+  return config_;
+}
+
+void FakeFlexfecReceiveStream::Start() {
+  receiving_ = true;
+}
+
+void FakeFlexfecReceiveStream::Stop() {
+  receiving_ = false;
+}
+
+// TODO(brandtr): Implement when the stats have been designed.
+webrtc::FlexfecReceiveStream::Stats FakeFlexfecReceiveStream::GetStats() const {
+  return webrtc::FlexfecReceiveStream::Stats();
+}
+
 FakeCall::FakeCall(const webrtc::Call::Config& config)
     : config_(config),
       audio_network_state_(webrtc::kNetworkUp),
@@ -348,6 +370,11 @@ const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) {
   return nullptr;
 }
 
+const std::vector<FakeFlexfecReceiveStream*>&
+FakeCall::GetFlexfecReceiveStreams() {
+  return flexfec_receive_streams_;
+}
+
 webrtc::NetworkState FakeCall::GetNetworkState(webrtc::MediaType media) const {
   switch (media) {
     case webrtc::MediaType::AUDIO:
@@ -451,13 +478,24 @@ void FakeCall::DestroyVideoReceiveStream(
 
 webrtc::FlexfecReceiveStream* FakeCall::CreateFlexfecReceiveStream(
     webrtc::FlexfecReceiveStream::Config config) {

[magjed_webrtc, 2016/11/18 14:28:00]

I would like this to be a const-ref as well (and/or
webrtc::FlexfecReceiveStream::Config&& if you want to have move functionality).

[brandtr, 2016/11/21 08:52:03]

Can't do it here alone, because FakeCall is a webrtc::Call, which has this
signature:
https://cs.chromium.org/chromium/src/third_party/webrtc/call.h?l=122. (Note that
the signature of CreateFlexfecReceiveStream is inspired by
CreateVideoReceiveStream. Now I noticed that CreateAudioReceiveStream takes a
const ref parameter however...)

I will make an AI to update the signature of CreateFlexfecReceiveStream (both in
call.h and here) to take a const ref instead. I don't think move functionality
is needed.

-  // TODO(brandtr): Implement when adding integration with WebRtcVideoEngine2.
-  return nullptr;
+  flexfec_receive_streams_.push_back(
+      new FakeFlexfecReceiveStream(std::move(config)));
+  ++num_created_receive_streams_;
+  return flexfec_receive_streams_.back();
 }
 
 void FakeCall::DestroyFlexfecReceiveStream(
     webrtc::FlexfecReceiveStream* receive_stream) {
-  // TODO(brandtr): Implement when adding integration with WebRtcVideoEngine2.
+  auto it = std::find(flexfec_receive_streams_.begin(),
+                      flexfec_receive_streams_.end(),
+                      static_cast<FakeFlexfecReceiveStream*>(receive_stream));
+  if (it == flexfec_receive_streams_.end()) {
+    ADD_FAILURE()
+        << "DestroyFlexfecReceiveStream called with unknown parameter.";
+  } else {
+    delete *it;
+    flexfec_receive_streams_.erase(it);
+  }
 }
 
 webrtc::PacketReceiver* FakeCall::Receiver() {