Build WebRTC with data channel only.

webrtc/pc/peerconnection.cc

Index: webrtc/pc/peerconnection.cc
diff --git a/webrtc/pc/peerconnection.cc b/webrtc/pc/peerconnection.cc
index 852dd39de5bc21eed1461340f796d4ef0132e55f..f9cff26fd2de5984d7e31020d14d3e190d91e5b9 100644
--- a/webrtc/pc/peerconnection.cc
+++ b/webrtc/pc/peerconnection.cc
@@ -220,6 +220,9 @@ bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
 
 namespace webrtc {
 
+std::unique_ptr<RtcEventLog> CreateRtcEventLog();
+Call* CreateCallImpl(const Call::Config& config);
+
 bool PeerConnectionInterface::RTCConfiguration::operator==(
     const PeerConnectionInterface::RTCConfiguration& o) const {
   // This static_assert prevents us from accidentally breaking operator==.
@@ -395,7 +398,7 @@ PeerConnection::PeerConnection(PeerConnectionFactory* factory)
     : factory_(factory),
       observer_(NULL),
       uma_observer_(NULL),
-      event_log_(RtcEventLog::Create()),
+      event_log_(CreateRtcEventLog()),
       signaling_state_(kStable),
       ice_connection_state_(kIceConnectionNew),
       ice_gathering_state_(kIceGatheringNew),
@@ -2335,13 +2338,18 @@ void PeerConnection::CreateCall_w() {
   const int kMaxBandwidthBps = 2000000;
 
   webrtc::Call::Config call_config(event_log_.get());
-  call_config.audio_state =
-      factory_->channel_manager() ->media_engine()->GetAudioState();
+  if (factory_->channel_manager()->media_engine()) {
+    call_config.audio_state =
+        factory_->channel_manager()->media_engine()->GetAudioState();
+  } else {
+    LOG(LS_WARNING) << "The Call::Config.audio_state is unset because the "
+                       "media engine is unset.";

[Taylor Brandstetter, 2017/05/18 17:57:05]

When will this warning be hit? If it's only hit when CreateCallImpl returns
nullptr, then it would be slightly simpler to move the body of this function
into CreateCallImpl, or just return nullptr early here.

[Zhi Huang, 2017/05/23 03:40:35]

Done.

+  }
+
   call_config.bitrate_config.min_bitrate_bps = kMinBandwidthBps;
   call_config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
   call_config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
-
-  call_.reset(webrtc::Call::Create(call_config));
+  call_.reset(CreateCallImpl(call_config));
 }
 
 }  // namespace webrtc