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..eed4c0d974bf03ed9c3a7af249b6b36502cd099f 100644
--- a/webrtc/pc/peerconnection.cc
+++ b/webrtc/pc/peerconnection.cc
@@ -220,6 +220,10 @@ bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
 
 namespace webrtc {
 
+std::unique_ptr<RtcEventLog> CreateRtcEventLog();
+
+Call* CreateCall(const Call::Config& config);
+
 bool PeerConnectionInterface::RTCConfiguration::operator==(
     const PeerConnectionInterface::RTCConfiguration& o) const {
   // This static_assert prevents us from accidentally breaking operator==.
@@ -395,7 +399,7 @@ 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 +2339,17 @@ void PeerConnection::CreateCall_w() {
   const int kMaxBandwidthBps = 2000000;
 
   webrtc::Call::Config call_config(event_log_.get());
+  if (!factory_->channel_manager()->media_engine()) {
+    LOG(LS_WARNING)
+        << "Failed to create the Call because the media engine is unset.";

[Taylor Brandstetter, 2017/05/30 17:30:53]

Other comment about log levels applies here too.

[Zhi Huang, 2017/05/31 00:03:29]

Done.

+    return;
+  }
   call_config.audio_state =
-      factory_->channel_manager() ->media_engine()->GetAudioState();
+      factory_->channel_manager()->media_engine()->GetAudioState();
   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(CreateCall(call_config));
 }
 
 }  // namespace webrtc