Build WebRTC with data channel only.

webrtc/pc/peerconnection.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 380 matching lines @@
   return mandatory_constraints_satisfied == constraints->GetMandatory().size();
 }
 
 PeerConnection::PeerConnection(PeerConnectionFactory* factory)
     : factory_(factory),
       observer_(NULL),
       uma_observer_(NULL),
       signaling_state_(kStable),
       ice_connection_state_(kIceConnectionNew),
       ice_gathering_state_(kIceGatheringNew),
+#if defined(HAVE_WEBRTC_VOICE) && defined(HAVE_WEBRTC_VIDEO)

[Taylor Brandstetter, 2017/05/11 04:43:11]

||?

[Zhi Huang, 2017/05/12 20:05:33]

Done.

       event_log_(RtcEventLog::Create()),
+#else
+      event_log_(std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl())),
+#endif
       rtcp_cname_(GenerateRtcpCname()),
       local_streams_(StreamCollection::Create()),
-      remote_streams_(StreamCollection::Create()) {}
+      remote_streams_(StreamCollection::Create()) {
+}
 
 PeerConnection::~PeerConnection() {
   TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
   RTC_DCHECK(signaling_thread()->IsCurrent());
   // Need to detach RTP senders/receivers from WebRtcSession,
   // since it's about to be destroyed.
   for (const auto& sender : senders_) {
     sender->internal()->Stop();
   }
   for (const auto& receiver : receivers_) {
@@ skipping 1904 matching lines @@
   return event_log_->StartLogging(file, max_size_bytes);
 }
 
 void PeerConnection::StopRtcEventLog_w() {
   if (event_log_) {
     event_log_->StopLogging();
   }
 }
 
 void PeerConnection::CreateCall_w() {
+#if defined(HAVE_WEBRTC_VOICE) && defined(HAVE_WEBRTC_VIDEO)

[Taylor Brandstetter, 2017/05/11 04:43:11]

||?

[Zhi Huang, 2017/05/12 20:05:33]

Done.

   RTC_DCHECK(!call_);
 
   const int kMinBandwidthBps = 30000;
   const int kStartBandwidthBps = 300000;
   const int kMaxBandwidthBps = 2000000;
 
   webrtc::Call::Config call_config(event_log_.get());
   call_config.audio_state =
       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));
+#endif
 }
 
 }  // namespace webrtc