Build WebRTC with data channel only.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 40 matching lines @@
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/send_delay_stats.h"
 #include "webrtc/video/stats_counter.h"
 #include "webrtc/video/video_receive_stream.h"
 #include "webrtc/video/video_send_stream.h"
 
 namespace webrtc {
 
+#ifdef HAVE_MEDIA
 const int Call::Config::kDefaultStartBitrateBps = 300000;
+#endif
 
 namespace {
 
 // TODO(nisse): This really begs for a shared context struct.
 bool UseSendSideBwe(const std::vector<RtpExtension>& extensions,
                     bool transport_cc) {
   if (!transport_cc)
     return false;
   for (const auto& extension : extensions) {
     if (extension.uri == RtpExtension::kTransportSequenceNumberUri)
@@ skipping 1210 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc