Build WebRTC with data channel only.

webrtc/call/call.h

 /*
  *  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.
  */
 #ifndef WEBRTC_CALL_CALL_H_
@@ skipping 42 matching lines @@
   virtual ~PacketReceiver() {}
 };
 
 // A Call instance can contain several send and/or receive streams. All streams
 // are assumed to have the same remote endpoint and will share bitrate estimates
 // etc.
 class Call {
  public:
   struct Config {
     explicit Config(RtcEventLog* event_log) : event_log(event_log) {
+#ifdef HAVE_MEDIA
       RTC_DCHECK(event_log);
+#endif
     }
 
-    static const int kDefaultStartBitrateBps;
+    static const int kDefaultStartBitrateBps = 300000;
 
     // Bitrate config used until valid bitrate estimates are calculated. Also
     // used to cap total bitrate used.
     struct BitrateConfig {
       int min_bitrate_bps = 0;
       int start_bitrate_bps = kDefaultStartBitrateBps;
       int max_bitrate_bps = -1;
     } bitrate_config;
 
     // AudioState which is possibly shared between multiple calls.
@@ skipping 80 matching lines @@
       const rtc::NetworkRoute& network_route) = 0;
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_CALL_H_