Reduce APK size by disabling WebRTC.

remoting/client/chromoting_client.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "remoting/client/chromoting_client.h"
 
 #include <utility>
 
 #include "remoting/base/capabilities.h"
 #include "remoting/client/audio_decode_scheduler.h"
@@ skipping 55 matching lines @@
 
   if (!protocol_config_)
     protocol_config_ = protocol::CandidateSessionConfig::CreateDefault();
   if (!audio_decode_scheduler_)
     protocol_config_->DisableAudioChannel();
 
   if (!connection_) {
     if (protocol_config_->webrtc_supported()) {
       DCHECK(!protocol_config_->ice_supported());
 #if defined(OS_NACL)
       LOG(FATAL) << "WebRTC is not supported in webapp.";

[Sergey Ulanov, 2016/03/16 19:12:36]

We want this LOG(FATAL) on all platforms:
#if !defined(ENABLED_WEBRTC_CLIENT) 
  LOG(FATAL) << "WebRTC is not supported";
#elif
   ....
#endif

[Yuwei, 2016/03/16 21:21:00]

Done.

-#else   // defined(OS_NACL)
+#elif !defined(DISABLE_WEBRTC)   // defined(OS_NACL)
       connection_.reset(new protocol::WebrtcConnectionToHost());
-#endif  // !defined(OS_NACL)
+#endif  // !defined(OS_NACL) && !defined(DISABLE_WEBRTC)
     } else {
       DCHECK(protocol_config_->ice_supported());
       connection_.reset(new protocol::IceConnectionToHost());
     }
   }
   connection_->set_client_stub(this);
   connection_->set_clipboard_stub(this);
   connection_->set_video_renderer(video_renderer_);
   connection_->set_audio_stub(audio_decode_scheduler_.get());
 
@@ skipping 144 matching lines @@
 
   // Negotiate capabilities with the host.
   VLOG(1) << "Client capabilities: " << local_capabilities_;
 
   protocol::Capabilities capabilities;
   capabilities.set_capabilities(local_capabilities_);
   connection_->host_stub()->SetCapabilities(capabilities);
 }
 
 }  // namespace remoting