Removing the dependency on //third_party/webrtc:webrtc and replacing with rtc_base.

remoting/protocol/ice_connection_to_client.cc

 // Copyright 2015 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/protocol/ice_connection_to_client.h"
 
 #include <utility>
 
 #include "base/bind.h"
 #include "base/location.h"
@@ skipping 14 matching lines @@
 #include "remoting/protocol/transport_context.h"
 #include "remoting/protocol/video_frame_pump.h"
 
 namespace remoting {
 namespace protocol {
 
 namespace {
 
 std::unique_ptr<AudioEncoder> CreateAudioEncoder(
     const protocol::SessionConfig& config) {
+#if defined(OS_IOS)
+  // TODO(nicholss): iOS should not use Opus. This is to prevent us from
+  // depending on //media. In the future we will use webrtc for conneciton
+  // and this will be a non-issue.
+  return nullptr;
+#else
   const protocol::ChannelConfig& audio_config = config.audio_config();
-
   if (audio_config.codec == protocol::ChannelConfig::CODEC_OPUS) {
     return base::WrapUnique(new AudioEncoderOpus());
   }
+#endif
 
   NOTREACHED();
   return nullptr;
 }
 
 }  // namespace
 
 IceConnectionToClient::IceConnectionToClient(
     std::unique_ptr<protocol::Session> session,
     scoped_refptr<TransportContext> transport_context,
@@ skipping 169 matching lines @@
 
 void IceConnectionToClient::CloseChannels() {
   control_dispatcher_.reset();
   event_dispatcher_.reset();
   video_dispatcher_.reset();
   audio_writer_.reset();
 }
 
 }  // namespace protocol
 }  // namespace remoting