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

remoting/protocol/ice_connection_to_client.cc

Index: remoting/protocol/ice_connection_to_client.cc
diff --git a/remoting/protocol/ice_connection_to_client.cc b/remoting/protocol/ice_connection_to_client.cc
index c4a42f16cd9cd2b9fbdd789eba31c38af444b4ff..037e80550fa138af9eab0b70c376a9d3388c141d 100644
--- a/remoting/protocol/ice_connection_to_client.cc
+++ b/remoting/protocol/ice_connection_to_client.cc
@@ -32,11 +32,17 @@ 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;