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

remoting/protocol/webrtc_audio_module.cc

 // Copyright 2016 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/webrtc_audio_module.h"
 
 #include "base/bind.h"
 #include "base/stl_util.h"
 #include "base/threading/thread_task_runner_handle.h"
 
@@ skipping 470 matching lines @@
   return -1;
 }
 
 int32_t WebrtcAudioModule::EnableBuiltInNS(bool enable) {
   NOTREACHED();
   return -1;
 }
 
 #if defined(WEBRTC_IOS)
 int WebrtcAudioModule::GetPlayoutAudioParameters(
-    AudioParameters* params) const {
+    webrtc::AudioParameters* params) const {
   NOTREACHED();
   return -1;
 }
 
-int WebrtcAudioModule::GetRecordAudioParameters(AudioParameters* params) const {
+int WebrtcAudioModule::GetRecordAudioParameters(
+    webrtc::AudioParameters* params) const {
   NOTREACHED();
   return -1;
 }
 #endif  // WEBRTC_IOS
 
 void WebrtcAudioModule::StartPlayoutOnAudioThread() {
   DCHECK(audio_task_runner_->BelongsToCurrentThread());
   poll_timer_.Start(
       FROM_HERE, kPollInterval,
       base::Bind(&WebrtcAudioModule::PollFromSource, base::Unretained(this)));
@@ skipping 16 matching lines @@
     int64_t ntp_time_ms = -1;
     char data[kBytesPerSample * kChannels * kSamplesPerFrame];
     audio_transport_->PullRenderData(kBytesPerSample * 8, kSamplingRate,
                                      kChannels, kSamplesPerFrame, data,
                                      &elapsed_time_ms, &ntp_time_ms);
   }
 }
 
 }  // namespace protocol
 }  // namespace remoting