Move content\browser\renderer_host\media to content namespace.

content/test/webrtc_audio_device_test.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 "content/test/webrtc_audio_device_test.h"
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/compiler_specific.h"
 #include "base/file_util.h"
@@ skipping 116 matching lines @@
   // Main parts are inspired by the RenderViewFakeResourcesTest.
   // Note that, the IPC part is not utilized in this test.
   saved_content_renderer_.reset(
       new ReplaceContentClientRenderer(&content_renderer_client_));
   mock_process_.reset(new WebRTCMockRenderProcess());
   ui_thread_.reset(new TestBrowserThread(BrowserThread::UI,
                                          MessageLoop::current()));
 
   // Create our own AudioManager and MediaStreamManager.
   audio_manager_.reset(media::AudioManager::Create());
-  media_stream_manager_.reset(
-      new media_stream::MediaStreamManager(audio_manager_.get()));
+  media_stream_manager_.reset(new MediaStreamManager(audio_manager_.get()));
 
   // Construct the resource context on the UI thread.
   resource_context_.reset(new MockRTCResourceContext);
 
   static const char kThreadName[] = "RenderThread";
   ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE,
       base::Bind(&WebRTCAudioDeviceTest::InitializeIOThread,
                  base::Unretained(this), kThreadName));
   WaitForIOThreadCompletion();
 
@@ skipping 218 matching lines @@
 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) {
   return network_->ReceivedRTPPacket(channel, data, len);
 }
 
 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data,
                                         int len) {
   return network_->ReceivedRTCPPacket(channel, data, len);
 }
 
 }  // namespace content