Tab Audio Capture: Browser-side connect/disconnect functionality.

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 211 matching lines @@
 
   test_request_context_.reset();
 
 #if defined(OS_WIN)
   initialize_com_.reset();
 #endif
 }
 
 void WebRTCAudioDeviceTest::CreateChannel(const char* name) {
   DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
+
+  static const int kRenderProcessId = 1;
   audio_render_host_ = new AudioRendererHost(
-      audio_manager_.get(), media_observer_.get());
+      kRenderProcessId, audio_manager_.get(), media_observer_.get());
   audio_render_host_->OnChannelConnected(base::GetCurrentProcId());
 
   audio_input_renderer_host_ = new AudioInputRendererHost(
       audio_manager_.get(), media_stream_manager_.get());
   audio_input_renderer_host_->OnChannelConnected(base::GetCurrentProcId());
 
   channel_.reset(new IPC::Channel(name, IPC::Channel::MODE_SERVER, this));
   ASSERT_TRUE(channel_->Connect());
 
   audio_render_host_->OnFilterAdded(channel_.get());
@@ skipping 121 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