Merge AssociateStreamWithProducer message into CreateStream message for both audio output and input.

content/renderer/media/webrtc_audio_device_unittest.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 "base/environment.h"
 #include "base/test/test_timeouts.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_audio_renderer.h"
 #include "content/renderer/render_thread_impl.h"
@@ skipping 86 matching lines @@
     return false;
 
   media::AudioHardwareConfig* hardware_config =
       RenderThreadImpl::current()->GetAudioHardwareConfig();
 
   // Use native capture sample rate and channel configuration to get some
   // action in this test.
   int sample_rate = hardware_config->GetInputSampleRate();
   media::ChannelLayout channel_layout =
       hardware_config->GetInputChannelLayout();
-  if (!capturer->Initialize(channel_layout, sample_rate, 1))
+  if (!capturer->Initialize(kRenderViewId, channel_layout, sample_rate, 1))
     return false;
 
   return true;
 }
 
 
 class WebRTCMediaProcessImpl : public webrtc::VoEMediaProcess {
  public:
   explicit WebRTCMediaProcessImpl(base::WaitableEvent* event)
       : event_(event),
@@ skipping 466 matching lines @@
   webrtc_audio_device->capturer()->Stop();
   renderer->Stop();
   EXPECT_EQ(0, base->StopSend(ch));
   EXPECT_EQ(0, base->StopPlayout(ch));
 
   EXPECT_EQ(0, base->DeleteChannel(ch));
   EXPECT_EQ(0, base->Terminate());
 }
 
 }  // namespace content