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

content/renderer/media/webrtc_audio_capturer.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/renderer/media/webrtc_audio_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
@@ skipping 91 matching lines @@
   friend class base::RefCounted<WebRtcAudioCapturer::ConfiguredBuffer>;
 
   scoped_ptr<int16[]> buffer_;
 
   // Cached values of utilized audio parameters.
   media::AudioParameters params_;
 };
 
 // static
 scoped_refptr<WebRtcAudioCapturer> WebRtcAudioCapturer::CreateCapturer() {
-  scoped_refptr<WebRtcAudioCapturer> capturer = new  WebRtcAudioCapturer();
+  scoped_refptr<WebRtcAudioCapturer> capturer = new WebRtcAudioCapturer();
   return capturer;
 }
 
 bool WebRtcAudioCapturer::Reconfigure(int sample_rate,
                                       media::ChannelLayout channel_layout) {
   scoped_refptr<ConfiguredBuffer> new_buffer(new ConfiguredBuffer());
   if (!new_buffer->Initialize(sample_rate, channel_layout))
     return false;
 
   SinkList sinks;
   {
     base::AutoLock auto_lock(lock_);
 
     buffer_ = new_buffer;
     sinks = sinks_;
   }
 
   // Tell all sinks which format we use.
   for (SinkList::const_iterator it = sinks.begin(); it != sinks.end(); ++it)
     (*it)->SetCaptureFormat(new_buffer->params());
 
   return true;
 }
 
-bool WebRtcAudioCapturer::Initialize(media::ChannelLayout channel_layout,
+bool WebRtcAudioCapturer::Initialize(int render_view_id,
+                                     media::ChannelLayout channel_layout,
                                      int sample_rate) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(!sinks_.empty());
   DVLOG(1) << "WebRtcAudioCapturer::Initialize()";
 
   DVLOG(1) << "Audio input hardware channel layout: " << channel_layout;
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputChannelLayout",
                             channel_layout, media::CHANNEL_LAYOUT_MAX);
 
   // Verify that the reported input channel configuration is supported.
@@ skipping 16 matching lines @@
           &kValidInputRates[arraysize(kValidInputRates)]) {
     DLOG(ERROR) << sample_rate << " is not a supported input rate.";
     return false;
   }
 
   if (!Reconfigure(sample_rate, channel_layout))
     return false;
 
   // Create and configure the default audio capturing source. The |source_|
   // will be overwritten if an external client later calls SetCapturerSource()
-  // providing an alternaive media::AudioCapturerSource.
-  SetCapturerSource(AudioDeviceFactory::NewInputDevice(),
+  // providing an alternative media::AudioCapturerSource.
+  SetCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id),
                     channel_layout,
                     static_cast<float>(sample_rate));
 
   return true;
 }
 
 WebRtcAudioCapturer::WebRtcAudioCapturer()
     : source_(NULL),
       running_(false),
       buffering_(false),
@@ skipping 302 matching lines @@
   if (!on_device_stopped_cb_.is_null())
     on_device_stopped_cb_.Run();
 }
 
 media::AudioParameters WebRtcAudioCapturer::audio_parameters() const {
   base::AutoLock auto_lock(lock_);
   return buffer_->params();
 }
 
 }  // namespace content