Link things together and enable the device selection for linux and mac.

content/renderer/media/webrtc_audio_device_impl.cc

 // Copyright (c) 2011 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_device_impl.h"
 
 #include "base/bind.h"
 #include "base/string_util.h"
+#include "content/browser/renderer_host/media/audio_input_device_manager.h"
 #include "content/common/view_messages.h"
 #include "content/renderer/render_thread_impl.h"
 #include "media/audio/audio_util.h"
 
 static const int64 kMillisecondsBetweenProcessCalls = 5000;
 static const char kVersion[] = "WebRTC AudioDevice 1.0.0.Chrome";
 
 static int GetAudioInputHardwareSampleRate() {
   static double input_sample_rate = 0;
   if (!input_sample_rate) {
@@ skipping 147 matching lines @@
         samples_per_sec,
         input_delay_ms_ + output_delay_ms_,
         0,  // clock_drift
         0,  // current_mic_level
         new_mic_level);  // not used
     accumulated_audio_samples += samples_per_10_msec;
     audio_byte_buffer += bytes_per_10_msec;
   }
 }
 
-void WebRtcAudioDeviceImpl::OnDeviceStarted(int device_index) {
-  VLOG(1) << "OnDeviceStarted (device_index=" << device_index << ")";
+void WebRtcAudioDeviceImpl::OnDeviceStarted(const std::string& device_uid) {
+  VLOG(1) << "OnDeviceStarted (device_uid=" << device_uid << ")";
   // -1 is an invalid device index. Do nothing if a valid device has
   // been started. Otherwise update the |recording_| state to false.
-  if (device_index != -1)
+  if (device_uid != media_stream::AudioInputDeviceManager::kInvalidDeviceUId)
     return;
 
   base::AutoLock auto_lock(lock_);
   if (recording_)
     recording_ = false;
 }
 
 void WebRtcAudioDeviceImpl::OnDeviceStopped() {
   VLOG(1) << "OnDeviceStopped";
   base::AutoLock auto_lock(lock_);
@@ skipping 173 matching lines @@
     // We do run at 44.1kHz at the actual audio layer, but ask for frames
     // at 44.0kHz to ensure that we can feed them to the webrtc::VoiceEngine.
     input_buffer_size = 440;
     output_buffer_size = 440;
   }
 #elif defined(OS_LINUX) || defined(OS_OPENBSD)
   if (output_sample_rate != 48000) {
     DLOG(ERROR) << "Only 48kHz sample rate is supported on Linux.";
     return -1;
   }
-  input_channels = 1;
+  input_channels = 2;
   output_channels = 1;
 
   // Based on tests using the current ALSA implementation in Chrome, we have
   // found that the best combination is 20ms on the input side and 10ms on the
   // output side.
   // TODO(henrika): It might be possible to reduce the input buffer
   // size and reduce the delay even more.
   input_buffer_size = 2 * 480;
   output_buffer_size = 480;
 #else
@@ skipping 559 matching lines @@
 }
 
 int32_t WebRtcAudioDeviceImpl::GetLoudspeakerStatus(bool* enabled) const {
   NOTIMPLEMENTED();
   return -1;
 }
 
 void WebRtcAudioDeviceImpl::SetSessionId(int session_id) {
   session_id_ = session_id;
 }