Ensures that WebRTC works for device selection using a different sample rate than default

content/renderer/media/webrtc_audio_device_impl.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 
 #include <string>
 
 #include "base/basictypes.h"
@@ skipping 351 matching lines @@
   virtual int32_t SetRecordingSampleRate(
       const uint32_t samples_per_sec) OVERRIDE;
   virtual int32_t RecordingSampleRate(uint32_t* samples_per_sec) const OVERRIDE;
   virtual int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) OVERRIDE;
   virtual int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const OVERRIDE;
 
   virtual int32_t ResetAudioDevice() OVERRIDE;
   virtual int32_t SetLoudspeakerStatus(bool enable) OVERRIDE;
   virtual int32_t GetLoudspeakerStatus(bool* enabled) const OVERRIDE;
 
-  // Sets the session id.
-  void SetSessionId(int session_id);
-
   // Sets the |renderer_|, returns false if |renderer_| has already existed.
   bool SetRenderer(WebRtcAudioRenderer* renderer);
 
   const scoped_refptr<WebRtcAudioCapturer>& capturer() const {
     return capturer_;
   }
 
   // Accessors.
   int input_buffer_size() const {
     return input_audio_parameters_.frames_per_buffer();
@@ skipping 47 matching lines @@
   // Cached value of the current audio delay on the input/capture side.
   int input_delay_ms_;
 
   // Cached value of the current audio delay on the output/renderer side.
   int output_delay_ms_;
 
   webrtc::AudioDeviceModule::ErrorCode last_error_;
 
   base::TimeTicks last_process_time_;
 
-  // Id of the media session to be started, it tells which device to be used
-  // on the input/capture side.
-  int session_id_;
-
   // Protects |recording_|, |output_delay_ms_|, |input_delay_ms_|, |renderer_|.
   mutable base::Lock lock_;
 
   bool initialized_;
   bool playing_;
 
   // Local copy of the current Automatic Gain Control state.
   bool agc_is_enabled_;
 
   // Used for histograms of total recording and playout times.
   base::Time start_capture_time_;
   base::Time start_render_time_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_