Remove unused fields from //content.

content/renderer/media/webrtc_audio_device_impl.h

 // Copyright 2013 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 <stdint.h>
 
 #include <list>
@@ skipping 362 matching lines @@
   // A list of raw pointer of WebRtcPlayoutDataSource::Sink objects which want
   // to get the playout data, the sink need to call RemovePlayoutSink()
   // before it goes away.
   PlayoutDataSinkList playout_sinks_;
 
   // Weak reference to the audio callback.
   // The webrtc client defines |audio_transport_callback_| by calling
   // RegisterAudioCallback().
   webrtc::AudioTransport* audio_transport_callback_;
 
-  // 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_;
 
   // Protects |recording_|, |output_delay_ms_|, |input_delay_ms_|, |renderer_|
   // |recording_|, |microphone_volume_| and |playout_sinks_|.
   mutable base::Lock lock_;
 
   // Used to protect the racing of calling OnData() since there can be more
   // than one input stream calling OnData().
   mutable base::Lock capture_callback_lock_;
 
   bool initialized_;
   bool playing_;
   bool recording_;
 
-  // Stores latest microphone volume received in a CaptureData() callback.
-  // Range is [0, 255].
-  uint32_t microphone_volume_;
-
   // Buffer used for temporary storage during render callback.
   // It is only accessed by the audio render thread.
   std::vector<int16_t> render_buffer_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_