Adds Analog Gain Control (AGC) to the WebRTC client.

media/audio/audio_low_latency_input_output_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/basictypes.h"
 #include "base/environment.h"
 #include "base/file_util.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/message_loop.h"
 #include "base/path_service.h"
@@ skipping 169 matching lines @@
               state.output_delay_ms);
       ++elements_written;
     }
 
     file_util::CloseFile(text_file);
   }
 
   // AudioInputStream::AudioInputCallback.
   virtual void OnData(AudioInputStream* stream,
                       const uint8* src, uint32 size,
-                      uint32 hardware_delay_bytes) OVERRIDE {
+                      uint32 hardware_delay_bytes,
+                      double volume) OVERRIDE {
     base::AutoLock lock(lock_);
 
     // Update three components in the AudioDelayState for this recorded
     // audio packet.
     base::Time now_time = base::Time::Now();
     int diff = (now_time - previous_write_time_).InMilliseconds();
     previous_write_time_ = now_time;
     if (input_elements_to_write_ < kMaxDelayMeasurements) {
       delay_states_[input_elements_to_write_].delta_time_ms = diff;
       delay_states_[input_elements_to_write_].buffer_delay_ms =
@@ skipping 248 matching lines @@
   aos->Stop();
   ais->Stop();
 
   // All Close() operations that run on the mocked audio thread,
   // should be synchronous and not post additional close tasks to
   // mocked the audio thread. Hence, there is no need to call
   // message_loop()->RunAllPending() after the Close() methods.
   aos->Close();
   ais->Close();
 }