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

media/audio/audio_input_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/memory/scoped_ptr.h"
 #include "base/message_loop.h"
 #include "base/threading/platform_thread.h"
 #include "media/audio/audio_io.h"
 #include "media/audio/audio_manager_base.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 static const int kSamplingRate = 8000;
 static const int kSamplesPerPacket = kSamplingRate / 20;
 
 // This class allows to find out if the callbacks are occurring as
 // expected and if any error has been reported.
 class TestInputCallback : public AudioInputStream::AudioInputCallback {
  public:
   explicit TestInputCallback(int max_data_bytes)
       : callback_count_(0),
         had_error_(0),
         was_closed_(0),
         max_data_bytes_(max_data_bytes) {
   }
   virtual void OnData(AudioInputStream* stream, const uint8* data,
-                      uint32 size, uint32 hardware_delay_bytes) {
+                      uint32 size, uint32 hardware_delay_bytes, double volume) {
     ++callback_count_;
     // Read the first byte to make sure memory is good.
     if (size) {
       ASSERT_LE(static_cast<int>(size), max_data_bytes_);
       int value = data[0];
       EXPECT_GE(value, 0);
     }
   }
   virtual void OnClose(AudioInputStream* stream) {
     ++was_closed_;
@@ skipping 124 matching lines @@
       FROM_HERE,
       MessageLoop::QuitClosure(),
       base::TimeDelta::FromMilliseconds(590));
   message_loop.Run();
   EXPECT_GE(test_callback.callback_count(), 10);
   EXPECT_FALSE(test_callback.had_error());
 
   ais->Stop();
   ais->Close();
 }