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

media/audio/audio_input_controller_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/bind.h"
 #include "base/message_loop.h"
 #include "base/synchronization/waitable_event.h"
 #include "base/test/test_timeouts.h"
 #include "media/audio/audio_input_controller.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 using ::testing::_;
 using ::testing::AtLeast;
 using ::testing::Exactly;
 using ::testing::InvokeWithoutArgs;
 using ::testing::NotNull;
 
 namespace media {
 
 static const int kSampleRate = AudioParameters::kAudioCDSampleRate;
 static const int kBitsPerSample = 16;
 static const int kChannels = 2;
 static const ChannelLayout kChannelLayout = CHANNEL_LAYOUT_STEREO;
 static const int kSamplesPerPacket = kSampleRate / 10;
 
+// Posts MessageLoop::QuitClosure() on specified message loop.
 ACTION_P(QuitMessageLoop, loop_or_proxy) {
   loop_or_proxy->PostTask(FROM_HERE, MessageLoop::QuitClosure());
 }
 
+// Posts MessageLoop::QuitClosure() on specified message loop after a certain
+// number of calls given by |limit|.
 ACTION_P3(CheckCountAndPostQuitTask, count, limit, loop_or_proxy) {
   if (++*count >= limit) {
     loop_or_proxy->PostTask(FROM_HERE, MessageLoop::QuitClosure());
   }
 }
 
 // Closes AudioOutputController synchronously.
 static void CloseAudioController(AudioInputController* controller) {
   base::WaitableEvent closed_event(true, false);
   controller->Close(base::Bind(&base::WaitableEvent::Signal,
@@ skipping 180 matching lines @@
 
   base::WaitableEvent closed_event_2(true, false);
   controller->Close(base::Bind(&base::WaitableEvent::Signal,
                                base::Unretained(&closed_event_2)));
 
   closed_event_1.Wait();
   closed_event_2.Wait();
 }
 
 }  // namespace media