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

media/audio/audio_input_controller_unittest.cc

Index: media/audio/audio_input_controller_unittest.cc
diff --git a/media/audio/audio_input_controller_unittest.cc b/media/audio/audio_input_controller_unittest.cc
index 279bbc25c72a9146b3684032a98868df588ccd4c..7498562bffd7a13364622d2521870b29beb23f94 100644
--- a/media/audio/audio_input_controller_unittest.cc
+++ b/media/audio/audio_input_controller_unittest.cc
@@ -25,14 +25,17 @@ 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());
 }
 
-ACTION_P3(CheckCountAndPostQuitTask, count, limit, loop_or_proxy) {
-  if (++*count >= limit) {
-    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.