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

media/audio/win/wavein_input_win.cc

Index: media/audio/win/wavein_input_win.cc
diff --git a/media/audio/win/wavein_input_win.cc b/media/audio/win/wavein_input_win.cc
index f8a4e7ded95e184b53eda5b64dae63ff8ce979b7..90cec5383efeec2240df9df451ae4ad1eaf2a398 100644
--- a/media/audio/win/wavein_input_win.cc
+++ b/media/audio/win/wavein_input_win.cc
@@ -201,6 +201,17 @@ double PCMWaveInAudioInputStream::GetVolume() {
   return 0.0;
 }
 
+void PCMWaveInAudioInputStream::SetAutomaticGainControl(bool enabled) {
+  // TODO(henrika): Add AGC support when volume control has been added.
+  NOTIMPLEMENTED();
+}
+
+bool PCMWaveInAudioInputStream::GetAutomaticGainControl() {
+  // TODO(henrika): Add AGC support when volume control has been added.
+  NOTIMPLEMENTED();
+  return false;
+}
+
 void PCMWaveInAudioInputStream::HandleError(MMRESULT error) {
   DLOG(WARNING) << "PCMWaveInAudio error " << error;
   callback_->OnError(this, error);
@@ -260,10 +271,12 @@ void PCMWaveInAudioInputStream::WaveCallback(HWAVEIN hwi, UINT msg,
     // to the callback and check if we need to stop playing.
     // It should be OK to assume the data in the buffer is what has been
     // recorded in the soundcard.
+    // TODO(henrika): comment on volume = 0.0.

[tommi (sloooow) - chröme, 2012/03/26 15:26:40]

ping

[henrika (OOO until Aug 14), 2012/03/27 09:20:38]

Done.

     WAVEHDR* buffer = reinterpret_cast<WAVEHDR*>(param1);
     obj->callback_->OnData(obj, reinterpret_cast<const uint8*>(buffer->lpData),
                            buffer->dwBytesRecorded,
-                           buffer->dwBytesRecorded);
+                           buffer->dwBytesRecorded,
+                           0.0);
 
     if (obj->state_ == kStateStopping) {
       // The main thread has called Stop() and is waiting to issue waveOutReset