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

media/audio/mac/audio_low_latency_input_mac.h

Index: media/audio/mac/audio_low_latency_input_mac.h
diff --git a/media/audio/mac/audio_low_latency_input_mac.h b/media/audio/mac/audio_low_latency_input_mac.h
index baea80160e7ee18ac94f6c1c317c5d0702167df4..cab1b6d9f9fda747d8e1936471f593948a68fc72 100644
--- a/media/audio/mac/audio_low_latency_input_mac.h
+++ b/media/audio/mac/audio_low_latency_input_mac.h
@@ -38,8 +38,10 @@
 
 #include <AudioUnit/AudioUnit.h>
 
+#include "base/atomicops.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/synchronization/lock.h"
+#include "base/time.h"
 #include "media/audio/audio_io.h"
 #include "media/audio/audio_parameters.h"
 
@@ -64,6 +66,8 @@ class AUAudioInputStream : public AudioInputStream {
   virtual double GetMaxVolume() OVERRIDE;
   virtual void SetVolume(double volume) OVERRIDE;
   virtual double GetVolume() OVERRIDE;
+  virtual void SetAutomaticGainControl(bool enabled) OVERRIDE;
+  virtual bool GetAutomaticGainControl() OVERRIDE;
 
   // Returns the current hardware sample rate for the default input device.
   static double HardwareSampleRate();
@@ -140,6 +144,19 @@ class AUAudioInputStream : public AudioInputStream {
   // when querying the volume of each channel.
   int number_of_channels_in_frame_;
 
+  // Keeps track of last time a new volume level was fed to the client in an
+  // OnData() callback.
+  base::Time last_volume_update_time_;
+
+  // Contains last result of internal call to GetVolume(). We save resources
+  // by not quering the capture volume for each callback.
+  double volume_;
+
+  // True when Automatic Gain Control is enabled, false otherwise.
+  // This member is modified on the audio thread and read on the internal
+  // Audio Unit thread in each capture callback.
+  base::subtle::Atomic32 agc_is_enabled_;

[scherkus (not reviewing), 2012/03/21 09:21:21]

please avoid using base::subtle -- it's intended to be used to build
synchronization primitives (i.e., Lock)

there's two ways around this:
  1) disallow changing AGC on the fly (i.e., must be set prior to recording)
  2) use a lock -- there will be no contention for the lock seeing as this value
is set once

if I had to pick I'd go with (1) assuming that WebRTC knows whether it wants AGC
or not prior to recording and does not need to change it on the fly

[henrika (OOO until Aug 14), 2012/03/21 10:16:04]

Good idea. I'll come back with a new version and will go for #1 as you propose.

+
   DISALLOW_COPY_AND_ASSIGN(AUAudioInputStream);
 };