Improved AGC update scheme for the audio backend in Chrome.

This CL serves two purposes:

1) Improve the existing "mic-volume-checking"-scheme by ensuring that we no longer calls native audio functions from the core capture thread.
2) Prepare for adding AGC to the live-audio backend.

TBR=tommi@chromium.org
BUG=none
TEST=WebRTC loopback tests in Chrome where I monitor the microphone volume while speaking into the mic.

Review URL: https://codereview.chromium.org/15563004

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@202538 0039d316-1c4b-4281-b951-d872f2087c98

2 files changed, 8 insertions, 6 deletions

diff --git a/media/audio/mac/audio_low_latency_input_mac.cc b/media/audio/mac/audio_low_latency_input_mac.cc
index cf9d180..ff2f83f 100644
--- a/media/audio/mac/audio_low_latency_input_mac.cc
+++ b/media/audio/mac/audio_low_latency_input_mac.cc
@@ -268,6 +268,7 @@ void AUAudioInputStream::Start(AudioInputCallback* callback) {
   if (started_ || !audio_unit_)
     return;
   sink_ = callback;
+  StartAgc();
   OSStatus result = AudioOutputUnitStart(audio_unit_);
   if (result == noErr) {
     started_ = true;
@@ -279,6 +280,7 @@ void AUAudioInputStream::Start(AudioInputCallback* callback) {
 void AUAudioInputStream::Stop() {
   if (!started_)
     return;
+  StopAgc();
   OSStatus result = AudioOutputUnitStop(audio_unit_);
   if (result == noErr) {
     started_ = false;
@@ -483,11 +485,11 @@ OSStatus AUAudioInputStream::Provide(UInt32 number_of_frames,
   // Update the capture latency.
   double capture_latency_frames = GetCaptureLatency(time_stamp);
 
-  // Update the AGC volume level once every second. Note that, |volume| is
-  // also updated each time SetVolume() is called through IPC by the
-  // render-side AGC.
+  // The AGC volume level is updated once every second on a separate thread.
+  // Note that, |volume| is also updated each time SetVolume() is called
+  // through IPC by the render-side AGC.
   double normalized_volume = 0.0;
-  QueryAgcVolume(&normalized_volume);
+  GetAgcVolume(&normalized_volume);
 
   AudioBuffer& buffer = io_data->mBuffers[0];
   uint8* audio_data = reinterpret_cast<uint8*>(buffer.mData);
diff --git a/media/audio/mac/audio_low_latency_input_mac.h b/media/audio/mac/audio_low_latency_input_mac.h
index 04a4ff8..07a727b 100644
--- a/media/audio/mac/audio_low_latency_input_mac.h
+++ b/media/audio/mac/audio_low_latency_input_mac.h
@@ -42,8 +42,8 @@
 #include "base/atomicops.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/synchronization/lock.h"
+#include "media/audio/agc_audio_stream.h"
 #include "media/audio/audio_io.h"
-#include "media/audio/audio_input_stream_impl.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/seekable_buffer.h"
 
@@ -52,7 +52,7 @@ namespace media {
 class AudioManagerMac;
 class DataBuffer;
 
-class AUAudioInputStream : public AudioInputStreamImpl {
+class AUAudioInputStream : public AgcAudioStream<AudioInputStream> {
  public:
   // The ctor takes all the usual parameters, plus |manager| which is the
   // the audio manager who is creating this object.