Moved AudioUtil static functions to AudioManager interfaces

content/renderer/media/webrtc_audio_capturer.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 "content/renderer/media/webrtc_audio_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
@@ skipping 33 matching lines @@
     buffer_size = (sample_rate / 100);
     DCHECK_EQ(buffer_size * 100, sample_rate) <<
         "Sample rate not supported";
   }
 #elif defined(OS_LINUX) || defined(OS_OPENBSD)
   // Based on tests using the current ALSA implementation in Chrome, we have
   // found that the best combination is 20ms on the input side and 10ms on the
   // output side.
   // TODO(henrika): It might be possible to reduce the input buffer
   // size and reduce the delay even more.
-  buffer_size = 2 * sample_rate / 100;
+  if (sample_rate == 44100)

[DaleCurtis, 2013/03/02 01:53:56]

Hmm, what is this?

[no longer working on chromium, 2013/03/04 14:55:04]

Oh, this change is making the webrtc audio work when the sample rate is 44100,
but it should be done in another CL.

Removed.

+    buffer_size = 2 * 440;
+  else
+    buffer_size = 2 * sample_rate / 100;
 #elif defined(OS_ANDROID)
   // TODO(leozwang): Tune and adjust buffer size on Android.
   buffer_size = 2 * sample_rate / 100;
 #endif
 
   return buffer_size;
 }
 
 // This is a temporary audio buffer with parameters used to send data to
 // callbacks.
@@ skipping 417 matching lines @@
   if (!on_device_stopped_cb_.is_null())
     on_device_stopped_cb_.Run();
 }
 
 media::AudioParameters WebRtcAudioCapturer::audio_parameters() const {
   base::AutoLock auto_lock(lock_);
   return buffer_->params();
 }
 
 }  // namespace content