Fix the maximum native sample rate in AudioProcessing

webrtc/modules/audio_processing/include/audio_processing.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 110 matching lines @@
 // AudioProcessing::SetExtraOptions().
 //
 // Note: If enabled and the reverse stream has more than one output channel,
 // the reverse stream will become an upmixed mono signal.
 struct Intelligibility {
   Intelligibility() : enabled(false) {}
   explicit Intelligibility(bool enabled) : enabled(enabled) {}
   bool enabled;
 };
 
-static const int kAudioProcMaxNativeSampleRateHz = 32000;
+static const int kAudioProcMaxNativeSampleRateHz = 48000;

[Andrew MacDonald, 2015/09/11 18:06:23]

Looks like this should only affect:
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/webrtc...

Have you tested that it works fine? It _looks_ like it should but my TODO there
worries me a bit :) I assume that was written before voice engine used the
SincResampler.

Assuming everything's fine, remove the TODO there.

[aluebs-webrtc, 2015/09/15 22:56:09]

You were right, it was not working for 44100, thank you for catching that! :)
Now I added some more logic to chose the destination sample rate.

 
 // The Audio Processing Module (APM) provides a collection of voice processing
 // components designed for real-time communications software.
 //
 // APM operates on two audio streams on a frame-by-frame basis. Frames of the
 // primary stream, on which all processing is applied, are passed to
 // |ProcessStream()|. Frames of the reverse direction stream, which are used for
 // analysis by some components, are passed to |AnalyzeReverseStream()|. On the
 // client-side, this will typically be the near-end (capture) and far-end
 // (render) streams, respectively. APM should be placed in the signal chain as
@@ skipping 794 matching lines @@
   // This does not impact the size of frames passed to |ProcessStream()|.
   virtual int set_frame_size_ms(int size) = 0;
   virtual int frame_size_ms() const = 0;
 
  protected:
   virtual ~VoiceDetection() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_