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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
 
 #include <stddef.h>  // size_t
 #include <stdio.h>  // FILE
 #include <vector>
 
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/common.h"
 #include "webrtc/modules/audio_processing/beamformer/array_util.h"
 #include "webrtc/typedefs.h"
 
 struct AecCore;
 
 namespace webrtc {
 
 class AudioFrame;
@@ skipping 93 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;
-
 // 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
 // close to the audio hardware abstraction layer (HAL) as possible.
@@ skipping 324 matching lines @@
   enum NativeRate {
     kSampleRate8kHz = 8000,
     kSampleRate16kHz = 16000,
     kSampleRate32kHz = 32000,
     kSampleRate48kHz = 48000
   };
 
   static const int kChunkSizeMs = 10;
 };
 
+static const int kAudioProcNativeSampleRatesHz[] = {

[Andrew MacDonald, 2015/09/16 02:35:08]

nit: Make these class members:

class AudioProcessing {
  static const int kNativeSampleRatesHz[];
  static const int kMaxNativeSampleRateHz;
};

in .cc
const int AudioProcessing::kNativeSampleRateHz[] = { ... }
const int AudioProcsesing::kMaxNativeSampleRateHz = ...;

[aluebs-webrtc, 2015/09/16 03:02:53]

I though about that, but there is no cc file. Is there another way around it?

[Andrew MacDonald, 2015/09/16 04:04:08]

You can use audio_processing_impl.cc.

[aluebs-webrtc, 2015/09/22 01:35:29]

Good point. Done.

+    AudioProcessing::kSampleRate8kHz,
+    AudioProcessing::kSampleRate16kHz,
+    AudioProcessing::kSampleRate32kHz,
+    AudioProcessing::kSampleRate48kHz};
+static const size_t kNumAudioProcNativeSampleRates =
+    arraysize(kAudioProcNativeSampleRatesHz);
+static const int kAudioProcMaxNativeSampleRateHz =
+    kAudioProcNativeSampleRatesHz[kNumAudioProcNativeSampleRates - 1];
+
 class StreamConfig {
  public:
   // sample_rate_hz: The sampling rate of the stream.
   //
   // num_channels: The number of audio channels in the stream, excluding the
   //               keyboard channel if it is present. When passing a
   //               StreamConfig with an array of arrays T*[N],
   //
   //                N == {num_channels + 1  if  has_keyboard
   //                     {num_channels      if  !has_keyboard
@@ skipping 449 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_