Fix the maximum native sample rate in AudioProcessing

webrtc/voice_engine/utility.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.
  */
 
 /*
  *  Contains functions often used by different parts of VoiceEngine.
  */
 
 #ifndef WEBRTC_VOICE_ENGINE_UTILITY_H_
 #define WEBRTC_VOICE_ENGINE_UTILITY_H_
 
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class AudioFrame;
 
 namespace voe {
 
 // Upmix or downmix and resample the audio in |src_frame| to |dst_frame|.
 // Expects |dst_frame| to have its sample rate and channels members set to the
 // desired values. Updates the samples per channel member accordingly. No other
 // members will be changed.

[Andrew MacDonald, 2015/09/23 02:35:52]

This comment has been false for awhile. It applies to the overload you added but
not the existing one. Could you clarify the difference between the two versions?

[aluebs-webrtc, 2015/09/23 17:13:43]

Done.

 void RemixAndResample(const AudioFrame& src_frame,
                       PushResampler<int16_t>* resampler,
                       AudioFrame* dst_frame);
-
-// Downmix and downsample the audio in |src_data| to |dst_af| as necessary,
-// specified by |codec_num_channels| and |codec_rate_hz|. |mono_buffer| is
-// temporary space and must be of sufficient size to hold the downmixed source
-// audio (recommend using a size of kMaxMonoDataSizeSamples).
-//
-// |dst_af| will have its data and format members (sample rate, channels and
-// samples per channel) set appropriately. No other members will be changed.
-// TODO(ajm): For now, this still calls Reset() on |dst_af|. Remove this, as
-// it shouldn't be needed.
-void DownConvertToCodecFormat(const int16_t* src_data,
-                              size_t samples_per_channel,
-                              int num_channels,
-                              int sample_rate_hz,
-                              int codec_num_channels,
-                              int codec_rate_hz,
-                              int16_t* mono_buffer,
-                              PushResampler<int16_t>* resampler,
-                              AudioFrame* dst_af);
+void RemixAndResample(const int16_t* src_data,
+                      size_t samples_per_channel,
+                      int num_channels,
+                      int sample_rate_hz,
+                      PushResampler<int16_t>* resampler,
+                      AudioFrame* dst_frame);
 
 void MixWithSat(int16_t target[],
                 int target_channel,
                 const int16_t source[],
                 int source_channel,
                 size_t source_len);
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_UTILITY_H_