Fix the maximum native sample rate in AudioProcessing

webrtc/voice_engine/transmit_mixer.cc

 /*
  *  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 1118 matching lines @@
     return _fileRecording;
 }
 
 void TransmitMixer::GenerateAudioFrame(const int16_t* audio,
                                        size_t samples_per_channel,
                                        int num_channels,
                                        int sample_rate_hz) {
   int codec_rate;
   int num_codec_channels;
   GetSendCodecInfo(&codec_rate, &num_codec_channels);
-  // TODO(ajm): This currently restricts the sample rate to 32 kHz.
-  // See: https://code.google.com/p/webrtc/issues/detail?id=3146
-  // When 48 kHz is supported natively by AudioProcessing, this will have
-  // to be changed to handle 44.1 kHz.
   int max_sample_rate_hz = kAudioProcMaxNativeSampleRateHz;
   if (audioproc_->echo_control_mobile()->is_enabled()) {
     // AECM only supports 8 and 16 kHz.
     max_sample_rate_hz = 16000;
   }
   codec_rate = std::min(codec_rate, max_sample_rate_hz);
   stereo_codec_ = num_codec_channels == 2;
 
   if (!mono_buffer_.get()) {
     // Temporary space for DownConvertToCodecFormat.
@@ skipping 184 matching lines @@
 void TransmitMixer::EnableStereoChannelSwapping(bool enable) {
   swap_stereo_channels_ = enable;
 }
 
 bool TransmitMixer::IsStereoChannelSwappingEnabled() {
   return swap_stereo_channels_;
 }
 
 }  // namespace voe
 }  // namespace webrtc