Fix the maximum native sample rate in AudioProcessing

webrtc/voice_engine/channel.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 3323 matching lines @@
     return 0;
 }
 
 void Channel::Demultiplex(const int16_t* audio_data,
                           int sample_rate,
                           size_t number_of_frames,
                           int number_of_channels) {
   CodecInst codec;
   GetSendCodec(codec);
 
-  if (!mono_recording_audio_.get()) {
-    // Temporary space for DownConvertToCodecFormat.
-    mono_recording_audio_.reset(new int16_t[kMaxMonoDataSizeSamples]);
-  }
-  DownConvertToCodecFormat(audio_data,
-                           number_of_frames,
-                           number_of_channels,
-                           sample_rate,
-                           codec.channels,
-                           codec.plfreq,
-                           mono_recording_audio_.get(),
-                           &input_resampler_,
-                           &_audioFrame);
+  // Never upsample the capture signal here. This should be done at the

[Andrew MacDonald, 2015/09/23 17:31:47]

add "or upmix"

[aluebs-webrtc, 2015/09/23 19:48:54]

Good point. Done.

+  // end of the send chain.
+  _audioFrame.sample_rate_hz_ = std::min(codec.plfreq, sample_rate);
+  _audioFrame.num_channels_ = std::min(number_of_channels, codec.channels);
+  RemixAndResample(audio_data, number_of_frames, number_of_channels,
+                   sample_rate, &input_resampler_, &_audioFrame);
 }
 
 uint32_t
 Channel::PrepareEncodeAndSend(int mixingFrequency)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::PrepareEncodeAndSend()");
 
     if (_audioFrame.samples_per_channel_ == 0)
     {
@@ skipping 780 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt)
       != 0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc