Remove additional channel constraints when Beamforming is enabled in AudioProcessing

webrtc/modules/audio_processing/audio_buffer.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 132 matching lines @@
   for (int i = 0; i < num_proc_channels_; ++i) {
     FloatToFloatS16(data_ptr[i],
                     proc_num_frames_,
                     data_->fbuf()->channels()[i]);
   }
 }
 
 void AudioBuffer::CopyTo(const StreamConfig& stream_config,
                          float* const* data) {
   assert(stream_config.num_frames() == output_num_frames_);
-  assert(stream_config.num_channels() == num_channels_);
+  assert(stream_config.num_channels() == num_channels_ || num_channels_ == 1);
 
   // Convert to the float range.
   float* const* data_ptr = data;
   if (output_num_frames_ != proc_num_frames_) {
     // Convert to an intermediate buffer for subsequent resampling.
     data_ptr = process_buffer_->channels();
   }
   for (int i = 0; i < num_channels_; ++i) {
     FloatS16ToFloat(data_->fbuf()->channels()[i],
                     proc_num_frames_,
                     data_ptr[i]);
   }
 
   // Resample.
   if (output_num_frames_ != proc_num_frames_) {
     for (int i = 0; i < num_channels_; ++i) {
       output_resamplers_[i]->Resample(data_ptr[i],
                                       proc_num_frames_,
                                       data[i],
                                       output_num_frames_);
     }
   }
+
+  // Upmix

[hlundin-webrtc, 2016/01/11 16:07:23]

Nit: Upmix.

+  for (int i = num_channels_; i < stream_config.num_channels(); ++i) {
+    memcpy(data[i], data[0], output_num_frames_ * sizeof(**data));
+  }
 }
 
 void AudioBuffer::InitForNewData() {
   keyboard_data_ = NULL;
   mixed_low_pass_valid_ = false;
   reference_copied_ = false;
   activity_ = AudioFrame::kVadUnknown;
   num_channels_ = num_proc_channels_;
 }
 
@@ skipping 267 matching lines @@
 
 void AudioBuffer::SplitIntoFrequencyBands() {
   splitting_filter_->Analysis(data_.get(), split_data_.get());
 }
 
 void AudioBuffer::MergeFrequencyBands() {
   splitting_filter_->Synthesis(split_data_.get(), data_.get());
 }
 
 }  // namespace webrtc