Extension and refactoring of the audioproc_f tool to be a fully fledged tool for audio processing m…

webrtc/modules/audio_processing/test/process_test.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 690 matching lines @@
           if (tick_diff.Microseconds() < min_time_reverse_us) {
             min_time_reverse_us = tick_diff.Microseconds();
           }
         }
 
       } else if (event_msg.type() == Event::STREAM) {
         ASSERT_TRUE(event_msg.has_stream());
         const Stream msg = event_msg.stream();
         primary_count++;
 
-        // ProcessStream could have changed this for the output frame.
-        near_frame.num_channels_ = apm->num_input_channels();
-
         ASSERT_TRUE(msg.has_input_data() ^ (msg.input_channel_size() > 0));
         if (msg.has_input_data()) {
           ASSERT_EQ(sizeof(int16_t) * samples_per_channel *
               near_frame.num_channels_, msg.input_data().size());
           memcpy(near_frame.data_,
                  msg.input_data().data(),
                  msg.input_data().size());
           near_read_bytes += msg.input_data().size();
         } else {
           for (int i = 0; i < msg.input_channel_size(); ++i) {
@@ skipping 426 matching lines @@
 }  // namespace
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   webrtc::void_main(argc, argv);
 
   // Optional, but removes memory leak noise from Valgrind.
   google::protobuf::ShutdownProtobufLibrary();
   return 0;
 }