Add aecdump support to audioproc_f

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 618 matching lines @@
           printf("  Primary rates: %d Hz (in), %d Hz (out)\n",
                  msg.sample_rate(), output_sample_rate);
           printf("  Primary channels: %d (in), %d (out)\n",
                  msg.num_input_channels(),
                  msg.num_output_channels());
           printf("  Reverse rate: %d\n", reverse_sample_rate);
           printf("  Reverse channels: %d\n", msg.num_reverse_channels());
         }
 
         if (!raw_output) {
-          // The WAV file needs to be reset every time, because it cant change
-          // it's sample rate or number of channels.
+          // The WAV file needs to be reset every time, because it can't change
+          // its sample rate or number of channels.
           output_wav_file.reset(new WavWriter(out_filename + ".wav",
                                               output_sample_rate,
                                               msg.num_output_channels()));
         }
 
       } else if (event_msg.type() == Event::REVERSE_STREAM) {
         ASSERT_TRUE(event_msg.has_reverse_stream());
         ReverseStream msg = event_msg.reverse_stream();
         reverse_count++;
 
@@ skipping 490 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;
 }