Unpack different wav files after each INIT event of the aecdump

webrtc/modules/audio_processing/test/unpack.cc

 /*
  *  Copyright (c) 2011 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 285 matching lines @@
       reverse_samples_per_channel =
           static_cast<size_t>(reverse_sample_rate / 100);
       input_samples_per_channel =
           static_cast<size_t>(input_sample_rate / 100);
       output_samples_per_channel =
           static_cast<size_t>(output_sample_rate / 100);
 
       if (!FLAGS_raw) {
         // The WAV files need to be reset every time, because they cant change
         // their sample rate or number of channels.
-        reverse_wav_file.reset(new WavWriter(FLAGS_reverse_file + ".wav",
+        std::stringstream reverse_name;
+        reverse_name << FLAGS_reverse_file << frame_count << ".wav";
+        reverse_wav_file.reset(new WavWriter(reverse_name.str(),
                                              reverse_sample_rate,
                                              num_reverse_channels));
-        input_wav_file.reset(new WavWriter(FLAGS_input_file + ".wav",
+        std::stringstream input_name;
+        input_name << FLAGS_input_file << frame_count << ".wav";
+        input_wav_file.reset(new WavWriter(input_name.str(),
                                            input_sample_rate,
                                            num_input_channels));
-        output_wav_file.reset(new WavWriter(FLAGS_output_file + ".wav",
+        std::stringstream output_name;
+        output_name << FLAGS_output_file << frame_count << ".wav";
+        output_wav_file.reset(new WavWriter(output_name.str(),
                                             output_sample_rate,
                                             num_output_channels));
       }
     }
   }
 
   return 0;
 }
 
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   return webrtc::do_main(argc, argv);
 }