Add an optional artificial nearend signal for aecdump call recreation

webrtc/modules/audio_processing/test/audio_processing_simulator.h

 /*
  *  Copyright (c) 2016 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 26 matching lines @@
   rtc::Optional<int> output_sample_rate_hz;
   rtc::Optional<int> output_num_channels;
   rtc::Optional<int> reverse_output_sample_rate_hz;
   rtc::Optional<int> reverse_output_num_channels;
   rtc::Optional<std::string> microphone_positions;
   int target_angle_degrees = 90;
   rtc::Optional<std::string> output_filename;
   rtc::Optional<std::string> reverse_output_filename;
   rtc::Optional<std::string> input_filename;
   rtc::Optional<std::string> reverse_input_filename;
+  rtc::Optional<std::string> artificial_nearend_filename;
   rtc::Optional<bool> use_aec;
   rtc::Optional<bool> use_aecm;
   rtc::Optional<bool> use_ed;  // Residual Echo Detector.
   rtc::Optional<std::string> ed_graph_output_filename;
   rtc::Optional<bool> use_agc;
   rtc::Optional<bool> use_hpf;
   rtc::Optional<bool> use_ns;
   rtc::Optional<bool> use_ts;
   rtc::Optional<bool> use_bf;
   rtc::Optional<bool> use_ie;
@@ skipping 85 matching lines @@
                                   int reverse_output_sample_rate_hz,
                                   int input_num_channels,
                                   int output_num_channels,
                                   int reverse_input_num_channels,
                                   int reverse_output_num_channels);
 
   const SimulationSettings settings_;
   std::unique_ptr<AudioProcessing> ap_;
 
   std::unique_ptr<ChannelBuffer<float>> in_buf_;
+  std::unique_ptr<ChannelBuffer<float>> artificial_nearend_buf_;

[aleloi, 2016/12/08 13:00:55]

The nearend buffer is only used in the aec-dump-based simulator. To limit the
scope in which they are visible, I think the buffer and reader should be private
in the derived class.

[peah-webrtc, 2016/12/09 07:19:25]

Great point! I move this and artificial_nearend_buffer_reader_ into the
aec_dump_based simulator class instead.

Done.

   std::unique_ptr<ChannelBuffer<float>> out_buf_;
   std::unique_ptr<ChannelBuffer<float>> reverse_in_buf_;
   std::unique_ptr<ChannelBuffer<float>> reverse_out_buf_;
   StreamConfig in_config_;
   StreamConfig out_config_;
   StreamConfig reverse_in_config_;
   StreamConfig reverse_out_config_;
   std::unique_ptr<ChannelBufferWavReader> buffer_reader_;
+  std::unique_ptr<ChannelBufferWavReader> artificial_nearend_buffer_reader_;
   std::unique_ptr<ChannelBufferWavReader> reverse_buffer_reader_;
   AudioFrame rev_frame_;
   AudioFrame fwd_frame_;
   bool bitexact_output_ = true;
 
  private:
   void SetupOutput();
 
   size_t num_process_stream_calls_ = 0;
   size_t num_reverse_process_stream_calls_ = 0;
   size_t output_reset_counter_ = 0;
   std::unique_ptr<ChannelBufferWavWriter> buffer_writer_;
   std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_writer_;
   TickIntervalStats proc_time_;
   std::ofstream residual_echo_likelihood_graph_writer_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_