audioproc_f with simulated mic analog gain

webrtc/modules/audio_processing/test/audioproc_float.cc

 /*
  *  Copyright (c) 2014 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 140 matching lines @@
              "Specify the VAD likelihood (0-3)");
 DEFINE_int32(ns_level,
              kParameterNotSpecifiedValue,
              "Specify the NS level (0-3)");
 DEFINE_int32(stream_delay,
              kParameterNotSpecifiedValue,
              "Specify the stream delay in ms to use");
 DEFINE_int32(stream_drift_samples,
              kParameterNotSpecifiedValue,
              "Specify the number of stream drift samples to use");
+DEFINE_bool(simulate_mic_gain,

[peah-webrtc, 2017/04/26 12:54:44]

I guess that in the long run you somehow need to specify what mic gain to use.
Have you looked into how to specify that to audioproc_f?

+            false,
+            "Simulate adjustable microphone analog gain");
 DEFINE_bool(performance_report, false, "Report the APM performance ");
 DEFINE_bool(verbose, false, "Produce verbose output");
 DEFINE_bool(bitexactness_report,
             false,
             "Report bitexactness for aec dump result reproduction");
 DEFINE_bool(discard_settings_in_aecdump,
             false,
             "Discard any config settings specified in the aec dump");
 DEFINE_bool(store_intermediate_output,
             false,
@@ skipping 86 matching lines @@
   SetSettingIfFlagSet(FLAGS_agc_limiter, &settings.use_agc_limiter);
   SetSettingIfSpecified(FLAGS_agc_compression_gain,
                         &settings.agc_compression_gain);
   SetSettingIfSpecified(FLAGS_vad_likelihood, &settings.vad_likelihood);
   SetSettingIfSpecified(FLAGS_ns_level, &settings.ns_level);
   SetSettingIfSpecified(FLAGS_stream_delay, &settings.stream_delay);
   SetSettingIfSpecified(FLAGS_stream_drift_samples,
                         &settings.stream_drift_samples);
   SetSettingIfSpecified(FLAGS_custom_call_order_file,
                         &settings.custom_call_order_filename);
+  settings.simulate_mic_gain = FLAGS_simulate_mic_gain;
   settings.report_performance = FLAGS_performance_report;
   settings.use_verbose_logging = FLAGS_verbose;
   settings.report_bitexactness = FLAGS_bitexactness_report;
   settings.discard_all_settings_in_aecdump = FLAGS_discard_settings_in_aecdump;
   settings.fixed_interface = FLAGS_fixed_interface;
   settings.store_intermediate_output = FLAGS_store_intermediate_output;
 
   return settings;
 }
 
@@ skipping 181 matching lines @@
 
   return 0;
 }
 
 }  // namespace test
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   return webrtc::test::main(argc, argv);
 }