audioproc_f with simulated mic analog gain

webrtc/modules/audio_processing/test/audioproc_float.cc

Index: webrtc/modules/audio_processing/test/audioproc_float.cc
diff --git a/webrtc/modules/audio_processing/test/audioproc_float.cc b/webrtc/modules/audio_processing/test/audioproc_float.cc
index 9f1abc430d88747d161c8ad260e151614a769311..46e294086a6c8fad4fe513d17d0f8bbdc129f2c7 100644
--- a/webrtc/modules/audio_processing/test/audioproc_float.cc
+++ b/webrtc/modules/audio_processing/test/audioproc_float.cc
@@ -158,6 +158,9 @@ DEFINE_int32(stream_delay,
 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,
@@ -264,6 +267,7 @@ SimulationSettings CreateSettings() {
                         &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;