audioproc_f with simulated mic analog gain

webrtc/modules/audio_processing/test/wav_based_simulator.cc

Index: webrtc/modules/audio_processing/test/wav_based_simulator.cc
diff --git a/webrtc/modules/audio_processing/test/wav_based_simulator.cc b/webrtc/modules/audio_processing/test/wav_based_simulator.cc
index 6cf0b744e6f5c040c81f7b8798e0992008c96b63..f5ad4b280e70349bf3ae7e1455c6bdfb1f42a48f 100644
--- a/webrtc/modules/audio_processing/test/wav_based_simulator.cc
+++ b/webrtc/modules/audio_processing/test/wav_based_simulator.cc
@@ -12,6 +12,8 @@
 
 #include <stdio.h>
 #include <iostream>
+#include <memory>
+#include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
@@ -79,10 +81,6 @@ void WavBasedSimulator::PrepareProcessStreamCall() {
 
   ap_->echo_cancellation()->set_stream_drift_samples(
       settings_.stream_drift_samples ? *settings_.stream_drift_samples : 0);

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

What you do here is that you move the set_stream_analog_level call to the main
simulator, and rely on the default parameter for it to always be called. That
makes sense, but I'd rather make the default parameter an ordinary parameter in
order to make this more explicit.

-
-  RTC_CHECK_EQ(AudioProcessing::kNoError,
-               ap_->gain_control()->set_stream_analog_level(
-                   last_specified_microphone_level_));
 }
 
 void WavBasedSimulator::PrepareReverseProcessStreamCall() {
@@ -143,10 +141,6 @@ bool WavBasedSimulator::HandleProcessStreamCall() {
   if (samples_left_to_process) {
     PrepareProcessStreamCall();
     ProcessStream(settings_.fixed_interface);
-    // Call stream analog level to ensure that any side-effects are triggered.
-    (void)ap_->gain_control()->stream_analog_level();
-    last_specified_microphone_level_ =
-        ap_->gain_control()->stream_analog_level();
   }
   return samples_left_to_process;
 }