audioproc_f with simulated mic analog gain

webrtc/modules/audio_processing/test/wav_based_simulator.cc

 /*
  *  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.
  */
 
 #include "webrtc/modules/audio_processing/test/wav_based_simulator.h"
 
 #include <stdio.h>
 #include <iostream>
+#include <memory>
+#include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/test/testsupport/trace_to_stderr.h"
 
 namespace webrtc {
 namespace test {
 
 std::vector<WavBasedSimulator::SimulationEventType>
 WavBasedSimulator::GetCustomEventChain(const std::string& filename) {
@@ skipping 46 matching lines @@
   if (settings_.fixed_interface) {
     CopyToAudioFrame(*in_buf_, &fwd_frame_);
   }
   ap_->set_stream_key_pressed(settings_.use_ts && (*settings_.use_ts));
 
   RTC_CHECK_EQ(AudioProcessing::kNoError,
                ap_->set_stream_delay_ms(
                    settings_.stream_delay ? *settings_.stream_delay : 0));
 
   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() {
   if (settings_.fixed_interface) {
     CopyToAudioFrame(*reverse_in_buf_, &rev_frame_);
   }
 }
 
 void WavBasedSimulator::Process() {
   std::unique_ptr<test::TraceToStderr> trace_to_stderr;
@@ skipping 40 matching lines @@
   }
 
   DestroyAudioProcessor();
 }
 
 bool WavBasedSimulator::HandleProcessStreamCall() {
   bool samples_left_to_process = buffer_reader_->Read(in_buf_.get());
   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;
 }
 
 bool WavBasedSimulator::HandleProcessReverseStreamCall() {
   bool samples_left_to_process =
       reverse_buffer_reader_->Read(reverse_in_buf_.get());
   if (samples_left_to_process) {
     PrepareReverseProcessStreamCall();
     ProcessReverseStream(settings_.fixed_interface);
@@ skipping 37 matching lines @@
   }
 
   SetupBuffersConfigsOutputs(
       input_sample_rate_hz, output_sample_rate_hz, reverse_sample_rate_hz,
       reverse_output_sample_rate_hz, input_num_channels, output_num_channels,
       reverse_num_channels, reverse_output_num_channels);
 }
 
 }  // namespace test
 }  // namespace webrtc