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 50 matching lines @@
 
   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);
 
   RTC_CHECK_EQ(AudioProcessing::kNoError,
                ap_->gain_control()->set_stream_analog_level(
-                   last_specified_microphone_level_));
+                  last_specified_microphone_level_));
+  // TODO(aleloi): No undo level to set, i.e., no call to

[peah-webrtc, 2017/05/02 21:27:33]

Is this todo really needed? It is a todo about not to do something which is not
done.

+  // FakeRecordingDevice::NotifyAudioDeviceLevel().
 }
 
 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();
+    // TODO(aleloi): If settings_.simulate_mic_gain, set the returned value into

[peah-webrtc, 2017/05/02 21:27:33]

This makes sense. What about putting that into AudioProcessingSimulator instead?

+    // a FakeRecordingDevice instance via
+    // FakeRecordingDevice::set_analog_level() instead of using
+    // last_specified_microphone_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