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>

[peah-webrtc, 2017/06/29 05:45:27]

Why is utility needed?

[AleBzk, 2017/06/29 11:43:36]

Good catch, thanks!

 
 #include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_processing/test/fake_recording_device.h"

[peah-webrtc, 2017/06/29 05:45:27]

This include is not needed.

[AleBzk, 2017/06/29 11:43:36]

Done.

 #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) {
   std::vector<WavBasedSimulator::SimulationEventType> call_chain;
   FILE* stream = OpenFile(filename.c_str(), "r");
@@ skipping 21 matching lines @@
     }
 
     num_read = fread(&c, sizeof(char), 1, stream);
   }
 
   fclose(stream);
   return call_chain;
 }
 
 WavBasedSimulator::WavBasedSimulator(const SimulationSettings& settings)
-      : AudioProcessingSimulator(settings) {}
+    : AudioProcessingSimulator(settings) {}

[peah-webrtc, 2017/06/29 05:45:27]

Is this caused by git cl format? Was the indendation wrong before?

[AleBzk, 2017/06/29 11:43:36]

Right.

 
 WavBasedSimulator::~WavBasedSimulator() = default;
 
 std::vector<WavBasedSimulator::SimulationEventType>
 WavBasedSimulator::GetDefaultEventChain() {
   std::vector<WavBasedSimulator::SimulationEventType> call_chain(2);
   call_chain[0] = SimulationEventType::kProcessStream;
   call_chain[1] = SimulationEventType::kProcessReverseStream;
   return call_chain;
 }
 
 void WavBasedSimulator::PrepareProcessStreamCall() {
   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);
-
-  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