audioproc_f with simulated mic analog gain

webrtc/modules/audio_processing/test/audio_processing_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/audio_processing_simulator.h"
 
 #include <algorithm>
 #include <iostream>
 #include <sstream>
 #include <string>
+#include <utility>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 
 namespace webrtc {
 namespace test {
 namespace {
@@ skipping 72 matching lines @@
   int64_t interval = rtc::TimeNanos() - start_time_;
   proc_time_->sum += interval;
   proc_time_->max = std::max(proc_time_->max, interval);
   proc_time_->min = std::min(proc_time_->min, interval);
 }
 
 void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
   if (fixed_interface) {
     {
       const auto st = ScopedTimer(mutable_proc_time());
+      // TODO(alessiob): Change fwd_frame_ simulating a mic with analog gain and
+      // an application gain.
       RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->ProcessStream(&fwd_frame_));
     }
     CopyFromAudioFrame(fwd_frame_, out_buf_.get());
   } else {
     const auto st = ScopedTimer(mutable_proc_time());
+    // TODO(alessiob): Change in_buf_->channels() simulating a mic with analog
+    // gain and an application gain.
     RTC_CHECK_EQ(AudioProcessing::kNoError,
                  ap_->ProcessStream(in_buf_->channels(), in_config_,
                                     out_config_, out_buf_->channels()));
   }
 
   if (buffer_writer_) {
     buffer_writer_->Write(*out_buf_);
   }
 
   if (residual_echo_likelihood_graph_writer_.is_open()) {
@@ skipping 264 matching lines @@
     size_t kMaxFilenameSize = AudioProcessing::kMaxFilenameSize;
     RTC_CHECK_LE(settings_.aec_dump_output_filename->size(), kMaxFilenameSize);
     RTC_CHECK_EQ(AudioProcessing::kNoError,
                  ap_->StartDebugRecording(
                      settings_.aec_dump_output_filename->c_str(), -1));
   }
 }
 
 }  // namespace test
 }  // namespace webrtc