audioproc_f with fake microphone.

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.
  */
 
@@ skipping 62 matching lines @@
     for (size_t sample = 0; sample < dest->samples_per_channel_; ++sample) {
       dest->data_[sample * dest->num_channels_ + ch] =
           src.channels()[ch][sample] * 32767;
     }
   }
 }
 
 AudioProcessingSimulator::AudioProcessingSimulator(
     const SimulationSettings& settings)
     : settings_(settings) {
+  if (settings_.simulate_mic_gain) {
+    fake_recording_device_.emplace(
+        static_cast<FakeRecordingDevice::LevelToScalingMappingKind>(
+            *settings_.simulated_mic_kind));
+  }
   if (settings_.ed_graph_output_filename &&
       settings_.ed_graph_output_filename->size() > 0) {
     residual_echo_likelihood_graph_writer_.open(
         *settings_.ed_graph_output_filename);
     RTC_CHECK(residual_echo_likelihood_graph_writer_.is_open());
     WriteEchoLikelihoodGraphFileHeader(&residual_echo_likelihood_graph_writer_);
   }
 }
 
 AudioProcessingSimulator::~AudioProcessingSimulator() {
   if (residual_echo_likelihood_graph_writer_.is_open()) {
     WriteEchoLikelihoodGraphFileFooter(&residual_echo_likelihood_graph_writer_);
     residual_echo_likelihood_graph_writer_.close();
   }
 }
 
 AudioProcessingSimulator::ScopedTimer::~ScopedTimer() {
   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.
+      if (fake_recording_device_) {
+        fake_recording_device_->ProcessStream(&fwd_frame_, &fwd_frame_);
+      }
       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.
+    if (fake_recording_device_) {
+      const size_t channel_size = in_config_.num_frames();
+
+      std::vector<rtc::ArrayView<const float>> data_view;
+      std::vector<rtc::ArrayView<float>> after_scaling_view;
+      for (size_t i = 0; i < in_config_.num_channels(); ++i) {
+        data_view.emplace_back(in_buf_->channels()[i], channel_size);
+        after_scaling_view.emplace_back(in_buf_->channels()[i], channel_size);
+      }
+
+      fake_recording_device_->ProcessStream(data_view, after_scaling_view);
+    }
     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