Add support to audioproc_f for running the residual echo detector and producing an echo likelihood …

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 24 matching lines @@
 }
 
 std::string GetIndexedOutputWavFilename(const std::string& wav_name,
                                         int counter) {
   std::stringstream ss;
   ss << wav_name.substr(0, wav_name.size() - 4) << "_" << counter
      << wav_name.substr(wav_name.size() - 4);
   return ss.str();
 }
 
+void WriteEchoLikelihoodGraphFileHeader(std::ofstream* output_file) {
+  (*output_file) << "import numpy as np" << std::endl
+                 << "import matplotlib.pyplot as plt" << std::endl
+                 << "y = np.array([";
+}
+
+void WriteEchoLikelihoodGraphFileFooter(std::ofstream* output_file) {
+  (*output_file) << "])" << std::endl
+                 << "x = np.arange(len(y))*.01" << std::endl
+                 << "plt.plot(x, y)" << std::endl
+                 << "plt.ylabel('Echo likelihood')" << std::endl
+                 << "plt.xlabel('Time (s)')" << std::endl
+                 << "plt.ylim([0,1])" << std::endl
+                 << "plt.show()" << std::endl;
+}
+
 }  // namespace
 
 SimulationSettings::SimulationSettings() = default;
 SimulationSettings::SimulationSettings(const SimulationSettings&) = default;
 SimulationSettings::~SimulationSettings() = default;
 
 void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) {
   RTC_CHECK_EQ(src.num_channels(), dest->num_channels_);
   RTC_CHECK_EQ(src.num_frames(), dest->samples_per_channel_);
   for (size_t ch = 0; ch < dest->num_channels_; ++ch) {
     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) {}
+    : settings_(settings) {
+  if (settings_.red_graph_output_filename &&
+      settings_.red_graph_output_filename->size() > 0) {
+    residual_echo_likelihood_graph_writer_.open(
+        *settings_.red_graph_output_filename);
+    RTC_CHECK(residual_echo_likelihood_graph_writer_.is_open());

[hlundin-webrtc, 2016/11/14 14:23:11]

Great! Now all you have to do is remember to #include "webrtc/base/checks.h". :)

[ivoc, 2016/11/14 14:55:04]

Done. I saw all those other CHECKS in the file so I didn't bother to check the
includes.

+    WriteEchoLikelihoodGraphFileHeader(&residual_echo_likelihood_graph_writer_);
+  }
+}
 
-AudioProcessingSimulator::~AudioProcessingSimulator() = default;
+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());
       RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->ProcessStream(&fwd_frame_));
     }
     CopyFromAudioFrame(fwd_frame_, out_buf_.get());
   } else {
     const auto st = ScopedTimer(mutable_proc_time());
     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()) {
+    auto stats = ap_->GetStatistics();
+    residual_echo_likelihood_graph_writer_ << stats.residual_echo_likelihood
+                                           << ", ";
+  }
+
   ++num_process_stream_calls_;
 }
 
 void AudioProcessingSimulator::ProcessReverseStream(bool fixed_interface) {
   if (fixed_interface) {
     const auto st = ScopedTimer(mutable_proc_time());
     RTC_CHECK_EQ(AudioProcessing::kNoError,
                  ap_->ProcessReverseStream(&rev_frame_));
     CopyFromAudioFrame(rev_frame_, reverse_out_buf_.get());
 
@@ skipping 135 matching lines @@
     apm_config.level_controller.enabled = *settings_.use_lc;
   }
   if (settings_.use_refined_adaptive_filter) {
     config.Set<RefinedAdaptiveFilter>(
         new RefinedAdaptiveFilter(*settings_.use_refined_adaptive_filter));
   }
   config.Set<ExtendedFilter>(new ExtendedFilter(
       !settings_.use_extended_filter || *settings_.use_extended_filter));
   config.Set<DelayAgnostic>(new DelayAgnostic(!settings_.use_delay_agnostic ||
                                               *settings_.use_delay_agnostic));
+  if (settings_.use_red) {
+    apm_config.residual_echo_detector.enabled = *settings_.use_red;
+  }
 
   ap_.reset(AudioProcessing::Create(config));
   RTC_CHECK(ap_);
 
   ap_->ApplyConfig(apm_config);
 
   if (settings_.use_aec) {
     RTC_CHECK_EQ(AudioProcessing::kNoError,
                  ap_->echo_cancellation()->Enable(*settings_.use_aec));
   }
@@ skipping 85 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