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/logging.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 {
 
+constexpr FakeRecordingDevice::LevelToScalingMappingKind kDefaultMicKind =
+    FakeRecordingDevice::LevelToScalingMappingKind::kIdentity;
+
 void CopyFromAudioFrame(const AudioFrame& src, ChannelBuffer<float>* dest) {
   RTC_CHECK_EQ(src.num_channels_, dest->num_channels());
   RTC_CHECK_EQ(src.samples_per_channel_, dest->num_frames());
   // Copy the data from the input buffer.
   std::vector<float> tmp(src.samples_per_channel_ * src.num_channels_);
   S16ToFloat(src.data_, tmp.size(), tmp.data());
   Deinterleave(tmp.data(), src.samples_per_channel_, src.num_channels_,
                dest->channels());
 }
 
@@ skipping 33 matching lines @@
   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),
+          last_specified_microphone_level_(settings.initial_mic_gain),
+          fake_recording_device_(settings_.simulate_mic_gain ?
+              static_cast<FakeRecordingDevice::LevelToScalingMappingKind>(
+                  *settings.simulated_mic_kind) : kDefaultMicKind) {
   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) {
+  // Optionally use the fake recording device to simulate analog gain.
+  if (settings_.simulate_mic_gain) {
+    if (fixed_interface) {
+      fake_recording_device_.SimulateAnalogGain(
+          &fwd_frame_, &fwd_frame_, last_specified_microphone_level_,
+          real_recording_device_level_);
+    } else {
+      const size_t channel_size = in_config_.num_frames();
+
+      std::vector<rtc::ArrayView<const float>> data_view;

[peah-webrtc, 2017/05/05 20:25:21]

The lines 122-129 are only there to conform to the API of SimulateAnalogGain. As
that API is only used here, I'd prefer to change SimulateAnalogGain to be able
to receive in_buf_ as it is, which would avoid the need for 122-129. 

[AleBzk, 2017/05/16 08:53:03]

Done.

+      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_.SimulateAnalogGain(
+          data_view, after_scaling_view, last_specified_microphone_level_,
+          real_recording_device_level_);
+    }
+  }
+
+  // Notify the mic gain level to AGC.
+  LOG(LS_VERBOSE) << "AGC set_stream_analog_level set to "
+      << last_specified_microphone_level_;
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               ap_->gain_control()->set_stream_analog_level(
+                   last_specified_microphone_level_));
+
+  // Process the current audio frame.
   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()));
   }
 
+  // Store the mic gain level suggested by AGC if required.
+  last_specified_microphone_level_ = ap_->gain_control()->stream_analog_level();

[peah-webrtc, 2017/05/05 20:25:21]

With the changes done in this CL, I think last_specified_microphone_level_
should be renamed as it is used a bit different. 
What about last_specified_microphone_level_ -> new_analog_level_ ?

[AleBzk, 2017/05/16 08:53:03]

Done.

+
   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
-                                           << ", ";
+    residual_echo_likelihood_graph_writer_
+        << stats.residual_echo_likelihood << ", ";

[peah-webrtc, 2017/05/05 20:25:21]

 Unrelated change?

[AleBzk, 2017/05/16 08:53:03]

yup, sorry. I switched back.

   }
 
   ++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_));
@@ skipping 49 matching lines @@
   fwd_frame_.samples_per_channel_ =
       rtc::CheckedDivExact(fwd_frame_.sample_rate_hz_, kChunksPerSecond);
   fwd_frame_.num_channels_ = input_num_channels;
 
   rev_frame_.sample_rate_hz_ = reverse_input_sample_rate_hz;
   rev_frame_.samples_per_channel_ =
       rtc::CheckedDivExact(rev_frame_.sample_rate_hz_, kChunksPerSecond);
   rev_frame_.num_channels_ = reverse_input_num_channels;
 
   if (settings_.use_verbose_logging) {
+    rtc::LogMessage::LogToDebug(rtc::LS_VERBOSE);
+
     std::cout << "Sample rates:" << std::endl;
     std::cout << " Forward input: " << input_sample_rate_hz << std::endl;
     std::cout << " Forward output: " << output_sample_rate_hz << std::endl;
     std::cout << " Reverse input: " << reverse_input_sample_rate_hz
               << std::endl;
     std::cout << " Reverse output: " << reverse_output_sample_rate_hz
               << std::endl;
     std::cout << "Number of channels: " << std::endl;
     std::cout << " Forward input: " << input_num_channels << std::endl;
     std::cout << " Forward output: " << output_num_channels << std::endl;
@@ skipping 182 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