audioproc_f with simulated mic analog gain

webrtc/modules/audio_processing/test/audio_processing_simulator.cc

Index: webrtc/modules/audio_processing/test/audio_processing_simulator.cc
diff --git a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
index d2c274f460010af7976fc2a2f1032c8bb72ecc8a..e2df73b3a42eb9f04d8318c69ddc51e871e95ad9 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
@@ -14,9 +14,11 @@
 #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"
@@ -25,6 +27,9 @@ 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());
@@ -78,7 +83,11 @@ void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) {
 
 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(
@@ -103,6 +112,36 @@ AudioProcessingSimulator::ScopedTimer::~ScopedTimer() {
 }
 
 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());
@@ -116,14 +155,17 @@ void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
                                     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_;
@@ -193,6 +235,8 @@ void AudioProcessingSimulator::SetupBuffersConfigsOutputs(
   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;