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..eff15cda1d7c3ffaacc72275f0121ae8dc64b7f2 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
@@ -14,17 +14,23 @@
 #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"
+#include "webrtc/modules/audio_processing/test/fake_recording_device.h"
 
 namespace webrtc {
 namespace test {
 namespace {
 
+constexpr FakeRecordingDevice::DeviceKind kDefaultFakeRecDeviceKind =
+    FakeRecordingDevice::DeviceKind::IDENTITY;
+
 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 +84,13 @@ void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) {
 
 AudioProcessingSimulator::AudioProcessingSimulator(
     const SimulationSettings& settings)
-    : settings_(settings) {
+        : settings_(settings),
+          fake_recording_device_(FakeRecordingDevice::GetFakeRecDevice(
+              settings_.simulate_mic_gain ? static_cast<
+                  FakeRecordingDevice::DeviceKind>(*settings.simulated_mic_kind)
+                      : kDefaultFakeRecDeviceKind,
+                      settings.initial_mic_level)) {
+  RTC_DCHECK(fake_recording_device_);
   if (settings_.ed_graph_output_filename &&
       settings_.ed_graph_output_filename->size() > 0) {
     residual_echo_likelihood_graph_writer_.open(
@@ -103,6 +115,22 @@ AudioProcessingSimulator::ScopedTimer::~ScopedTimer() {
 }
 
 void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
+  // Optionally use the fake recording device to simulate analog gain.
+  RTC_DCHECK(fake_recording_device_);
+  if (settings_.simulate_mic_gain) {
+    if (fixed_interface) {
+      fake_recording_device_->SimulateAnalogGain(&fwd_frame_);
+    } else {
+      fake_recording_device_->SimulateAnalogGain(in_buf_.get());
+    }
+  }
+
+  // Notify the current mic level to AGC.
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               ap_->gain_control()->set_stream_analog_level(
+                  fake_recording_device_->mic_level()));
+
+  // Process the current audio frame.
   if (fixed_interface) {
     {
       const auto st = ScopedTimer(mutable_proc_time());
@@ -116,6 +144,10 @@ void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
                                     out_config_, out_buf_->channels()));
   }
 
+  // Store the mic level suggested by AGC if required.
+  fake_recording_device_->set_mic_level(
+      ap_->gain_control()->stream_analog_level());
+
   if (buffer_writer_) {
     buffer_writer_->Write(*out_buf_);
   }
@@ -193,6 +225,8 @@ 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;