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 461fc71c5e16d272df8d7a23fb7d8e8eaba0a72f..667c11c7cd1b6600c0fa6216ef197b3a4f75217e 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
@@ -14,12 +14,15 @@
 #include <iostream>
 #include <sstream>
 #include <string>
+#include <utility>
 #include <vector>
 
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/test/fake_recording_device.h"
 #include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"
 #include "webrtc/rtc_base/stringutils.h"
 
 namespace webrtc {
@@ -80,7 +83,12 @@ void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) {
 
 AudioProcessingSimulator::AudioProcessingSimulator(
     const SimulationSettings& settings)
-    : settings_(settings), worker_queue_("file_writer_task_queue") {
+    : settings_(settings),
+      analog_mic_level_(settings.initial_mic_level),
+      fake_recording_device_(
+          settings.initial_mic_level,
+          settings_.simulate_mic_gain ? *settings.simulated_mic_kind : 0),
+      worker_queue_("file_writer_task_queue") {
   if (settings_.ed_graph_output_filename &&
       settings_.ed_graph_output_filename->size() > 0) {
     residual_echo_likelihood_graph_writer_.open(
@@ -105,6 +113,30 @@ 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 (settings_.aec_dump_input_filename) {
+      // When the analog gain is sumulated and an AEC dump is used as input, set
+      // the undo level to |aec_dump_mic_level_| to virtually restore the
+      // unmodified microphone signal level.
+      RTC_DCHECK(aec_dump_mic_level_);
+      fake_recording_device_.SetUndoMicLevel(aec_dump_mic_level_);
+    }
+
+    if (fixed_interface) {
+      fake_recording_device_.SimulateAnalogGain(&fwd_frame_);
+    } else {
+      fake_recording_device_.SimulateAnalogGain(in_buf_.get());
+    }
+
+    analog_mic_level_ = fake_recording_device_.MicLevel();
+  }
+
+  // Notify the current mic level to AGC.
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               ap_->gain_control()->set_stream_analog_level(analog_mic_level_));
+
+  // Process the current audio frame.
   if (fixed_interface) {
     {
       const auto st = ScopedTimer(mutable_proc_time());
@@ -118,6 +150,12 @@ void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
                                     out_config_, out_buf_->channels()));
   }
 
+  // Store the mic level suggested by AGC.
+  analog_mic_level_ = ap_->gain_control()->stream_analog_level();
+  if (settings_.simulate_mic_gain) {
+    fake_recording_device_.SetMicLevel(analog_mic_level_);
+  }
+
   if (buffer_writer_) {
     buffer_writer_->Write(*out_buf_);
   }
@@ -195,6 +233,8 @@ 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;