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 35e2d2c4d8c4e289105e9455f860932b83ad5ef4..f4737e69555bfa811dd305a97958675063622989 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
@@ -14,18 +14,24 @@
 #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/aec_dump/aec_dump_factory.h"

[AleBzk, 2017/06/29 11:43:36]

Unrelated (merge)

 #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());
@@ -80,7 +86,13 @@ void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) {
 
 AudioProcessingSimulator::AudioProcessingSimulator(
     const SimulationSettings& settings)
-    : settings_(settings), worker_queue_("file_writer_task_queue") {
+    : settings_(settings),
+      fake_recording_device_(settings.initial_mic_level,
+                             settings_.simulate_mic_gain
+                                 ? static_cast<FakeRecordingDevice::DeviceKind>(

[peah-webrtc, 2017/06/29 22:04:00]

I think you should move the selection of DeviceKind into the
fake_recording_device. 
Having it here just constitutes a cast to something specified elsewhere and I
think the mapping between the parameter value and the actual DeviceKind would be
more clear if the DeviceKind and the mapping was done at the same place.

Please check the constructor for more comments.

[AleBzk, 2017/07/26 13:42:30]

Done.

+                                       *settings.simulated_mic_kind)
+                                 : kDefaultFakeRecDeviceKind),
+      worker_queue_("file_writer_task_queue") {

[AleBzk, 2017/06/29 11:43:36]

last line is unrelated (merge)

   if (settings_.ed_graph_output_filename &&
       settings_.ed_graph_output_filename->size() > 0) {
     residual_echo_likelihood_graph_writer_.open(
@@ -105,6 +117,31 @@ AudioProcessingSimulator::ScopedTimer::~ScopedTimer() {
 }
 
 void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
+  if (settings_.aec_dump_input_filename && settings_.simulate_mic_gain) {

[peah-webrtc, 2017/06/29 22:04:00]

This is better, but I'd rephrase it as

  if (settings_.simulate_mic_gain) {
     if (settings_.aec_dump_input_filename) { 
        RTC_DCHECK(aec_dump_mic_level_);
        fake_recording_device_.set_undo_mic_level(aec_dump_mic_level_);
     }
 
      if (fixed_interface) {
        fake_recording_device_.SimulateAnalogGain(&fwd_frame_);
      } else {
        fake_recording_device_.SimulateAnalogGain(in_buf_.get());
      }
    }

No need to have two different if-s on simulate_mic_gain

[AleBzk, 2017/07/26 13:42:30]

Done.

+    // 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_.set_undo_mic_level(aec_dump_mic_level_);
+  }
+
+  // Optionally use the fake recording device to simulate analog gain.
+  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.
+  if (settings_.simulate_mic_gain) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->gain_control()->set_stream_analog_level(

[peah-webrtc, 2017/06/29 22:04:00]

No, I don't think this is ok. The fake recording device should not be involved
if the microphone gain is not being simulated.

Furthermore, I doubt that this is correct now, as the  previous call to
set_mic_level was removed.

[AleBzk, 2017/07/26 13:42:30]

The code below is executed only when the mic gain is simulated.
So it should be ok.

RTC_CHECK_EQ(AudioProcessing::kNoError,
    ap_->gain_control()->set_stream_analog_level(
        fake_recording_device_.mic_level()));
It hasn't been removed; look line 151 on the right (latest patch set).

+                     fake_recording_device_.mic_level()));
+  }
+
+  // Process the current audio frame.
   if (fixed_interface) {
     {
       const auto st = ScopedTimer(mutable_proc_time());
@@ -118,6 +155,10 @@ void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
                                     out_config_, out_buf_->channels()));
   }
 
+  // Store the mic level suggested by AGC.
+  fake_recording_device_.set_mic_level(
+      ap_->gain_control()->stream_analog_level());
+
   if (buffer_writer_) {
     buffer_writer_->Write(*out_buf_);
   }
@@ -195,6 +236,8 @@ 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;