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/stringutils.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 
 namespace webrtc {
 namespace test {
 namespace {
@@ skipping 68 matching lines @@
   }
 }
 
 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) {
+void AudioProcessingSimulator::ProcessStream(bool fixed_interface,
+                                             bool skip_analog_level_update) {
+  if (!skip_analog_level_update)

[aleloi, 2017/04/21 11:46:42]

Add brackets. Also, suggest changing |if not skip_analog| into |if
do_level_update|.

[AleBzk, 2017/04/24 09:40:26]

Done.

+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->gain_control()->set_stream_analog_level(
+                     last_specified_microphone_level_));
   if (fixed_interface) {
     {
       const auto st = ScopedTimer(mutable_proc_time());
+      // TODO(alessiob): Apply last_specified_microphone_level_ to fwd_frame_
+      // simulating a mic with analog gain.
       RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->ProcessStream(&fwd_frame_));
     }
     CopyFromAudioFrame(fwd_frame_, out_buf_.get());
   } else {
     const auto st = ScopedTimer(mutable_proc_time());
+    // TODO(alessiob): Apply last_specified_microphone_level_ to
+    // in_buf_->channels() simulating a mic with analog gain.
     RTC_CHECK_EQ(AudioProcessing::kNoError,
                  ap_->ProcessStream(in_buf_->channels(), in_config_,
                                     out_config_, out_buf_->channels()));
   }
+  // Update last_specified_microphone_level_ using the value suggested by AGC
+  // or the default if settings_.simulate_mic_gain is false.
+  // TODO(alessiob): Check what the default value should be, probably 100 is ok.
+  if (!skip_analog_level_update)

[aleloi, 2017/04/21 11:46:42]

Add brackets 

[AleBzk, 2017/04/24 09:40:26]

Done.

+    last_specified_microphone_level_ = (settings_.simulate_mic_gain) ?

[aleloi, 2017/04/21 11:46:42]

Suggest drop parentheses.

[AleBzk, 2017/04/24 09:40:26]

Done.

+        ap_->gain_control()->stream_analog_level() : 100;
 
   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
                                            << ", ";
   }
@@ skipping 260 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