Adding APM configuration in AEC dump.

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 221 matching lines @@
 
   noise_suppression_ = new NoiseSuppressionImpl(this, crit_);
   component_list_.push_back(noise_suppression_);
 
   voice_detection_ = new VoiceDetectionImpl(this, crit_);
   component_list_.push_back(voice_detection_);
 
   gain_control_for_new_agc_.reset(new GainControlForNewAgc(gain_control_));
 
   SetExtraOptions(config);
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+  config_ = GetCurrentConfig();
+#endif
 }
 
 AudioProcessingImpl::~AudioProcessingImpl() {
   {
     CriticalSectionScoped crit_scoped(crit_);
     // Depends on gain_control_ and gain_control_for_new_agc_.
     agc_manager_.reset();
     // Depends on gain_control_.
     gain_control_for_new_agc_.reset();
     while (!component_list_.empty()) {
@@ skipping 293 matching lines @@
   ProcessingConfig processing_config = api_format_;
   processing_config.input_stream() = input_config;
   processing_config.output_stream() = output_config;
 
   RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
   assert(processing_config.input_stream().num_frames() ==
          api_format_.input_stream().num_frames());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_file_->Open()) {
+    ApmConfig current_config = GetCurrentConfig();
+    if (current_config != config_) {
+      config_ = current_config;
+      RETURN_ON_ERR(WriteConfigMessage());
+    }
+  }
+#endif
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+  if (debug_file_->Open()) {
     event_msg_->set_type(audioproc::Event::STREAM);
     audioproc::Stream* msg = event_msg_->mutable_stream();
     const size_t channel_size =
         sizeof(float) * api_format_.input_stream().num_frames();
     for (int i = 0; i < api_format_.input_stream().num_channels(); ++i)
       msg->add_input_channel(src[i], channel_size);
   }
 #endif
 
   capture_audio_->CopyFrom(src, api_format_.input_stream());
@@ skipping 365 matching lines @@
     if (debug_file_->CloseFile() == -1) {
       return kFileError;
     }
   }
 
   if (debug_file_->OpenFile(filename, false) == -1) {
     debug_file_->CloseFile();
     return kFileError;
   }
 
-  int err = WriteInitMessage();
+  config_ = GetCurrentConfig();
+  int err = WriteConfigMessage();
   if (err != kNoError) {
     return err;
   }
+
+  err = WriteInitMessage();
+  if (err != kNoError) {
+    return err;
+  }
   return kNoError;
 #else
   return kUnsupportedFunctionError;
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
 int AudioProcessingImpl::StartDebugRecording(FILE* handle) {
   CriticalSectionScoped crit_scoped(crit_);
 
   if (handle == NULL) {
     return kNullPointerError;
   }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // Stop any ongoing recording.
   if (debug_file_->Open()) {
     if (debug_file_->CloseFile() == -1) {
       return kFileError;
     }
   }
 
   if (debug_file_->OpenFromFileHandle(handle, true, false) == -1) {
     return kFileError;
   }
 
-  int err = WriteInitMessage();
+  config_ = GetCurrentConfig();
+  int err = WriteConfigMessage();
   if (err != kNoError) {
     return err;
   }
+
+  err = WriteInitMessage();
+  if (err != kNoError) {
+    return err;
+  }
   return kNoError;
 #else
   return kUnsupportedFunctionError;
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 }
 
 int AudioProcessingImpl::StartDebugRecordingForPlatformFile(
     rtc::PlatformFile handle) {
   FILE* stream = rtc::FdopenPlatformFileForWriting(handle);
   return StartDebugRecording(stream);
@@ skipping 256 matching lines @@
   msg->set_output_sample_rate(api_format_.output_stream().sample_rate_hz());
   // TODO(ekmeyerson): Add reverse output fields to event_msg_.
 
   int err = WriteMessageToDebugFile();
   if (err != kNoError) {
     return err;
   }
 
   return kNoError;
 }
+
+AudioProcessingImpl::ApmConfig AudioProcessingImpl::GetCurrentConfig() const {

[ivoc, 2015/09/25 09:24:21]

Wouldn't it be better to return an audioproc::Config from this function, so you
don't have to convert it later in the WriteConfigMessage function?

[minyue-webrtc, 2015/09/25 09:49:09]

I think it is very valid. I also felt it cumbersome adding ApmConfig.

+  ApmConfig config;
+  // Acoustic echo canceler
+  config.aec_enabled = echo_cancellation_->is_enabled();
+  config.aec_delay_agnostic = echo_cancellation_->is_delay_agnostic_enabled();
+  config.aec_drift_compensation =
+      echo_cancellation_->is_drift_compensation_enabled();
+  config.aec_extended_filter =
+      echo_cancellation_->is_extended_filter_enabled();
+  config.aec_suppression_level = echo_cancellation_->suppression_level();
+
+  // Mobile AEC
+  config.aecm_enabled = echo_control_mobile_->is_enabled();
+  config.aecm_comfort_noise =
+      echo_control_mobile_->is_comfort_noise_enabled();
+  config.aecm_routing_mode = echo_control_mobile_->routing_mode();
+
+  // Automatic gain controller
+  config.agc_enabled = gain_control_->is_enabled();
+  config.agc_experiment = use_new_agc_;
+  config.agc_mode = gain_control_->mode();
+  config.agc_limiter = gain_control_->is_limiter_enabled();
+
+  // High pass filter
+  config.hpf_enabled = high_pass_filter_->is_enabled();
+
+  // Noise suppression
+  config.ns_enabled = noise_suppression_->is_enabled();
+  config.ns_experiment = transient_suppressor_enabled_;
+  config.ns_level = noise_suppression_->level();
+
+  return config;
+}
+
+int AudioProcessingImpl::WriteConfigMessage() {
+  event_msg_->set_type(audioproc::Event::CONFIG);
+  audioproc::Config* msg = event_msg_->mutable_config();
+
+  // Acoustic echo canceler
+  msg->set_aec_enabled(config_.aec_enabled);
+  msg->set_aec_delay_agnostic(config_.aec_delay_agnostic);
+
+  msg->set_aec_drift_compensation(config_.aec_drift_compensation);
+  msg->set_aec_extended_filter(config_.aec_extended_filter);
+  msg->set_aec_suppression_level(config_.aec_suppression_level);
+
+  // Mobile AEC
+  msg->set_aecm_enabled(config_.aecm_enabled);
+  msg->set_aecm_comfort_noise(config_.aecm_comfort_noise);
+  msg->set_aecm_routing_mode(config_.aecm_routing_mode);
+
+  // Automatic gain controller
+  msg->set_agc_enabled(config_.agc_enabled);
+  msg->set_agc_experiment(config_.agc_experiment);
+  msg->set_agc_mode(config_.agc_mode);
+  msg->set_agc_limiter(config_.agc_limiter);
+
+  // High pass filter
+  msg->set_hpf_enabled(config_.hpf_enabled);
+
+  // Noise suppression
+  msg->set_ns_enabled(config_.ns_enabled);
+  msg->set_ns_experiment(config_.ns_experiment);
+  msg->set_ns_level(config_.ns_level);
+
+  int err = WriteMessageToDebugFile();
+  if (err != kNoError) {
+    return err;
+  }
+
+  return kNoError;
+}
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc