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 543 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()) {
+    RETURN_ON_ERR(WriteConfigMessage(false));
+
     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();
-  if (err != kNoError) {
-    return err;
-  }
+  RETURN_ON_ERR(WriteConfigMessage(true));
+  RETURN_ON_ERR(WriteInitMessage());
   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();
-  if (err != kNoError) {
-    return err;
-  }
+  RETURN_ON_ERR(WriteConfigMessage(true));
+  RETURN_ON_ERR(WriteInitMessage());
   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 249 matching lines @@
   msg->set_sample_rate(api_format_.input_stream().sample_rate_hz());
   msg->set_num_input_channels(api_format_.input_stream().num_channels());
   msg->set_num_output_channels(api_format_.output_stream().num_channels());
   msg->set_num_reverse_channels(
       api_format_.reverse_input_stream().num_channels());
   msg->set_reverse_sample_rate(
       api_format_.reverse_input_stream().sample_rate_hz());
   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_ON_ERR(WriteMessageToDebugFile());
+  return kNoError;
+}
+
+int AudioProcessingImpl::WriteConfigMessage(bool forced) {
+  audioproc::Config config;
+
+  config.set_aec_enabled(echo_cancellation_->is_enabled());
+  config.set_aec_delay_agnostic_enabled(
+      echo_cancellation_->is_delay_agnostic_enabled());
+  config.set_aec_drift_compensation_enabled(
+      echo_cancellation_->is_drift_compensation_enabled());
+  config.set_aec_extended_filter_enabled(
+      echo_cancellation_->is_extended_filter_enabled());
+  config.set_aec_suppression_level(
+      static_cast<int>(echo_cancellation_->suppression_level()));
+
+  config.set_aecm_enabled(echo_control_mobile_->is_enabled());
+  config.set_aecm_comfort_noise_enabled(
+      echo_control_mobile_->is_comfort_noise_enabled());
+  config.set_aecm_routing_mode(
+      static_cast<int>(echo_control_mobile_->routing_mode()));
+
+  config.set_agc_enabled(gain_control_->is_enabled());
+  config.set_agc_mode(static_cast<int>(gain_control_->mode()));
+  config.set_agc_limiter_enabled(gain_control_->is_limiter_enabled());
+  config.set_noise_robust_enabled(use_new_agc_);

[Andrew MacDonald, 2015/10/02 22:00:01]

Did you build this? This is using the wrong field name: it's missing "agc".

[minyue-webrtc, 2015/10/02 22:01:41]

oh, I thought I changed it. sorry, was too hasty.

+
+  config.set_hpf_enabled(high_pass_filter_->is_enabled());
+
+  config.set_ns_enabled(noise_suppression_->is_enabled());
+  config.set_ns_level(static_cast<int>(noise_suppression_->level()));
+
+  config.set_transient_suppression_enabled(transient_suppressor_enabled_);
+
+  std::string serialized_config = config.SerializeAsString();
+  if (!forced && last_serialized_config_ == serialized_config) {
+    return kNoError;
   }
 
+  last_serialized_config_ = serialized_config;
+
+  event_msg_->set_type(audioproc::Event::CONFIG);
+  event_msg_->mutable_config()->CopyFrom(config);
+
+  RETURN_ON_ERR(WriteMessageToDebugFile());
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc