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Side by Side Diff: webrtc/modules/audio_processing/audio_processing_impl.cc

Issue 1348903004: Adding APM configuration in AEC dump. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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457 void AudioProcessingImpl::SetExtraOptions(const Config& config) { 457 void AudioProcessingImpl::SetExtraOptions(const Config& config) {
458 CriticalSectionScoped crit_scoped(crit_); 458 CriticalSectionScoped crit_scoped(crit_);
459 for (auto item : component_list_) { 459 for (auto item : component_list_) {
460 item->SetExtraOptions(config); 460 item->SetExtraOptions(config);
461 } 461 }
462 462
463 if (transient_suppressor_enabled_ != config.Get<ExperimentalNs>().enabled) { 463 if (transient_suppressor_enabled_ != config.Get<ExperimentalNs>().enabled) {
464 transient_suppressor_enabled_ = config.Get<ExperimentalNs>().enabled; 464 transient_suppressor_enabled_ = config.Get<ExperimentalNs>().enabled;
465 InitializeTransient(); 465 InitializeTransient();
466 } 466 }
467
468 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
469 if (debug_file_->Open()) {
470 int err = WriteInitMessage();
ivoc 2015/09/22 11:18:54 Can you explain why this call needs to be added he
minyue-webrtc 2015/09/22 11:39:28 Yes, it may lead to multiple init messages, but wo
peah-webrtc 2015/09/22 12:05:29 I'm very new to the protobuf concept, what is the
minyue-webrtc 2015/09/22 13:19:29 I do not know better, and so I included Ivo in the
peah-webrtc 2015/09/24 09:28:01 Acknowledged.
peah-webrtc 2015/09/24 09:28:01 It definitely makes sense to have unpack.cc restar
471 if (err != kNoError) {
472 LOG(LS_ERROR) << "Failed to update debug dump during SetExtraOptions.";
473 }
474 }
475 #endif
467 } 476 }
468 477
469 int AudioProcessingImpl::input_sample_rate_hz() const { 478 int AudioProcessingImpl::input_sample_rate_hz() const {
470 CriticalSectionScoped crit_scoped(crit_); 479 CriticalSectionScoped crit_scoped(crit_);
471 return api_format_.input_stream().sample_rate_hz(); 480 return api_format_.input_stream().sample_rate_hz();
472 } 481 }
473 482
474 int AudioProcessingImpl::sample_rate_hz() const { 483 int AudioProcessingImpl::sample_rate_hz() const {
475 CriticalSectionScoped crit_scoped(crit_); 484 CriticalSectionScoped crit_scoped(crit_);
476 return api_format_.input_stream().sample_rate_hz(); 485 return api_format_.input_stream().sample_rate_hz();
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1232 msg->set_sample_rate(api_format_.input_stream().sample_rate_hz()); 1241 msg->set_sample_rate(api_format_.input_stream().sample_rate_hz());
1233 msg->set_num_input_channels(api_format_.input_stream().num_channels()); 1242 msg->set_num_input_channels(api_format_.input_stream().num_channels());
1234 msg->set_num_output_channels(api_format_.output_stream().num_channels()); 1243 msg->set_num_output_channels(api_format_.output_stream().num_channels());
1235 msg->set_num_reverse_channels( 1244 msg->set_num_reverse_channels(
1236 api_format_.reverse_input_stream().num_channels()); 1245 api_format_.reverse_input_stream().num_channels());
1237 msg->set_reverse_sample_rate( 1246 msg->set_reverse_sample_rate(
1238 api_format_.reverse_input_stream().sample_rate_hz()); 1247 api_format_.reverse_input_stream().sample_rate_hz());
1239 msg->set_output_sample_rate(api_format_.output_stream().sample_rate_hz()); 1248 msg->set_output_sample_rate(api_format_.output_stream().sample_rate_hz());
1240 // TODO(ekmeyerson): Add reverse output fields to event_msg_. 1249 // TODO(ekmeyerson): Add reverse output fields to event_msg_.
1241 1250
1251 // Acoustic echo canceler
1252 msg->set_aec_enabled(echo_cancellation_->is_enabled());
1253 msg->set_aec_delay_agnostic(echo_cancellation_->is_delay_logging_enabled());
peah-webrtc 2015/09/22 12:05:29 The delay logging is set if the delay agnostic aec
minyue-webrtc 2015/09/22 13:19:29 Sorry, it is my bad, I chose a wrong flag to repor
1254
1255 msg->set_aec_drift_compensation(
1256 echo_cancellation_->is_drift_compensation_enabled());
1257 msg->set_aec_extended_filter(
1258 echo_cancellation_->is_extended_filter_enabled());
1259 msg->set_aec_suppression_level(echo_cancellation_->suppression_level());
1260
1261 // Mobile AEC
1262 msg->set_aecm_enabled(echo_control_mobile_->is_enabled());
1263 msg->set_aecm_comfort_noise(
1264 echo_control_mobile_->is_comfort_noise_enabled());
1265 msg->set_aecm_routing_mode(echo_control_mobile_->routing_mode());
1266
1267 // Automatic gain controller
1268 msg->set_agc_enabled(gain_control_->is_enabled());
1269 msg->set_agc_mode(gain_control_->mode());
1270 msg->set_agc_limiter(gain_control_->is_limiter_enabled());
1271
1272 // High pass filter
1273 msg->set_hpf_enabled(high_pass_filter_->is_enabled());
1274
1275 // Noise suppression
1276 msg->set_ns_enabled(noise_suppression_->is_enabled());
1277 msg->set_ns_experiment(noise_suppression_->is_enabled());
minyue-webrtc 2015/09/24 17:09:58 This was not right. noise suppression experiment i
1278 msg->set_ns_level(noise_suppression_->level());
1279
1242 int err = WriteMessageToDebugFile(); 1280 int err = WriteMessageToDebugFile();
1243 if (err != kNoError) { 1281 if (err != kNoError) {
1244 return err; 1282 return err;
1245 } 1283 }
1246 1284
1247 return kNoError; 1285 return kNoError;
1248 } 1286 }
1249 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP 1287 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
1250 1288
1251 } // namespace webrtc 1289 } // namespace webrtc
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