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Side by Side Diff: webrtc/modules/audio_processing/test/audio_processing_simulator.cc

Issue 2090583002: New module for the adaptive level controlling functionality in the audio processing module (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Temporarily deactivated the level controller until the CL with the proper tuning has been landed Created 4 years, 5 months ago
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1 /* 1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2016 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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216 } 216 }
217 if (settings_.use_ts) { 217 if (settings_.use_ts) {
218 config.Set<ExperimentalNs>(new ExperimentalNs(*settings_.use_ts)); 218 config.Set<ExperimentalNs>(new ExperimentalNs(*settings_.use_ts));
219 } 219 }
220 if (settings_.use_ie) { 220 if (settings_.use_ie) {
221 config.Set<Intelligibility>(new Intelligibility(*settings_.use_ie)); 221 config.Set<Intelligibility>(new Intelligibility(*settings_.use_ie));
222 } 222 }
223 if (settings_.use_aec3) { 223 if (settings_.use_aec3) {
224 config.Set<EchoCanceller3>(new EchoCanceller3(*settings_.use_aec3)); 224 config.Set<EchoCanceller3>(new EchoCanceller3(*settings_.use_aec3));
225 } 225 }
226 if (settings_.use_lc) {
227 config.Set<LevelControl>(new LevelControl(true));
228 }
226 if (settings_.use_refined_adaptive_filter) { 229 if (settings_.use_refined_adaptive_filter) {
227 config.Set<RefinedAdaptiveFilter>( 230 config.Set<RefinedAdaptiveFilter>(
228 new RefinedAdaptiveFilter(*settings_.use_refined_adaptive_filter)); 231 new RefinedAdaptiveFilter(*settings_.use_refined_adaptive_filter));
229 } 232 }
230 config.Set<ExtendedFilter>(new ExtendedFilter( 233 config.Set<ExtendedFilter>(new ExtendedFilter(
231 !settings_.use_extended_filter || *settings_.use_extended_filter)); 234 !settings_.use_extended_filter || *settings_.use_extended_filter));
232 config.Set<DelayAgnostic>(new DelayAgnostic(!settings_.use_delay_agnostic || 235 config.Set<DelayAgnostic>(new DelayAgnostic(!settings_.use_delay_agnostic ||
233 *settings_.use_delay_agnostic)); 236 *settings_.use_delay_agnostic));
234 237
235 ap_.reset(AudioProcessing::Create(config)); 238 ap_.reset(AudioProcessing::Create(config));
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326 size_t kMaxFilenameSize = AudioProcessing::kMaxFilenameSize; 329 size_t kMaxFilenameSize = AudioProcessing::kMaxFilenameSize;
327 RTC_CHECK_LE(settings_.aec_dump_output_filename->size(), kMaxFilenameSize); 330 RTC_CHECK_LE(settings_.aec_dump_output_filename->size(), kMaxFilenameSize);
328 RTC_CHECK_EQ(AudioProcessing::kNoError, 331 RTC_CHECK_EQ(AudioProcessing::kNoError,
329 ap_->StartDebugRecording( 332 ap_->StartDebugRecording(
330 settings_.aec_dump_output_filename->c_str(), -1)); 333 settings_.aec_dump_output_filename->c_str(), -1));
331 } 334 }
332 } 335 }
333 336
334 } // namespace test 337 } // namespace test
335 } // namespace webrtc 338 } // namespace webrtc
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