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Side by Side Diff: webrtc/media/engine/webrtcvoiceengine.cc

Issue 2334583002: Revert of Introduced new scheme for controlling the functionality inside the audio processing module (Closed)
Patch Set: Created 4 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2004 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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861 config.Set<webrtc::Intelligibility>( 861 config.Set<webrtc::Intelligibility>(
862 new webrtc::Intelligibility(*intelligibility_enhancer_)); 862 new webrtc::Intelligibility(*intelligibility_enhancer_));
863 } 863 }
864 864
865 if (options.level_control) { 865 if (options.level_control) {
866 level_control_ = options.level_control; 866 level_control_ = options.level_control;
867 } 867 }
868 868
869 LOG(LS_INFO) << "Level control: " 869 LOG(LS_INFO) << "Level control: "
870 << (!!level_control_ ? *level_control_ : -1); 870 << (!!level_control_ ? *level_control_ : -1);
871 webrtc::AudioProcessing::Config apm_config;
872 if (level_control_) { 871 if (level_control_) {
873 apm_config.level_controller.enabled = *level_control_; 872 config.Set<webrtc::LevelControl>(new webrtc::LevelControl(*level_control_));
874 } 873 }
875 874
876 // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine 875 // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
877 // returns NULL on audio_processing(). 876 // returns NULL on audio_processing().
878 webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing(); 877 webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing();
879 if (audioproc) { 878 if (audioproc) {
880 audioproc->SetExtraOptions(config); 879 audioproc->SetExtraOptions(config);
881 audioproc->ApplyConfig(apm_config);
882 } 880 }
883 881
884 if (options.recording_sample_rate) { 882 if (options.recording_sample_rate) {
885 LOG(LS_INFO) << "Recording sample rate is " 883 LOG(LS_INFO) << "Recording sample rate is "
886 << *options.recording_sample_rate; 884 << *options.recording_sample_rate;
887 if (adm()->SetRecordingSampleRate(*options.recording_sample_rate)) { 885 if (adm()->SetRecordingSampleRate(*options.recording_sample_rate)) {
888 LOG_RTCERR1(SetRecordingSampleRate, *options.recording_sample_rate); 886 LOG_RTCERR1(SetRecordingSampleRate, *options.recording_sample_rate);
889 } 887 }
890 } 888 }
891 889
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2661 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 2659 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
2662 const auto it = send_streams_.find(ssrc); 2660 const auto it = send_streams_.find(ssrc);
2663 if (it != send_streams_.end()) { 2661 if (it != send_streams_.end()) {
2664 return it->second->channel(); 2662 return it->second->channel();
2665 } 2663 }
2666 return -1; 2664 return -1;
2667 } 2665 }
2668 } // namespace cricket 2666 } // namespace cricket
2669 2667
2670 #endif // HAVE_WEBRTC_VOICE 2668 #endif // HAVE_WEBRTC_VOICE
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