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

Issue 1952123003: Surface the IntelligibilityEnhancer on MediaConstraints (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Change to use MediaConstraints Created 4 years, 7 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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570 options.highpass_filter = rtc::Optional<bool>(true); 570 options.highpass_filter = rtc::Optional<bool>(true);
571 options.stereo_swapping = rtc::Optional<bool>(false); 571 options.stereo_swapping = rtc::Optional<bool>(false);
572 options.audio_jitter_buffer_max_packets = rtc::Optional<int>(50); 572 options.audio_jitter_buffer_max_packets = rtc::Optional<int>(50);
573 options.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(false); 573 options.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(false);
574 options.typing_detection = rtc::Optional<bool>(true); 574 options.typing_detection = rtc::Optional<bool>(true);
575 options.adjust_agc_delta = rtc::Optional<int>(0); 575 options.adjust_agc_delta = rtc::Optional<int>(0);
576 options.experimental_agc = rtc::Optional<bool>(false); 576 options.experimental_agc = rtc::Optional<bool>(false);
577 options.extended_filter_aec = rtc::Optional<bool>(false); 577 options.extended_filter_aec = rtc::Optional<bool>(false);
578 options.delay_agnostic_aec = rtc::Optional<bool>(false); 578 options.delay_agnostic_aec = rtc::Optional<bool>(false);
579 options.experimental_ns = rtc::Optional<bool>(false); 579 options.experimental_ns = rtc::Optional<bool>(false);
580 options.intelligibility_enhancer = rtc::Optional<bool>(false);
580 bool error = ApplyOptions(options); 581 bool error = ApplyOptions(options);
581 RTC_DCHECK(error); 582 RTC_DCHECK(error);
582 } 583 }
583 584
584 SetDefaultDevices(); 585 SetDefaultDevices();
585 } 586 }
586 587
587 WebRtcVoiceEngine::~WebRtcVoiceEngine() { 588 WebRtcVoiceEngine::~WebRtcVoiceEngine() {
588 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 589 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
589 LOG(LS_INFO) << "WebRtcVoiceEngine::~WebRtcVoiceEngine"; 590 LOG(LS_INFO) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
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836 837
837 if (options.experimental_ns) { 838 if (options.experimental_ns) {
838 experimental_ns_ = options.experimental_ns; 839 experimental_ns_ = options.experimental_ns;
839 } 840 }
840 if (experimental_ns_) { 841 if (experimental_ns_) {
841 LOG(LS_INFO) << "Experimental ns is enabled? " << *experimental_ns_; 842 LOG(LS_INFO) << "Experimental ns is enabled? " << *experimental_ns_;
842 config.Set<webrtc::ExperimentalNs>( 843 config.Set<webrtc::ExperimentalNs>(
843 new webrtc::ExperimentalNs(*experimental_ns_)); 844 new webrtc::ExperimentalNs(*experimental_ns_));
844 } 845 }
845 846
847 if (options.intelligibility_enhancer) {
henrika_webrtc 2016/05/11 08:50:36 You have access to the ADM here. Why can't you use
aluebs-webrtc 2016/05/11 21:22:05 Great point! I didn't realize there was a 2 way co
848 intelligibility_enhancer_ = options.intelligibility_enhancer;
849 }
850 if (intelligibility_enhancer_) {
851 LOG(LS_INFO) << "Intelligibility Enhancer is enabled? "
852 << *intelligibility_enhancer_;
853 config.Set<webrtc::Intelligibility>(
854 new webrtc::Intelligibility(*intelligibility_enhancer_));
the sun 2016/05/11 11:23:41 nit: the call to adm()->EnableBuiltInNS(false) sho
aluebs-webrtc 2016/05/11 21:22:05 Actually I think the best thing is to do it above
855 }
856
846 // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine 857 // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
847 // returns NULL on audio_processing(). 858 // returns NULL on audio_processing().
848 webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing(); 859 webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing();
849 if (audioproc) { 860 if (audioproc) {
850 audioproc->SetExtraOptions(config); 861 audioproc->SetExtraOptions(config);
851 } 862 }
852 863
853 if (options.recording_sample_rate) { 864 if (options.recording_sample_rate) {
854 LOG(LS_INFO) << "Recording sample rate is " 865 LOG(LS_INFO) << "Recording sample rate is "
855 << *options.recording_sample_rate; 866 << *options.recording_sample_rate;
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2556 } 2567 }
2557 } else { 2568 } else {
2558 LOG(LS_INFO) << "Stopping playout for channel #" << channel; 2569 LOG(LS_INFO) << "Stopping playout for channel #" << channel;
2559 engine()->voe()->base()->StopPlayout(channel); 2570 engine()->voe()->base()->StopPlayout(channel);
2560 } 2571 }
2561 return true; 2572 return true;
2562 } 2573 }
2563 } // namespace cricket 2574 } // namespace cricket
2564 2575
2565 #endif // HAVE_WEBRTC_VOICE 2576 #endif // HAVE_WEBRTC_VOICE
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