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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
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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
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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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