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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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774 } | 774 } |
775 if (voep->SetNsStatus(*options.noise_suppression, ns_mode) == -1) { | 775 if (voep->SetNsStatus(*options.noise_suppression, ns_mode) == -1) { |
776 LOG_RTCERR2(SetNsStatus, *options.noise_suppression, ns_mode); | 776 LOG_RTCERR2(SetNsStatus, *options.noise_suppression, ns_mode); |
777 return false; | 777 return false; |
778 } else { | 778 } else { |
779 LOG(LS_INFO) << "Noise suppression set to " << *options.noise_suppression | 779 LOG(LS_INFO) << "Noise suppression set to " << *options.noise_suppression |
780 << " with mode " << ns_mode; | 780 << " with mode " << ns_mode; |
781 } | 781 } |
782 } | 782 } |
783 | 783 |
784 if (options.highpass_filter) { | |
785 LOG(LS_INFO) << "High pass filter enabled? " << *options.highpass_filter; | |
786 if (voep->EnableHighPassFilter(*options.highpass_filter) == -1) { | |
787 LOG_RTCERR1(SetHighpassFilterStatus, *options.highpass_filter); | |
788 return false; | |
789 } | |
790 } | |
791 | |
792 if (options.stereo_swapping) { | 784 if (options.stereo_swapping) { |
793 LOG(LS_INFO) << "Stereo swapping enabled? " << *options.stereo_swapping; | 785 LOG(LS_INFO) << "Stereo swapping enabled? " << *options.stereo_swapping; |
794 voep->EnableStereoChannelSwapping(*options.stereo_swapping); | 786 voep->EnableStereoChannelSwapping(*options.stereo_swapping); |
795 if (voep->IsStereoChannelSwappingEnabled() != *options.stereo_swapping) { | 787 if (voep->IsStereoChannelSwappingEnabled() != *options.stereo_swapping) { |
796 LOG_RTCERR1(EnableStereoChannelSwapping, *options.stereo_swapping); | 788 LOG_RTCERR1(EnableStereoChannelSwapping, *options.stereo_swapping); |
797 return false; | 789 return false; |
798 } | 790 } |
799 } | 791 } |
800 | 792 |
801 if (options.audio_jitter_buffer_max_packets) { | 793 if (options.audio_jitter_buffer_max_packets) { |
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870 << (!!level_control_ ? *level_control_ : -1); | 862 << (!!level_control_ ? *level_control_ : -1); |
871 webrtc::AudioProcessing::Config apm_config; | 863 webrtc::AudioProcessing::Config apm_config; |
872 if (level_control_) { | 864 if (level_control_) { |
873 apm_config.level_controller.enabled = *level_control_; | 865 apm_config.level_controller.enabled = *level_control_; |
874 if (options.level_control_initial_peak_level_dbfs) { | 866 if (options.level_control_initial_peak_level_dbfs) { |
875 apm_config.level_controller.initial_peak_level_dbfs = | 867 apm_config.level_controller.initial_peak_level_dbfs = |
876 *options.level_control_initial_peak_level_dbfs; | 868 *options.level_control_initial_peak_level_dbfs; |
877 } | 869 } |
878 } | 870 } |
879 | 871 |
872 LOG(LS_INFO) << "High pass filter: " | |
the sun
2016/10/26 09:06:43
If the AudioProcessing::Config has a ToString() me
peah-webrtc
2016/10/28 05:50:28
I totally agree! I remove this log line as well as
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873 << (!!options.highpass_filter ? *options.highpass_filter : -1); | |
874 if (options.highpass_filter) { | |
875 apm_config.high_pass_filter.enabled = *options.highpass_filter; | |
876 } | |
877 | |
880 // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine | 878 // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine |
881 // returns NULL on audio_processing(). | 879 // returns NULL on audio_processing(). |
882 webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing(); | 880 webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing(); |
883 if (audioproc) { | 881 if (audioproc) { |
884 audioproc->SetExtraOptions(config); | 882 audioproc->SetExtraOptions(config); |
885 audioproc->ApplyConfig(apm_config); | 883 audioproc->ApplyConfig(apm_config); |
886 } | 884 } |
887 | 885 |
888 if (options.recording_sample_rate) { | 886 if (options.recording_sample_rate) { |
889 LOG(LS_INFO) << "Recording sample rate is " | 887 LOG(LS_INFO) << "Recording sample rate is " |
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2515 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); | 2513 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
2516 const auto it = send_streams_.find(ssrc); | 2514 const auto it = send_streams_.find(ssrc); |
2517 if (it != send_streams_.end()) { | 2515 if (it != send_streams_.end()) { |
2518 return it->second->channel(); | 2516 return it->second->channel(); |
2519 } | 2517 } |
2520 return -1; | 2518 return -1; |
2521 } | 2519 } |
2522 } // namespace cricket | 2520 } // namespace cricket |
2523 | 2521 |
2524 #endif // HAVE_WEBRTC_VOICE | 2522 #endif // HAVE_WEBRTC_VOICE |
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