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Issue 1568853002: Revert changes to default option setting in https://codereview.webrtc.org/1500633002/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 11 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2004 Google Inc. 3 * Copyright 2004 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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525 } 525 }
526 webrtc::Trace::set_level_filter(kDefaultTraceFilter); 526 webrtc::Trace::set_level_filter(kDefaultTraceFilter);
527 527
528 // Save the default AGC configuration settings. This must happen before 528 // Save the default AGC configuration settings. This must happen before
529 // calling ApplyOptions or the default will be overwritten. 529 // calling ApplyOptions or the default will be overwritten.
530 if (voe_wrapper_->processing()->GetAgcConfig(default_agc_config_) == -1) { 530 if (voe_wrapper_->processing()->GetAgcConfig(default_agc_config_) == -1) {
531 LOG_RTCERR0(GetAgcConfig); 531 LOG_RTCERR0(GetAgcConfig);
532 return false; 532 return false;
533 } 533 }
534 534
535 // Set default engine options.
536 {
537 AudioOptions options;
538 options.echo_cancellation = rtc::Optional<bool>(true);
539 options.auto_gain_control = rtc::Optional<bool>(true);
540 options.noise_suppression = rtc::Optional<bool>(true);
541 options.highpass_filter = rtc::Optional<bool>(true);
542 options.stereo_swapping = rtc::Optional<bool>(false);
543 options.audio_jitter_buffer_max_packets = rtc::Optional<int>(50);
544 options.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(false);
545 options.typing_detection = rtc::Optional<bool>(true);
546 options.adjust_agc_delta = rtc::Optional<int>(0);
547 options.experimental_agc = rtc::Optional<bool>(false);
548 options.extended_filter_aec = rtc::Optional<bool>(false);
549 options.delay_agnostic_aec = rtc::Optional<bool>(false);
550 options.experimental_ns = rtc::Optional<bool>(false);
551 options.aec_dump = rtc::Optional<bool>(false);
552 if (!ApplyOptions(options)) {
553 return false;
554 }
555 }
556
535 // Print our codec list again for the call diagnostic log 557 // Print our codec list again for the call diagnostic log
536 LOG(LS_INFO) << "WebRtc VoiceEngine codecs:"; 558 LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
537 for (const AudioCodec& codec : codecs_) { 559 for (const AudioCodec& codec : codecs_) {
538 LOG(LS_INFO) << ToString(codec); 560 LOG(LS_INFO) << ToString(codec);
539 } 561 }
540 562
541 SetDefaultDevices(); 563 SetDefaultDevices();
542 564
543 initialized_ = true; 565 initialized_ = true;
544 return true; 566 return true;
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562 584
563 VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(webrtc::Call* call, 585 VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(webrtc::Call* call,
564 const AudioOptions& options) { 586 const AudioOptions& options) {
565 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 587 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
566 return new WebRtcVoiceMediaChannel(this, options, call); 588 return new WebRtcVoiceMediaChannel(this, options, call);
567 } 589 }
568 590
569 bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { 591 bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
570 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 592 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
571 LOG(LS_INFO) << "ApplyOptions: " << options_in.ToString(); 593 LOG(LS_INFO) << "ApplyOptions: " << options_in.ToString();
572 594 AudioOptions options = options_in; // The options are modified below.
573 // Default engine options.
574 AudioOptions options;
575 options.echo_cancellation = rtc::Optional<bool>(true);
576 options.auto_gain_control = rtc::Optional<bool>(true);
577 options.noise_suppression = rtc::Optional<bool>(true);
578 options.highpass_filter = rtc::Optional<bool>(true);
579 options.stereo_swapping = rtc::Optional<bool>(false);
580 options.audio_jitter_buffer_max_packets = rtc::Optional<int>(50);
581 options.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(false);
582 options.typing_detection = rtc::Optional<bool>(true);
583 options.adjust_agc_delta = rtc::Optional<int>(0);
584 options.experimental_agc = rtc::Optional<bool>(false);
585 options.extended_filter_aec = rtc::Optional<bool>(false);
586 options.delay_agnostic_aec = rtc::Optional<bool>(false);
587 options.experimental_ns = rtc::Optional<bool>(false);
588 options.aec_dump = rtc::Optional<bool>(false);
589
590 // Apply any given options on top.
591 options.SetAll(options_in);
592 595
593 // kEcConference is AEC with high suppression. 596 // kEcConference is AEC with high suppression.
594 webrtc::EcModes ec_mode = webrtc::kEcConference; 597 webrtc::EcModes ec_mode = webrtc::kEcConference;
595 webrtc::AecmModes aecm_mode = webrtc::kAecmSpeakerphone; 598 webrtc::AecmModes aecm_mode = webrtc::kAecmSpeakerphone;
596 webrtc::AgcModes agc_mode = webrtc::kAgcAdaptiveAnalog; 599 webrtc::AgcModes agc_mode = webrtc::kAgcAdaptiveAnalog;
597 webrtc::NsModes ns_mode = webrtc::kNsHighSuppression; 600 webrtc::NsModes ns_mode = webrtc::kNsHighSuppression;
598 if (options.aecm_generate_comfort_noise) { 601 if (options.aecm_generate_comfort_noise) {
599 LOG(LS_VERBOSE) << "Comfort noise explicitly set to " 602 LOG(LS_VERBOSE) << "Comfort noise explicitly set to "
600 << *options.aecm_generate_comfort_noise 603 << *options.aecm_generate_comfort_noise
601 << " (default is false)."; 604 << " (default is false).";
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2500 } 2503 }
2501 } else { 2504 } else {
2502 LOG(LS_INFO) << "Stopping playout for channel #" << channel; 2505 LOG(LS_INFO) << "Stopping playout for channel #" << channel;
2503 engine()->voe()->base()->StopPlayout(channel); 2506 engine()->voe()->base()->StopPlayout(channel);
2504 } 2507 }
2505 return true; 2508 return true;
2506 } 2509 }
2507 } // namespace cricket 2510 } // namespace cricket
2508 2511
2509 #endif // HAVE_WEBRTC_VOICE 2512 #endif // HAVE_WEBRTC_VOICE
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