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Issue 1479023002: Prepare the AudioSendStream to be hooked up to send-side BWE. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Comments addressed. Created 5 years 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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530 kRtpAudioLevelHeaderExtensionDefaultId)); 530 kRtpAudioLevelHeaderExtensionDefaultId));
531 rtp_header_extensions_.push_back( 531 rtp_header_extensions_.push_back(
532 RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension, 532 RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
533 kRtpAbsoluteSenderTimeHeaderExtensionDefaultId)); 533 kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
534 if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") { 534 if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
535 rtp_header_extensions_.push_back(RtpHeaderExtension( 535 rtp_header_extensions_.push_back(RtpHeaderExtension(
536 kRtpTransportSequenceNumberHeaderExtension, 536 kRtpTransportSequenceNumberHeaderExtension,
537 kRtpTransportSequenceNumberHeaderExtensionDefaultId)); 537 kRtpTransportSequenceNumberHeaderExtensionDefaultId));
538 } 538 }
539 options_ = GetDefaultEngineOptions(); 539 options_ = GetDefaultEngineOptions();
540 voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
540 } 541 }
541 542
542 WebRtcVoiceEngine::~WebRtcVoiceEngine() { 543 WebRtcVoiceEngine::~WebRtcVoiceEngine() {
543 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 544 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
544 LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine"; 545 LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
545 if (adm_) { 546 if (adm_) {
546 voe_wrapper_.reset(); 547 voe_wrapper_.reset();
547 adm_->Release(); 548 adm_->Release();
548 adm_ = NULL; 549 adm_ = NULL;
549 } 550 }
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2662 } 2663 }
2663 } else { 2664 } else {
2664 LOG(LS_INFO) << "Stopping playout for channel #" << channel; 2665 LOG(LS_INFO) << "Stopping playout for channel #" << channel;
2665 engine()->voe()->base()->StopPlayout(channel); 2666 engine()->voe()->base()->StopPlayout(channel);
2666 } 2667 }
2667 return true; 2668 return true;
2668 } 2669 }
2669 } // namespace cricket 2670 } // namespace cricket
2670 2671
2671 #endif // HAVE_WEBRTC_VOICE 2672 #endif // HAVE_WEBRTC_VOICE
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