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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: Remove incorrect thread check. 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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523 kRtpAudioLevelHeaderExtensionDefaultId)); 523 kRtpAudioLevelHeaderExtensionDefaultId));
524 rtp_header_extensions_.push_back( 524 rtp_header_extensions_.push_back(
525 RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension, 525 RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
526 kRtpAbsoluteSenderTimeHeaderExtensionDefaultId)); 526 kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
527 if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") { 527 if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
528 rtp_header_extensions_.push_back(RtpHeaderExtension( 528 rtp_header_extensions_.push_back(RtpHeaderExtension(
529 kRtpTransportSequenceNumberHeaderExtension, 529 kRtpTransportSequenceNumberHeaderExtension,
530 kRtpTransportSequenceNumberHeaderExtensionDefaultId)); 530 kRtpTransportSequenceNumberHeaderExtensionDefaultId));
531 } 531 }
532 options_ = GetDefaultEngineOptions(); 532 options_ = GetDefaultEngineOptions();
533 voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
533 } 534 }
534 535
535 WebRtcVoiceEngine::~WebRtcVoiceEngine() { 536 WebRtcVoiceEngine::~WebRtcVoiceEngine() {
536 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 537 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
537 LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine"; 538 LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
538 if (adm_) { 539 if (adm_) {
539 voe_wrapper_.reset(); 540 voe_wrapper_.reset();
540 adm_->Release(); 541 adm_->Release();
541 adm_ = NULL; 542 adm_ = NULL;
542 } 543 }
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2637 } 2638 }
2638 } else { 2639 } else {
2639 LOG(LS_INFO) << "Stopping playout for channel #" << channel; 2640 LOG(LS_INFO) << "Stopping playout for channel #" << channel;
2640 engine()->voe()->base()->StopPlayout(channel); 2641 engine()->voe()->base()->StopPlayout(channel);
2641 } 2642 }
2642 return true; 2643 return true;
2643 } 2644 }
2644 } // namespace cricket 2645 } // namespace cricket
2645 2646
2646 #endif // HAVE_WEBRTC_VOICE 2647 #endif // HAVE_WEBRTC_VOICE
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