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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: Magnus' 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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517 kRtpAudioLevelHeaderExtensionDefaultId)); 517 kRtpAudioLevelHeaderExtensionDefaultId));
518 rtp_header_extensions_.push_back( 518 rtp_header_extensions_.push_back(
519 RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension, 519 RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
520 kRtpAbsoluteSenderTimeHeaderExtensionDefaultId)); 520 kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
521 if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") { 521 if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
522 rtp_header_extensions_.push_back(RtpHeaderExtension( 522 rtp_header_extensions_.push_back(RtpHeaderExtension(
523 kRtpTransportSequenceNumberHeaderExtension, 523 kRtpTransportSequenceNumberHeaderExtension,
524 kRtpTransportSequenceNumberHeaderExtensionDefaultId)); 524 kRtpTransportSequenceNumberHeaderExtensionDefaultId));
525 } 525 }
526 options_ = GetDefaultEngineOptions(); 526 options_ = GetDefaultEngineOptions();
527 voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
the sun 2015/12/03 11:10:28 Remove
stefan-webrtc 2015/12/04 10:31:42 Done.
527 } 528 }
528 529
529 WebRtcVoiceEngine::~WebRtcVoiceEngine() { 530 WebRtcVoiceEngine::~WebRtcVoiceEngine() {
530 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 531 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
531 LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine"; 532 LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
532 if (adm_) { 533 if (adm_) {
533 voe_wrapper_.reset(); 534 voe_wrapper_.reset();
534 adm_->Release(); 535 adm_->Release();
535 adm_ = NULL; 536 adm_ = NULL;
536 } 537 }
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2651 } 2652 }
2652 } else { 2653 } else {
2653 LOG(LS_INFO) << "Stopping playout for channel #" << channel; 2654 LOG(LS_INFO) << "Stopping playout for channel #" << channel;
2654 engine()->voe()->base()->StopPlayout(channel); 2655 engine()->voe()->base()->StopPlayout(channel);
2655 } 2656 }
2656 return true; 2657 return true;
2657 } 2658 }
2658 } // namespace cricket 2659 } // namespace cricket
2659 2660
2660 #endif // HAVE_WEBRTC_VOICE 2661 #endif // HAVE_WEBRTC_VOICE
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