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https://github.com/ZLMediaKit/ZLMediaKit.git
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Merge branch 'master' of github.com:ZLMediaKit/ZLMediaKit into transcode2
# Conflicts: # CMakeLists.txt # conf/config.ini # src/Common/MediaSink.cpp # src/Common/MediaSink.h # src/Common/MediaSource.cpp # src/Common/MultiMediaSourceMuxer.h # src/Common/config.cpp # src/Common/config.h # src/Extension/AAC.cpp # src/Extension/AAC.h # src/Rtsp/RtpCodec.h # src/Rtsp/RtspMuxer.cpp # src/Rtsp/RtspMuxer.h # webrtc/Nack.cpp # webrtc/WebRtcTransport.cpp
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@@ -1,9 +1,9 @@
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/*
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* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
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* Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
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*
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* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
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* This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
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*
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* Use of this source code is governed by MIT license that can be found in the
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* Use of this source code is governed by MIT-like license that can be found in the
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* LICENSE file in the root of the source tree. All contributing project authors
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* may be found in the AUTHORS file in the root of the source tree.
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*/
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@@ -20,9 +20,8 @@ WebRtcPusher::Ptr WebRtcPusher::create(const EventPoller::Ptr &poller,
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const RtspMediaSource::Ptr &src,
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const std::shared_ptr<void> &ownership,
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const MediaInfo &info,
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const ProtocolOption &option,
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bool preferred_tcp) {
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WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option,preferred_tcp), [](WebRtcPusher *ptr) {
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const ProtocolOption &option) {
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WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option), [](WebRtcPusher *ptr) {
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ptr->onDestory();
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delete ptr;
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});
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@@ -34,8 +33,7 @@ WebRtcPusher::WebRtcPusher(const EventPoller::Ptr &poller,
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const RtspMediaSource::Ptr &src,
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const std::shared_ptr<void> &ownership,
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const MediaInfo &info,
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const ProtocolOption &option,
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bool preferred_tcp) : WebRtcTransportImp(poller,preferred_tcp) {
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const ProtocolOption &option) : WebRtcTransportImp(poller) {
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_media_info = info;
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_push_src = src;
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_push_src_ownership = ownership;
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@@ -129,7 +127,7 @@ void WebRtcPusher::onDestory() {
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if (getSession()) {
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WarnL << "RTC推流器(" << _media_info.shortUrl() << ")结束推流,耗时(s):" << duration;
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if (bytes_usage >= iFlowThreshold * 1024) {
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NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, bytes_usage, duration, false, static_cast<SockInfo &>(*getSession()));
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NOTICE_EMIT(BroadcastFlowReportArgs, Broadcast::kBroadcastFlowReport, _media_info, bytes_usage, duration, false, *getSession());
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}
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}
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