ZLMediaKit/webrtc/WebRtcPusher.cpp

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/*
* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
*
* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
*
* Use of this source code is governed by MIT license that can be found in the
* LICENSE file in the root of the source tree. All contributing project authors
* may be found in the AUTHORS file in the root of the source tree.
*/
#include "WebRtcPusher.h"
using namespace std;
using namespace mediakit;
WebRtcPusher::Ptr WebRtcPusher::create(const EventPoller::Ptr &poller,
const RtspMediaSourceImp::Ptr &src,
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const std::shared_ptr<void> &ownership,
const MediaInfo &info,
const mediakit::ProtocolOption &option) {
WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option), [](WebRtcPusher *ptr) {
ptr->onDestory();
delete ptr;
});
ret->onCreate();
return ret;
}
WebRtcPusher::WebRtcPusher(const EventPoller::Ptr &poller,
const RtspMediaSourceImp::Ptr &src,
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const std::shared_ptr<void> &ownership,
const MediaInfo &info,
const mediakit::ProtocolOption &option) : WebRtcTransportImp(poller) {
_media_info = info;
_push_src = src;
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_push_src_ownership = ownership;
_continue_push_ms = option.continue_push_ms;
CHECK(_push_src);
}
bool WebRtcPusher::close(MediaSource &sender, bool force) {
//此回调在其他线程触发
if (!force && totalReaderCount(sender)) {
return false;
}
string err = StrPrinter << "close media:" << sender.getUrl() << " " << force;
weak_ptr<WebRtcPusher> weak_self = static_pointer_cast<WebRtcPusher>(shared_from_this());
getPoller()->async([weak_self, err]() {
auto strong_self = weak_self.lock();
if (strong_self) {
strong_self->onShutdown(SockException(Err_shutdown, err));
//主动关闭推流,那么不延时注销
strong_self->_push_src = nullptr;
}
});
return true;
}
int WebRtcPusher::totalReaderCount(MediaSource &sender) {
auto total_count = 0;
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for (auto &src : _push_src_sim) {
total_count += src.second->totalReaderCount();
}
return total_count + _push_src->totalReaderCount();
}
MediaOriginType WebRtcPusher::getOriginType(MediaSource &sender) const {
return MediaOriginType::rtc_push;
}
string WebRtcPusher::getOriginUrl(MediaSource &sender) const {
return _media_info._full_url;
}
std::shared_ptr<SockInfo> WebRtcPusher::getOriginSock(MediaSource &sender) const {
return static_pointer_cast<SockInfo>(getSession());
}
void WebRtcPusher::onRecvRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp) {
if (!_simulcast) {
assert(_push_src);
_push_src->onWrite(rtp, false);
return;
}
if (rtp->type == TrackAudio) {
//音频
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for (auto &pr : _push_src_sim) {
pr.second->onWrite(rtp, false);
}
} else {
//视频
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auto &src = _push_src_sim[rid];
if (!src) {
auto stream_id = rid.empty() ? _push_src->getId() : _push_src->getId() + "_" + rid;
auto src_imp = std::make_shared<RtspMediaSourceImp>(_push_src->getVhost(), _push_src->getApp(), stream_id);
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_push_src_sim_ownership[rid] = src_imp->getOwnership();
src_imp->setSdp(_push_src->getSdp());
src_imp->setProtocolOption(_push_src->getProtocolOption());
src_imp->setListener(static_pointer_cast<WebRtcPusher>(shared_from_this()));
src = src_imp;
}
src->onWrite(std::move(rtp), false);
}
}
void WebRtcPusher::onStartWebRTC() {
WebRtcTransportImp::onStartWebRTC();
_simulcast = _answer_sdp->supportSimulcast();
if (canRecvRtp()) {
_push_src->setSdp(_answer_sdp->toRtspSdp());
}
}
void WebRtcPusher::onDestory() {
WebRtcTransportImp::onDestory();
auto duration = getDuration();
auto bytes_usage = getBytesUsage();
//流量统计事件广播
GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold);
if (getSession()) {
WarnL << "RTC推流器("
<< _media_info.shortUrl()
<< ")结束推流,耗时(s):" << duration;
if (bytes_usage >= iFlowThreshold * 1024) {
NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, bytes_usage, duration,
false, static_cast<SockInfo &>(*getSession()));
}
}
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if (_push_src && _continue_push_ms) {
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//取消所有权
_push_src_ownership = nullptr;
//延时10秒注销流
auto push_src = std::move(_push_src);
getPoller()->doDelayTask(_continue_push_ms, [push_src]() { return 0; });
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}
}
void WebRtcPusher::onRtcConfigure(RtcConfigure &configure) const {
WebRtcTransportImp::onRtcConfigure(configure);
//这只是推流
configure.audio.direction = configure.video.direction = RtpDirection::recvonly;
}
float WebRtcPusher::getLossRate(MediaSource &sender,mediakit::TrackType type){
return WebRtcTransportImp::getLossRate(type);
}