ZLMediaKit/webrtc/WebRtcPusher.cpp
夏楚 c72cf4cbcc
整理命名空间 (#1409)
* feat: remove using namespace mediakit in header files.

(cherry picked from commit d44aeb339a8a0e1f0455be82b21fe4b1b536299f)

* feat: remove using namespace mediakit in FFmpegSource.h

* feat: remove using namespace mediakit in RtpExt.h

* feat: remove using namespace mediakit in header files.

* feat: remove using namespace std in header files.

* feat: remove using namespace std in header files when zltoolkit remove std in header

* 补充命名空间

* 整理命名空间

* 整理命名空间2

* 修复macos ci

* 修复编译问题

* 修复编译问题2

* 修复编译问题3

Co-authored-by: Johnny <hellojinqiang@gmail.com>
Co-authored-by: Xiaofeng Wang <wasphin@gmail.com>
2022-02-02 20:34:50 +08:00

146 lines
5.2 KiB
C++

/*
* 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 RtspMediaSource::Ptr &src,
const std::shared_ptr<void> &ownership,
const MediaInfo &info) {
WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info), [](WebRtcPusher *ptr) {
ptr->onDestory();
delete ptr;
});
ret->onCreate();
return ret;
}
WebRtcPusher::WebRtcPusher(const EventPoller::Ptr &poller,
const RtspMediaSource::Ptr &src,
const std::shared_ptr<void> &ownership,
const MediaInfo &info) : WebRtcTransportImp(poller) {
_media_info = info;
_push_src = src;
_push_src_ownership = ownership;
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.getSchema() << "/" << sender.getVhost() << "/"
<< sender.getApp() << "/" << sender.getId() << " " << 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));
}
});
return true;
}
int WebRtcPusher::totalReaderCount(MediaSource &sender) {
auto total_count = 0;
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
for (auto &pr : _push_src_sim) {
pr.second->onWrite(rtp, false);
}
} else {
//视频
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);
_push_src_sim_ownership[rid] = src_imp->getOwnership();
src_imp->setSdp(_push_src->getSdp());
src_imp->setProtocolTranslation(_push_src->isRecording(Recorder::type_hls),
_push_src->isRecording(Recorder::type_mp4));
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._vhost << "/"
<< _media_info._app << "/"
<< _media_info._streamid
<< ")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()));
}
}
GET_CONFIG(uint32_t, continue_push_ms, General::kContinuePushMS);
if (_push_src && continue_push_ms) {
//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; });
}
}
void WebRtcPusher::onRtcConfigure(RtcConfigure &configure) const {
WebRtcTransportImp::onRtcConfigure(configure);
//这只是推流
configure.audio.direction = configure.video.direction = RtpDirection::recvonly;
}