mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-30 07:26:53 +08:00
c72cf4cbcc
* 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>
91 lines
3.4 KiB
C++
91 lines
3.4 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 "WebRtcPlayer.h"
|
|
|
|
using namespace std;
|
|
using namespace mediakit;
|
|
|
|
WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
|
|
const RtspMediaSource::Ptr &src,
|
|
const MediaInfo &info) {
|
|
WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info), [](WebRtcPlayer *ptr) {
|
|
ptr->onDestory();
|
|
delete ptr;
|
|
});
|
|
ret->onCreate();
|
|
return ret;
|
|
}
|
|
|
|
WebRtcPlayer::WebRtcPlayer(const EventPoller::Ptr &poller,
|
|
const RtspMediaSource::Ptr &src,
|
|
const MediaInfo &info) : WebRtcTransportImp(poller) {
|
|
_media_info = info;
|
|
_play_src = src;
|
|
CHECK(_play_src);
|
|
}
|
|
|
|
void WebRtcPlayer::onStartWebRTC() {
|
|
CHECK(_play_src);
|
|
WebRtcTransportImp::onStartWebRTC();
|
|
if (canSendRtp()) {
|
|
_play_src->pause(false);
|
|
_reader = _play_src->getRing()->attach(getPoller(), true);
|
|
weak_ptr<WebRtcPlayer> weak_self = static_pointer_cast<WebRtcPlayer>(shared_from_this());
|
|
_reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt) {
|
|
auto strongSelf = weak_self.lock();
|
|
if (!strongSelf) {
|
|
return;
|
|
}
|
|
size_t i = 0;
|
|
pkt->for_each([&](const RtpPacket::Ptr &rtp) {
|
|
//TraceL<<"send track type:"<<rtp->type<<" ts:"<<rtp->getStamp()<<" ntp:"<<rtp->ntp_stamp<<" size:"<<rtp->getPayloadSize()<<" i:"<<i;
|
|
strongSelf->onSendRtp(rtp, ++i == pkt->size());
|
|
});
|
|
});
|
|
_reader->setDetachCB([weak_self]() {
|
|
auto strongSelf = weak_self.lock();
|
|
if (!strongSelf) {
|
|
return;
|
|
}
|
|
strongSelf->onShutdown(SockException(Err_shutdown, "rtsp ring buffer detached"));
|
|
});
|
|
}
|
|
//使用完毕后，释放强引用，这样确保推流器断开后能及时注销媒体
|
|
_play_src = nullptr;
|
|
}
|
|
void WebRtcPlayer::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 (_reader && getSession()) {
|
|
WarnL << "RTC播放器("
|
|
<< _media_info._vhost << "/"
|
|
<< _media_info._app << "/"
|
|
<< _media_info._streamid
|
|
<< ")结束播放,耗时(s):" << duration;
|
|
if (bytes_usage >= iFlowThreshold * 1024) {
|
|
NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, bytes_usage, duration,
|
|
true, static_cast<SockInfo &>(*getSession()));
|
|
}
|
|
}
|
|
}
|
|
|
|
void WebRtcPlayer::onRtcConfigure(RtcConfigure &configure) const {
|
|
CHECK(_play_src);
|
|
WebRtcTransportImp::onRtcConfigure(configure);
|
|
//这是播放
|
|
configure.audio.direction = configure.video.direction = RtpDirection::sendonly;
|
|
configure.setPlayRtspInfo(_play_src->getSdp());
|
|
}
|