mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-22 19:00:01 +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++
/*
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* Copyright (c) 2016 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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*
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* Use of this source code is governed by MIT 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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#include "WebRtcPlayer.h"
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using namespace std;
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using namespace mediakit;
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WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
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const RtspMediaSource::Ptr &src,
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const MediaInfo &info) {
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WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info), [](WebRtcPlayer *ptr) {
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ptr->onDestory();
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delete ptr;
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});
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ret->onCreate();
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return ret;
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}
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WebRtcPlayer::WebRtcPlayer(const EventPoller::Ptr &poller,
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const RtspMediaSource::Ptr &src,
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const MediaInfo &info) : WebRtcTransportImp(poller) {
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_media_info = info;
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_play_src = src;
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CHECK(_play_src);
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}
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void WebRtcPlayer::onStartWebRTC() {
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CHECK(_play_src);
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WebRtcTransportImp::onStartWebRTC();
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if (canSendRtp()) {
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_play_src->pause(false);
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_reader = _play_src->getRing()->attach(getPoller(), true);
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weak_ptr<WebRtcPlayer> weak_self = static_pointer_cast<WebRtcPlayer>(shared_from_this());
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_reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt) {
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auto strongSelf = weak_self.lock();
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if (!strongSelf) {
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return;
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}
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size_t i = 0;
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pkt->for_each([&](const RtpPacket::Ptr &rtp) {
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//TraceL<<"send track type:"<<rtp->type<<" ts:"<<rtp->getStamp()<<" ntp:"<<rtp->ntp_stamp<<" size:"<<rtp->getPayloadSize()<<" i:"<<i;
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strongSelf->onSendRtp(rtp, ++i == pkt->size());
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});
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});
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_reader->setDetachCB([weak_self]() {
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auto strongSelf = weak_self.lock();
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if (!strongSelf) {
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return;
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}
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strongSelf->onShutdown(SockException(Err_shutdown, "rtsp ring buffer detached"));
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});
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}
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//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
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_play_src = nullptr;
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}
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void WebRtcPlayer::onDestory() {
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WebRtcTransportImp::onDestory();
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auto duration = getDuration();
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auto bytes_usage = getBytesUsage();
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//流量统计事件广播
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GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold);
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if (_reader && getSession()) {
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WarnL << "RTC播放器("
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<< _media_info._vhost << "/"
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<< _media_info._app << "/"
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<< _media_info._streamid
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<< ")结束播放,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,
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true, static_cast<SockInfo &>(*getSession()));
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}
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}
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}
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void WebRtcPlayer::onRtcConfigure(RtcConfigure &configure) const {
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CHECK(_play_src);
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WebRtcTransportImp::onRtcConfigure(configure);
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//这是播放
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configure.audio.direction = configure.video.direction = RtpDirection::sendonly;
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configure.setPlayRtspInfo(_play_src->getSdp());
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}
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