ZLMediaKit/webrtc/WebRtcPlayer.cpp
johzzy 029813402d
feat: update negotiateSdp and WebRtcArgs (#3371)
- update negotiateSdp
- update HttpAllArgs and alias
- update onRtcConfigure
- define setWebRtcArgs, handle set_webrtc_cands and setLocalIp

---------

Co-authored-by: xiongziliang <771730766@qq.com>
Co-authored-by: KkemChen <kkemchen@qq.com>
2024-03-23 22:46:30 +08:00

114 lines
4.1 KiB
C++

/*
* Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
*
* This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
*
* Use of this source code is governed by MIT-like 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"
#include "Common/config.h"
using namespace std;
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(src);
}
void WebRtcPlayer::onStartWebRTC() {
auto playSrc = _play_src.lock();
if(!playSrc){
onShutdown(SockException(Err_shutdown, "rtsp media source was shutdown"));
return ;
}
WebRtcTransportImp::onStartWebRTC();
if (canSendRtp()) {
playSrc->pause(false);
_reader = playSrc->getRing()->attach(getPoller(), true);
weak_ptr<WebRtcPlayer> weak_self = static_pointer_cast<WebRtcPlayer>(shared_from_this());
weak_ptr<Session> weak_session = static_pointer_cast<Session>(getSession());
_reader->setGetInfoCB([weak_session]() {
Any ret;
ret.set(static_pointer_cast<SockInfo>(weak_session.lock()));
return ret;
});
_reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt) {
auto strong_self = weak_self.lock();
if (!strong_self) {
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;
strong_self->onSendRtp(rtp, ++i == pkt->size());
});
});
_reader->setDetachCB([weak_self]() {
auto strong_self = weak_self.lock();
if (!strong_self) {
return;
}
strong_self->onShutdown(SockException(Err_shutdown, "rtsp ring buffer detached"));
});
_reader->setMessageCB([weak_self] (const toolkit::Any &data) {
auto strong_self = weak_self.lock();
if (!strong_self) {
return;
}
if (data.is<Buffer>()) {
auto &buffer = data.get<Buffer>();
// PPID 51: 文本string
// PPID 53: 二进制
strong_self->sendDatachannel(0, 51, buffer.data(), buffer.size());
} else {
WarnL << "Send unknown message type to webrtc player: " << data.type_name();
}
});
}
}
void WebRtcPlayer::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.shortUrl() << ")结束播放,耗时(s):" << duration;
if (bytes_usage >= iFlowThreshold * 1024) {
NOTICE_EMIT(BroadcastFlowReportArgs, Broadcast::kBroadcastFlowReport, _media_info, bytes_usage, duration, true, *getSession());
}
}
WebRtcTransportImp::onDestory();
}
void WebRtcPlayer::onRtcConfigure(RtcConfigure &configure) const {
auto playSrc = _play_src.lock();
if(!playSrc){
return ;
}
WebRtcTransportImp::onRtcConfigure(configure);
//这是播放
configure.audio.direction = configure.video.direction = RtpDirection::sendonly;
configure.setPlayRtspInfo(playSrc->getSdp());
}
}// namespace mediakit