ZLMediaKit/webrtc/WebRtcTransport.h
2021-04-07 17:51:47 +08:00

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#pragma once
#include <memory>
#include <string>
#include "DtlsTransport.hpp"
#include "IceServer.hpp"
#include "SrtpSession.hpp"
#include "StunPacket.hpp"
#include "Sdp.h"
#include "Poller/EventPoller.h"
#include "Network/Socket.h"
#include "Rtsp/RtspMediaSourceImp.h"
#include "Rtcp/RtcpContext.h"
using namespace toolkit;
using namespace mediakit;
class WebRtcTransport : public RTC::DtlsTransport::Listener, public RTC::IceServer::Listener {
public:
using Ptr = std::shared_ptr<WebRtcTransport>;
WebRtcTransport(const EventPoller::Ptr &poller);
~WebRtcTransport() override = default;
/**
* 创建对象
*/
virtual void onCreate();
/**
* 销毁对象
*/
virtual void onDestory();
/**
* 创建webrtc answer sdp
* @param offer offer sdp
* @return answer sdp
*/
std::string getAnswerSdp(const string &offer);
/**
* socket收到udp数据
* @param buf 数据指针
* @param len 数据长度
* @param tuple 数据来源
*/
void inputSockData(char *buf, size_t len, RTC::TransportTuple *tuple);
/**
* 发送rtp
* @param buf rtcp内容
* @param len rtcp长度
* @param flush 是否flush socket
* @param pt rtp payload type
*/
void sendRtpPacket(char *buf, size_t len, bool flush, uint8_t pt);
void sendRtcpPacket(char *buf, size_t len, bool flush);
const EventPoller::Ptr& getPoller() const;
protected:
//// dtls相关的回调 ////
void OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) override;
void OnDtlsTransportConnected(const RTC::DtlsTransport *dtlsTransport,
RTC::SrtpSession::CryptoSuite srtpCryptoSuite,
uint8_t *srtpLocalKey,
size_t srtpLocalKeyLen,
uint8_t *srtpRemoteKey,
size_t srtpRemoteKeyLen,
std::string &remoteCert) override;
void OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) override;
void OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) override;
void OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
protected:
//// ice相关的回调 ///
void OnIceServerSendStunPacket(const RTC::IceServer *iceServer, const RTC::StunPacket *packet, RTC::TransportTuple *tuple) override;
void OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) override;
void OnIceServerConnected(const RTC::IceServer *iceServer) override;
void OnIceServerCompleted(const RTC::IceServer *iceServer) override;
void OnIceServerDisconnected(const RTC::IceServer *iceServer) override;
protected:
virtual void onStartWebRTC() = 0;
virtual void onRtcConfigure(RtcConfigure &configure) const {}
virtual void onCheckSdp(SdpType type, RtcSession &sdp);
virtual void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) = 0;
virtual void onRtp(const char *buf, size_t len) = 0;
virtual void onRtcp(const char *buf, size_t len) = 0;
virtual void onShutdown(const SockException &ex) = 0;
protected:
const RtcSession& getSdp(SdpType type) const;
private:
void onSendSockData(const char *buf, size_t len, bool flush = true);
void setRemoteDtlsFingerprint(const RtcSession &remote);
private:
EventPoller::Ptr _poller;
std::shared_ptr<RTC::IceServer> _ice_server;
std::shared_ptr<RTC::DtlsTransport> _dtls_transport;
std::shared_ptr<RTC::SrtpSession> _srtp_session_send;
std::shared_ptr<RTC::SrtpSession> _srtp_session_recv;
RtcSession::Ptr _offer_sdp;
RtcSession::Ptr _answer_sdp;
};
class RtpReceiverImp;
class WebRtcTransportImp : public WebRtcTransport, public std::enable_shared_from_this<WebRtcTransportImp>{
public:
using Ptr = std::shared_ptr<WebRtcTransportImp>;
~WebRtcTransportImp() override;
/**
* 创建WebRTC对象
* @param poller 改对象需要绑定的线程
* @return 对象
*/
static Ptr create(const EventPoller::Ptr &poller);
/**
* 绑定rtsp媒体源
* @param src 媒体源
* @param is_play 是播放还是推流
*/
void attach(const RtspMediaSource::Ptr &src, bool is_play = true);
protected:
void onStartWebRTC() override;
void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) override;
void onCheckSdp(SdpType type, RtcSession &sdp) override;
void onRtcConfigure(RtcConfigure &configure) const override;
void onRtp(const char *buf, size_t len) override;
void onRtcp(const char *buf, size_t len) override;
void onShutdown(const SockException &ex) override;
private:
WebRtcTransportImp(const EventPoller::Ptr &poller);
void onCreate() override;
void onDestory() override;
void onSendRtp(const RtpPacket::Ptr &rtp, bool flush);
SdpAttrCandidate::Ptr getIceCandidate() const;
bool canSendRtp() const;
bool canRecvRtp() const;
class RtpPayloadInfo {
public:
bool is_common_rtp;
const RtcCodecPlan *plan;
const RtcMedia *media;
std::shared_ptr<RtpReceiverImp> receiver;
RtcpContext::Ptr rtcp_context_recv;
RtcpContext::Ptr rtcp_context_send;
};
void onSortedRtp(const RtpPayloadInfo &info,RtpPacket::Ptr rtp);
void onBeforeSortedRtp(const RtpPayloadInfo &info,const RtpPacket::Ptr &rtp);
private:
//保持自我强引用
Ptr _self;
//检测超时的定时器
Timer::Ptr _timer;
//刷新计时器
Ticker _alive_ticker;
//pli rtcp计时器
Ticker _pli_ticker;
//rtc rtp推流的视频ssrc
uint32_t _recv_video_ssrc;
//记录协商的rtp的pt类型
uint8_t _send_rtp_pt[2] = {0, 0};
//复合udp端口接收一切rtp与rtcp
Socket::Ptr _socket;
//推流的rtsp源
RtspMediaSource::Ptr _push_src;
//播放的rtsp源
RtspMediaSource::Ptr _play_src;
//播放rtsp源的reader对象
RtspMediaSource::RingType::RingReader::Ptr _reader;
//根据rtp的pt获取相关信息
unordered_map<uint8_t, RtpPayloadInfo> _rtp_info_pt;
//根据推流端rtp的ssrc获取相关信息
unordered_map<uint32_t, RtpPayloadInfo*> _rtp_info_ssrc;
};