ZLMediaKit/webrtc/WebRtcTransport.h

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#pragma once
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#include <memory>
#include <string>
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#include "DtlsTransport.hpp"
#include "IceServer.hpp"
#include "SrtpSession.hpp"
#include "StunPacket.hpp"
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#include "Sdp.h"
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#include "Poller/EventPoller.h"
#include "Network/Socket.h"
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#include "Rtsp/RtspMediaSourceImp.h"
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#include "Rtcp/RtcpContext.h"
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using namespace toolkit;
using namespace mediakit;
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class WebRtcTransport : public RTC::DtlsTransport::Listener, public RTC::IceServer::Listener {
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public:
using Ptr = std::shared_ptr<WebRtcTransport>;
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WebRtcTransport(const EventPoller::Ptr &poller);
~WebRtcTransport() override = default;
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/**
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*
*/
virtual void onCreate();
/**
*
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*/
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virtual void onDestory();
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/**
* webrtc answer sdp
* @param offer offer sdp
* @return answer sdp
*/
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std::string getAnswerSdp(const string &offer);
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/**
* socket收到udp数据
* @param buf
* @param len
* @param tuple
*/
void inputSockData(char *buf, size_t len, RTC::TransportTuple *tuple);
/**
* rtp
* @param buf rtcp内容
* @param len rtcp长度
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* @param flush flush socket
* @param pt rtp payload type
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*/
void sendRtpPacket(char *buf, size_t len, bool flush, uint8_t pt);
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void sendRtcpPacket(char *buf, size_t len, bool flush);
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const EventPoller::Ptr& getPoller() const;
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protected:
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//// dtls相关的回调 ////
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void OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) override;
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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;
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void OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) override;
void OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) override;
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void OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
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void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
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protected:
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//// ice相关的回调 ///
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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;
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protected:
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virtual void onStartWebRTC() = 0;
virtual void onRtcConfigure(RtcConfigure &configure) const {}
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virtual void onCheckSdp(SdpType type, RtcSession &sdp);
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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;
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virtual void onShutdown(const SockException &ex) = 0;
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protected:
const RtcSession& getSdp(SdpType type) const;
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private:
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void onSendSockData(const char *buf, size_t len, bool flush = true);
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void setRemoteDtlsFingerprint(const RtcSession &remote);
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private:
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EventPoller::Ptr _poller;
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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;
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RtcSession::Ptr _offer_sdp;
RtcSession::Ptr _answer_sdp;
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};
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class RtpReceiverImp;
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class WebRtcTransportImp : public WebRtcTransport, public std::enable_shared_from_this<WebRtcTransportImp>{
public:
using Ptr = std::shared_ptr<WebRtcTransportImp>;
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~WebRtcTransportImp() override;
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/**
* WebRTC对象
* @param poller 线
* @return
*/
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static Ptr create(const EventPoller::Ptr &poller);
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/**
* rtsp媒体源
* @param src
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* @param is_play
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*/
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void attach(const RtspMediaSource::Ptr &src, bool is_play = true);
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protected:
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void onStartWebRTC() override;
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void onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush = true) override;
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void onCheckSdp(SdpType type, RtcSession &sdp) override;
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void onRtcConfigure(RtcConfigure &configure) const override;
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void onRtp(const char *buf, size_t len) override;
void onRtcp(const char *buf, size_t len) override;
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void onShutdown(const SockException &ex) override;
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private:
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WebRtcTransportImp(const EventPoller::Ptr &poller);
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void onCreate() override;
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void onDestory() override;
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void onSendRtp(const RtpPacket::Ptr &rtp, bool flush);
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SdpAttrCandidate::Ptr getIceCandidate() const;
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bool canSendRtp() const;
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bool canRecvRtp() const;
class RtpPayloadInfo {
public:
bool is_common_rtp;
const RtcCodecPlan *plan;
const RtcMedia *media;
std::shared_ptr<RtpReceiverImp> receiver;
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RtcpContext::Ptr rtcp_context_recv;
RtcpContext::Ptr rtcp_context_send;
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};
void onSortedRtp(const RtpPayloadInfo &info,RtpPacket::Ptr rtp);
void onBeforeSortedRtp(const RtpPayloadInfo &info,const RtpPacket::Ptr &rtp);
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private:
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//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;
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//pli rtcp计时器
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Ticker _pli_ticker;
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//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
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Socket::Ptr _socket;
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//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
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RtspMediaSource::RingType::RingReader::Ptr _reader;
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//根据rtp的pt获取相关信息
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unordered_map<uint8_t, RtpPayloadInfo> _rtp_info_pt;
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//根据推流端rtp的ssrc获取相关信息
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unordered_map<uint32_t, RtpPayloadInfo*> _rtp_info_ssrc;
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};