mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-29 22:55:52 +08:00
250 lines
8.9 KiB
C++
250 lines
8.9 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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#pragma once
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#include <memory>
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#include <string>
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#include "DtlsTransport.hpp"
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#include "IceServer.hpp"
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#include "SrtpSession.hpp"
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#include "StunPacket.hpp"
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#include "Sdp.h"
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#include "Poller/EventPoller.h"
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#include "Network/Socket.h"
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#include "Rtsp/RtspMediaSourceImp.h"
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#include "Rtcp/RtcpContext.h"
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#include "Rtcp/RtcpFCI.h"
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#include "Nack.h"
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using namespace toolkit;
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using namespace mediakit;
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class WebRtcTransport : public RTC::DtlsTransport::Listener, public RTC::IceServer::Listener {
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public:
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using Ptr = std::shared_ptr<WebRtcTransport>;
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WebRtcTransport(const EventPoller::Ptr &poller);
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~WebRtcTransport() override = default;
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/**
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* 创建对象
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*/
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virtual void onCreate();
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/**
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* 销毁对象
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*/
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virtual void onDestory();
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/**
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* 创建webrtc answer sdp
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* @param offer offer sdp
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* @return answer sdp
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*/
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std::string getAnswerSdp(const string &offer);
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/**
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* socket收到udp数据
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* @param buf 数据指针
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* @param len 数据长度
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* @param tuple 数据来源
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*/
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void inputSockData(char *buf, size_t len, RTC::TransportTuple *tuple);
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/**
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* 发送rtp
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* @param buf rtcp内容
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* @param len rtcp长度
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* @param flush 是否flush socket
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* @param ctx 用户指针
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*/
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void sendRtpPacket(const char *buf, size_t len, bool flush, void *ctx = nullptr);
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void sendRtcpPacket(const char *buf, size_t len, bool flush, void *ctx = nullptr);
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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,
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RTC::SrtpSession::CryptoSuite srtpCryptoSuite,
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uint8_t *srtpLocalKey,
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size_t srtpLocalKeyLen,
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uint8_t *srtpRemoteKey,
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size_t srtpRemoteKeyLen,
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std::string &remoteCert) override;
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void OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) override;
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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;
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void OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) override;
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void OnIceServerConnected(const RTC::IceServer *iceServer) override;
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void OnIceServerCompleted(const RTC::IceServer *iceServer) override;
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void OnIceServerDisconnected(const RTC::IceServer *iceServer) override;
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protected:
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virtual void onStartWebRTC() = 0;
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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;
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virtual void onRtp(const char *buf, size_t len) = 0;
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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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virtual void onBeforeEncryptRtp(const char *buf, size_t &len, void *ctx) = 0;
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virtual void onBeforeEncryptRtcp(const char *buf, size_t &len, void *ctx) = 0;
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protected:
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const RtcSession& getSdp(SdpType type) const;
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RTC::TransportTuple* getSelectedTuple() const;
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void sendRtcpRemb(uint32_t ssrc, size_t bit_rate);
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void sendRtcpPli(uint32_t ssrc);
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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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uint8_t _srtp_buf[2000];
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EventPoller::Ptr _poller;
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std::shared_ptr<RTC::IceServer> _ice_server;
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std::shared_ptr<RTC::DtlsTransport> _dtls_transport;
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std::shared_ptr<RTC::SrtpSession> _srtp_session_send;
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std::shared_ptr<RTC::SrtpSession> _srtp_session_recv;
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RtcSession::Ptr _offer_sdp;
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RtcSession::Ptr _answer_sdp;
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};
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class RtpChannel;
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class MediaTrack {
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public:
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using Ptr = std::shared_ptr<MediaTrack>;
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const RtcCodecPlan *plan_rtp;
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const RtcCodecPlan *plan_rtx;
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uint32_t offer_ssrc_rtp = 0;
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uint32_t offer_ssrc_rtx = 0;
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uint32_t answer_ssrc_rtp = 0;
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uint32_t answer_ssrc_rtx = 0;
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const RtcMedia *media;
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RtpExtContext::Ptr rtp_ext_ctx;
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//for send rtp
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NackList nack_list;
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RtcpContext::Ptr rtcp_context_send;
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//for recv rtp
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unordered_map<string/*rid*/, std::shared_ptr<RtpChannel> > rtp_channel;
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std::shared_ptr<RtpChannel> getRtpChannel(uint32_t ssrc) const;
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};
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class WebRtcTransportImp : public WebRtcTransport, public MediaSourceEvent, public SockInfo, public std::enable_shared_from_this<WebRtcTransportImp>{
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public:
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using Ptr = std::shared_ptr<WebRtcTransportImp>;
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~WebRtcTransportImp() override;
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/**
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* 创建WebRTC对象
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* @param poller 改对象需要绑定的线程
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* @return 对象
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*/
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static Ptr create(const EventPoller::Ptr &poller);
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/**
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* 绑定rtsp媒体源
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* @param src 媒体源
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* @param is_play 是播放还是推流
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*/
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void attach(const RtspMediaSource::Ptr &src, const MediaInfo &info, 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;
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void onRtcp(const char *buf, size_t len) override;
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void onBeforeEncryptRtp(const char *buf, size_t &len, void *ctx) override;
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void onBeforeEncryptRtcp(const char *buf, size_t &len, void *ctx) override {};
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void onShutdown(const SockException &ex) override;
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///////MediaSourceEvent override///////
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// 关闭
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bool close(MediaSource &sender, bool force) override;
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// 播放总人数
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int totalReaderCount(MediaSource &sender) override;
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// 获取媒体源类型
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MediaOriginType getOriginType(MediaSource &sender) const override;
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// 获取媒体源url或者文件路径
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string getOriginUrl(MediaSource &sender) const override;
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// 获取媒体源客户端相关信息
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std::shared_ptr<SockInfo> getOriginSock(MediaSource &sender) const override;
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///////SockInfo override///////
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//获取本机ip
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string get_local_ip() override;
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//获取本机端口号
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uint16_t get_local_port() override;
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//获取对方ip
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string get_peer_ip() override;
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//获取对方端口号
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uint16_t get_peer_port() override;
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//获取标识符
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string getIdentifier() const 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, bool rtx = false);
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SdpAttrCandidate::Ptr getIceCandidate() const;
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bool canSendRtp() const;
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bool canRecvRtp() const;
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void onSortedRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp);
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void onSendNack(MediaTrack &track, const FCI_NACK &nack, uint32_t ssrc);
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void createRtpChannel(const string &rid, uint32_t ssrc, const MediaTrack::Ptr &track);
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private:
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uint16_t _rtx_seq[2] = {0, 0};
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//用掉的总流量
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uint64_t _bytes_usage = 0;
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//媒体相关元数据
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MediaInfo _media_info;
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//保持自我强引用
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Ptr _self;
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//检测超时的定时器
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Timer::Ptr _timer;
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//刷新计时器
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Ticker _alive_ticker;
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//pli rtcp计时器
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Ticker _pli_ticker;
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//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
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RtspMediaSource::Ptr _push_src;
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unordered_map<string/*rid*/, RtspMediaSource::Ptr> _push_src_simulcast;
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//播放的rtsp源
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RtspMediaSource::Ptr _play_src;
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//播放rtsp源的reader对象
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RtspMediaSource::RingType::RingReader::Ptr _reader;
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//根据发送rtp的track类型获取相关信息
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MediaTrack::Ptr _type_to_track[2];
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//根据接收rtp的pt获取相关信息
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unordered_map<uint8_t/*pt*/, std::pair<bool/*is rtx*/,MediaTrack::Ptr> > _pt_to_track;
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//根据rtcp的ssrc获取相关信息，收发rtp和rtx的ssrc都会记录
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unordered_map<uint32_t/*ssrc*/, MediaTrack::Ptr> _ssrc_to_track;
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}; |