mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-22 19:00:01 +08:00
256 lines
8.4 KiB
C++
256 lines
8.4 KiB
C++
/*
|
||
* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
|
||
*
|
||
* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
|
||
*
|
||
* Use of this source code is governed by MIT 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.
|
||
*/
|
||
|
||
#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 type rtp类型
|
||
*/
|
||
void sendRtpPacket(char *buf, size_t len, bool flush, TrackType type);
|
||
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;
|
||
virtual void onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) = 0;
|
||
|
||
protected:
|
||
const RtcSession& getSdp(SdpType type) const;
|
||
RTC::TransportTuple* getSelectedTuple() const;
|
||
void sendRtcpRemb(uint32_t ssrc, size_t bit_rate);
|
||
void sendRtcpPli(uint32_t ssrc);
|
||
|
||
private:
|
||
void onSendSockData(const char *buf, size_t len, bool flush = true);
|
||
void setRemoteDtlsFingerprint(const RtcSession &remote);
|
||
|
||
private:
|
||
uint8_t _srtp_buf[2000];
|
||
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 MediaSourceEvent, public SockInfo, 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, const MediaInfo &info, 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 onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) override;
|
||
|
||
void onShutdown(const SockException &ex) override;
|
||
|
||
///////MediaSourceEvent override///////
|
||
// 关闭
|
||
bool close(MediaSource &sender, bool force) override;
|
||
// 播放总人数
|
||
int totalReaderCount(MediaSource &sender) override;
|
||
// 获取媒体源类型
|
||
MediaOriginType getOriginType(MediaSource &sender) const override;
|
||
// 获取媒体源url或者文件路径
|
||
string getOriginUrl(MediaSource &sender) const override;
|
||
// 获取媒体源客户端相关信息
|
||
std::shared_ptr<SockInfo> getOriginSock(MediaSource &sender) const override;
|
||
|
||
///////SockInfo override///////
|
||
//获取本机ip
|
||
string get_local_ip() override;
|
||
//获取本机端口号
|
||
uint16_t get_local_port() override;
|
||
//获取对方ip
|
||
string get_peer_ip() override;
|
||
//获取对方端口号
|
||
uint16_t get_peer_port() override;
|
||
//获取标识符
|
||
string getIdentifier() const 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;
|
||
const RtcSession& getSdpWithSSRC() 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:
|
||
//用掉的总流量
|
||
uint64_t _bytes_usage = 0;
|
||
//媒体相关元数据
|
||
MediaInfo _media_info;
|
||
//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;
|
||
//记录协商的发送rtp的pt和ssrc
|
||
RtpPayloadInfo* _send_rtp_info[2] = {nullptr, nullptr};
|
||
//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/*pt*/, RtpPayloadInfo> _rtp_info_pt;
|
||
//根据rtcp的ssrc获取相关信息
|
||
unordered_map<uint32_t/*ssrc*/, RtpPayloadInfo*> _rtp_info_ssrc;
|
||
//发送rtp时需要修改rtp ext id
|
||
map<RtpExtType, uint8_t> _rtp_ext_type_to_id;
|
||
//接收rtp时需要修改rtp ext id
|
||
unordered_map<uint8_t, RtpExtType> _rtp_ext_id_to_type;
|
||
};
|
||
|
||
|
||
|
||
|
||
|
||
|
||
|
||
|
||
|
||
|
||
|
||
|
||
|
||
|
||
|
||
|
||
|
||
|