mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-29 22:55:52 +08:00
240 lines
8.4 KiB
C++
240 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"
|
||
#include "Rtcp/RtcpFCI.h"
|
||
#include "Nack.h"
|
||
#include "Network/Session.h"
|
||
#include "TwccContext.h"
|
||
|
||
using namespace toolkit;
|
||
using namespace mediakit;
|
||
|
||
//RTC配置项目
|
||
namespace RTC {
|
||
extern const string kPort;
|
||
extern const string kTimeOutSec;
|
||
}//namespace RTC
|
||
|
||
class WebRtcTransport : public RTC::DtlsTransport::Listener, public RTC::IceServer::Listener, public std::enable_shared_from_this<WebRtcTransport> {
|
||
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, int len, RTC::TransportTuple *tuple);
|
||
|
||
/**
|
||
* 发送rtp
|
||
* @param buf rtcp内容
|
||
* @param len rtcp长度
|
||
* @param flush 是否flush socket
|
||
* @param ctx 用户指针
|
||
*/
|
||
void sendRtpPacket(const char *buf, int len, bool flush, void *ctx = nullptr);
|
||
void sendRtcpPacket(const char *buf, int len, bool flush, void *ctx = nullptr);
|
||
|
||
const EventPoller::Ptr& getPoller() const;
|
||
const string& getIdentifier() 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) = 0;
|
||
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, uint64_t stamp_ms) = 0;
|
||
virtual void onRtcp(const char *buf, size_t len) = 0;
|
||
virtual void onShutdown(const SockException &ex) = 0;
|
||
virtual void onBeforeEncryptRtp(const char *buf, int &len, void *ctx) = 0;
|
||
virtual void onBeforeEncryptRtcp(const char *buf, int &len, void *ctx) = 0;
|
||
|
||
protected:
|
||
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);
|
||
|
||
protected:
|
||
RtcSession::Ptr _offer_sdp;
|
||
RtcSession::Ptr _answer_sdp;
|
||
|
||
private:
|
||
uint8_t _srtp_buf[2000];
|
||
string _identifier;
|
||
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;
|
||
Ticker _ticker;
|
||
};
|
||
|
||
class RtpChannel;
|
||
class MediaTrack {
|
||
public:
|
||
using Ptr = std::shared_ptr<MediaTrack>;
|
||
const RtcCodecPlan *plan_rtp;
|
||
const RtcCodecPlan *plan_rtx;
|
||
uint32_t offer_ssrc_rtp = 0;
|
||
uint32_t offer_ssrc_rtx = 0;
|
||
uint32_t answer_ssrc_rtp = 0;
|
||
uint32_t answer_ssrc_rtx = 0;
|
||
const RtcMedia *media;
|
||
RtpExtContext::Ptr rtp_ext_ctx;
|
||
|
||
//for send rtp
|
||
NackList nack_list;
|
||
RtcpContext::Ptr rtcp_context_send;
|
||
|
||
//for recv rtp
|
||
unordered_map<string/*rid*/, std::shared_ptr<RtpChannel> > rtp_channel;
|
||
std::shared_ptr<RtpChannel> getRtpChannel(uint32_t ssrc) const;
|
||
};
|
||
|
||
class WebRtcTransportImp : public WebRtcTransport {
|
||
public:
|
||
using Ptr = std::shared_ptr<WebRtcTransportImp>;
|
||
~WebRtcTransportImp() override;
|
||
|
||
void setSession(Session::Ptr session);
|
||
const Session::Ptr& getSession() const;
|
||
uint64_t getBytesUsage() const;
|
||
uint64_t getDuration() const;
|
||
bool canSendRtp() const;
|
||
bool canRecvRtp() const;
|
||
void onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx = false);
|
||
|
||
protected:
|
||
WebRtcTransportImp(const EventPoller::Ptr &poller);
|
||
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, uint64_t stamp_ms) override;
|
||
void onRtcp(const char *buf, size_t len) override;
|
||
void onBeforeEncryptRtp(const char *buf, int &len, void *ctx) override;
|
||
void onBeforeEncryptRtcp(const char *buf, int &len, void *ctx) override {};
|
||
void onCreate() override;
|
||
void onDestory() override;
|
||
void onShutdown(const SockException &ex) override;
|
||
virtual void onRecvRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp) = 0;
|
||
void updateTicker();
|
||
|
||
private:
|
||
SdpAttrCandidate::Ptr getIceCandidate() const;
|
||
void onSortedRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp);
|
||
void onSendNack(MediaTrack &track, const FCI_NACK &nack, uint32_t ssrc);
|
||
void onSendTwcc(uint32_t ssrc, const string &twcc_fci);
|
||
void createRtpChannel(const string &rid, uint32_t ssrc, MediaTrack &track);
|
||
void registerSelf();
|
||
void unregisterSelf();
|
||
void unrefSelf();
|
||
void onCheckAnswer(RtcSession &sdp);
|
||
|
||
private:
|
||
uint16_t _rtx_seq[2] = {0, 0};
|
||
//用掉的总流量
|
||
uint64_t _bytes_usage = 0;
|
||
//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;
|
||
//udp session
|
||
Session::Ptr _session;
|
||
//twcc rtcp发送上下文对象
|
||
TwccContext _twcc_ctx;
|
||
//根据发送rtp的track类型获取相关信息
|
||
MediaTrack::Ptr _type_to_track[2];
|
||
//根据rtcp的ssrc获取相关信息，收发rtp和rtx的ssrc都会记录
|
||
unordered_map<uint32_t/*ssrc*/, MediaTrack::Ptr> _ssrc_to_track;
|
||
//根据接收rtp的pt获取相关信息
|
||
unordered_map<uint8_t/*pt*/, std::pair<bool/*is rtx*/,MediaTrack::Ptr> > _pt_to_track;
|
||
};
|
||
|
||
class WebRtcTransportManager {
|
||
public:
|
||
static WebRtcTransportManager &Instance();
|
||
void addItem(string key, const WebRtcTransportImp::Ptr &ptr);
|
||
void removeItem(string key);
|
||
WebRtcTransportImp::Ptr getItem(const string &key);
|
||
|
||
private:
|
||
WebRtcTransportManager() = default;
|
||
mutable mutex _mtx;
|
||
unordered_map<string, weak_ptr<WebRtcTransportImp> > _map;
|
||
}; |