mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-22 19:00:01 +08:00
398 lines
13 KiB
C++
398 lines
13 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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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 RtpReceiverImp;
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class NackList {
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public:
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void push_back(RtpPacket::Ptr rtp) {
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auto seq = rtp->getSeq();
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_nack_cache_seq.emplace_back(seq);
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_nack_cache_pkt.emplace(seq, std::move(rtp));
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while (get_cache_ms() > kMaxNackMS) {
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//需要清除部分nack缓存
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pop_front();
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}
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}
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template<typename FUNC>
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void for_each_nack(const FCI_NACK &nack, const FUNC &func) {
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auto seq = nack.getPid();
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for (auto bit : nack.getBitArray()) {
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if (bit) {
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//丢包
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RtpPacket::Ptr *ptr = get_rtp(seq);
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if (ptr) {
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func(*ptr);
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}
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}
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++seq;
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}
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}
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private:
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void pop_front() {
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if (_nack_cache_seq.empty()) {
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return;
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}
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_nack_cache_pkt.erase(_nack_cache_seq.front());
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_nack_cache_seq.pop_front();
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}
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RtpPacket::Ptr *get_rtp(uint16_t seq) {
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auto it = _nack_cache_pkt.find(seq);
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if (it == _nack_cache_pkt.end()) {
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return nullptr;
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}
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return &it->second;
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}
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uint32_t get_cache_ms() {
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if (_nack_cache_seq.size() < 2) {
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return 0;
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}
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uint32_t back = _nack_cache_pkt[_nack_cache_seq.back()]->getStampMS();
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uint32_t front = _nack_cache_pkt[_nack_cache_seq.front()]->getStampMS();
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if (back > front) {
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return back - front;
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}
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//很有可能回环了
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return back + (UINT32_MAX - front);
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}
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private:
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static constexpr uint32_t kMaxNackMS = 10 * 1000;
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deque<uint16_t> _nack_cache_seq;
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unordered_map<uint16_t, RtpPacket::Ptr > _nack_cache_pkt;
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};
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class NackContext {
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public:
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using onNack = function<void(const FCI_NACK &nack)>;
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void received(uint16_t seq) {
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if (!_last_max_seq && _seq.empty()) {
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_last_max_seq = seq - 1;
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}
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_seq.emplace(seq);
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auto max_seq = *_seq.rbegin();
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auto min_seq = *_seq.begin();
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auto diff = max_seq - min_seq;
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if (!diff) {
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return;
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}
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if (diff > UINT32_MAX / 2) {
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//回环
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_seq.clear();
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_last_max_seq = min_seq;
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return;
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}
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if (_seq.size() == diff + 1 && _last_max_seq + 1 == min_seq) {
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//都是连续的seq，未丢包
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_seq.clear();
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_last_max_seq = max_seq;
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} else {
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//seq不连续，有丢包
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if (min_seq == _last_max_seq + 1) {
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//前面部分seq是连续的，未丢包，移除之
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eraseFrontSeq();
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}
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//有丢包，丢包从_last_max_seq开始
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if (max_seq - _last_max_seq > FCI_NACK::kBitSize) {
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vector<bool> vec;
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vec.resize(FCI_NACK::kBitSize);
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for (auto i = 0; i < FCI_NACK::kBitSize; ++i) {
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vec[i] = _seq.find(_last_max_seq + i + 2) == _seq.end();
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}
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doNack(FCI_NACK(_last_max_seq + 1, vec));
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_last_max_seq += FCI_NACK::kBitSize + 1;
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if (_last_max_seq >= max_seq) {
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_seq.clear();
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} else {
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auto it = _seq.emplace_hint(_seq.begin(), _last_max_seq);
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_seq.erase(_seq.begin(), it);
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}
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}
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}
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}
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void setOnNack(onNack cb) {
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_cb = std::move(cb);
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}
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private:
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void doNack(const FCI_NACK &nack) {
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if (_cb) {
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_cb(nack);
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}
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}
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void eraseFrontSeq(){
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//前面部分seq是连续的，未丢包，移除之
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for (auto it = _seq.begin(); it != _seq.end();) {
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if (*it != _last_max_seq + 1) {
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//seq不连续，丢包了
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break;
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}
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_last_max_seq = *it;
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it = _seq.erase(it);
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}
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}
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private:
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onNack _cb;
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set<uint16_t> _seq;
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uint16_t _last_max_seq = 0;
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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 onRtp_l(const char *buf, size_t len, bool rtx);
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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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class RtpPayloadInfo {
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public:
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using Ptr = std::shared_ptr<RtpPayloadInfo>;
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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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NackList nack_list;
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RtcpContext::Ptr rtcp_context_send;
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unordered_map<string/*rid*/, std::pair<uint32_t/*rtp ssrc*/, uint32_t/*rtx ssrc*/> > rid_ssrc;
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unordered_map<uint32_t/*rtx ssrc*/, uint32_t/*rtp ssrc*/> rtx_ssrc_to_rtp_ssrc;
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unordered_map<uint32_t/*simulcast ssrc*/, NackContext> nack_ctx;
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unordered_map<uint32_t/*simulcast ssrc*/, RtcpContext::Ptr> rtcp_context_recv;
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unordered_map<uint32_t/*simulcast ssrc*/, std::shared_ptr<RtpReceiverImp> > receiver;
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};
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void onSortedRtp(RtpPayloadInfo &info, const string &rid, RtpPacket::Ptr rtp);
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void onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack, uint32_t ssrc);
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void changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *header, bool is_recv, bool is_rtx = false, string *rid_ptr = nullptr) const;
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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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RtpPayloadInfo::Ptr _send_rtp_info[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*/,RtpPayloadInfo::Ptr> > _rtp_info_pt;
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//根据rtcp的ssrc获取相关信息
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unordered_map<uint32_t/*ssrc*/, std::pair<bool/*is rtx*/,RtpPayloadInfo::Ptr> > _rtp_info_ssrc;
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//发送rtp时需要修改rtp ext id
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map<RtpExtType, uint8_t> _rtp_ext_type_to_id;
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//接收rtp时需要修改rtp ext id
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unordered_map<uint8_t, RtpExtType> _rtp_ext_id_to_type;
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}; |