ZLMediaKit/webrtc/WebRtcTransport.cpp

/*
 * 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.
 */

#include "WebRtcTransport.h"
#include <iostream>
#include "RtpExt.h"
#include "Rtcp/Rtcp.h"
#include "Rtcp/RtcpFCI.h"
#include "Rtsp/RtpReceiver.h"

#define RTP_SSRC_OFFSET 1
#define RTX_SSRC_OFFSET 2
#define RTP_CNAME "zlmediakit-rtp"
#define RTP_LABEL "zlmediakit-label"
#define RTP_MSLABEL "zlmediakit-mslabel"
#define RTP_MSID RTP_MSLABEL " " RTP_LABEL

using namespace std;
using namespace mediakit;

//RTC配置项目
namespace RTC {
#define RTC_FIELD "rtc."
//rtp和rtcp接受超时时间
const string kTimeOutSec = RTC_FIELD"timeoutSec";
//服务器外网ip
const string kExternIP = RTC_FIELD"externIP";
//设置remb比特率，非0时关闭twcc并开启remb。该设置在rtc推流时有效，可以控制推流画质
const string kRembBitRate = RTC_FIELD"rembBitRate";
//webrtc单端口udp服务器
const string kPort =  RTC_FIELD"port";

static onceToken token([]() {
    mINI::Instance()[kTimeOutSec] = 15;
    mINI::Instance()[kExternIP] = "";
    mINI::Instance()[kRembBitRate] = 0;
    mINI::Instance()[kPort] = 8000;
});

}//namespace RTC

static atomic<uint64_t> s_key{0};

WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) {
    _poller = poller;
    _identifier = "zlm_" + to_string(++s_key);
    _packet_pool.setSize(64);
}

void WebRtcTransport::onCreate(){
    _dtls_transport = std::make_shared<RTC::DtlsTransport>(_poller, this);
    _ice_server = std::make_shared<RTC::IceServer>(this, _identifier, makeRandStr(24));
}

void WebRtcTransport::onDestory(){
    _dtls_transport = nullptr;
    _ice_server = nullptr;
}

const EventPoller::Ptr& WebRtcTransport::getPoller() const{
    return _poller;
}

const string &WebRtcTransport::getIdentifier() const {
    return _identifier;
}

//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////

void WebRtcTransport::OnIceServerSendStunPacket(const RTC::IceServer *iceServer, const RTC::StunPacket *packet, RTC::TransportTuple *tuple) {
    sendSockData((char *) packet->GetData(), packet->GetSize(), tuple);
}

void WebRtcTransport::OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) {
    InfoL;
}

void WebRtcTransport::OnIceServerConnected(const RTC::IceServer *iceServer) {
    InfoL;
}

void WebRtcTransport::OnIceServerCompleted(const RTC::IceServer *iceServer) {
    InfoL;
    if (_answer_sdp->media[0].role == DtlsRole::passive) {
        _dtls_transport->Run(RTC::DtlsTransport::Role::SERVER);
    } else {
        _dtls_transport->Run(RTC::DtlsTransport::Role::CLIENT);
    }
}

void WebRtcTransport::OnIceServerDisconnected(const RTC::IceServer *iceServer) {
    InfoL;
}

//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////

void WebRtcTransport::OnDtlsTransportConnected(
        const RTC::DtlsTransport *dtlsTransport,
        RTC::SrtpSession::CryptoSuite srtpCryptoSuite,
        uint8_t *srtpLocalKey,
        size_t srtpLocalKeyLen,
        uint8_t *srtpRemoteKey,
        size_t srtpRemoteKeyLen,
        std::string &remoteCert) {
    InfoL;
    _srtp_session_send = std::make_shared<RTC::SrtpSession>(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpLocalKey, srtpLocalKeyLen);
    _srtp_session_recv = std::make_shared<RTC::SrtpSession>(RTC::SrtpSession::Type::INBOUND, srtpCryptoSuite, srtpRemoteKey, srtpRemoteKeyLen);
    onStartWebRTC();
}

void WebRtcTransport::OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) {
    sendSockData((char *)data, len, nullptr);
}

void WebRtcTransport::OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) {
    InfoL;
}

void WebRtcTransport::OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) {
    InfoL;
    onShutdown(SockException(Err_shutdown, "dtls transport failed"));
}

void WebRtcTransport::OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) {
    InfoL;
    onShutdown(SockException(Err_shutdown, "dtls close notify received"));
}

void WebRtcTransport::OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) {
    InfoL << hexdump(data, len);
}
//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////

void WebRtcTransport::sendSockData(const char *buf, size_t len, RTC::TransportTuple *tuple){
    auto pkt = _packet_pool.obtain2();
    pkt->assign(buf, len);
    onSendSockData(std::move(pkt), true, tuple ? tuple : _ice_server->GetSelectedTuple());
}

RTC::TransportTuple* WebRtcTransport::getSelectedTuple() const{
    return  _ice_server->GetSelectedTuple();
}

void WebRtcTransport::sendRtcpRemb(uint32_t ssrc, size_t bit_rate) {
    auto remb = FCI_REMB::create({ssrc}, (uint32_t)bit_rate);
    auto fb = RtcpFB::create(PSFBType::RTCP_PSFB_REMB, remb.data(), remb.size());
    fb->ssrc = htonl(0);
    fb->ssrc_media = htonl(ssrc);
    sendRtcpPacket((char *) fb.get(), fb->getSize(), true);
}

void WebRtcTransport::sendRtcpPli(uint32_t ssrc) {
    auto pli = RtcpFB::create(PSFBType::RTCP_PSFB_PLI);
    pli->ssrc = htonl(0);
    pli->ssrc_media = htonl(ssrc);
    sendRtcpPacket((char *) pli.get(), pli->getSize(), true);
}

string getFingerprint(const string &algorithm_str, const std::shared_ptr<RTC::DtlsTransport> &transport){
    auto algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm(algorithm_str);
    for (auto &finger_prints : transport->GetLocalFingerprints()) {
        if (finger_prints.algorithm == algorithm) {
            return finger_prints.value;
        }
    }
    throw std::invalid_argument(StrPrinter << "不支持的加密算法:" << algorithm_str);
}

void WebRtcTransport::setRemoteDtlsFingerprint(const RtcSession &remote){
    //设置远端dtls签名
    RTC::DtlsTransport::Fingerprint remote_fingerprint;
    remote_fingerprint.algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm(_offer_sdp->media[0].fingerprint.algorithm);
    remote_fingerprint.value = _offer_sdp->media[0].fingerprint.hash;
    _dtls_transport->SetRemoteFingerprint(remote_fingerprint);
}

void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const {
    //开启remb后关闭twcc，因为开启twcc后remb无效
    GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate);
    configure.enableTWCC(!remb_bit_rate);
}

std::string WebRtcTransport::getAnswerSdp(const string &offer){
    try {
        //// 解析offer sdp ////
        _offer_sdp = std::make_shared<RtcSession>();
        _offer_sdp->loadFrom(offer);
        onCheckSdp(SdpType::offer, *_offer_sdp);
        _offer_sdp->checkValid();
        setRemoteDtlsFingerprint(*_offer_sdp);

        //// sdp 配置 ////
        SdpAttrFingerprint fingerprint;
        fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm;
        fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
        RtcConfigure configure;
        configure.setDefaultSetting(_ice_server->GetUsernameFragment(), _ice_server->GetPassword(),
                                    RtpDirection::sendrecv, fingerprint);
        onRtcConfigure(configure);

        //// 生成answer sdp ////
        _answer_sdp = configure.createAnswer(*_offer_sdp);
        onCheckSdp(SdpType::answer, *_answer_sdp);
        _answer_sdp->checkValid();
        return _answer_sdp->toString();
    } catch (exception &ex) {
        onShutdown(SockException(Err_shutdown, ex.what()));
        throw;
    }
}

static bool is_dtls(char *buf) {
    return ((*buf > 19) && (*buf < 64));
}

static bool is_rtp(char *buf) {
    RtpHeader *header = (RtpHeader *) buf;
    return ((header->pt < 64) || (header->pt >= 96));
}

static bool is_rtcp(char *buf) {
    RtpHeader *header = (RtpHeader *) buf;
    return ((header->pt >= 64) && (header->pt < 96));
}

static string getPeerAddress(RTC::TransportTuple *tuple){
    return SockUtil::inet_ntoa(((struct sockaddr_in *)tuple)->sin_addr) + ":" + to_string(ntohs(((struct sockaddr_in *)tuple)->sin_port));
}

void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tuple) {
    if (RTC::StunPacket::IsStun((const uint8_t *) buf, len)) {
        std::unique_ptr<RTC::StunPacket> packet(RTC::StunPacket::Parse((const uint8_t *) buf, len));
        if (!packet) {
            WarnL << "parse stun error" << std::endl;
            return;
        }
        _ice_server->ProcessStunPacket(packet.get(), tuple);
        return;
    }
    if (is_dtls(buf)) {
        _dtls_transport->ProcessDtlsData((uint8_t *) buf, len);
        return;
    }
    if (is_rtp(buf)) {
        if (!_srtp_session_recv) {
            WarnL << "received rtp packet when dtls not completed from:" << getPeerAddress(tuple);
            return;
        }
        if (_srtp_session_recv->DecryptSrtp((uint8_t *) buf, &len)) {
            onRtp(buf, len, _ticker.createdTime());
        }
        return;
    }
    if (is_rtcp(buf)) {
        if (!_srtp_session_recv) {
            WarnL << "received rtcp packet when dtls not completed from:" << getPeerAddress(tuple);
            return;
        }
        if (_srtp_session_recv->DecryptSrtcp((uint8_t *) buf, &len)) {
            onRtcp(buf, len);
        }
        return;
    }
}

void WebRtcTransport::sendRtpPacket(const char *buf, int len, bool flush, void *ctx) {
    if (_srtp_session_send) {
        auto pkt = _packet_pool.obtain2();
        //预留rtx加入的两个字节
        pkt->setCapacity((size_t) len + SRTP_MAX_TRAILER_LEN + 2);
        pkt->assign(buf, len);
        onBeforeEncryptRtp(pkt->data(), len, ctx);
        if (_srtp_session_send->EncryptRtp(reinterpret_cast<uint8_t *>(pkt->data()), &len)) {
            pkt->setSize(len);
            onSendSockData(std::move(pkt), flush);
        }
    }
}

void WebRtcTransport::sendRtcpPacket(const char *buf, int len, bool flush, void *ctx) {
    if (_srtp_session_send) {
        auto pkt = _packet_pool.obtain2();
        //预留rtx加入的两个字节
        pkt->setCapacity((size_t) len + SRTP_MAX_TRAILER_LEN + 2);
        pkt->assign(buf, len);
        onBeforeEncryptRtcp(pkt->data(), len, ctx);
        if (_srtp_session_send->EncryptRtcp(reinterpret_cast<uint8_t *>(pkt->data()), &len)) {
            pkt->setSize(len);
            onSendSockData(std::move(pkt), flush);
        }
    }
}

///////////////////////////////////////////////////////////////////////////////////

void WebRtcTransportImp::onCreate(){
    WebRtcTransport::onCreate();
    registerSelf();

    weak_ptr<WebRtcTransportImp> weak_self = static_pointer_cast<WebRtcTransportImp>(shared_from_this());
    GET_CONFIG(float, timeoutSec, RTC::kTimeOutSec);
    _timer = std::make_shared<Timer>(timeoutSec / 2, [weak_self]() {
        auto strong_self = weak_self.lock();
        if (!strong_self) {
            return false;
        }
        if (strong_self->_alive_ticker.elapsedTime() > timeoutSec * 1000) {
            strong_self->onShutdown(SockException(Err_timeout, "接受rtp和rtcp超时"));
        }
        return true;
    }, getPoller());

    _twcc_ctx.setOnSendTwccCB([this](uint32_t ssrc, string fci) {
        onSendTwcc(ssrc, fci);
    });
}

WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcTransport(poller) {
    InfoL << getIdentifier();
}

WebRtcTransportImp::~WebRtcTransportImp() {
    InfoL << getIdentifier();
}

void WebRtcTransportImp::onDestory() {
    WebRtcTransport::onDestory();
    unregisterSelf();
}

void WebRtcTransportImp::onSendSockData(Buffer::Ptr buf, bool flush, RTC::TransportTuple *tuple) {
    if (!_selected_session) {
        WarnL << "send data failed:" << buf->size();
        return;
    }
    //一次性发送一帧的rtp数据，提高网络io性能
    _selected_session->setSendFlushFlag(flush);
    _selected_session->send(std::move(buf));
}

///////////////////////////////////////////////////////////////////

bool WebRtcTransportImp::canSendRtp() const{
    for (auto &m : _answer_sdp->media) {
        if (m.direction == RtpDirection::sendrecv || m.direction == RtpDirection::sendonly) {
            return true;
        }
    }
    return false;
}

bool WebRtcTransportImp::canRecvRtp() const{
    for (auto &m : _answer_sdp->media) {
        if (m.direction == RtpDirection::sendrecv || m.direction == RtpDirection::recvonly) {
            return true;
        }
    }
    return false;
}

void WebRtcTransportImp::onStartWebRTC() {
    //获取ssrc和pt相关信息,届时收到rtp和rtcp时分别可以根据pt和ssrc找到相关的信息
    for (auto &m_answer : _answer_sdp->media) {
        auto m_offer = _offer_sdp->getMedia(m_answer.type);
        auto track = std::make_shared<MediaTrack>();

        track->media = &m_answer;
        track->answer_ssrc_rtp = m_answer.getRtpSSRC();
        track->answer_ssrc_rtx = m_answer.getRtxSSRC();
        track->offer_ssrc_rtp = m_offer->getRtpSSRC();
        track->offer_ssrc_rtx = m_offer->getRtxSSRC();
        track->plan_rtp = &m_answer.plan[0];;
        track->plan_rtx = m_answer.getRelatedRtxPlan(track->plan_rtp->pt);
        track->rtcp_context_send = std::make_shared<RtcpContextForSend>();

        //rtp track type --> MediaTrack
        if (m_answer.direction == RtpDirection::sendonly || m_answer.direction == RtpDirection::sendrecv) {
            //该类型的track 才支持发送
            _type_to_track[m_answer.type] = track;
        }
        //send ssrc --> MediaTrack
        _ssrc_to_track[track->answer_ssrc_rtp] = track;
        _ssrc_to_track[track->answer_ssrc_rtx] = track;

        //recv ssrc --> MediaTrack
        _ssrc_to_track[track->offer_ssrc_rtp] = track;
        _ssrc_to_track[track->offer_ssrc_rtx] = track;

        //rtp pt --> MediaTrack
        _pt_to_track.emplace(track->plan_rtp->pt, std::unique_ptr<WrappedMediaTrack>(new WrappedRtpTrack(track, _twcc_ctx, *this)));
        if (track->plan_rtx) {
            //rtx pt --> MediaTrack
            _pt_to_track.emplace(track->plan_rtx->pt, std::unique_ptr<WrappedMediaTrack>(new WrappedRtxTrack(track)));
        }
        if (m_offer->type != TrackApplication) {
            //记录rtp ext类型与id的关系，方便接收或发送rtp时修改rtp ext id
            track->rtp_ext_ctx = std::make_shared<RtpExtContext>(*m_offer);
            weak_ptr<MediaTrack> weak_track = track;
            track->rtp_ext_ctx->setOnGetRtp([this, weak_track](uint8_t pt, uint32_t ssrc, const string &rid) {
                //ssrc --> MediaTrack
                auto track = weak_track.lock();
                assert(track);
                _ssrc_to_track[ssrc] = std::move(track);
                InfoL << "get rtp, pt:" << (int) pt << ", ssrc:" << ssrc << ", rid:" << rid;
            });

            size_t index = 0;
            for (auto &ssrc : m_offer->rtp_ssrc_sim) {
                //记录ssrc对应的MediaTrack
                _ssrc_to_track[ssrc.ssrc] = track;
                if (m_offer->rtp_rids.size() > index) {
                    //支持firefox的simulcast, 提前映射好ssrc和rid的关系
                    track->rtp_ext_ctx->setRid(ssrc.ssrc, m_offer->rtp_rids[index]);
                }
                ++index;
            }
        }
    }
}

void WebRtcTransportImp::onCheckAnswer(RtcSession &sdp) {
    //修改answer sdp的ip、端口信息
    GET_CONFIG(string, extern_ip, RTC::kExternIP);
    for (auto &m : sdp.media) {
        m.addr.reset();
        m.addr.address = extern_ip.empty() ? SockUtil::get_local_ip() : extern_ip;
        m.rtcp_addr.reset();
        m.rtcp_addr.address = m.addr.address;

        GET_CONFIG(uint16_t, local_port, RTC::kPort);
        m.rtcp_addr.port = local_port;
        m.port = m.rtcp_addr.port;
        sdp.origin.address = m.addr.address;
    }

    if (!canSendRtp()) {
        //设置我们发送的rtp的ssrc
        return;
    }

    for (auto &m : sdp.media) {
        if (m.type == TrackApplication) {
            continue;
        }
        if (!m.rtp_rtx_ssrc.empty()) {
            //已经生成了ssrc
            continue;
        }
        //添加answer sdp的ssrc信息
        m.rtp_rtx_ssrc.emplace_back();
        auto &ssrc = m.rtp_rtx_ssrc.back();
        //发送的ssrc我们随便定义，因为在发送rtp时会修改为此值
        ssrc.ssrc = m.type + RTP_SSRC_OFFSET;
        ssrc.cname = RTP_CNAME;
        ssrc.label = RTP_LABEL;
        ssrc.mslabel = RTP_MSLABEL;
        ssrc.msid = RTP_MSID;

        if (m.getRelatedRtxPlan(m.plan[0].pt)) {
            //rtx ssrc
            ssrc.rtx_ssrc = ssrc.ssrc + RTX_SSRC_OFFSET;
        }
    }
}

void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) {
    switch (type) {
        case SdpType::answer: onCheckAnswer(sdp); break;
        case SdpType::offer: break;
        default: /*不可达*/ assert(0); break;
    }
}

void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
    WebRtcTransport::onRtcConfigure(configure);
    //添加接收端口candidate信息
    configure.addCandidate(*getIceCandidate());
}

SdpAttrCandidate::Ptr WebRtcTransportImp::getIceCandidate() const{
    auto candidate = std::make_shared<SdpAttrCandidate>();
    candidate->foundation = "udpcandidate";
    //rtp端口
    candidate->component = 1;
    candidate->transport = "udp";
    //优先级，单candidate时随便
    candidate->priority = 100;
    GET_CONFIG(string, extern_ip, RTC::kExternIP);
    candidate->address = extern_ip.empty() ? SockUtil::get_local_ip() : extern_ip;
    GET_CONFIG(uint16_t, local_port, RTC::kPort);
    candidate->port = local_port;
    candidate->type = "host";
    return candidate;
}

///////////////////////////////////////////////////////////////////

class RtpChannel : public RtpTrackImp, public std::enable_shared_from_this<RtpChannel> {
public:
    RtpChannel(EventPoller::Ptr poller, RtpTrackImp::OnSorted cb, function<void(const FCI_NACK &nack)> on_nack) {
        _poller = std::move(poller);
        _on_nack = std::move(on_nack);
        setOnSorted(std::move(cb));

        _nack_ctx.setOnNack([this](const FCI_NACK &nack) {
            onNack(nack);
        });
    }

    ~RtpChannel() override = default;

    RtpPacket::Ptr inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t len, bool is_rtx) {
        auto rtp = RtpTrack::inputRtp(type, sample_rate, ptr, len);
        if (!rtp) {
            return rtp;
        }
        auto seq = rtp->getSeq();
        _nack_ctx.received(seq, is_rtx);
        if (!is_rtx) {
            //统计rtp接受情况，便于生成nack rtcp包
            _rtcp_context.onRtp(seq, rtp->getStamp(), rtp->ntp_stamp, sample_rate, len);
        }
        return rtp;
    }

    Buffer::Ptr createRtcpRR(RtcpHeader *sr, uint32_t ssrc) {
        _rtcp_context.onRtcp(sr);
        return _rtcp_context.createRtcpRR(ssrc, getSSRC());
    }

    int getLossRate() {
        return _rtcp_context.geLostInterval() * 100 / _rtcp_context.getExpectedPacketsInterval();
    }

private:
    void starNackTimer(){
        if (_delay_task) {
            return;
        }
        weak_ptr<RtpChannel> weak_self = shared_from_this();
        _delay_task = _poller->doDelayTask(10, [weak_self]() -> uint64_t {
            auto strong_self = weak_self.lock();
            if (!strong_self) {
                return 0;
            }
            auto ret = strong_self->_nack_ctx.reSendNack();
            if (!ret) {
                strong_self->_delay_task = nullptr;
            }
            return ret;
        });
    }

    void onNack(const FCI_NACK &nack) {
        _on_nack(nack);
        starNackTimer();
    }

private:
    NackContext _nack_ctx;
    RtcpContextForRecv _rtcp_context;
    EventPoller::Ptr _poller;
    EventPoller::DelayTask::Ptr _delay_task;
    function<void(const FCI_NACK &nack)> _on_nack;
};

std::shared_ptr<RtpChannel> MediaTrack::getRtpChannel(uint32_t ssrc) const{
    auto it_chn = rtp_channel.find(rtp_ext_ctx->getRid(ssrc));
    if (it_chn == rtp_channel.end()) {
        return nullptr;
    }
    return it_chn->second;
}

void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
    _bytes_usage += len;
    auto rtcps = RtcpHeader::loadFromBytes((char *) buf, len);
    for (auto rtcp : rtcps) {
        switch ((RtcpType) rtcp->pt) {
            case RtcpType::RTCP_SR : {
                //对方汇报rtp发送情况
                RtcpSR *sr = (RtcpSR *) rtcp;
                auto it = _ssrc_to_track.find(sr->ssrc);
                if (it != _ssrc_to_track.end()) {
                    auto &track = it->second;
                    auto rtp_chn = track->getRtpChannel(sr->ssrc);
                    if(!rtp_chn){
                        WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
                    } else {
                        //InfoL << "接收丢包率,ssrc:" << sr->ssrc << ",loss rate(%):" << rtp_chn->getLossRate();
                        //设置rtp时间戳与ntp时间戳的对应关系
                        rtp_chn->setNtpStamp(sr->rtpts, sr->getNtpUnixStampMS());
                        auto rr = rtp_chn->createRtcpRR(sr, track->answer_ssrc_rtp);
                        sendRtcpPacket(rr->data(), rr->size(), true);
                    }
                } else {
                    WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
                }
                break;
            }
            case RtcpType::RTCP_RR : {
                _alive_ticker.resetTime();
                //对方汇报rtp接收情况
                RtcpRR *rr = (RtcpRR *) rtcp;
                for (auto item : rr->getItemList()) {
                    auto it = _ssrc_to_track.find(item->ssrc);
                    if (it != _ssrc_to_track.end()) {
                        auto &track = it->second;
                        track->rtcp_context_send->onRtcp(rtcp);
                        auto sr = track->rtcp_context_send->createRtcpSR(track->answer_ssrc_rtp);
                        sendRtcpPacket(sr->data(), sr->size(), true);
                    } else {
                        WarnL << "未识别的rr rtcp包:" << rtcp->dumpString();
                    }
                }
                break;
            }
            case RtcpType::RTCP_BYE : {
                //对方汇报停止发送rtp
                RtcpBye *bye = (RtcpBye *) rtcp;
                for (auto ssrc : bye->getSSRC()) {
                    auto it = _ssrc_to_track.find(*ssrc);
                    if (it == _ssrc_to_track.end()) {
                        WarnL << "未识别的bye rtcp包:" << rtcp->dumpString();
                        continue;
                    }
                    _ssrc_to_track.erase(it);
                }
                onShutdown(SockException(Err_eof, "rtcp bye message received"));
                break;
            }
            case RtcpType::RTCP_PSFB:
            case RtcpType::RTCP_RTPFB: {
                if ((RtcpType) rtcp->pt == RtcpType::RTCP_PSFB) {
                    break;
                }
                //RTPFB
                switch ((RTPFBType) rtcp->report_count) {
                    case RTPFBType::RTCP_RTPFB_NACK : {
                        RtcpFB *fb = (RtcpFB *) rtcp;
                        auto it = _ssrc_to_track.find(fb->ssrc_media);
                        if (it == _ssrc_to_track.end()) {
                            WarnL << "未识别的 rtcp包:" << rtcp->dumpString();
                            return;
                        }
                        auto &track = it->second;
                        auto &fci = fb->getFci<FCI_NACK>();
                        track->nack_list.forEach(fci, [&](const RtpPacket::Ptr &rtp) {
                            //rtp重传
                            onSendRtp(rtp, true, true);
                        });
                        break;
                    }
                    default: break;
                }
                break;
            }
            default: break;
        }
    }
}

///////////////////////////////////////////////////////////////////

void WebRtcTransportImp::createRtpChannel(const string &rid, uint32_t ssrc, MediaTrack &track) {
    //rid --> RtpReceiverImp
    auto &ref = track.rtp_channel[rid];
    weak_ptr<WebRtcTransportImp> weak_self = dynamic_pointer_cast<WebRtcTransportImp>(shared_from_this());
    ref = std::make_shared<RtpChannel>(getPoller(), [&track, this, rid](RtpPacket::Ptr rtp) mutable {
        onSortedRtp(track, rid, std::move(rtp));
    }, [&track, weak_self, ssrc](const FCI_NACK &nack) mutable {
        //nack发送可能由定时器异步触发
        auto strong_self = weak_self.lock();
        if (strong_self) {
            strong_self->onSendNack(track, nack, ssrc);
        }
    });
    InfoL << "create rtp receiver of ssrc:" << ssrc << ", rid:" << rid << ", codec:" << track.plan_rtp->codec;
}

void WebRtcTransportImp::updateTicker() {
    _alive_ticker.resetTime();
}

void WebRtcTransportImp::onRtp(const char *buf, size_t len, uint64_t stamp_ms) {
    _bytes_usage += len;
    _alive_ticker.resetTime();

    RtpHeader *rtp = (RtpHeader *) buf;
    //根据接收到的rtp的pt信息，找到该流的信息
    auto it = _pt_to_track.find(rtp->pt);
    if (it == _pt_to_track.end()) {
        WarnL << "unknown rtp pt:" << (int)rtp->pt;
        return;
    }
    it->second->inputRtp(buf, len, stamp_ms, rtp);
}

void WrappedRtpTrack::inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) {
#if 0
    auto seq = ntohs(rtp->seq);
    if (track->media->type == TrackVideo && seq % 100 == 0) {
        //此处模拟接受丢包
        return;
    }
#endif
    
    auto ssrc = ntohl(rtp->ssrc);

    //修改ext id至统一
    string rid;
    auto twcc_ext = track->rtp_ext_ctx->changeRtpExtId(rtp, true, &rid, RtpExtType::transport_cc);

    if (twcc_ext) {
        _twcc_ctx.onRtp(ssrc, twcc_ext.getTransportCCSeq(), stamp_ms);
    }

    auto &ref = track->rtp_channel[rid];
    if (!ref) {
        _transport.createRtpChannel(rid, ssrc, *track);
    }

    //解析并排序rtp
    ref->inputRtp(track->media->type, track->plan_rtp->sample_rate, (uint8_t *) buf, len, false);
}

void WrappedRtxTrack::inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) {
    //修改ext id至统一
    string rid;
    track->rtp_ext_ctx->changeRtpExtId(rtp, true, &rid, RtpExtType::transport_cc);

    auto &ref = track->rtp_channel[rid];
    if (!ref) {
        //再接收到对应的rtp前，丢弃rtx包
        WarnL << "unknown rtx rtp, rid:" << rid << ", ssrc:" << ntohl(rtp->ssrc) << ", codec:" << track->plan_rtp->codec << ", seq:" << ntohs(rtp->seq);
        return;
    }

    //这里是rtx重传包
    // https://datatracker.ietf.org/doc/html/rfc4588#section-4
    auto payload = rtp->getPayloadData();
    auto size = rtp->getPayloadSize(len);
    if (size < 2) {
        return;
    }

    //前两个字节是原始的rtp的seq
    auto origin_seq = payload[0] << 8 | payload[1];
    // rtx 转换为 rtp
    rtp->pt = track->plan_rtp->pt;
    rtp->seq = htons(origin_seq);
    rtp->ssrc = htonl(ref->getSSRC());

    memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf);
    buf += 2;
    len -= 2;
    ref->inputRtp(track->media->type, track->plan_rtp->sample_rate, (uint8_t *) buf, len, true);
}

void WebRtcTransportImp::onSendNack(MediaTrack &track, const FCI_NACK &nack, uint32_t ssrc) {
    auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_NACK, &nack, FCI_NACK::kSize);
    rtcp->ssrc = htonl(track.answer_ssrc_rtp);
    rtcp->ssrc_media = htonl(ssrc);
    sendRtcpPacket((char *) rtcp.get(), rtcp->getSize(), true);
}

void WebRtcTransportImp::onSendTwcc(uint32_t ssrc, const string &twcc_fci) {
    auto rtcp = RtcpFB::create(RTPFBType::RTCP_RTPFB_TWCC, twcc_fci.data(), twcc_fci.size());
    rtcp->ssrc = htonl(0);
    rtcp->ssrc_media = htonl(ssrc);
    sendRtcpPacket((char *) rtcp.get(), rtcp->getSize(), true);
}

///////////////////////////////////////////////////////////////////

void WebRtcTransportImp::onSortedRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp) {
    if (track.media->type == TrackVideo && _pli_ticker.elapsedTime() > 2000) {
        //定期发送pli请求关键帧，方便非rtc等协议
        _pli_ticker.resetTime();
        sendRtcpPli(rtp->getSSRC());

        //开启remb，则发送remb包调节比特率
        GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate);
        if (remb_bit_rate && _answer_sdp->supportRtcpFb(SdpConst::kRembRtcpFb)) {
            sendRtcpRemb(rtp->getSSRC(), remb_bit_rate);
        }
    }

    onRecvRtp(track, rid, std::move(rtp));
}

///////////////////////////////////////////////////////////////////

void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx) {
    auto &track = _type_to_track[rtp->type];
    if (!track) {
        //忽略，对方不支持该编码类型
        return;
    }
    if (!rtx) {
        //统计rtp发送情况，好做sr汇报
        track->rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStamp(), rtp->ntp_stamp, rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
        track->nack_list.pushBack(rtp);
#if 0
        //此处模拟发送丢包
        if (rtp->type == TrackVideo && rtp->getSeq() % 100 == 0) {
            return;
        }
#endif
    } else {
        //发送rtx重传包
        //TraceL << "send rtx rtp:" << rtp->getSeq();
    }
    pair<bool/*rtx*/, MediaTrack *> ctx{rtx, track.get()};
    sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, &ctx);
    _bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
}

void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, int &len, void *ctx) {
    auto pr = (pair<bool/*rtx*/, MediaTrack *> *) ctx;
    auto header = (RtpHeader *) buf;

    if (!pr->first || !pr->second->plan_rtx) {
        //普通的rtp,或者不支持rtx, 修改目标pt和ssrc
        pr->second->rtp_ext_ctx->changeRtpExtId(header, false);
        header->pt = pr->second->plan_rtp->pt;
        header->ssrc = htonl(pr->second->answer_ssrc_rtp);
    } else {
        //重传的rtp, rtx
        pr->second->rtp_ext_ctx->changeRtpExtId(header, false);
        header->pt = pr->second->plan_rtx->pt;
        if (pr->second->answer_ssrc_rtx) {
            //有rtx单独的ssrc,有些情况下，浏览器支持rtx，但是未指定rtx单独的ssrc
            header->ssrc = htonl(pr->second->answer_ssrc_rtx);
        } else {
            //未单独指定rtx的ssrc，那么使用rtp的ssrc
            header->ssrc = htonl(pr->second->answer_ssrc_rtp);
        }

        auto origin_seq = ntohs(header->seq);
        //seq跟原来的不一样
        header->seq = htons(_rtx_seq[pr->second->media->type]);
        ++_rtx_seq[pr->second->media->type];

        auto payload = header->getPayloadData();
        auto payload_size = header->getPayloadSize(len);
        if (payload_size) {
            //rtp负载后移两个字节，这两个字节用于存放osn
            //https://datatracker.ietf.org/doc/html/rfc4588#section-4
            memmove(payload + 2, payload, payload_size);
        }
        payload[0] = origin_seq >> 8;
        payload[1] = origin_seq & 0xFF;
        len += 2;
    }
}

void WebRtcTransportImp::onShutdown(const SockException &ex){
    WarnL << ex.what();
    unrefSelf();
    for (auto &pr : _history_sessions) {
        auto session = pr.second.lock();
        if (session) {
            session->shutdown(ex);
        }
    }
}

void WebRtcTransportImp::setSession(Session::Ptr session) {
    _history_sessions.emplace(session.get(), session);
    if (_selected_session) {
        InfoL << "rtc network changed: " << _selected_session->get_peer_ip() << ":" << _selected_session->get_peer_port()
              << " -> " << session->get_peer_ip() << ":" << session->get_peer_port() << ", id:" << getIdentifier();
    }
    _selected_session = std::move(session);
    unrefSelf();
}

const Session::Ptr &WebRtcTransportImp::getSession() const {
    return _selected_session;
}

uint64_t WebRtcTransportImp::getBytesUsage() const{
    return _bytes_usage;
}

uint64_t WebRtcTransportImp::getDuration() const{
    return _alive_ticker.createdTime() / 1000;
}

/////////////////////////////////////////////////////////////////////////////////////////////

void WebRtcTransportImp::registerSelf() {
    _self =  static_pointer_cast<WebRtcTransportImp>(shared_from_this());
    WebRtcTransportManager::Instance().addItem(getIdentifier(), _self);
}

void WebRtcTransportImp::unrefSelf() {
    _self = nullptr;
}

void WebRtcTransportImp::unregisterSelf() {
    unrefSelf();
    WebRtcTransportManager::Instance().removeItem(getIdentifier());
}

WebRtcTransportManager &WebRtcTransportManager::Instance() {
    static WebRtcTransportManager s_instance;
    return s_instance;
}

void WebRtcTransportManager::addItem(const string &key, const WebRtcTransportImp::Ptr &ptr) {
    lock_guard<mutex> lck(_mtx);
    _map[key] = ptr;
}

WebRtcTransportImp::Ptr WebRtcTransportManager::getItem(const string &key) {
    if (key.empty()) {
        return nullptr;
    }
    lock_guard<mutex> lck(_mtx);
    auto it = _map.find(key);
    if (it == _map.end()) {
        return nullptr;
    }
    return it->second.lock();
}

void WebRtcTransportManager::removeItem(const string &key) {
    lock_guard<mutex> lck(_mtx);
    _map.erase(key);
}

//////////////////////////////////////////////////////////////////////////////////////////////

WebRtcPluginManager &WebRtcPluginManager::Instance() {
    static WebRtcPluginManager s_instance;
    return s_instance;
}

void WebRtcPluginManager::registerPlugin(const string &type, Plugin cb) {
    lock_guard<mutex> lck(_mtx_creator);
    _map_creator[type] = std::move(cb);
}

void WebRtcPluginManager::getAnswerSdp(Session &sender, const string &type, const string &offer, const WebRtcArgs &args,
                                       const onCreateRtc &cb) {
    lock_guard<mutex> lck(_mtx_creator);
    auto it = _map_creator.find(type);
    if (it == _map_creator.end()) {
        cb(WebRtcException(SockException(Err_other, "the type can not supported")));
        return;
    }
    it->second(sender, offer, args, cb);
}

#include "WebRtcPlayer.h"
#include "WebRtcPusher.h"
#include "WebRtcEchoTest.h"

void echo_plugin(Session &sender, const string &offer, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
    cb(*WebRtcEchoTest::create(EventPollerPool::Instance().getPoller()));
}

void push_plugin(Session &sender, const string &offer_sdp, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
    MediaInfo info(args["url"]);
    Broadcast::PublishAuthInvoker invoker = [cb, offer_sdp, info](const string &err, bool enable_hls, bool enable_mp4) mutable {
        if (!err.empty()) {
            cb(WebRtcException(SockException(Err_other, err)));
            return;
        }

        RtspMediaSourceImp::Ptr push_src;
        std::shared_ptr<void> push_src_ownership;
        auto src = MediaSource::find(RTSP_SCHEMA, info._vhost, info._app, info._streamid);
        auto push_failed = (bool)src;

        while (src) {
            //尝试断连后继续推流
            auto rtsp_src = dynamic_pointer_cast<RtspMediaSourceImp>(src);
            if (!rtsp_src) {
                //源不是rtsp推流产生的
                break;
            }
            auto ownership = rtsp_src->getOwnership();
            if (!ownership) {
                //获取推流源所有权失败
                break;
            }
            push_src = std::move(rtsp_src);
            push_src_ownership = std::move(ownership);
            push_failed = false;
            break;
        }

        if (push_failed) {
            cb(WebRtcException(SockException(Err_other, "already publishing")));
            return;
        }

        if (!push_src) {
            push_src = std::make_shared<RtspMediaSourceImp>(info._vhost, info._app, info._streamid);
            push_src_ownership = push_src->getOwnership();
            push_src->setProtocolTranslation(enable_hls, enable_mp4);
        }
        auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info);
        push_src->setListener(rtc);
        cb(*rtc);
    };

    //rtsp推流需要鉴权
    auto flag = NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastMediaPublish, MediaOriginType::rtc_push, info, invoker, static_cast<SockInfo &>(sender));
    if (!flag) {
        //该事件无人监听,默认不鉴权
        GET_CONFIG(bool, to_hls, General::kPublishToHls);
        GET_CONFIG(bool, to_mp4, General::kPublishToMP4);
        invoker("", to_hls, to_mp4);
    }
}

void play_plugin(Session &sender, const string &offer_sdp, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
    MediaInfo info(args["url"]);
    auto session_ptr = sender.shared_from_this();
    Broadcast::AuthInvoker invoker = [cb, offer_sdp, info, session_ptr](const string &err) mutable {
        if (!err.empty()) {
            cb(WebRtcException(SockException(Err_other, err)));
            return;
        }

        //webrtc播放的是rtsp的源
        info._schema = RTSP_SCHEMA;
        MediaSource::findAsync(info, session_ptr, [=](const MediaSource::Ptr &src_in) mutable {
            auto src = dynamic_pointer_cast<RtspMediaSource>(src_in);
            if (!src) {
                cb(WebRtcException(SockException(Err_other, "stream not found")));
                return;
            }
            //还原成rtc，目的是为了hook时识别哪种播放协议
            info._schema = RTC_SCHEMA;
            auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info);
            cb(*rtc);
        });
    };

    //广播通用播放url鉴权事件
    auto flag = NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastMediaPlayed, info, invoker, static_cast<SockInfo &>(sender));
    if (!flag) {
        //该事件无人监听,默认不鉴权
        invoker("");
    }
}

static onceToken s_rtc_auto_register([](){
    WebRtcPluginManager::Instance().registerPlugin("echo", echo_plugin);
    WebRtcPluginManager::Instance().registerPlugin("push", push_plugin);
    WebRtcPluginManager::Instance().registerPlugin("play", play_plugin);
});