#include "WebRtcTransport.h"
#include <iostream>
#include "Rtcp/Rtcp.h"
#include "Rtsp/RtpReceiver.h"
#define RTX_SSRC_OFFSET 2
#define RTP_CNAME "zlmediakit-rtp"
#define RTX_CNAME "zlmediakit-rtx"


WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) {
    _dtls_transport = std::make_shared<RTC::DtlsTransport>(poller, this);
    _ice_server = std::make_shared<RTC::IceServer>(this, makeRandStr(4), makeRandStr(24));
}

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

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

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

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

void WebRtcTransport::onSendSockData(const char *buf, size_t len, bool flush){
    auto tuple = _ice_server->GetSelectedTuple();
    assert(tuple);
    onSendSockData(buf, len, (struct sockaddr_in *) tuple, flush);
}

const RtcSession& WebRtcTransport::getSdp(SdpType type) const{
    switch (type) {
        case SdpType::offer: return *_offer_sdp;
        case SdpType::answer: return *_answer_sdp;
        default: throw std::invalid_argument("不识别的sdp类型");
    }
}

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::onCheckSdp(SdpType type, RtcSession &sdp) const{
    for (auto &m : sdp.media) {
        if (m.type != TrackApplication && !m.rtcp_mux) {
            throw std::invalid_argument("只支持rtcp-mux模式");
        }
    }
    if (sdp.group.mids.empty()) {
        throw std::invalid_argument("只支持group BUNDLE模式");
    }
}

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

    auto str = _answer_sdp->toString();
    TraceL << "\r\n" << str;
    return str;
}

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

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

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

void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple *tuple) {
    if (RTC::StunPacket::IsStun((const uint8_t *) buf, len)) {
        RTC::StunPacket *packet = RTC::StunPacket::Parse((const uint8_t *) buf, len);
        if (packet == nullptr) {
            WarnL << "parse stun error" << std::endl;
            return;
        }
        _ice_server->ProcessStunPacket(packet, tuple);
        return;
    }
    if (is_dtls(buf)) {
        _dtls_transport->ProcessDtlsData((uint8_t *) buf, len);
        return;
    }
    if (is_rtp(buf)) {
        if (_srtp_session_recv->DecryptSrtp((uint8_t *) buf, &len)) {
            onRtp(buf, len);
        } else {
            WarnL;
        }
        return;
    }
    if (is_rtcp(buf)) {
        if (_srtp_session_recv->DecryptSrtcp((uint8_t *) buf, &len)) {
            onRtcp(buf, len);
        } else {
            WarnL;
        }
        return;
    }
}

void WebRtcTransport::sendRtpPacket(char *buf, size_t len, bool flush, uint8_t pt) {
    const uint8_t *p = (uint8_t *) buf;
    bool ret = false;
    if (_srtp_session_send) {
        ret = _srtp_session_send->EncryptRtp(&p, &len, pt);
    }
    if (ret) {
        onSendSockData((char *) p, len, flush);
    }
}

void WebRtcTransport::sendRtcpPacket(char *buf, size_t len, bool flush){
    const uint8_t *p = (uint8_t *) buf;
    bool ret = false;
    if (_srtp_session_send) {
        ret = _srtp_session_send->EncryptRtcp(&p, &len);
    }
    if (ret) {
        onSendSockData((char *) p, len, flush);
    }
}

///////////////////////////////////////////////////////////////////////////////////
WebRtcTransportImp::Ptr WebRtcTransportImp::create(const EventPoller::Ptr &poller){
    WebRtcTransportImp::Ptr ret(new WebRtcTransportImp(poller), [](WebRtcTransportImp *ptr){
        ptr->onDestory();
       delete ptr;
    });
    return ret;
}

WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcTransport(poller) {
    _socket = Socket::createSocket(poller, false);
    //随机端口，绑定全部网卡
    _socket->bindUdpSock(0);
    _socket->setOnRead([this](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) mutable {
        inputSockData(buf->data(), buf->size(), addr);
    });
}

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

void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src) {
    assert(src);
    _src = src;
}

void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush) {
    auto ptr = BufferRaw::create();
    ptr->assign(buf, len);
    _socket->send(ptr, (struct sockaddr *)(dst), sizeof(struct sockaddr), flush);
}

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

bool WebRtcTransportImp::canSendRtp() const{
    auto &sdp = getSdp(SdpType::answer);
    return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::sendonly;
}

bool WebRtcTransportImp::canRecvRtp() const{
    auto &sdp = getSdp(SdpType::answer);
    return sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::recvonly;
}

void WebRtcTransportImp::onStartWebRTC() {
    if (canRecvRtp()) {
        _push_src = std::make_shared<RtspMediaSourceImp>(DEFAULT_VHOST, "live", "push");
        auto rtsp_sdp = getSdp(SdpType::answer).toRtspSdp();
        _push_src->setSdp(rtsp_sdp);

        for (auto &m : getSdp(SdpType::offer).media) {
            if (m.type == TrackVideo) {
                _recv_video_ssrc = m.rtp_ssrc.ssrc;
            }
            for (auto &plan : m.plan) {
                auto hit_pan = getSdp(SdpType::answer).getMedia(m.type)->getPlan(plan.pt);
                if (!hit_pan) {
                    continue;
                }
                //获取offer端rtp的ssrc和pt相关信息
                auto &ref = _rtp_receiver[plan.pt];
                _ssrc_info[m.rtp_ssrc.ssrc] = &ref;
                ref.plan = &plan;
                ref.media = &m;
                ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid;
                ref.rtcp_context_recv = std::make_shared<RtcpContext>(ref.plan->sample_rate, true);
                ref.rtcp_context_send = std::make_shared<RtcpContext>(ref.plan->sample_rate, false);
                ref.receiver = std::make_shared<RtpReceiverImp>([&ref, this](RtpPacket::Ptr rtp) {
                    onSortedRtp(ref, std::move(rtp));
                }, [ref, this](const RtpPacket::Ptr &rtp) {
                    onBeforeSortedRtp(ref, rtp);
                });
            }
        }
    }
    if (canSendRtp()) {
        _reader = _src->getRing()->attach(_socket->getPoller(), true);
        weak_ptr<WebRtcTransportImp> weak_self = shared_from_this();
        _reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt) {
            auto strongSelf = weak_self.lock();
            if (!strongSelf) {
                return;
            }
            size_t i = 0;
            pkt->for_each([&](const RtpPacket::Ptr &rtp) {
                strongSelf->onSendRtp(rtp, ++i == pkt->size());
            });
        });
    }
}

void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp) const{
    WebRtcTransport::onCheckSdp(type, sdp);
    if (type != SdpType::answer || !canSendRtp()) {
        return;
    }

    //添加answer sdp的ssrc信息，并且记录发送rtp的pt
    for (auto &m : sdp.media) {
        if (m.type == TrackApplication) {
            continue;
        }
        m.rtp_ssrc.ssrc = _src->getSsrc(m.type);
        m.rtp_ssrc.cname = RTP_CNAME;
        //todo 先屏蔽rtx，因为chrome报错
        if (false && m.getRelatedRtxPlan(m.plan[0].pt)) {
            m.rtx_ssrc.ssrc = RTX_SSRC_OFFSET + m.rtp_ssrc.ssrc;
            m.rtx_ssrc.cname = RTX_CNAME;
        }
        auto rtsp_media = _rtsp_send_sdp.getMedia(m.type);
        if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) {
            _send_rtp_pt[m.type] = m.plan[0].pt;
        }
    }
}

void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
    WebRtcTransport::onRtcConfigure(configure);
    _rtsp_send_sdp.loadFrom(_src->getSdp(), false);

    //根据rtsp流的相关信息，设置rtc最佳编码
    for (auto &m : _rtsp_send_sdp.media) {
        switch (m.type) {
            case TrackVideo: {
                configure.video.preferred_codec.insert(configure.video.preferred_codec.begin(), getCodecId(m.plan[0].codec));
                break;
            }
            case TrackAudio: {
                configure.audio.preferred_codec.insert(configure.audio.preferred_codec.begin(),getCodecId(m.plan[0].codec));
                break;
            }
            default:
                break;
        }
    }

    //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;
    //todo 此处修改为配置文件
    candidate->address = SockUtil::get_local_ip();
    candidate->port = _socket->get_local_port();
    candidate->type = "host";
    return candidate;
}

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

class RtpReceiverImp : public RtpReceiver {
public:
    RtpReceiverImp( function<void(RtpPacket::Ptr rtp)> cb,  function<void(const RtpPacket::Ptr &rtp)> cb_before = nullptr){
        _on_sort = std::move(cb);
        _on_before_sort = std::move(cb_before);
    }

    ~RtpReceiverImp() override = default;

    bool inputRtp(TrackType type, int samplerate, uint8_t *ptr, size_t len){
        return handleOneRtp((int) type, type, samplerate, ptr, len);
    }

protected:
    void onRtpSorted(RtpPacket::Ptr rtp, int track_index) override {
        _on_sort(std::move(rtp));
    }

    void onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_index) override {
        if (_on_before_sort) {
            _on_before_sort(rtp);
        }
    }

private:
    function<void(RtpPacket::Ptr rtp)> _on_sort;
    function<void(const RtpPacket::Ptr &rtp)> _on_before_sort;
};

void WebRtcTransportImp::onRtcp(const char *buf, size_t 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_info.find(sr->ssrc);
                if (it != _ssrc_info.end()) {
                    it->second->rtcp_context_recv->onRtcp(sr);
                    auto rr = it->second->rtcp_context_recv->createRtcpRR(sr->items.ssrc, sr->ssrc);
                    sendRtcpPacket(rr->data(), rr->size(), true);
                    InfoL << "send rtcp rr";
                }
                break;
            }
            case RtcpType::RTCP_RR : {
                //对方汇报rtp接收情况
                RtcpRR *rr = (RtcpRR *) rtcp;
                auto it = _ssrc_info.find(rr->ssrc);
                if (it != _ssrc_info.end()) {
                    auto sr = it->second->rtcp_context_send->createRtcpSR(rr->items.ssrc);
                    sendRtcpPacket(sr->data(), sr->size(), true);
                    InfoL << "send rtcp sr";
                }
                break;
            }
            case RtcpType::RTCP_BYE : {
                //todo 此处应该销毁对象
                break;
            }
            case RtcpType::RTCP_PSFB: {
//                InfoL << rtcp->dumpString();
                break;
            }
            default: break;
        }
    }
}

void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
    RtpHeader *rtp = (RtpHeader *) buf;
    //根据接收到的rtp的pt信息，找到该流的信息
    auto it = _rtp_receiver.find(rtp->pt);
    if (it == _rtp_receiver.end()) {
        WarnL;
        return;
    }
    auto &info = it->second;
    //解析并排序rtp
    info.receiver->inputRtp(info.media->type, info.plan->sample_rate, (uint8_t *) buf, len);
}

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

void WebRtcTransportImp::onSortedRtp(const RtpPayloadInfo &info, RtpPacket::Ptr rtp) {
    if(!info.is_common_rtp){
        //todo rtx/red/ulpfec类型的rtp先未处理
        WarnL;
        return;
    }
    if (_pli_ticker.elapsedTime() > 2000) {
        //todo 定期发送pli
        _pli_ticker.resetTime();
        auto pli = RtcpPli::create();
        pli->ssrc = htonl(0);
        pli->ssrc_media = htonl(_recv_video_ssrc);
        sendRtcpPacket((char *) pli.get(), sizeof(RtcpPli), true);
        InfoL << "send pli";
    }
    _push_src->onWrite(std::move(rtp), false);
}

void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const RtpPacket::Ptr &rtp) {
    //统计rtp收到的情况，好做rr汇报
    info.rtcp_context_recv->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
}

void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){
    auto &pt = _send_rtp_pt[rtp->type];
    if (!pt) {
        //忽略，对方不支持该编码类型
        return;
    }
    sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, pt);
    //统计rtp发送情况，好做sr汇报
    _rtp_receiver[pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
}