mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-29 14:45:55 +08:00
943 lines
34 KiB
C++
943 lines
34 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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#include "WebRtcTransport.h"
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#include <iostream>
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#include "RtpExt.h"
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#include "Rtcp/Rtcp.h"
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#include "Rtcp/RtcpFCI.h"
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#include "Rtsp/RtpReceiver.h"
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#define RTX_SSRC_OFFSET 2
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#define RTP_CNAME "zlmediakit-rtp"
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#define RTP_LABEL "zlmediakit-label"
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#define RTP_MSLABEL "zlmediakit-mslabel"
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#define RTP_MSID RTP_MSLABEL " " RTP_LABEL
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//RTC配置项目
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namespace RTC {
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#define RTC_FIELD "rtc."
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//rtp和rtcp接受超时时间
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const string kTimeOutSec = RTC_FIELD"timeoutSec";
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//服务器外网ip
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const string kExternIP = RTC_FIELD"externIP";
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//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
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const string kRembBitRate = RTC_FIELD"rembBitRate";
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static onceToken token([]() {
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mINI::Instance()[kTimeOutSec] = 15;
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mINI::Instance()[kExternIP] = "";
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mINI::Instance()[kRembBitRate] = 0;
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});
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}//namespace RTC
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WebRtcTransport::WebRtcTransport(const EventPoller::Ptr &poller) {
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_poller = poller;
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_dtls_transport = std::make_shared<RTC::DtlsTransport>(poller, this);
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_ice_server = std::make_shared<RTC::IceServer>(this, makeRandStr(4), makeRandStr(28).substr(4));
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}
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void WebRtcTransport::onCreate(){
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}
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void WebRtcTransport::onDestory(){
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_dtls_transport = nullptr;
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_ice_server = nullptr;
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}
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const EventPoller::Ptr& WebRtcTransport::getPoller() const{
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return _poller;
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}
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//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
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void WebRtcTransport::OnIceServerSendStunPacket(const RTC::IceServer *iceServer, const RTC::StunPacket *packet, RTC::TransportTuple *tuple) {
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onSendSockData((char *) packet->GetData(), packet->GetSize(), (struct sockaddr_in *) tuple);
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}
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void WebRtcTransport::OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) {
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InfoL;
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}
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void WebRtcTransport::OnIceServerConnected(const RTC::IceServer *iceServer) {
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InfoL;
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}
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void WebRtcTransport::OnIceServerCompleted(const RTC::IceServer *iceServer) {
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InfoL;
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if (_answer_sdp->media[0].role == DtlsRole::passive) {
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_dtls_transport->Run(RTC::DtlsTransport::Role::SERVER);
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} else {
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_dtls_transport->Run(RTC::DtlsTransport::Role::CLIENT);
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}
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}
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void WebRtcTransport::OnIceServerDisconnected(const RTC::IceServer *iceServer) {
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InfoL;
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}
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//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
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void WebRtcTransport::OnDtlsTransportConnected(
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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) {
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InfoL;
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_srtp_session_send = std::make_shared<RTC::SrtpSession>(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpLocalKey, srtpLocalKeyLen);
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_srtp_session_recv = std::make_shared<RTC::SrtpSession>(RTC::SrtpSession::Type::INBOUND, srtpCryptoSuite, srtpRemoteKey, srtpRemoteKeyLen);
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onStartWebRTC();
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}
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void WebRtcTransport::OnDtlsTransportSendData(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) {
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onSendSockData((char *)data, len);
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}
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void WebRtcTransport::OnDtlsTransportConnecting(const RTC::DtlsTransport *dtlsTransport) {
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InfoL;
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}
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void WebRtcTransport::OnDtlsTransportFailed(const RTC::DtlsTransport *dtlsTransport) {
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InfoL;
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onShutdown(SockException(Err_shutdown, "dtls transport failed"));
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}
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void WebRtcTransport::OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) {
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InfoL;
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onShutdown(SockException(Err_shutdown, "dtls close notify received"));
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}
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void WebRtcTransport::OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) {
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InfoL << hexdump(data, len);
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}
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//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
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void WebRtcTransport::onSendSockData(const char *buf, size_t len, bool flush){
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auto tuple = _ice_server->GetSelectedTuple();
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assert(tuple);
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onSendSockData(buf, len, (struct sockaddr_in *) tuple, flush);
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}
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const RtcSession& WebRtcTransport::getSdp(SdpType type) const{
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switch (type) {
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case SdpType::offer: return *_offer_sdp;
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case SdpType::answer: return *_answer_sdp;
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default: throw std::invalid_argument("不识别的sdp类型");
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}
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}
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RTC::TransportTuple* WebRtcTransport::getSelectedTuple() const{
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return _ice_server->GetSelectedTuple();
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}
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void WebRtcTransport::sendRtcpRemb(uint32_t ssrc, size_t bit_rate) {
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auto remb = FCI_REMB::create({ssrc}, (uint32_t)bit_rate);
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auto fb = RtcpFB::create(PSFBType::RTCP_PSFB_REMB, remb.data(), remb.size());
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fb->ssrc = htonl(0);
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fb->ssrc_media = htonl(ssrc);
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sendRtcpPacket((char *) fb.get(), fb->getSize(), true);
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}
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void WebRtcTransport::sendRtcpPli(uint32_t ssrc) {
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auto pli = RtcpFB::create(PSFBType::RTCP_PSFB_PLI);
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pli->ssrc = htonl(0);
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pli->ssrc_media = htonl(ssrc);
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sendRtcpPacket((char *) pli.get(), pli->getSize(), true);
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}
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string getFingerprint(const string &algorithm_str, const std::shared_ptr<RTC::DtlsTransport> &transport){
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auto algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm(algorithm_str);
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for (auto &finger_prints : transport->GetLocalFingerprints()) {
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if (finger_prints.algorithm == algorithm) {
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return finger_prints.value;
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}
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}
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throw std::invalid_argument(StrPrinter << "不支持的加密算法:" << algorithm_str);
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}
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void WebRtcTransport::setRemoteDtlsFingerprint(const RtcSession &remote){
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//设置远端dtls签名
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RTC::DtlsTransport::Fingerprint remote_fingerprint;
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remote_fingerprint.algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm(_offer_sdp->media[0].fingerprint.algorithm);
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remote_fingerprint.value = _offer_sdp->media[0].fingerprint.hash;
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_dtls_transport->SetRemoteFingerprint(remote_fingerprint);
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}
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void WebRtcTransport::onCheckSdp(SdpType type, RtcSession &sdp){
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for (auto &m : sdp.media) {
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if (m.type != TrackApplication && !m.rtcp_mux) {
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throw std::invalid_argument("只支持rtcp-mux模式");
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}
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}
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if (sdp.group.mids.empty()) {
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throw std::invalid_argument("只支持group BUNDLE模式");
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}
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if (type == SdpType::offer) {
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sdp.checkValidSSRC();
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}
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}
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void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const {
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//开启remb后关闭twcc，因为开启twcc后remb无效
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GET_CONFIG(size_t, remb_bit_rate, RTC::kRembBitRate);
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configure.enableTWCC(!remb_bit_rate);
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}
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std::string WebRtcTransport::getAnswerSdp(const string &offer){
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try {
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//// 解析offer sdp ////
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_offer_sdp = std::make_shared<RtcSession>();
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_offer_sdp->loadFrom(offer);
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onCheckSdp(SdpType::offer, *_offer_sdp);
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setRemoteDtlsFingerprint(*_offer_sdp);
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//// sdp 配置 ////
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SdpAttrFingerprint fingerprint;
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fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm;
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fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
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RtcConfigure configure;
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configure.setDefaultSetting(_ice_server->GetUsernameFragment(), _ice_server->GetPassword(),
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RtpDirection::sendrecv, fingerprint);
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onRtcConfigure(configure);
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//// 生成answer sdp ////
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_answer_sdp = configure.createAnswer(*_offer_sdp);
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onCheckSdp(SdpType::answer, *_answer_sdp);
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return _answer_sdp->toString();
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} catch (exception &ex) {
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onShutdown(SockException(Err_shutdown, ex.what()));
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throw;
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}
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}
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bool is_dtls(char *buf) {
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return ((*buf > 19) && (*buf < 64));
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}
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bool is_rtp(char *buf) {
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RtpHeader *header = (RtpHeader *) buf;
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return ((header->pt < 64) || (header->pt >= 96));
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}
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bool is_rtcp(char *buf) {
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RtpHeader *header = (RtpHeader *) buf;
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return ((header->pt >= 64) && (header->pt < 96));
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}
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void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple *tuple) {
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if (RTC::StunPacket::IsStun((const uint8_t *) buf, len)) {
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RTC::StunPacket *packet = RTC::StunPacket::Parse((const uint8_t *) buf, len);
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if (packet == nullptr) {
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WarnL << "parse stun error" << std::endl;
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return;
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}
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_ice_server->ProcessStunPacket(packet, tuple);
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return;
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}
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if (is_dtls(buf)) {
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_dtls_transport->ProcessDtlsData((uint8_t *) buf, len);
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return;
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}
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if (is_rtp(buf)) {
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if (_srtp_session_recv->DecryptSrtp((uint8_t *) buf, &len)) {
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onRtp(buf, len);
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} else {
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RtpHeader *rtp = (RtpHeader *) buf;
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WarnL << "srtp_unprotect rtp failed, pt:" << (int)rtp->pt;
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}
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return;
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}
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if (is_rtcp(buf)) {
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if (_srtp_session_recv->DecryptSrtcp((uint8_t *) buf, &len)) {
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onRtcp(buf, len);
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} else {
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WarnL;
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}
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return;
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}
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}
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void WebRtcTransport::sendRtpPacket(const char *buf, size_t len, bool flush, void *ctx) {
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if (_srtp_session_send) {
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//预留rtx加入的两个字节
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CHECK(len + SRTP_MAX_TRAILER_LEN + 2 <= sizeof(_srtp_buf));
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memcpy(_srtp_buf, buf, len);
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onBeforeEncryptRtp((char *) _srtp_buf, len, ctx);
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if (_srtp_session_send->EncryptRtp(_srtp_buf, &len)) {
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onSendSockData((char *) _srtp_buf, len, flush);
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}
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}
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}
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void WebRtcTransport::sendRtcpPacket(const char *buf, size_t len, bool flush, void *ctx){
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if (_srtp_session_send) {
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CHECK(len + SRTP_MAX_TRAILER_LEN <= sizeof(_srtp_buf));
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memcpy(_srtp_buf, buf, len);
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onBeforeEncryptRtcp((char *) _srtp_buf, len, ctx);
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if (_srtp_session_send->EncryptRtcp(_srtp_buf, &len)) {
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onSendSockData((char *) _srtp_buf, len, flush);
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}
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}
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}
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///////////////////////////////////////////////////////////////////////////////////
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WebRtcTransportImp::Ptr WebRtcTransportImp::create(const EventPoller::Ptr &poller){
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WebRtcTransportImp::Ptr ret(new WebRtcTransportImp(poller), [](WebRtcTransportImp *ptr){
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ptr->onDestory();
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delete ptr;
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});
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ret->onCreate();
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return ret;
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}
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void WebRtcTransportImp::onCreate(){
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WebRtcTransport::onCreate();
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_socket = Socket::createSocket(getPoller(), false);
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//随机端口，绑定全部网卡
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_socket->bindUdpSock(0);
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weak_ptr<WebRtcTransportImp> weak_self = shared_from_this();
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_socket->setOnRead([weak_self](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) mutable {
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auto strong_self = weak_self.lock();
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if (strong_self) {
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strong_self->inputSockData(buf->data(), buf->size(), addr);
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}
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});
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_self = shared_from_this();
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GET_CONFIG(float, timeoutSec, RTC::kTimeOutSec);
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_timer = std::make_shared<Timer>(timeoutSec / 2, [weak_self]() {
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auto strong_self = weak_self.lock();
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if (!strong_self) {
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return false;
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}
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if (strong_self->_alive_ticker.elapsedTime() > timeoutSec * 1000) {
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strong_self->onShutdown(SockException(Err_timeout, "接受rtp和rtcp超时"));
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}
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return true;
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}, getPoller());
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}
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WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcTransport(poller) {
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InfoL << this;
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}
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WebRtcTransportImp::~WebRtcTransportImp() {
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InfoL << this;
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}
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void WebRtcTransportImp::onDestory() {
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WebRtcTransport::onDestory();
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uint64_t duration = _alive_ticker.createdTime() / 1000;
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//流量统计事件广播
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GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold);
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if (_reader) {
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WarnL << "RTC播放器("
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<< _media_info._vhost << "/"
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<< _media_info._app << "/"
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<< _media_info._streamid
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<< ")结束播放,耗时(s):" << duration;
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if (_bytes_usage >= iFlowThreshold * 1024) {
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NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, true, static_cast<SockInfo &>(*_socket));
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}
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}
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if (_push_src) {
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WarnL << "RTC推流器("
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<< _media_info._vhost << "/"
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<< _media_info._app << "/"
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<< _media_info._streamid
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<< ")结束推流,耗时(s):" << duration;
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if (_bytes_usage >= iFlowThreshold * 1024) {
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NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, false, static_cast<SockInfo &>(*_socket));
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}
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}
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}
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void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src, const MediaInfo &info, bool is_play) {
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assert(src);
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_media_info = info;
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if (is_play) {
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_play_src = src;
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} else {
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_push_src = src;
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}
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}
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void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush) {
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auto ptr = BufferRaw::create();
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ptr->assign(buf, len);
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_socket->send(ptr, (struct sockaddr *)(dst), sizeof(struct sockaddr), flush);
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}
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///////////////////////////////////////////////////////////////////
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bool WebRtcTransportImp::canSendRtp() const{
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auto &sdp = getSdp(SdpType::answer);
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return _play_src && (sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::sendonly);
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}
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bool WebRtcTransportImp::canRecvRtp() const{
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auto &sdp = getSdp(SdpType::answer);
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return _push_src && (sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::recvonly);
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}
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void WebRtcTransportImp::onStartWebRTC() {
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//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
|
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for (auto &m_answer : getSdp(SdpType::answer).media) {
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auto m_offer = getSdp(SdpType::offer).getMedia(m_answer.type);
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auto track = std::make_shared<MediaTrack>();
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track->media = &m_answer;
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track->answer_ssrc_rtp = m_answer.getRtpSSRC();
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track->answer_ssrc_rtx = m_answer.getRtxSSRC();
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track->offer_ssrc_rtp = m_offer->getRtpSSRC();
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track->offer_ssrc_rtx = m_offer->getRtxSSRC();
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track->plan_rtp = &m_answer.plan[0];;
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track->plan_rtx = m_answer.getRelatedRtxPlan(track->plan_rtp->pt);
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track->rtcp_context_send = std::make_shared<RtcpContext>(false);
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//send ssrc --> MediaTrack
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_ssrc_to_track[track->answer_ssrc_rtp] = track;
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_ssrc_to_track[track->answer_ssrc_rtx] = track;
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//recv ssrc --> MediaTrack
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_ssrc_to_track[track->offer_ssrc_rtp] = track;
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_ssrc_to_track[track->offer_ssrc_rtx] = track;
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//rtp pt --> MediaTrack
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_pt_to_track.emplace(track->plan_rtp->pt, std::make_pair(false, track));
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if (track->plan_rtx) {
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//rtx pt --> MediaTrack
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_pt_to_track.emplace(track->plan_rtx->pt, std::make_pair(true, track));
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}
|
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if (m_offer->type != TrackApplication) {
|
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//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);
|
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track->rtp_ext_ctx->setOnGetRtp([this, track](uint8_t pt, uint32_t ssrc, const string &rid) {
|
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//ssrc --> MediaTrack
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_ssrc_to_track[ssrc] = track;
|
||
InfoL << "get rtp, pt:" << (int) pt << ", ssrc:" << ssrc << ", rid:" << rid;
|
||
});
|
||
|
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int index = 0;
|
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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;
|
||
}
|
||
}
|
||
}
|
||
|
||
if (canRecvRtp()) {
|
||
_push_src->setSdp(getSdp(SdpType::answer).toRtspSdp());
|
||
}
|
||
if (canSendRtp()) {
|
||
_reader = _play_src->getRing()->attach(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());
|
||
});
|
||
});
|
||
_reader->setDetachCB([weak_self](){
|
||
auto strongSelf = weak_self.lock();
|
||
if (!strongSelf) {
|
||
return;
|
||
}
|
||
strongSelf->onShutdown(SockException(Err_eof, "rtsp ring buffer detached"));
|
||
});
|
||
|
||
RtcSession rtsp_send_sdp;
|
||
rtsp_send_sdp.loadFrom(_play_src->getSdp(), false);
|
||
for (auto &m : getSdp(SdpType::answer).media) {
|
||
if (m.type == TrackApplication) {
|
||
continue;
|
||
}
|
||
auto rtsp_media = rtsp_send_sdp.getMedia(m.type);
|
||
if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) {
|
||
auto it = _pt_to_track.find(m.plan[0].pt);
|
||
CHECK(it != _pt_to_track.end());
|
||
//记录发送rtp时约定的信息，届时发送rtp时需要修改pt和ssrc
|
||
_type_to_track[m.type] = it->second.second;
|
||
}
|
||
}
|
||
}
|
||
//使用完毕后，释放强引用，这样确保推流器断开后能及时注销媒体
|
||
_play_src = nullptr;
|
||
}
|
||
|
||
void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){
|
||
WebRtcTransport::onCheckSdp(type, sdp);
|
||
if (type != SdpType::answer) {
|
||
//我们只修改answer sdp
|
||
return;
|
||
}
|
||
|
||
//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;
|
||
m.rtcp_addr.port = _socket->get_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;
|
||
}
|
||
//添加answer sdp的ssrc信息
|
||
m.rtp_rtx_ssrc.emplace_back();
|
||
m.rtp_rtx_ssrc[0].ssrc = _play_src->getSsrc(m.type);
|
||
m.rtp_rtx_ssrc[0].cname = RTP_CNAME;
|
||
m.rtp_rtx_ssrc[0].label = RTP_LABEL;
|
||
m.rtp_rtx_ssrc[0].mslabel = RTP_MSLABEL;
|
||
m.rtp_rtx_ssrc[0].msid = RTP_MSID;
|
||
|
||
if (m.getRelatedRtxPlan(m.plan[0].pt)) {
|
||
m.rtp_rtx_ssrc.emplace_back();
|
||
m.rtp_rtx_ssrc[1] = m.rtp_rtx_ssrc[0];
|
||
m.rtp_rtx_ssrc[1].ssrc += RTX_SSRC_OFFSET;
|
||
}
|
||
}
|
||
}
|
||
|
||
void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
|
||
WebRtcTransport::onRtcConfigure(configure);
|
||
|
||
if (_play_src) {
|
||
//这是播放,同时也可能有推流
|
||
configure.video.direction = _push_src ? RtpDirection::sendrecv : RtpDirection::sendonly;
|
||
configure.audio.direction = configure.video.direction;
|
||
configure.setPlayRtspInfo(_play_src->getSdp());
|
||
} else if (_push_src) {
|
||
//这只是推流
|
||
configure.video.direction = RtpDirection::recvonly;
|
||
configure.audio.direction = RtpDirection::recvonly;
|
||
} else {
|
||
throw std::invalid_argument("未设置播放或推流的媒体源");
|
||
}
|
||
|
||
//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;
|
||
candidate->port = _socket->get_local_port();
|
||
candidate->type = "host";
|
||
return candidate;
|
||
}
|
||
|
||
///////////////////////////////////////////////////////////////////
|
||
|
||
class RtpChannel : public RtpTrackImp {
|
||
public:
|
||
RtpChannel(RtpTrackImp::OnSorted cb, function<void(const FCI_NACK &nack)> on_nack) {
|
||
setOnSorted(std::move(cb));
|
||
_nack_ctx.setOnNack(std::move(on_nack));
|
||
}
|
||
|
||
~RtpChannel() override = default;
|
||
|
||
bool inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t len, bool is_rtx){
|
||
if (!is_rtx) {
|
||
RtpHeader *rtp = (RtpHeader *) ptr;
|
||
auto seq = ntohs(rtp->seq);
|
||
//统计rtp接受情况，便于生成nack rtcp包
|
||
_nack_ctx.received(seq);
|
||
//统计rtp收到的情况，好做rr汇报
|
||
_rtcp_context.onRtp(seq, ntohl(rtp->stamp), len);
|
||
}
|
||
return RtpTrack::inputRtp(type, sample_rate, ptr, len);
|
||
}
|
||
|
||
Buffer::Ptr createRtcpRR(RtcpHeader *sr, uint32_t ssrc) {
|
||
_rtcp_context.onRtcp(sr);
|
||
return _rtcp_context.createRtcpRR(ssrc, getSSRC());
|
||
}
|
||
|
||
private:
|
||
NackContext _nack_ctx;
|
||
RtcpContext _rtcp_context{true};
|
||
};
|
||
|
||
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 {
|
||
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;
|
||
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.for_each_nack(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, const MediaTrack::Ptr &track) {
|
||
//rid --> RtpReceiverImp
|
||
auto &ref = track->rtp_channel[rid];
|
||
ref = std::make_shared<RtpChannel>([track, this, rid](RtpPacket::Ptr rtp) mutable {
|
||
onSortedRtp(*track, rid, std::move(rtp));
|
||
}, [track, this, ssrc](const FCI_NACK &nack) mutable {
|
||
onSendNack(*track, nack, ssrc);
|
||
});
|
||
InfoL << "create rtp receiver of ssrc:" << ssrc << ", rid:" << rid << ", codec:" << track->plan_rtp->codec;
|
||
}
|
||
|
||
void WebRtcTransportImp::onRtp(const char *buf, size_t len) {
|
||
_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;
|
||
}
|
||
bool is_rtx = it->second.first;
|
||
auto ssrc = ntohl(rtp->ssrc);
|
||
auto &track = it->second.second;
|
||
|
||
//修改ext id至统一
|
||
string rid;
|
||
track->rtp_ext_ctx->changeRtpExtId(rtp, true, &rid);
|
||
|
||
auto &ref = track->rtp_channel[rid];
|
||
if (!ref) {
|
||
if (is_rtx) {
|
||
//再接收到对应的rtp前，丢弃rtx包
|
||
WarnL << "unknown rtx rtp, rid:" << rid << ", ssrc:" << ssrc << ", codec:" << track->plan_rtp->codec << ", seq:" << ntohs(rtp->seq);
|
||
return;
|
||
}
|
||
createRtpChannel(rid, ssrc, track);
|
||
}
|
||
|
||
if (!is_rtx) {
|
||
//这是普通的rtp数据
|
||
#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
|
||
//解析并排序rtp
|
||
if(!ref){
|
||
InfoL << "ignore no rtp receiver of ssrc:" << ssrc<<" is rtx:"<<rtx;
|
||
return;
|
||
}
|
||
assert(ref);
|
||
ref->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len);
|
||
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 = htons(track.answer_ssrc_rtp);
|
||
rtcp->ssrc_media = htonl(ssrc);
|
||
DebugL << htonl(ssrc) << " " << nack.getPid();
|
||
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 && getSdp(SdpType::answer).supportRtcpFb(SdpConst::kRembRtcpFb)) {
|
||
sendRtcpRemb(rtp->getSSRC(), remb_bit_rate);
|
||
}
|
||
}
|
||
|
||
if (_push_src) {
|
||
if (rtp->type == TrackAudio) {
|
||
//音频
|
||
for (auto &pr : _push_src_simulcast) {
|
||
pr.second->onWrite(rtp, false);
|
||
}
|
||
} else {
|
||
//视频
|
||
auto &src = _push_src_simulcast[rid];
|
||
if (!src) {
|
||
auto stream_id = rid.empty() ? _push_src->getId() : _push_src->getId() + "_" + rid;
|
||
auto src_imp = std::make_shared<RtspMediaSourceImp>(_push_src->getVhost(), _push_src->getApp(), stream_id);
|
||
src_imp->setSdp(_push_src->getSdp());
|
||
src_imp->setProtocolTranslation(_push_src->isRecording(Recorder::type_hls),_push_src->isRecording(Recorder::type_mp4));
|
||
src_imp->setListener(shared_from_this());
|
||
src = src_imp;
|
||
}
|
||
src->onWrite(std::move(rtp), false);
|
||
}
|
||
}
|
||
}
|
||
|
||
///////////////////////////////////////////////////////////////////
|
||
|
||
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(), ntohl(rtp->getHeader()->stamp), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
|
||
track->nack_list.push_back(rtp);
|
||
#if 0
|
||
//此处模拟发送丢包
|
||
if (rtp->type == TrackVideo && rtp->getSeq() % 100 == 0) {
|
||
return;
|
||
}
|
||
#endif
|
||
} else {
|
||
WarnL << "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, size_t &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]++);
|
||
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();
|
||
_self = nullptr;
|
||
}
|
||
|
||
/////////////////////////////////////////////////////////////////////////////////////////////
|
||
|
||
bool WebRtcTransportImp::close(MediaSource &sender, bool force) {
|
||
//此回调在其他线程触发
|
||
if (!force && totalReaderCount(sender)) {
|
||
return false;
|
||
}
|
||
string err = StrPrinter << "close media:" << sender.getSchema() << "/" << sender.getVhost() << "/" << sender.getApp() << "/" << sender.getId() << " " << force;
|
||
onShutdown(SockException(Err_shutdown,err));
|
||
return true;
|
||
}
|
||
|
||
int WebRtcTransportImp::totalReaderCount(MediaSource &sender) {
|
||
auto total_count = 0;
|
||
for (auto &src : _push_src_simulcast) {
|
||
total_count += src.second->totalReaderCount();
|
||
}
|
||
return total_count;
|
||
}
|
||
|
||
MediaOriginType WebRtcTransportImp::getOriginType(MediaSource &sender) const {
|
||
return MediaOriginType::rtc_push;
|
||
}
|
||
|
||
string WebRtcTransportImp::getOriginUrl(MediaSource &sender) const {
|
||
return "";
|
||
}
|
||
|
||
std::shared_ptr<SockInfo> WebRtcTransportImp::getOriginSock(MediaSource &sender) const {
|
||
return const_cast<WebRtcTransportImp *>(this)->shared_from_this();
|
||
}
|
||
|
||
/////////////////////////////////////////////////////////////////////////////////////////////
|
||
|
||
string WebRtcTransportImp::get_local_ip() {
|
||
return getSdp(SdpType::answer).media[0].candidate[0].address;
|
||
}
|
||
|
||
uint16_t WebRtcTransportImp::get_local_port() {
|
||
return _socket->get_local_port();
|
||
}
|
||
|
||
string WebRtcTransportImp::get_peer_ip() {
|
||
return SockUtil::inet_ntoa(((struct sockaddr_in *) getSelectedTuple())->sin_addr);
|
||
}
|
||
|
||
uint16_t WebRtcTransportImp::get_peer_port() {
|
||
return ntohs(((struct sockaddr_in *) getSelectedTuple())->sin_port);
|
||
}
|
||
|
||
string WebRtcTransportImp::getIdentifier() const {
|
||
return StrPrinter << this;
|
||
} |