mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-22 19:00:01 +08:00
705 lines
24 KiB
C++
705 lines
24 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 "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 RTX_CNAME "zlmediakit-rtx"
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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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TraceL << ssrc << " " << bit_rate;
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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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}
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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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WarnL;
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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(char *buf, size_t len, bool flush, uint8_t pt) {
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const uint8_t *p = (uint8_t *) buf;
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bool ret = false;
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if (_srtp_session_send) {
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ret = _srtp_session_send->EncryptRtp(&p, &len, pt);
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}
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if (ret) {
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onSendSockData((char *) p, len, flush);
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}
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}
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void WebRtcTransport::sendRtcpPacket(char *buf, size_t len, bool flush){
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const uint8_t *p = (uint8_t *) buf;
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bool ret = false;
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if (_srtp_session_send) {
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ret = _srtp_session_send->EncryptRtcp(&p, &len);
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}
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if (ret) {
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onSendSockData((char *) p, len, flush);
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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 (_play_src) {
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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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for (auto &m : getSdp(SdpType::offer).media) {
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if (m.type == TrackVideo) {
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_recv_video_ssrc = m.rtp_ssrc.ssrc;
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}
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for (auto &plan : m.plan) {
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auto hit_pan = getSdp(SdpType::answer).getMedia(m.type)->getPlan(plan.pt);
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if (!hit_pan) {
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continue;
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}
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//获取offer端rtp的ssrc和pt相关信息
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auto &ref = _rtp_info_pt[plan.pt];
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_rtp_info_ssrc[m.rtp_ssrc.ssrc] = &ref;
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ref.plan = &plan;
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ref.media = &m;
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ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid;
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ref.rtcp_context_recv = std::make_shared<RtcpContext>(ref.plan->sample_rate, true);
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ref.rtcp_context_send = std::make_shared<RtcpContext>(ref.plan->sample_rate, false);
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ref.receiver = std::make_shared<RtpReceiverImp>([&ref, this](RtpPacket::Ptr rtp) {
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onSortedRtp(ref, std::move(rtp));
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}, [ref, this](const RtpPacket::Ptr &rtp) {
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onBeforeSortedRtp(ref, rtp);
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});
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}
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}
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if (canRecvRtp()) {
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_push_src->setSdp(getSdp(SdpType::answer).toRtspSdp());
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}
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if (canSendRtp()) {
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_reader = _play_src->getRing()->attach(getPoller(), true);
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weak_ptr<WebRtcTransportImp> weak_self = shared_from_this();
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_reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt) {
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auto strongSelf = weak_self.lock();
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if (!strongSelf) {
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return;
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}
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size_t i = 0;
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pkt->for_each([&](const RtpPacket::Ptr &rtp) {
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strongSelf->onSendRtp(rtp, ++i == pkt->size());
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});
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});
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}
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}
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void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){
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WebRtcTransport::onCheckSdp(type, sdp);
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if (type != SdpType::answer) {
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return;
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}
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GET_CONFIG(string, extern_ip, RTC::kExternIP);
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for (auto &m : sdp.media) {
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m.addr.reset();
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m.addr.address = extern_ip.empty() ? SockUtil::get_local_ip() : extern_ip;
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m.rtcp_addr.reset();
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m.rtcp_addr.address = m.addr.address;
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m.rtcp_addr.port = _socket->get_local_port();
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m.port = m.rtcp_addr.port;
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sdp.origin.address = m.addr.address;
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}
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if (!canSendRtp()) {
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return;
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||
}
|
||
|
||
RtcSession rtsp_send_sdp;
|
||
rtsp_send_sdp.loadFrom(_play_src->getSdp(), false);
|
||
|
||
for (auto &m : sdp.media) {
|
||
if (m.type == TrackApplication) {
|
||
continue;
|
||
}
|
||
//添加answer sdp的ssrc信息
|
||
m.rtp_ssrc.ssrc = _play_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)) {
|
||
//记录发送rtp的pt
|
||
_send_rtp_pt[m.type] = m.plan[0].pt;
|
||
}
|
||
}
|
||
}
|
||
|
||
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 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) {
|
||
_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 = _rtp_info_ssrc.find(sr->ssrc);
|
||
if (it != _rtp_info_ssrc.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);
|
||
}
|
||
break;
|
||
}
|
||
case RtcpType::RTCP_RR : {
|
||
_alive_ticker.resetTime();
|
||
//对方汇报rtp接收情况
|
||
RtcpRR *rr = (RtcpRR *) rtcp;
|
||
auto it = _rtp_info_ssrc.find(rr->ssrc);
|
||
if (it != _rtp_info_ssrc.end()) {
|
||
auto sr = it->second->rtcp_context_send->createRtcpSR(rr->items.ssrc);
|
||
sendRtcpPacket(sr->data(), sr->size(), true);
|
||
}
|
||
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 = _rtp_info_ssrc.find(*ssrc);
|
||
if (it == _rtp_info_ssrc.end()) {
|
||
continue;
|
||
}
|
||
_rtp_info_pt.erase(it->second->plan->pt);
|
||
_rtp_info_ssrc.erase(it);
|
||
}
|
||
onShutdown(SockException(Err_eof, "rtcp bye message received"));
|
||
break;
|
||
}
|
||
case RtcpType::RTCP_PSFB:
|
||
case RtcpType::RTCP_RTPFB: {
|
||
InfoL << "\n" << rtcp->dumpString();
|
||
break;
|
||
}
|
||
default: break;
|
||
}
|
||
}
|
||
}
|
||
|
||
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 = _rtp_info_pt.find(rtp->pt);
|
||
if (it == _rtp_info_pt.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
|
||
return;
|
||
}
|
||
if (_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(_recv_video_ssrc);
|
||
|
||
//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) {
|
||
sendRtcpRemb(_recv_video_ssrc, remb_bit_rate);
|
||
}
|
||
}
|
||
if (_push_src) {
|
||
_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 == 0xFF) {
|
||
//忽略，对方不支持该编码类型
|
||
return;
|
||
}
|
||
_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
|
||
sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, pt);
|
||
//统计rtp发送情况，好做sr汇报
|
||
_rtp_info_pt[pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
|
||
}
|
||
|
||
void WebRtcTransportImp::onShutdown(const SockException &ex){
|
||
InfoL << ex.what();
|
||
_self = nullptr;
|
||
}
|
||
|
||
/////////////////////////////////////////////////////////////////////////////////////////////
|
||
|
||
bool WebRtcTransportImp::close(MediaSource &sender, bool force) {
|
||
//此回调在其他线程触发
|
||
if(!_push_src || (!force && _push_src->totalReaderCount())){
|
||
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) {
|
||
return _push_src ? _push_src->totalReaderCount() : sender.readerCount();
|
||
}
|
||
|
||
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;
|
||
} |