mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-27 05:38:31 +08:00
824 lines
29 KiB
C++
824 lines
29 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 <set>
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#include <cmath>
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#include <algorithm>
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#include <iomanip>
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#include "Common/config.h"
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#include "RtspPlayer.h"
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#include "Util/MD5.h"
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#include "Util/util.h"
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#include "Util/base64.h"
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#include "Network/sockutil.h"
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using namespace toolkit;
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using namespace mediakit::Client;
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namespace mediakit {
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enum PlayType {
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type_play = 0,
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type_pause,
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type_seek
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};
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RtspPlayer::RtspPlayer(const EventPoller::Ptr &poller) : TcpClient(poller){
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}
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RtspPlayer::~RtspPlayer(void) {
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DebugL << endl;
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}
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void RtspPlayer::sendTeardown(){
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if (alive()) {
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if (!_content_base.empty()) {
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sendRtspRequest("TEARDOWN", _content_base);
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}
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shutdown(SockException(Err_shutdown, "teardown"));
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}
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}
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void RtspPlayer::teardown(){
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sendTeardown();
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_md5_nonce.clear();
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_realm.clear();
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_sdp_track.clear();
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_session_id.clear();
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_content_base.clear();
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RtpReceiver::clear();
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CLEAR_ARR(_rtp_sock);
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CLEAR_ARR(_rtcp_sock);
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CLEAR_ARR(_rtp_seq_start)
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CLEAR_ARR(_rtp_recv_count)
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CLEAR_ARR(_rtp_recv_count)
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CLEAR_ARR(_rtp_seq_now)
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_play_check_timer.reset();
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_rtp_check_timer.reset();
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_cseq_send = 1;
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_on_response = nullptr;
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}
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void RtspPlayer::play(const string &strUrl){
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RtspUrl url;
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if(!url.parse(strUrl)){
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onPlayResult_l(SockException(Err_other,StrPrinter << "illegal rtsp url:" << strUrl),false);
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return;
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}
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teardown();
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if (url._user.size()) {
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(*this)[kRtspUser] = url._user;
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}
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if (url._passwd.size()) {
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(*this)[kRtspPwd] = url._passwd;
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(*this)[kRtspPwdIsMD5] = false;
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}
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_play_url = url._url;
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_rtp_type = (Rtsp::eRtpType)(int)(*this)[kRtpType];
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DebugL << url._url << " " << (url._user.size() ? url._user : "null") << " " << (url._passwd.size() ? url._passwd : "null") << " " << _rtp_type;
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weak_ptr<RtspPlayer> weakSelf = dynamic_pointer_cast<RtspPlayer>(shared_from_this());
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float playTimeOutSec = (*this)[kTimeoutMS].as<int>() / 1000.0f;
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_play_check_timer.reset(new Timer(playTimeOutSec, [weakSelf]() {
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auto strongSelf=weakSelf.lock();
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if(!strongSelf) {
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return false;
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}
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strongSelf->onPlayResult_l(SockException(Err_timeout,"play rtsp timeout"),false);
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return false;
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}, getPoller()));
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if(!(*this)[kNetAdapter].empty()){
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setNetAdapter((*this)[kNetAdapter]);
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}
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startConnect(url._host, url._port, playTimeOutSec);
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}
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void RtspPlayer::onConnect(const SockException &err){
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if(err.getErrCode() != Err_success) {
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onPlayResult_l(err,false);
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return;
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}
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sendOptions();
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}
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void RtspPlayer::onRecv(const Buffer::Ptr& buf) {
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if(_benchmark_mode && !_play_check_timer){
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//在性能测试模式下，如果rtsp握手完毕后，不再解析rtp包
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_rtp_recv_ticker.resetTime();
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return;
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}
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input(buf->data(), buf->size());
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}
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void RtspPlayer::onErr(const SockException &ex) {
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//定时器_pPlayTimer为空后表明握手结束了
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onPlayResult_l(ex,!_play_check_timer);
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}
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// from live555
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bool RtspPlayer::handleAuthenticationFailure(const string ¶msStr) {
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if(!_realm.empty()){
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//已经认证过了
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return false;
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}
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char *realm = new char[paramsStr.size()];
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char *nonce = new char[paramsStr.size()];
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char *stale = new char[paramsStr.size()];
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onceToken token(nullptr,[&](){
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delete[] realm;
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delete[] nonce;
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delete[] stale;
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});
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if (sscanf(paramsStr.data(), "Digest realm=\"%[^\"]\", nonce=\"%[^\"]\", stale=%[a-zA-Z]", realm, nonce, stale) == 3) {
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_realm = (const char *)realm;
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_md5_nonce = (const char *)nonce;
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return true;
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}
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if (sscanf(paramsStr.data(), "Digest realm=\"%[^\"]\", nonce=\"%[^\"]\"", realm, nonce) == 2) {
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_realm = (const char *)realm;
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_md5_nonce = (const char *)nonce;
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return true;
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}
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if (sscanf(paramsStr.data(), "Basic realm=\"%[^\"]\"", realm) == 1) {
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_realm = (const char *)realm;
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return true;
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}
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return false;
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}
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bool RtspPlayer::handleResponse(const string &cmd, const Parser &parser){
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string authInfo = parser["WWW-Authenticate"];
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//发送DESCRIBE命令后的回复
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if ((parser.Url() == "401") && handleAuthenticationFailure(authInfo)) {
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sendOptions();
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return false;
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}
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if(parser.Url() == "302" || parser.Url() == "301"){
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auto newUrl = parser["Location"];
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if(newUrl.empty()){
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throw std::runtime_error("未找到Location字段(跳转url)");
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}
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play(newUrl);
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return false;
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}
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if (parser.Url() != "200") {
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throw std::runtime_error(StrPrinter << cmd << ":" << parser.Url() << " " << parser.Tail() << endl);
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}
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return true;
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}
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void RtspPlayer::handleResDESCRIBE(const Parser& parser) {
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if (!handleResponse("DESCRIBE", parser)) {
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return;
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}
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_content_base = parser["Content-Base"];
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if(_content_base.empty()){
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_content_base = _play_url;
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}
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if (_content_base.back() == '/') {
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_content_base.pop_back();
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}
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SdpParser sdpParser(parser.Content());
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//解析sdp
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_sdp_track = sdpParser.getAvailableTrack();
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auto title = sdpParser.getTrack(TrackTitle);
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if (_sdp_track.empty()) {
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throw std::runtime_error("无有效的Sdp Track");
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}
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if (!onCheckSDP(sdpParser.toString())) {
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throw std::runtime_error("onCheckSDP faied");
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}
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sendSetup(0);
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}
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//有必要的情况下创建udp端口
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void RtspPlayer::createUdpSockIfNecessary(int track_idx){
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auto &rtpSockRef = _rtp_sock[track_idx];
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auto &rtcpSockRef = _rtcp_sock[track_idx];
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if (!rtpSockRef || !rtcpSockRef) {
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std::pair<Socket::Ptr, Socket::Ptr> pr = std::make_pair(createSocket(), createSocket());
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makeSockPair(pr, get_local_ip());
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rtpSockRef = pr.first;
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rtcpSockRef = pr.second;
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}
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}
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//发送SETUP命令
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void RtspPlayer::sendSetup(unsigned int track_idx) {
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_on_response = std::bind(&RtspPlayer::handleResSETUP, this, placeholders::_1, track_idx);
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auto &track = _sdp_track[track_idx];
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auto baseUrl = _content_base + "/" + track->_control_surffix;
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if (track->_control.find("://") != string::npos) {
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baseUrl = track->_control;
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}
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switch (_rtp_type) {
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case Rtsp::RTP_TCP: {
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sendRtspRequest("SETUP",baseUrl,{"Transport",StrPrinter << "RTP/AVP/TCP;unicast;interleaved=" << track->_type * 2 << "-" << track->_type * 2 + 1});
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}
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break;
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case Rtsp::RTP_MULTICAST: {
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sendRtspRequest("SETUP",baseUrl,{"Transport","RTP/AVP;multicast"});
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}
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break;
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case Rtsp::RTP_UDP: {
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createUdpSockIfNecessary(track_idx);
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sendRtspRequest("SETUP", baseUrl, {"Transport",
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StrPrinter << "RTP/AVP;unicast;client_port="
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<< _rtp_sock[track_idx]->get_local_port() << "-"
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<< _rtcp_sock[track_idx]->get_local_port()});
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}
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break;
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default:
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break;
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}
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}
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void RtspPlayer::handleResSETUP(const Parser &parser, unsigned int track_idx) {
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if (parser.Url() != "200") {
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throw std::runtime_error(StrPrinter << "SETUP:" << parser.Url() << " " << parser.Tail() << endl);
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}
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if (track_idx == 0) {
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_session_id = parser["Session"];
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_session_id.append(";");
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_session_id = FindField(_session_id.data(), nullptr, ";");
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}
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auto strTransport = parser["Transport"];
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if (strTransport.find("TCP") != string::npos || strTransport.find("interleaved") != string::npos) {
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_rtp_type = Rtsp::RTP_TCP;
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} else if (strTransport.find("multicast") != string::npos) {
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_rtp_type = Rtsp::RTP_MULTICAST;
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} else {
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_rtp_type = Rtsp::RTP_UDP;
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}
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auto transport_map = Parser::parseArgs(strTransport, ";", "=");
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RtspSplitter::enableRecvRtp(_rtp_type == Rtsp::RTP_TCP);
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string ssrc = transport_map["ssrc"];
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if(!ssrc.empty()){
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sscanf(ssrc.data(), "%x", &_sdp_track[track_idx]->_ssrc);
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} else{
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_sdp_track[track_idx]->_ssrc = 0;
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}
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if (_rtp_type == Rtsp::RTP_TCP) {
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int interleaved_rtp, interleaved_rtcp;
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sscanf(transport_map["interleaved"].data(), "%d-%d", &interleaved_rtp, &interleaved_rtcp);
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_sdp_track[track_idx]->_interleaved = interleaved_rtp;
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} else {
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auto port_str = transport_map[(_rtp_type == Rtsp::RTP_MULTICAST ? "port" : "server_port")];
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int rtp_port, rtcp_port;
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sscanf(port_str.data(), "%d-%d", &rtp_port, &rtcp_port);
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auto &pRtpSockRef = _rtp_sock[track_idx];
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auto &pRtcpSockRef = _rtcp_sock[track_idx];
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if (_rtp_type == Rtsp::RTP_MULTICAST) {
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//udp组播
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auto multiAddr = transport_map["destination"];
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pRtpSockRef = createSocket();
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if (!pRtpSockRef->bindUdpSock(rtp_port, multiAddr.data())) {
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pRtpSockRef.reset();
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throw std::runtime_error("open udp sock err");
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}
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auto fd = pRtpSockRef->rawFD();
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if (-1 == SockUtil::joinMultiAddrFilter(fd, multiAddr.data(), get_peer_ip().data(),get_local_ip().data())) {
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SockUtil::joinMultiAddr(fd, multiAddr.data(),get_local_ip().data());
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}
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} else {
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createUdpSockIfNecessary(track_idx);
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//udp单播
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struct sockaddr_in rtpto;
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rtpto.sin_port = ntohs(rtp_port);
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rtpto.sin_family = AF_INET;
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rtpto.sin_addr.s_addr = inet_addr(get_peer_ip().data());
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pRtpSockRef->setSendPeerAddr((struct sockaddr *)&(rtpto));
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//发送rtp打洞包
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pRtpSockRef->send("\xce\xfa\xed\xfe", 4);
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//设置rtcp发送目标，为后续发送rtcp做准备
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rtpto.sin_port = ntohs(rtcp_port);
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rtpto.sin_family = AF_INET;
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rtpto.sin_addr.s_addr = inet_addr(get_peer_ip().data());
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pRtcpSockRef->setSendPeerAddr((struct sockaddr *)&(rtpto));
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}
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auto srcIP = inet_addr(get_peer_ip().data());
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weak_ptr<RtspPlayer> weakSelf = dynamic_pointer_cast<RtspPlayer>(shared_from_this());
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//设置rtp over udp接收回调处理函数
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pRtpSockRef->setOnRead([srcIP, track_idx, weakSelf](const Buffer::Ptr &buf, struct sockaddr *addr , int addr_len) {
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auto strongSelf = weakSelf.lock();
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if (!strongSelf) {
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return;
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}
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if (((struct sockaddr_in *) addr)->sin_addr.s_addr != srcIP) {
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WarnL << "收到其他地址的rtp数据:" << SockUtil::inet_ntoa(((struct sockaddr_in *) addr)->sin_addr);
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return;
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}
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strongSelf->handleOneRtp(track_idx, strongSelf->_sdp_track[track_idx]->_type,
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strongSelf->_sdp_track[track_idx]->_samplerate, (unsigned char *) buf->data(), buf->size());
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});
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if(pRtcpSockRef) {
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//设置rtcp over udp接收回调处理函数
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pRtcpSockRef->setOnRead([srcIP, track_idx, weakSelf](const Buffer::Ptr &buf, struct sockaddr *addr , int addr_len) {
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auto strongSelf = weakSelf.lock();
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if (!strongSelf) {
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return;
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}
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if (((struct sockaddr_in *) addr)->sin_addr.s_addr != srcIP) {
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WarnL << "收到其他地址的rtcp数据:" << SockUtil::inet_ntoa(((struct sockaddr_in *) addr)->sin_addr);
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return;
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}
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strongSelf->onRtcpPacket(track_idx, strongSelf->_sdp_track[track_idx], (unsigned char *) buf->data(), buf->size());
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});
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}
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}
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if (track_idx < _sdp_track.size() - 1) {
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//需要继续发送SETUP命令
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sendSetup(track_idx + 1);
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return;
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}
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//所有setup命令发送完毕
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//发送play命令
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sendPause(type_play, 0);
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}
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void RtspPlayer::sendDescribe() {
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//发送DESCRIBE命令后处理函数:handleResDESCRIBE
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_on_response = std::bind(&RtspPlayer::handleResDESCRIBE, this, placeholders::_1);
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sendRtspRequest("DESCRIBE", _play_url, {"Accept", "application/sdp"});
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}
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void RtspPlayer::sendOptions(){
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_on_response = [this](const Parser& parser){
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if (!handleResponse("OPTIONS", parser)) {
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return;
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}
|
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//获取服务器支持的命令
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_supported_cmd.clear();
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auto public_val = split(parser["Public"],",");
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for(auto &cmd : public_val){
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trim(cmd);
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_supported_cmd.emplace(cmd);
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}
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//发送Describe请求，获取sdp
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sendDescribe();
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};
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sendRtspRequest("OPTIONS", _play_url);
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}
|
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void RtspPlayer::sendKeepAlive(){
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_on_response = [](const Parser &parser) {};
|
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if(_supported_cmd.find("GET_PARAMETER") != _supported_cmd.end()){
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//支持GET_PARAMETER，用此命令保活
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sendRtspRequest("GET_PARAMETER", _content_base);
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}else{
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//不支持GET_PARAMETER，用OPTIONS命令保活
|
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sendRtspRequest("OPTIONS", _play_url);
|
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}
|
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}
|
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|
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void RtspPlayer::sendPause(int type , uint32_t seekMS){
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_on_response = std::bind(&RtspPlayer::handleResPAUSE, this, placeholders::_1, type);
|
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//开启或暂停rtsp
|
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switch (type){
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case type_pause:
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sendRtspRequest("PAUSE", _content_base);
|
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break;
|
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case type_play:
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sendRtspRequest("PLAY", _content_base);
|
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break;
|
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case type_seek:
|
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sendRtspRequest("PLAY", _content_base, {"Range",StrPrinter << "npt=" << setiosflags(ios::fixed) << setprecision(2) << seekMS / 1000.0 << "-"});
|
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break;
|
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default:
|
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WarnL << "unknown type : " << type;
|
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_on_response = nullptr;
|
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break;
|
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}
|
||
}
|
||
|
||
void RtspPlayer::pause(bool pause_flag) {
|
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sendPause(pause_flag ? type_pause : type_seek, getProgressMilliSecond());
|
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}
|
||
|
||
void RtspPlayer::handleResPAUSE(const Parser& parser,int type) {
|
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if (parser.Url() != "200") {
|
||
switch (type) {
|
||
case type_pause:
|
||
WarnL << "Pause failed:" << parser.Url() << " " << parser.Tail() << endl;
|
||
break;
|
||
case type_play:
|
||
WarnL << "Play failed:" << parser.Url() << " " << parser.Tail() << endl;
|
||
onPlayResult_l(SockException(Err_shutdown, StrPrinter << "rtsp play failed:" << parser.Url() << " " << parser.Tail() ), !_play_check_timer);
|
||
break;
|
||
case type_seek:
|
||
WarnL << "Seek failed:" << parser.Url() << " " << parser.Tail() << endl;
|
||
break;
|
||
}
|
||
return;
|
||
}
|
||
|
||
if (type == type_pause) {
|
||
//暂停成功！
|
||
_rtp_check_timer.reset();
|
||
return;
|
||
}
|
||
|
||
//play或seek成功
|
||
uint32_t iSeekTo = 0;
|
||
//修正时间轴
|
||
auto strRange = parser["Range"];
|
||
if (strRange.size()) {
|
||
auto strStart = FindField(strRange.data(), "npt=", "-");
|
||
if (strStart == "now") {
|
||
strStart = "0";
|
||
}
|
||
iSeekTo = (uint32_t)(1000 * atof(strStart.data()));
|
||
DebugL << "seekTo(ms):" << iSeekTo;
|
||
}
|
||
|
||
onPlayResult_l(SockException(Err_success, type == type_seek ? "resum rtsp success" : "rtsp play success"), !_play_check_timer);
|
||
}
|
||
|
||
void RtspPlayer::onWholeRtspPacket(Parser &parser) {
|
||
try {
|
||
decltype(_on_response) func;
|
||
_on_response.swap(func);
|
||
if(func){
|
||
func(parser);
|
||
}
|
||
parser.Clear();
|
||
} catch (std::exception &err) {
|
||
//定时器_pPlayTimer为空后表明握手结束了
|
||
onPlayResult_l(SockException(Err_other, err.what()),!_play_check_timer);
|
||
}
|
||
}
|
||
|
||
void RtspPlayer::onRtpPacket(const char *data, size_t len) {
|
||
int trackIdx = -1;
|
||
uint8_t interleaved = data[1];
|
||
if(interleaved %2 == 0){
|
||
trackIdx = getTrackIndexByInterleaved(interleaved);
|
||
handleOneRtp(trackIdx, _sdp_track[trackIdx]->_type, _sdp_track[trackIdx]->_samplerate, (unsigned char *)data + 4, len - 4);
|
||
}else{
|
||
trackIdx = getTrackIndexByInterleaved(interleaved - 1);
|
||
onRtcpPacket(trackIdx, _sdp_track[trackIdx], (unsigned char *) data + 4, len - 4);
|
||
}
|
||
}
|
||
|
||
//此处预留rtcp处理函数
|
||
void RtspPlayer::onRtcpPacket(int track_idx, SdpTrack::Ptr &track, unsigned char *data, size_t len){}
|
||
|
||
#if 0
|
||
//改代码提取自FFmpeg，参考之
|
||
// Receiver Report
|
||
avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
|
||
avio_w8(pb, RTCP_RR);
|
||
avio_wb16(pb, 7); /* length in words - 1 */
|
||
// our own SSRC: we use the server's SSRC + 1 to avoid conflicts
|
||
avio_wb32(pb, s->ssrc + 1);
|
||
avio_wb32(pb, s->ssrc); // server SSRC
|
||
// some placeholders we should really fill...
|
||
// RFC 1889/p64
|
||
extended_max = stats->cycles + stats->max_seq;
|
||
expected = extended_max - stats->base_seq;
|
||
lost = expected - stats->received;
|
||
lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
|
||
expected_interval = expected - stats->expected_prior;
|
||
stats->expected_prior = expected;
|
||
received_interval = stats->received - stats->received_prior;
|
||
stats->received_prior = stats->received;
|
||
lost_interval = expected_interval - received_interval;
|
||
if (expected_interval == 0 || lost_interval <= 0)
|
||
fraction = 0;
|
||
else
|
||
fraction = (lost_interval << 8) / expected_interval;
|
||
|
||
fraction = (fraction << 24) | lost;
|
||
|
||
avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
|
||
avio_wb32(pb, extended_max); /* max sequence received */
|
||
avio_wb32(pb, stats->jitter >> 4); /* jitter */
|
||
|
||
if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) {
|
||
avio_wb32(pb, 0); /* last SR timestamp */
|
||
avio_wb32(pb, 0); /* delay since last SR */
|
||
} else {
|
||
uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
|
||
uint32_t delay_since_last = av_rescale(av_gettime_relative() - s->last_rtcp_reception_time,
|
||
65536, AV_TIME_BASE);
|
||
|
||
avio_wb32(pb, middle_32_bits); /* last SR timestamp */
|
||
avio_wb32(pb, delay_since_last); /* delay since last SR */
|
||
}
|
||
|
||
// CNAME
|
||
avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
|
||
avio_w8(pb, RTCP_SDES);
|
||
len = strlen(s->hostname);
|
||
avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
|
||
avio_wb32(pb, s->ssrc + 1);
|
||
avio_w8(pb, 0x01);
|
||
avio_w8(pb, len);
|
||
avio_write(pb, s->hostname, len);
|
||
avio_w8(pb, 0); /* END */
|
||
// padding
|
||
for (len = (7 + len) % 4; len % 4; len++)
|
||
avio_w8(pb, 0);
|
||
#endif
|
||
|
||
void RtspPlayer::sendReceiverReport(bool over_tcp, int track_idx){
|
||
static const char s_cname[] = "ZLMediaKitRtsp";
|
||
uint8_t aui8Rtcp[4 + 32 + 10 + sizeof(s_cname) + 1] = {0};
|
||
uint8_t *pui8Rtcp_RR = aui8Rtcp + 4, *pui8Rtcp_SDES = pui8Rtcp_RR + 32;
|
||
auto &track = _sdp_track[track_idx];
|
||
auto &counter = _rtcp_counter[track_idx];
|
||
|
||
aui8Rtcp[0] = '$';
|
||
aui8Rtcp[1] = track->_interleaved + 1;
|
||
aui8Rtcp[2] = (sizeof(aui8Rtcp) - 4) >> 8;
|
||
aui8Rtcp[3] = (sizeof(aui8Rtcp) - 4) & 0xFF;
|
||
|
||
pui8Rtcp_RR[0] = 0x81;/* 1 report block */
|
||
pui8Rtcp_RR[1] = 0xC9;//RTCP_RR
|
||
pui8Rtcp_RR[2] = 0x00;
|
||
pui8Rtcp_RR[3] = 0x07;/* length in words - 1 */
|
||
|
||
auto track_ssrc = track->_ssrc ? track->_ssrc : getSSRC(track_idx);
|
||
// our own SSRC: we use the server's SSRC + 1 to avoid conflicts
|
||
uint32_t ssrc = htonl(track_ssrc + 1);
|
||
memcpy(&pui8Rtcp_RR[4], &ssrc, 4);
|
||
|
||
// server SSRC
|
||
ssrc = htonl(track_ssrc);
|
||
memcpy(&pui8Rtcp_RR[8], &ssrc, 4);
|
||
|
||
//FIXME: 8 bits of fraction, 24 bits of total packets lost
|
||
pui8Rtcp_RR[12] = 0x00;
|
||
pui8Rtcp_RR[13] = 0x00;
|
||
pui8Rtcp_RR[14] = 0x00;
|
||
pui8Rtcp_RR[15] = 0x00;
|
||
|
||
//FIXME: max sequence received
|
||
uint16_t cycleCount = (uint16_t)getCycleCount(track_idx);
|
||
pui8Rtcp_RR[16] = (cycleCount >> 8) & 0xFF;
|
||
pui8Rtcp_RR[17] = cycleCount & 0xFF;
|
||
pui8Rtcp_RR[18] = counter.pktCnt >> 8;
|
||
pui8Rtcp_RR[19] = counter.pktCnt & 0xFF;
|
||
|
||
uint32_t jitter = htonl((uint32_t)getJitterSize(track_idx));
|
||
//FIXME: jitter
|
||
memcpy(pui8Rtcp_RR + 20, &jitter, 4);
|
||
/* last SR timestamp */
|
||
memcpy(pui8Rtcp_RR + 24, &counter.lastTimeStamp, 4);
|
||
uint32_t msInc = htonl(ntohl(counter.timeStamp) - ntohl(counter.lastTimeStamp));
|
||
/* delay since last SR */
|
||
memcpy(pui8Rtcp_RR + 28, &msInc, 4);
|
||
|
||
// CNAME
|
||
pui8Rtcp_SDES[0] = 0x81;
|
||
pui8Rtcp_SDES[1] = 0xCA;
|
||
pui8Rtcp_SDES[2] = 0x00;
|
||
pui8Rtcp_SDES[3] = 0x06;
|
||
|
||
memcpy(&pui8Rtcp_SDES[4], &ssrc, 4);
|
||
|
||
pui8Rtcp_SDES[8] = 0x01;
|
||
pui8Rtcp_SDES[9] = 0x0f;
|
||
memcpy(&pui8Rtcp_SDES[10], s_cname, sizeof(s_cname));
|
||
pui8Rtcp_SDES[10 + sizeof(s_cname)] = 0x00;
|
||
|
||
if (over_tcp) {
|
||
send(obtainBuffer((char *) aui8Rtcp, sizeof(aui8Rtcp)));
|
||
} else if (_rtcp_sock[track_idx]) {
|
||
_rtcp_sock[track_idx]->send((char *) aui8Rtcp + 4, sizeof(aui8Rtcp) - 4);
|
||
}
|
||
}
|
||
|
||
void RtspPlayer::onRtpSorted(const RtpPacket::Ptr &rtppt, int trackidx){
|
||
//统计丢包率
|
||
if (_rtp_seq_start[trackidx] == 0 || rtppt->sequence < _rtp_seq_start[trackidx]) {
|
||
_rtp_seq_start[trackidx] = rtppt->sequence;
|
||
_rtp_recv_count[trackidx] = 0;
|
||
}
|
||
_rtp_recv_count[trackidx] ++;
|
||
_rtp_seq_now[trackidx] = rtppt->sequence;
|
||
_stamp[trackidx] = rtppt->timeStamp;
|
||
//计算相对时间戳
|
||
onRecvRTP_l(rtppt, _sdp_track[trackidx]);
|
||
}
|
||
|
||
float RtspPlayer::getPacketLossRate(TrackType type) const{
|
||
try {
|
||
auto track_idx = getTrackIndexByTrackType(type);
|
||
if (_rtp_seq_now[track_idx] - _rtp_seq_start[track_idx] + 1 == 0) {
|
||
return 0;
|
||
}
|
||
return (float)(1.0f - (double) _rtp_recv_count[track_idx] / (_rtp_seq_now[track_idx] - _rtp_seq_start[track_idx] + 1));
|
||
} catch (...) {
|
||
uint64_t totalRecv = 0;
|
||
uint64_t totalSend = 0;
|
||
for (unsigned int i = 0; i < _sdp_track.size(); i++) {
|
||
totalRecv += _rtp_recv_count[i];
|
||
totalSend += (_rtp_seq_now[i] - _rtp_seq_start[i] + 1);
|
||
}
|
||
if (totalSend == 0) {
|
||
return 0;
|
||
}
|
||
return (float)(1.0f - (double) totalRecv / totalSend);
|
||
}
|
||
}
|
||
|
||
uint32_t RtspPlayer::getProgressMilliSecond() const{
|
||
return MAX(_stamp[0],_stamp[1]);
|
||
}
|
||
|
||
void RtspPlayer::seekToMilliSecond(uint32_t ms) {
|
||
sendPause(type_seek,ms);
|
||
}
|
||
|
||
void RtspPlayer::sendRtspRequest(const string &cmd, const string &url, const std::initializer_list<string> &header) {
|
||
string key;
|
||
StrCaseMap header_map;
|
||
int i = 0;
|
||
for(auto &val : header){
|
||
if(++i % 2 == 0){
|
||
header_map.emplace(key,val);
|
||
}else{
|
||
key = val;
|
||
}
|
||
}
|
||
sendRtspRequest(cmd,url,header_map);
|
||
}
|
||
|
||
void RtspPlayer::sendRtspRequest(const string &cmd, const string &url,const StrCaseMap &header_const) {
|
||
auto header = header_const;
|
||
header.emplace("CSeq",StrPrinter << _cseq_send++);
|
||
header.emplace("User-Agent",SERVER_NAME);
|
||
|
||
if(!_session_id.empty()){
|
||
header.emplace("Session", _session_id);
|
||
}
|
||
|
||
if(!_realm.empty() && !(*this)[kRtspUser].empty()){
|
||
if(!_md5_nonce.empty()){
|
||
//MD5认证
|
||
/*
|
||
response计算方法如下：
|
||
RTSP客户端应该使用username + password并计算response如下:
|
||
(1)当password为MD5编码,则
|
||
response = md5( password:nonce:md5(public_method:url) );
|
||
(2)当password为ANSI字符串,则
|
||
response= md5( md5(username:realm:password):nonce:md5(public_method:url) );
|
||
*/
|
||
string encrypted_pwd = (*this)[kRtspPwd];
|
||
if(!(*this)[kRtspPwdIsMD5].as<bool>()){
|
||
encrypted_pwd = MD5((*this)[kRtspUser] + ":" + _realm + ":" + encrypted_pwd).hexdigest();
|
||
}
|
||
auto response = MD5(encrypted_pwd + ":" + _md5_nonce + ":" + MD5(cmd + ":" + url).hexdigest()).hexdigest();
|
||
_StrPrinter printer;
|
||
printer << "Digest ";
|
||
printer << "username=\"" << (*this)[kRtspUser] << "\", ";
|
||
printer << "realm=\"" << _realm << "\", ";
|
||
printer << "nonce=\"" << _md5_nonce << "\", ";
|
||
printer << "uri=\"" << url << "\", ";
|
||
printer << "response=\"" << response << "\"";
|
||
header.emplace("Authorization",printer);
|
||
}else if(!(*this)[kRtspPwdIsMD5].as<bool>()){
|
||
//base64认证
|
||
string authStr = StrPrinter << (*this)[kRtspUser] << ":" << (*this)[kRtspPwd];
|
||
char authStrBase64[1024] = {0};
|
||
av_base64_encode(authStrBase64, sizeof(authStrBase64), (uint8_t *) authStr.data(), (int) authStr.size());
|
||
header.emplace("Authorization",StrPrinter << "Basic " << authStrBase64 );
|
||
}
|
||
}
|
||
|
||
_StrPrinter printer;
|
||
printer << cmd << " " << url << " RTSP/1.0\r\n";
|
||
for (auto &pr : header){
|
||
printer << pr.first << ": " << pr.second << "\r\n";
|
||
}
|
||
printer << "\r\n";
|
||
SockSender::send(std::move(printer));
|
||
}
|
||
|
||
void RtspPlayer::onRecvRTP_l(const RtpPacket::Ptr &rtp, const SdpTrack::Ptr &track) {
|
||
_rtp_recv_ticker.resetTime();
|
||
onRecvRTP(rtp, track);
|
||
|
||
int track_idx = getTrackIndexByTrackType(rtp->type);
|
||
RtcpCounter &counter = _rtcp_counter[track_idx];
|
||
counter.pktCnt = rtp->sequence;
|
||
auto &ticker = _rtcp_send_ticker[track_idx];
|
||
if (ticker.elapsedTime() > 3 * 1000) {
|
||
//每3秒发送一次心跳，rtcp与rtsp信令轮流心跳，该特性用于兼容issue:642
|
||
if (_send_rtcp) {
|
||
counter.lastTimeStamp = counter.timeStamp;
|
||
//直接保存网络字节序
|
||
memcpy(&counter.timeStamp, rtp->data() + 8, 4);
|
||
if (counter.lastTimeStamp != 0) {
|
||
sendReceiverReport(_rtp_type == Rtsp::RTP_TCP, track_idx);
|
||
ticker.resetTime();
|
||
_send_rtcp = false;
|
||
}
|
||
} else {
|
||
//有些rtsp服务器需要rtcp保活，有些需要发送信令保活
|
||
if (track_idx == 0) {
|
||
//只需要发送一次心跳信令包
|
||
sendKeepAlive();
|
||
ticker.resetTime();
|
||
_send_rtcp = true;
|
||
}
|
||
}
|
||
}
|
||
}
|
||
|
||
void RtspPlayer::onPlayResult_l(const SockException &ex , bool handshake_done) {
|
||
if (ex.getErrCode() == Err_shutdown) {
|
||
//主动shutdown的，不触发回调
|
||
return;
|
||
}
|
||
|
||
WarnL << ex.getErrCode() << " " << ex.what();
|
||
if (!handshake_done) {
|
||
//开始播放阶段
|
||
_play_check_timer.reset();
|
||
onPlayResult(ex);
|
||
//是否为性能测试模式
|
||
_benchmark_mode = (*this)[Client::kBenchmarkMode].as<int>();
|
||
} else if (ex) {
|
||
//播放成功后异常断开回调
|
||
onShutdown(ex);
|
||
} else {
|
||
//恢复播放
|
||
onResume();
|
||
}
|
||
|
||
if (!ex) {
|
||
//播放成功，恢复rtp接收超时定时器
|
||
_rtp_recv_ticker.resetTime();
|
||
auto timeoutMS = (*this)[kMediaTimeoutMS].as<uint64_t>();
|
||
weak_ptr<RtspPlayer> weakSelf = dynamic_pointer_cast<RtspPlayer>(shared_from_this());
|
||
auto lam = [weakSelf, timeoutMS]() {
|
||
auto strongSelf = weakSelf.lock();
|
||
if (!strongSelf) {
|
||
return false;
|
||
}
|
||
if (strongSelf->_rtp_recv_ticker.elapsedTime() > timeoutMS) {
|
||
//接收rtp媒体数据包超时
|
||
strongSelf->onPlayResult_l(SockException(Err_timeout, "receive rtp timeout"), true);
|
||
return false;
|
||
}
|
||
return true;
|
||
};
|
||
//创建rtp数据接收超时检测定时器
|
||
_rtp_check_timer = std::make_shared<Timer>(timeoutMS / 2000.0f, lam, getPoller());
|
||
} else {
|
||
sendTeardown();
|
||
}
|
||
}
|
||
|
||
int RtspPlayer::getTrackIndexByInterleaved(int interleaved) const {
|
||
for (unsigned int i = 0; i < _sdp_track.size(); i++) {
|
||
if (_sdp_track[i]->_interleaved == interleaved) {
|
||
return i;
|
||
}
|
||
}
|
||
if (_sdp_track.size() == 1) {
|
||
return 0;
|
||
}
|
||
throw SockException(Err_shutdown, StrPrinter << "no such track with interleaved:" << interleaved);
|
||
}
|
||
|
||
int RtspPlayer::getTrackIndexByTrackType(TrackType track_type) const {
|
||
for (unsigned int i = 0; i < _sdp_track.size(); i++) {
|
||
if (_sdp_track[i]->_type == track_type) {
|
||
return i;
|
||
}
|
||
}
|
||
if (_sdp_track.size() == 1) {
|
||
return 0;
|
||
}
|
||
throw SockException(Err_shutdown, StrPrinter << "no such track with type:" << (int) track_type);
|
||
}
|
||
|
||
} /* namespace mediakit */
|