876 lines
32 KiB
C++
876 lines
32 KiB
C++
/*
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* Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
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*
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* This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
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*
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* Use of this source code is governed by MIT-like 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 "RtspPlayer.h"
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#include "Common/config.h"
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#include "Rtcp/Rtcp.h"
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#include "Rtcp/RtcpContext.h"
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#include "RtspDemuxer.h"
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#include "RtspMediaSource.h"
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#include "RtspPlayerImp.h"
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#include "Util/MD5.h"
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#include "Util/base64.h"
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#include <algorithm>
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#include <cmath>
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#include <iomanip>
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#include <set>
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using namespace toolkit;
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using namespace std;
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namespace mediakit {
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enum PlayType { type_play = 0, type_pause, type_seek, type_speed };
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enum class BeatType : uint32_t { both = 0, rtcp, cmd };
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RtspPlayer::RtspPlayer(const EventPoller::Ptr &poller)
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: TcpClient(poller) {}
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RtspPlayer::~RtspPlayer(void) {
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DebugL;
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}
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void RtspPlayer::sendTeardown() {
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if (alive()) {
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if (!_control_url.empty()) {
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sendRtspRequest("TEARDOWN", _control_url);
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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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_rtcp_context.clear();
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CLEAR_ARR(_rtp_sock);
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CLEAR_ARR(_rtcp_sock);
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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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try {
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url.parse(strUrl);
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} catch (std::exception &ex) {
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onPlayResult_l(SockException(Err_other, StrPrinter << "illegal rtsp url:" << ex.what()), 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)[Client::kRtspUser] = url._user;
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}
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if (url._passwd.size()) {
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(*this)[Client::kRtspPwd] = url._passwd;
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(*this)[Client::kRtspPwdIsMD5] = false;
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}
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_play_url = url._url;
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_rtp_type = (Rtsp::eRtpType)(int)(*this)[Client::kRtpType];
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_beat_type = (*this)[Client::kRtspBeatType].as<int>();
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_beat_interval_ms = (*this)[Client::kBeatIntervalMS].as<int>();
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_speed = (*this)[Client::kRtspSpeed].as<float>();
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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 = static_pointer_cast<RtspPlayer>(shared_from_this());
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float playTimeOutSec = (*this)[Client::kTimeoutMS].as<int>() / 1000.0f;
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_play_check_timer.reset(new Timer(
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playTimeOutSec,
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[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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},
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getPoller()));
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if (!(*this)[Client::kNetAdapter].empty()) {
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setNetAdapter((*this)[Client::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 [AUTO-TRANSLATED:747b5399]
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// In performance test mode, if the RTSP handshake is complete, no RTP packets will be parsed
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_rtp_recv_ticker.resetTime();
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return;
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}
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try {
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input(buf->data(), buf->size());
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} catch (exception &e) {
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SockException ex(Err_other, e.what());
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onPlayResult_l(ex, !_play_check_timer);
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}
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}
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void RtspPlayer::onError(const SockException &ex) {
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// 定时器_pPlayTimer为空后表明握手结束了 [AUTO-TRANSLATED:06a369c2]
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// After the timer _pPlayTimer is empty, it indicates that the handshake is complete
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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 [AUTO-TRANSLATED:f2db5f9c]
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// It has been authenticated
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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 [AUTO-TRANSLATED:39629cf0]
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// The response after sending the DESCRIBE command
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if ((parser.status() == "401") && handleAuthenticationFailure(authInfo)) {
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sendOptions();
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return false;
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}
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if (parser.status() == "302" || parser.status() == "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.status() != "200") {
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throw std::runtime_error(StrPrinter << cmd << ":" << parser.status() << " " << parser.statusStr() << 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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// 解析sdp [AUTO-TRANSLATED:ed3f07fe]
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// Parse SDP
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SdpParser sdpParser(parser.content());
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_control_url = sdpParser.getControlUrl(_content_base);
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string sdp;
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auto play_track = (TrackType)((int)(*this)[Client::kPlayTrack] - 1);
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if (play_track != TrackInvalid) {
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auto track = sdpParser.getTrack(play_track);
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_sdp_track.emplace_back(track);
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auto title_track = sdpParser.getTrack(TrackTitle);
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sdp = (title_track ? title_track->toString() : "") + track->toString();
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} else {
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_sdp_track = sdpParser.getAvailableTrack();
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sdp = sdpParser.toString();
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}
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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(sdp)) {
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throw std::runtime_error("onCheckSDP faied");
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}
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_rtcp_context.clear();
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for (auto &track : _sdp_track) {
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if (track->_pt != 0xff) {
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setPayloadType(_rtcp_context.size(), track->_pt);
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}
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_rtcp_context.emplace_back(std::make_shared<RtcpContextForRecv>());
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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 [AUTO-TRANSLATED:651202fc]
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// Create UDP port if necessary
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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 [AUTO-TRANSLATED:68a7ca33]
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// Send SETUP command
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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 control_url = track->getControlUrl(_content_base);
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switch (_rtp_type) {
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case Rtsp::RTP_TCP: {
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sendRtspRequest(
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"SETUP", control_url,
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{ "Transport", StrPrinter << "RTP/AVP/TCP;unicast;interleaved=" << track->_type * 2 << "-" << track->_type * 2 + 1 << ";mode=play" });
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} break;
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case Rtsp::RTP_MULTICAST: {
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sendRtspRequest("SETUP", control_url, { "Transport", "RTP/AVP;multicast;mode=play" });
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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(
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"SETUP", control_url,
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{ "Transport",
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StrPrinter << "RTP/AVP;unicast;client_port=" << _rtp_sock[track_idx]->get_local_port() << "-" << _rtcp_sock[track_idx]->get_local_port()
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<< ";mode=play" });
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} break;
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default: 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.status() != "200") {
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throw std::runtime_error(StrPrinter << "SETUP:" << parser.status() << " " << parser.statusStr() << 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 = findSubString(_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 [AUTO-TRANSLATED:ccc90d1f]
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// UDP multicast
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auto multiAddr = transport_map["destination"];
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pRtpSockRef = createSocket();
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// 目前组播仅支持ipv4 [AUTO-TRANSLATED:8215bfd2]
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// Currently, multicast only supports IPv4
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if (!pRtpSockRef->bindUdpSock(rtp_port, "0.0.0.0")) {
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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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// 设置rtcp发送端口 [AUTO-TRANSLATED:f39b07bd]
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// Set RTCP send port
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pRtcpSockRef = createSocket();
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// 目前组播仅支持ipv4 [AUTO-TRANSLATED:8215bfd2]
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// Currently, multicast only supports IPv4
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if (!pRtcpSockRef->bindUdpSock(0, "0.0.0.0")) {
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// 分配端口失败 [AUTO-TRANSLATED:59ecd25d]
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// Port allocation failed
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throw runtime_error("open udp socket failed");
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}
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// 设置发送地址和发送端口 [AUTO-TRANSLATED:67e1cb6e]
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// Set send address and send port
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auto dst = SockUtil::make_sockaddr(get_peer_ip().data(), rtcp_port);
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pRtcpSockRef->bindPeerAddr((struct sockaddr *)&(dst));
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} else {
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createUdpSockIfNecessary(track_idx);
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// udp单播 [AUTO-TRANSLATED:7d16a875]
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// UDP unicast
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auto dst = SockUtil::make_sockaddr(get_peer_ip().data(), rtp_port);
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pRtpSockRef->bindPeerAddr((struct sockaddr *)&(dst));
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// 发送rtp打洞包 [AUTO-TRANSLATED:9a79d94f]
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// Send RTP hole punching packet
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pRtpSockRef->send("\xce\xfa\xed\xfe", 4);
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dst = SockUtil::make_sockaddr(get_peer_ip().data(), rtcp_port);
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// 设置rtcp发送目标，为后续发送rtcp做准备 [AUTO-TRANSLATED:70929b8e]
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// Set RTCP send target, prepare for subsequent RTCP sending
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pRtcpSockRef->bindPeerAddr((struct sockaddr *)&(dst));
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}
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auto peer_ip = get_peer_ip();
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weak_ptr<RtspPlayer> weakSelf = static_pointer_cast<RtspPlayer>(shared_from_this());
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// 设置rtp over udp接收回调处理函数 [AUTO-TRANSLATED:6e74b593]
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// Set RTP over UDP receive callback handler
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pRtpSockRef->setOnRead([peer_ip, 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 (SockUtil::inet_ntoa(addr) != peer_ip) {
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WarnL << "收到其他地址的rtp数据:" << SockUtil::inet_ntoa(addr);
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return;
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}
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strongSelf->handleOneRtp(
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track_idx, strongSelf->_sdp_track[track_idx]->_type, strongSelf->_sdp_track[track_idx]->_samplerate, (uint8_t *)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 [AUTO-TRANSLATED:eed55b8e]
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// Set RTCP over UDP receive callback handler
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pRtcpSockRef->setOnRead([peer_ip, 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 (SockUtil::inet_ntoa(addr) != peer_ip) {
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WarnL << "收到其他地址的rtcp数据:" << SockUtil::inet_ntoa(addr);
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return;
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}
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strongSelf->onRtcpPacket(track_idx, strongSelf->_sdp_track[track_idx], (uint8_t *)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 [AUTO-TRANSLATED:d7ea1a7a]
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// Need to continue sending SETUP command
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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 [AUTO-TRANSLATED:be499080]
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// All SETUP commands have been sent
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// 发送play命令 [AUTO-TRANSLATED:47a826d1]
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// Send PLAY command
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if (_speed==0.0f) {
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sendPause(type_play, 0);
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} else {
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sendPause(type_speed, 0);
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}
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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 [AUTO-TRANSLATED:3c2b0ffe]
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// Handle the response to the DESCRIBE command: 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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}
|
||
// 获取服务器支持的命令 [AUTO-TRANSLATED:8a6a12f1]
|
||
// Get the commands supported by the server
|
||
_supported_cmd.clear();
|
||
auto public_val = split(parser["Public"], ",");
|
||
for (auto &cmd : public_val) {
|
||
trim(cmd);
|
||
_supported_cmd.emplace(cmd);
|
||
}
|
||
// 发送Describe请求，获取sdp [AUTO-TRANSLATED:f2e291d1]
|
||
// Send Describe request to get SDP
|
||
sendDescribe();
|
||
};
|
||
sendRtspRequest("OPTIONS", _play_url);
|
||
}
|
||
|
||
void RtspPlayer::sendKeepAlive() {
|
||
_on_response = [](const Parser &parser) {};
|
||
if (_supported_cmd.find("GET_PARAMETER") != _supported_cmd.end()) {
|
||
// 支持GET_PARAMETER，用此命令保活 [AUTO-TRANSLATED:b45cd737]
|
||
// Support GET_PARAMETER, use this command to keep alive
|
||
sendRtspRequest("GET_PARAMETER", _control_url);
|
||
} else {
|
||
// 不支持GET_PARAMETER，用OPTIONS命令保活 [AUTO-TRANSLATED:3391350c]
|
||
// Do not support GET_PARAMETER, use OPTIONS command to keep alive
|
||
sendRtspRequest("OPTIONS", _play_url);
|
||
}
|
||
}
|
||
|
||
void RtspPlayer::sendPause(int type, uint32_t seekMS) {
|
||
_on_response = std::bind(&RtspPlayer::handleResPAUSE, this, placeholders::_1, type);
|
||
// 开启或暂停rtsp [AUTO-TRANSLATED:8ba5b594]
|
||
// Start or pause RTSP
|
||
switch (type) {
|
||
case type_pause: sendRtspRequest("PAUSE", _control_url, {}); break;
|
||
case type_play:
|
||
// sendRtspRequest("PLAY", _content_base);
|
||
// break;
|
||
case type_seek:
|
||
sendRtspRequest("PLAY", _control_url, { "Range", StrPrinter << "npt=" << setiosflags(ios::fixed) << setprecision(2) << seekMS / 1000.0 << "-" });
|
||
break;
|
||
case type_speed:
|
||
speed(_speed);
|
||
break;
|
||
default:
|
||
WarnL << "unknown type : " << type;
|
||
_on_response = nullptr;
|
||
break;
|
||
}
|
||
}
|
||
|
||
void RtspPlayer::pause(bool bPause) {
|
||
sendPause(bPause ? type_pause : type_seek, getProgressMilliSecond());
|
||
}
|
||
|
||
void RtspPlayer::speed(float speed) {
|
||
sendRtspRequest("PLAY", _control_url, { "Scale", StrPrinter << speed });
|
||
}
|
||
|
||
void RtspPlayer::handleResPAUSE(const Parser &parser, int type) {
|
||
if (parser.status() != "200") {
|
||
switch (type) {
|
||
case type_pause: WarnL << "Pause failed:" << parser.status() << " " << parser.statusStr(); break;
|
||
case type_play:
|
||
WarnL << "Play failed:" << parser.status() << " " << parser.statusStr();
|
||
onPlayResult_l(SockException(Err_other, StrPrinter << "rtsp play failed:" << parser.status() << " " << parser.statusStr()), !_play_check_timer);
|
||
break;
|
||
case type_seek: WarnL << "Seek failed:" << parser.status() << " " << parser.statusStr(); break;
|
||
}
|
||
return;
|
||
}
|
||
|
||
if (type == type_pause) {
|
||
// 暂停成功！ [AUTO-TRANSLATED:782cea77]
|
||
// Pause successfully!
|
||
_rtp_check_timer.reset();
|
||
return;
|
||
}
|
||
|
||
// play或seek成功 [AUTO-TRANSLATED:ba7b0da3]
|
||
// Play or seek successfully
|
||
uint32_t iSeekTo = 0;
|
||
// 修正时间轴 [AUTO-TRANSLATED:5ab341f9]
|
||
// Correct the timeline
|
||
auto strRange = parser["Range"];
|
||
if (strRange.size()) {
|
||
auto strStart = findSubString(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 ? "resume rtsp success" : "rtsp play success"), !_play_check_timer);
|
||
}
|
||
|
||
void RtspPlayer::onWholeRtspPacket(Parser &parser) {
|
||
if (!start_with(parser.method(), "RTSP")) {
|
||
// 不是rtsp回复，忽略 [AUTO-TRANSLATED:1dca8f64]
|
||
// Not an RTSP response, ignore
|
||
WarnL << "Not rtsp response: " << parser.method();
|
||
return;
|
||
}
|
||
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 [AUTO-TRANSLATED:06a369c2]
|
||
// _pPlayTimer is empty after handshake ends
|
||
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);
|
||
if (trackIdx == -1) {
|
||
return;
|
||
}
|
||
handleOneRtp(
|
||
trackIdx, _sdp_track[trackIdx]->_type, _sdp_track[trackIdx]->_samplerate, (uint8_t *)data + RtpPacket::kRtpTcpHeaderSize,
|
||
len - RtpPacket::kRtpTcpHeaderSize);
|
||
} else {
|
||
trackIdx = getTrackIndexByInterleaved(interleaved - 1);
|
||
if (trackIdx == -1) {
|
||
return;
|
||
}
|
||
onRtcpPacket(trackIdx, _sdp_track[trackIdx], (uint8_t *)data + RtpPacket::kRtpTcpHeaderSize, len - RtpPacket::kRtpTcpHeaderSize);
|
||
}
|
||
}
|
||
|
||
// 此处预留rtcp处理函数 [AUTO-TRANSLATED:30c3afa8]
|
||
// Reserved for RTCP processing function
|
||
void RtspPlayer::onRtcpPacket(int track_idx, SdpTrack::Ptr &track, uint8_t *data, size_t len) {
|
||
auto rtcp_arr = RtcpHeader::loadFromBytes((char *)data, len);
|
||
for (auto &rtcp : rtcp_arr) {
|
||
_rtcp_context[track_idx]->onRtcp(rtcp);
|
||
if ((RtcpType)rtcp->pt == RtcpType::RTCP_SR) {
|
||
auto sr = (RtcpSR *)(rtcp);
|
||
// 设置rtp时间戳与ntp时间戳的对应关系 [AUTO-TRANSLATED:e92f4749]
|
||
// Set the correspondence between RTP timestamp and NTP timestamp
|
||
setNtpStamp(track_idx, sr->rtpts, sr->getNtpUnixStampMS());
|
||
}
|
||
}
|
||
}
|
||
|
||
void RtspPlayer::onRtpSorted(RtpPacket::Ptr rtppt, int trackidx) {
|
||
_stamp[trackidx] = rtppt->getStampMS();
|
||
_rtp_recv_ticker.resetTime();
|
||
onRecvRTP(std::move(rtppt), _sdp_track[trackidx]);
|
||
}
|
||
|
||
float RtspPlayer::getPacketLossRate(TrackType type) const {
|
||
size_t lost = 0, expected = 0;
|
||
try {
|
||
auto track_idx = getTrackIndexByTrackType(type);
|
||
if (_rtcp_context.empty()) {
|
||
return 0;
|
||
}
|
||
auto ctx = _rtcp_context[track_idx];
|
||
lost = ctx->getLost();
|
||
expected = ctx->getExpectedPackets();
|
||
} catch (...) {
|
||
for (auto &ctx : _rtcp_context) {
|
||
lost += ctx->getLost();
|
||
expected += ctx->getExpectedPackets();
|
||
}
|
||
}
|
||
if (!expected) {
|
||
return 0;
|
||
}
|
||
return (float)(double(lost) / double(expected));
|
||
}
|
||
|
||
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", kServerName);
|
||
|
||
if (!_session_id.empty()) {
|
||
header.emplace("Session", _session_id);
|
||
}
|
||
|
||
if (!_realm.empty() && !(*this)[Client::kRtspUser].empty()) {
|
||
if (!_md5_nonce.empty()) {
|
||
// MD5认证 [AUTO-TRANSLATED:0640fa6a]
|
||
// MD5 authentication
|
||
/*
|
||
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) );
|
||
/*
|
||
The response calculation method is as follows:
|
||
The RTSP client should use username + password and calculate the response as follows:
|
||
(1) When password is MD5 encoded, then
|
||
response = md5( password:nonce:md5(public_method:url) );
|
||
(2) When password is ANSI string, then
|
||
response= md5( md5(username:realm:password):nonce:md5(public_method:url) );
|
||
|
||
* [AUTO-TRANSLATED:7858b67d]
|
||
*/
|
||
string encrypted_pwd = (*this)[Client::kRtspPwd];
|
||
if (!(*this)[Client::kRtspPwdIsMD5].as<bool>()) {
|
||
encrypted_pwd = MD5((*this)[Client::kRtspUser] + ":" + _realm + ":" + encrypted_pwd).hexdigest();
|
||
}
|
||
auto response = MD5(encrypted_pwd + ":" + _md5_nonce + ":" + MD5(cmd + ":" + url).hexdigest()).hexdigest();
|
||
_StrPrinter printer;
|
||
printer << "Digest ";
|
||
printer << "username=\"" << (*this)[Client::kRtspUser] << "\", ";
|
||
printer << "realm=\"" << _realm << "\", ";
|
||
printer << "nonce=\"" << _md5_nonce << "\", ";
|
||
printer << "uri=\"" << url << "\", ";
|
||
printer << "response=\"" << response << "\"";
|
||
header.emplace("Authorization", printer);
|
||
} else if (!(*this)[Client::kRtspPwdIsMD5].as<bool>()) {
|
||
// base64认证 [AUTO-TRANSLATED:06d26447]
|
||
// Base64 authentication
|
||
auto authStrBase64 = encodeBase64((*this)[Client::kRtspUser] + ":" + (*this)[Client::kRtspPwd]);
|
||
header.emplace("Authorization", StrPrinter << "Basic " << authStrBase64);
|
||
}
|
||
}
|
||
|
||
_StrPrinter printer;
|
||
printer << cmd << " " << url << " RTSP/1.0\r\n";
|
||
|
||
TraceL << cmd << " "<< url;
|
||
for (auto &pr : header) {
|
||
printer << pr.first << ": " << pr.second << "\r\n";
|
||
}
|
||
printer << "\r\n";
|
||
SockSender::send(std::move(printer));
|
||
}
|
||
|
||
void RtspPlayer::onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_idx) {
|
||
auto &rtcp_ctx = _rtcp_context[track_idx];
|
||
rtcp_ctx->onRtp(rtp->getSeq(), rtp->getStamp(), rtp->ntp_stamp, rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
|
||
|
||
auto &ticker = _rtcp_send_ticker[track_idx];
|
||
if (ticker.elapsedTime() < _beat_interval_ms) {
|
||
// 心跳时间未到 [AUTO-TRANSLATED:265d4e62]
|
||
// Heartbeat time not reached
|
||
return;
|
||
}
|
||
|
||
// 有些rtsp服务器需要rtcp保活，有些需要发送信令保活; rtcp与rtsp信令轮流心跳，该特性用于兼容issue:#642 [AUTO-TRANSLATED:a36070c5]
|
||
// Some RTSP servers require RTCP keep-alive, some require sending signaling keep-alive; RTCP and RTSP signaling alternate heartbeat, this feature is used to be compatible with issue:#642
|
||
auto &rtcp_flag = _send_rtcp[track_idx];
|
||
ticker.resetTime();
|
||
|
||
switch ((BeatType)_beat_type) {
|
||
case BeatType::cmd: rtcp_flag = false; break;
|
||
case BeatType::rtcp: rtcp_flag = true; break;
|
||
case BeatType::both:
|
||
default: rtcp_flag = !rtcp_flag; break;
|
||
}
|
||
|
||
// 发送信令保活 [AUTO-TRANSLATED:0ef0747e]
|
||
// Send signaling keep-alive
|
||
if (!rtcp_flag) {
|
||
if (track_idx == 0) {
|
||
// 两个track无需同时触发发送信令保活 [AUTO-TRANSLATED:7dde4aec]
|
||
// Two tracks do not need to trigger sending signaling keep-alive at the same time
|
||
sendKeepAlive();
|
||
}
|
||
return;
|
||
}
|
||
|
||
// 发送rtcp [AUTO-TRANSLATED:5c7aad87]
|
||
// Send RTCP
|
||
static auto send_rtcp = [](RtspPlayer *thiz, int index, Buffer::Ptr ptr) {
|
||
if (thiz->_rtp_type == Rtsp::RTP_TCP) {
|
||
auto &track = thiz->_sdp_track[index];
|
||
thiz->send(makeRtpOverTcpPrefix((uint16_t)(ptr->size()), track->_interleaved + 1));
|
||
thiz->send(std::move(ptr));
|
||
} else {
|
||
thiz->_rtcp_sock[index]->send(std::move(ptr));
|
||
}
|
||
};
|
||
|
||
auto ssrc = rtp->getSSRC();
|
||
auto rtcp = rtcp_ctx->createRtcpRR(ssrc + 1, ssrc);
|
||
auto rtcp_sdes = RtcpSdes::create({ kServerName });
|
||
rtcp_sdes->chunks.type = (uint8_t)SdesType::RTCP_SDES_CNAME;
|
||
rtcp_sdes->chunks.ssrc = htonl(ssrc);
|
||
send_rtcp(this, track_idx, std::move(rtcp));
|
||
send_rtcp(this, track_idx, RtcpHeader::toBuffer(rtcp_sdes));
|
||
}
|
||
|
||
void RtspPlayer::onPlayResult_l(const SockException &ex, bool handshake_done) {
|
||
if (ex.getErrCode() == Err_shutdown) {
|
||
// 主动shutdown的，不触发回调 [AUTO-TRANSLATED:21f9c396]
|
||
// Active shutdown, do not trigger callback
|
||
return;
|
||
}
|
||
|
||
WarnL << ex.getErrCode() << " " << ex.what();
|
||
if (!handshake_done) {
|
||
// 开始播放阶段 [AUTO-TRANSLATED:7ef385fc]
|
||
// Start playback stage
|
||
_play_check_timer.reset();
|
||
onPlayResult(ex);
|
||
// 是否为性能测试模式 [AUTO-TRANSLATED:871a0e65]
|
||
// Whether it is performance test mode
|
||
_benchmark_mode = (*this)[Client::kBenchmarkMode].as<int>();
|
||
} else if (ex) {
|
||
// 播放成功后异常断开回调 [AUTO-TRANSLATED:3fe28e4f]
|
||
// Callback for abnormal disconnection after successful playback
|
||
onShutdown(ex);
|
||
} else {
|
||
// 恢复播放 [AUTO-TRANSLATED:7a99afd6]
|
||
// Resume playback
|
||
onResume();
|
||
}
|
||
|
||
if (!ex) {
|
||
// 播放成功，恢复rtp接收超时定时器 [AUTO-TRANSLATED:0ebefcb5]
|
||
// Playback successful, restore RTP receive timeout timer
|
||
_rtp_recv_ticker.resetTime();
|
||
auto timeoutMS = (*this)[Client::kMediaTimeoutMS].as<uint64_t>();
|
||
weak_ptr<RtspPlayer> weakSelf = static_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 [AUTO-TRANSLATED:601b8c0c]
|
||
// Receive RTP media data packet timeout
|
||
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 [AUTO-TRANSLATED:edbffc19]
|
||
// Create RTP data receive timeout detection timer
|
||
_rtp_check_timer = std::make_shared<Timer>(timeoutMS / 2000.0f, lam, getPoller());
|
||
} else {
|
||
sendTeardown();
|
||
}
|
||
}
|
||
|
||
int RtspPlayer::getTrackIndexByInterleaved(int interleaved) const {
|
||
for (size_t i = 0; i < _sdp_track.size(); ++i) {
|
||
if (_sdp_track[i]->_interleaved == interleaved) {
|
||
return i;
|
||
}
|
||
}
|
||
if (_sdp_track.size() == 1) {
|
||
return 0;
|
||
}
|
||
WarnL << "no such track with interleaved:" << interleaved;
|
||
return -1;
|
||
}
|
||
|
||
int RtspPlayer::getTrackIndexByTrackType(TrackType track_type) const {
|
||
for (size_t 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_other, StrPrinter << "no such track with type:" << getTrackString(track_type));
|
||
}
|
||
|
||
///////////////////////////////////////////////////
|
||
// RtspPlayerImp
|
||
float RtspPlayerImp::getDuration() const {
|
||
return _demuxer ? _demuxer->getDuration() : 0;
|
||
}
|
||
|
||
void RtspPlayerImp::onPlayResult(const toolkit::SockException &ex) {
|
||
if (!(*this)[Client::kWaitTrackReady].as<bool>() || ex) {
|
||
Super::onPlayResult(ex);
|
||
return;
|
||
}
|
||
}
|
||
|
||
void RtspPlayerImp::addTrackCompleted() {
|
||
if ((*this)[Client::kWaitTrackReady].as<bool>()) {
|
||
Super::onPlayResult(toolkit::SockException(toolkit::Err_success, "play success"));
|
||
}
|
||
}
|
||
|
||
std::vector<Track::Ptr> RtspPlayerImp::getTracks(bool ready /*= true*/) const {
|
||
return _demuxer ? _demuxer->getTracks(ready) : Super::getTracks(ready);
|
||
}
|
||
|
||
bool RtspPlayerImp::onCheckSDP(const std::string &sdp) {
|
||
_rtsp_media_src = std::dynamic_pointer_cast<RtspMediaSource>(_media_src);
|
||
if (_rtsp_media_src) {
|
||
_rtsp_media_src->setSdp(sdp);
|
||
}
|
||
_demuxer = std::make_shared<RtspDemuxer>();
|
||
_demuxer->setTrackListener(this, (*this)[Client::kWaitTrackReady].as<bool>());
|
||
_demuxer->loadSdp(sdp);
|
||
return true;
|
||
}
|
||
|
||
void RtspPlayerImp::onRecvRTP(RtpPacket::Ptr rtp, const SdpTrack::Ptr &track) {
|
||
// rtp解复用时可以判断是否为关键帧起始位置 [AUTO-TRANSLATED:fb7d9b6e]
|
||
// When demultiplexing RTP, it can be determined whether it is the starting position of the key frame
|
||
auto key_pos = _demuxer->inputRtp(rtp);
|
||
if (_rtsp_media_src) {
|
||
_rtsp_media_src->onWrite(std::move(rtp), key_pos);
|
||
}
|
||
}
|
||
|
||
} /* namespace mediakit */
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