mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
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754073918a
* 优化MultiMediaSourceMuxer头文件包含 * 将MediaSinkDelegate和Demux移到MediaSink中 * MediaSource头文件重构, 独立出PacketCache.h 精简Frame和Track的头文件 * Rtmp头文件重构 * Rtsp头文件重构 * webrtc头文件重构 * 规范.h头文件包含,并将其移到.cpp中: - 尽量不包含Common\config.h - Util\File.h - Rtsp/RtspPlayer.h - Rtmp/RtmpPlayer.h * 删除多余的Stamp.h和Base64包含
187 lines
6.3 KiB
C++
187 lines
6.3 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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#if defined(ENABLE_RTPPROXY)
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#include "RtpSession.h"
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#include "RtpSelector.h"
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#include "Network/TcpServer.h"
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#include "Rtsp/RtpReceiver.h"
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#include "Common/config.h"
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using namespace std;
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using namespace toolkit;
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namespace mediakit{
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const string RtpSession::kStreamID = "stream_id";
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const string RtpSession::kSSRC = "ssrc";
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void RtpSession::attachServer(const Server &server) {
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_stream_id = const_cast<Server &>(server)[kStreamID];
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_ssrc = const_cast<Server &>(server)[kSSRC];
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}
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RtpSession::RtpSession(const Socket::Ptr &sock) : Session(sock) {
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DebugP(this);
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socklen_t addr_len = sizeof(_addr);
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getpeername(sock->rawFD(), (struct sockaddr *)&_addr, &addr_len);
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_is_udp = sock->sockType() == SockNum::Sock_UDP;
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if (_is_udp) {
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// 设置udp socket读缓存
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SockUtil::setRecvBuf(getSock()->rawFD(), 4 * 1024 * 1024);
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}
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}
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RtpSession::~RtpSession() {
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DebugP(this);
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if(_process){
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RtpSelector::Instance().delProcess(_stream_id,_process.get());
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}
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}
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void RtpSession::onRecv(const Buffer::Ptr &data) {
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if (_is_udp) {
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onRtpPacket(data->data(), data->size());
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return;
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}
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RtpSplitter::input(data->data(), data->size());
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}
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void RtpSession::onError(const SockException &err) {
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WarnP(this) << _stream_id << " " << err.what();
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}
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void RtpSession::onManager() {
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if(_process && !_process->alive()){
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shutdown(SockException(Err_timeout, "receive rtp timeout"));
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}
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if(!_process && _ticker.createdTime() > 10 * 1000){
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shutdown(SockException(Err_timeout, "illegal connection"));
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}
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}
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void RtpSession::onRtpPacket(const char *data, size_t len) {
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if (!_is_udp) {
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if (_search_rtp) {
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//搜索上下文期间,数据丢弃
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if (_search_rtp_finished) {
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//下个包开始就是正确的rtp包了
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_search_rtp_finished = false;
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_search_rtp = false;
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}
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return;
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}
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GET_CONFIG(uint32_t, rtpMaxSize, Rtp::kRtpMaxSize);
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if (len > 1024 * rtpMaxSize) {
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_search_rtp = true;
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WarnL << "rtp包长度异常(" << len << "),发送端可能缓存溢出并覆盖,开始搜索ssrc以便恢复上下文";
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return;
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}
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}
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if (!_process) {
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//未设置ssrc时,尝试获取ssrc
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if (!_ssrc && !RtpSelector::getSSRC(data, len, _ssrc)) {
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return;
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}
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if (_stream_id.empty()) {
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//未指定流id就使用ssrc为流id
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_stream_id = printSSRC(_ssrc);
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}
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//tcp情况下,一个tcp链接只可能是一路流,不需要通过多个ssrc来区分,所以不需要频繁getProcess
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_process = RtpSelector::Instance().getProcess(_stream_id, true);
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_process->setDelegate(dynamic_pointer_cast<RtpSession>(shared_from_this()));
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}
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try {
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uint32_t rtp_ssrc = 0;
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RtpSelector::getSSRC(data, len, rtp_ssrc);
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if (rtp_ssrc != _ssrc) {
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WarnP(this) << "ssrc不匹配,rtp已丢弃:" << rtp_ssrc << " != " << _ssrc;
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return;
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}
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_process->inputRtp(false, getSock(), data, len, (struct sockaddr *)&_addr);
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} catch (RtpTrack::BadRtpException &ex) {
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if (!_is_udp) {
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WarnL << ex.what() << ",开始搜索ssrc以便恢复上下文";
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_search_rtp = true;
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} else {
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throw;
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}
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} catch (...) {
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throw;
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}
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_ticker.resetTime();
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}
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bool RtpSession::close(MediaSource &sender) {
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//此回调在其他线程触发
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string err = StrPrinter << "close media: " << sender.getUrl();
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safeShutdown(SockException(Err_shutdown, err));
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return true;
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}
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static const char *findSSRC(const char *data, ssize_t len, uint32_t ssrc) {
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//rtp前面必须预留两个字节的长度字段
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for (ssize_t i = 2; i <= len - 4; ++i) {
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auto ptr = (const uint8_t *) data + i;
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if (ptr[0] == (ssrc >> 24) && ptr[1] == ((ssrc >> 16) & 0xFF) &&
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ptr[2] == ((ssrc >> 8) & 0xFF) && ptr[3] == (ssrc & 0xFF)) {
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return (const char *) ptr;
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}
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}
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return nullptr;
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}
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//rtp长度到ssrc间的长度固定为10
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static size_t constexpr kSSRCOffset = 2 + 4 + 4;
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const char *RtpSession::onSearchPacketTail(const char *data, size_t len) {
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if (!_search_rtp) {
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//tcp上下文正常,不用搜索ssrc
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return RtpSplitter::onSearchPacketTail(data, len);
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}
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if (!_process) {
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throw SockException(Err_shutdown, "ssrc未获取到,无法通过ssrc恢复tcp上下文");
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}
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//搜索第一个rtp的ssrc
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auto ssrc_ptr0 = findSSRC(data, len, _ssrc);
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if (!ssrc_ptr0) {
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//未搜索到任意rtp,返回数据不够
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return nullptr;
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}
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//这两个字节是第一个rtp的长度字段
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auto rtp_len_ptr = (ssrc_ptr0 - kSSRCOffset);
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auto rtp_len = ((uint8_t *)rtp_len_ptr)[0] << 8 | ((uint8_t *)rtp_len_ptr)[1];
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//搜索第二个rtp的ssrc
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auto ssrc_ptr1 = findSSRC(ssrc_ptr0 + rtp_len, data + (ssize_t) len - ssrc_ptr0 - rtp_len, _ssrc);
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if (!ssrc_ptr1) {
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//未搜索到第二个rtp,返回数据不够
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return nullptr;
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}
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//两个ssrc的间隔正好等于rtp的长度(外加rtp长度字段),那么说明找到rtp
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auto ssrc_offset = ssrc_ptr1 - ssrc_ptr0;
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if (ssrc_offset == rtp_len + 2 || ssrc_offset == rtp_len + 4) {
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InfoL << "rtp搜索成功,tcp上下文恢复成功,丢弃的rtp残余数据为:" << rtp_len_ptr - data;
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_search_rtp_finished = true;
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if (rtp_len_ptr == data) {
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//停止搜索rtp,否则会进入死循环
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_search_rtp = false;
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}
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//前面的数据都需要丢弃,这个是rtp的起始
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return rtp_len_ptr;
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}
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//第一个rtp长度不匹配,说明第一个找到的ssrc不是rtp,丢弃之,我们从第二个ssrc所在rtp开始搜索
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return ssrc_ptr1 - kSSRCOffset;
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}
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}//namespace mediakit
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#endif//defined(ENABLE_RTPPROXY)
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