ZLMediaKit/src/Rtp/RtpSession.cpp
mtdxc 754073918a
Header refactor (#2115)
* 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
2022-11-29 11:07:13 +08:00

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/*
* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
*
* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
*
* Use of this source code is governed by MIT license that can be found in the
* LICENSE file in the root of the source tree. All contributing project authors
* may be found in the AUTHORS file in the root of the source tree.
*/
#if defined(ENABLE_RTPPROXY)
#include "RtpSession.h"
#include "RtpSelector.h"
#include "Network/TcpServer.h"
#include "Rtsp/RtpReceiver.h"
#include "Common/config.h"
using namespace std;
using namespace toolkit;
namespace mediakit{
const string RtpSession::kStreamID = "stream_id";
const string RtpSession::kSSRC = "ssrc";
void RtpSession::attachServer(const Server &server) {
_stream_id = const_cast<Server &>(server)[kStreamID];
_ssrc = const_cast<Server &>(server)[kSSRC];
}
RtpSession::RtpSession(const Socket::Ptr &sock) : Session(sock) {
DebugP(this);
socklen_t addr_len = sizeof(_addr);
getpeername(sock->rawFD(), (struct sockaddr *)&_addr, &addr_len);
_is_udp = sock->sockType() == SockNum::Sock_UDP;
if (_is_udp) {
// 设置udp socket读缓存
SockUtil::setRecvBuf(getSock()->rawFD(), 4 * 1024 * 1024);
}
}
RtpSession::~RtpSession() {
DebugP(this);
if(_process){
RtpSelector::Instance().delProcess(_stream_id,_process.get());
}
}
void RtpSession::onRecv(const Buffer::Ptr &data) {
if (_is_udp) {
onRtpPacket(data->data(), data->size());
return;
}
RtpSplitter::input(data->data(), data->size());
}
void RtpSession::onError(const SockException &err) {
WarnP(this) << _stream_id << " " << err.what();
}
void RtpSession::onManager() {
if(_process && !_process->alive()){
shutdown(SockException(Err_timeout, "receive rtp timeout"));
}
if(!_process && _ticker.createdTime() > 10 * 1000){
shutdown(SockException(Err_timeout, "illegal connection"));
}
}
void RtpSession::onRtpPacket(const char *data, size_t len) {
if (!_is_udp) {
if (_search_rtp) {
//搜索上下文期间,数据丢弃
if (_search_rtp_finished) {
//下个包开始就是正确的rtp包了
_search_rtp_finished = false;
_search_rtp = false;
}
return;
}
GET_CONFIG(uint32_t, rtpMaxSize, Rtp::kRtpMaxSize);
if (len > 1024 * rtpMaxSize) {
_search_rtp = true;
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";
return;
}
}
if (!_process) {
//未设置ssrc时尝试获取ssrc
if (!_ssrc && !RtpSelector::getSSRC(data, len, _ssrc)) {
return;
}
if (_stream_id.empty()) {
//未指定流id就使用ssrc为流id
_stream_id = printSSRC(_ssrc);
}
//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
_process = RtpSelector::Instance().getProcess(_stream_id, true);
_process->setDelegate(dynamic_pointer_cast<RtpSession>(shared_from_this()));
}
try {
uint32_t rtp_ssrc = 0;
RtpSelector::getSSRC(data, len, rtp_ssrc);
if (rtp_ssrc != _ssrc) {
WarnP(this) << "ssrc不匹配,rtp已丢弃:" << rtp_ssrc << " != " << _ssrc;
return;
}
_process->inputRtp(false, getSock(), data, len, (struct sockaddr *)&_addr);
} catch (RtpTrack::BadRtpException &ex) {
if (!_is_udp) {
WarnL << ex.what() << "开始搜索ssrc以便恢复上下文";
_search_rtp = true;
} else {
throw;
}
} catch (...) {
throw;
}
_ticker.resetTime();
}
bool RtpSession::close(MediaSource &sender) {
//此回调在其他线程触发
string err = StrPrinter << "close media: " << sender.getUrl();
safeShutdown(SockException(Err_shutdown, err));
return true;
}
static const char *findSSRC(const char *data, ssize_t len, uint32_t ssrc) {
//rtp前面必须预留两个字节的长度字段
for (ssize_t i = 2; i <= len - 4; ++i) {
auto ptr = (const uint8_t *) data + i;
if (ptr[0] == (ssrc >> 24) && ptr[1] == ((ssrc >> 16) & 0xFF) &&
ptr[2] == ((ssrc >> 8) & 0xFF) && ptr[3] == (ssrc & 0xFF)) {
return (const char *) ptr;
}
}
return nullptr;
}
//rtp长度到ssrc间的长度固定为10
static size_t constexpr kSSRCOffset = 2 + 4 + 4;
const char *RtpSession::onSearchPacketTail(const char *data, size_t len) {
if (!_search_rtp) {
//tcp上下文正常不用搜索ssrc
return RtpSplitter::onSearchPacketTail(data, len);
}
if (!_process) {
throw SockException(Err_shutdown, "ssrc未获取到无法通过ssrc恢复tcp上下文");
}
//搜索第一个rtp的ssrc
auto ssrc_ptr0 = findSSRC(data, len, _ssrc);
if (!ssrc_ptr0) {
//未搜索到任意rtp返回数据不够
return nullptr;
}
//这两个字节是第一个rtp的长度字段
auto rtp_len_ptr = (ssrc_ptr0 - kSSRCOffset);
auto rtp_len = ((uint8_t *)rtp_len_ptr)[0] << 8 | ((uint8_t *)rtp_len_ptr)[1];
//搜索第二个rtp的ssrc
auto ssrc_ptr1 = findSSRC(ssrc_ptr0 + rtp_len, data + (ssize_t) len - ssrc_ptr0 - rtp_len, _ssrc);
if (!ssrc_ptr1) {
//未搜索到第二个rtp返回数据不够
return nullptr;
}
//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
auto ssrc_offset = ssrc_ptr1 - ssrc_ptr0;
if (ssrc_offset == rtp_len + 2 || ssrc_offset == rtp_len + 4) {
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;
_search_rtp_finished = true;
if (rtp_len_ptr == data) {
//停止搜索rtp否则会进入死循环
_search_rtp = false;
}
//前面的数据都需要丢弃这个是rtp的起始
return rtp_len_ptr;
}
//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
return ssrc_ptr1 - kSSRCOffset;
}
}//namespace mediakit
#endif//defined(ENABLE_RTPPROXY)