Files
ustreamer/janus/src/rtp.h
Maxim Devaev c505a423af refactoring
2022-06-03 07:25:41 +03:00

59 lines
2.4 KiB
C

/*****************************************************************************
# #
# uStreamer - Lightweight and fast MJPEG-HTTP streamer. #
# #
# This source file is partially based on this code: #
# - https://github.com/catid/kvm/blob/master/kvm_pipeline/src #
# #
# Copyright (C) 2018-2022 Maxim Devaev <mdevaev@gmail.com> #
# #
# This program is free software: you can redistribute it and/or modify #
# it under the terms of the GNU General Public License as published by #
# the Free Software Foundation, either version 3 of the License, or #
# (at your option) any later version. #
# #
# This program is distributed in the hope that it will be useful, #
# but WITHOUT ANY WARRANTY; without even the implied warranty of #
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the #
# GNU General Public License for more details. #
# #
# You should have received a copy of the GNU General Public License #
# along with this program. If not, see <https://www.gnu.org/licenses/>. #
# #
*****************************************************************************/
#pragma once
#include <stdlib.h>
#include <stdint.h>
#include <stdbool.h>
#include <sys/types.h>
#include "uslibs/tools.h"
// https://stackoverflow.com/questions/47635545/why-webrtc-chose-rtp-max-packet-size-to-1200-bytes
#define RTP_DATAGRAM_SIZE 1200
#define RTP_HEADER_SIZE 12
typedef struct {
unsigned payload;
bool video;
uint32_t ssrc;
uint16_t seq;
uint8_t datagram[RTP_DATAGRAM_SIZE];
size_t used;
} rtp_s;
typedef void (*rtp_callback_f)(const rtp_s *rtp);
rtp_s *rtp_init(unsigned payload, bool video);
void rtp_destroy(rtp_s *rtp);
void rtp_write_header(rtp_s *rtp, uint32_t pts, bool marked);