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58 lines
2.3 KiB
C
58 lines
2.3 KiB
C
/*****************************************************************************
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# #
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# uStreamer - Lightweight and fast MJPEG-HTTP streamer. #
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# #
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# Copyright (C) 2018-2023 Maxim Devaev <mdevaev@gmail.com> #
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# #
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# This program is free software: you can redistribute it and/or modify #
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# it under the terms of the GNU General Public License as published by #
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# the Free Software Foundation, either version 3 of the License, or #
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# (at your option) any later version. #
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# #
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# This program is distributed in the hope that it will be useful, #
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# but WITHOUT ANY WARRANTY; without even the implied warranty of #
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# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the #
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# GNU General Public License for more details. #
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# #
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# You should have received a copy of the GNU General Public License #
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# along with this program. If not, see <https://www.gnu.org/licenses/>. #
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# #
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*****************************************************************************/
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#pragma once
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#include <stdlib.h>
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#include <stdint.h>
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#include <stdbool.h>
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#include <sys/types.h>
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#include "uslibs/tools.h"
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// https://stackoverflow.com/questions/47635545/why-webrtc-chose-rtp-max-packet-size-to-1200-bytes
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#define US_RTP_DATAGRAM_SIZE 1200
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#define US_RTP_HEADER_SIZE 12
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typedef struct {
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unsigned payload;
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bool video;
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uint32_t ssrc;
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uint16_t seq;
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uint8_t datagram[US_RTP_DATAGRAM_SIZE];
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size_t used;
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bool zero_playout_delay;
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} us_rtp_s;
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typedef void (*us_rtp_callback_f)(const us_rtp_s *rtp);
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us_rtp_s *us_rtp_init(unsigned payload, bool video);
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us_rtp_s *us_rtp_dup(const us_rtp_s *rtp);
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void us_rtp_destroy(us_rtp_s *rtp);
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void us_rtp_write_header(us_rtp_s *rtp, uint32_t pts, bool marked);
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