/***************************************************************************** # # # uStreamer - Lightweight and fast MJPG-HTTP streamer. # # # # This source file is partially based on this code: # # - https://github.com/catid/kvm/blob/master/kvm_pipeline/src # # # # Copyright (C) 2018-2021 Maxim Devaev # # # # 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 . # # # *****************************************************************************/ #pragma once #include #include #include #include #include #include #include #include #include #include "tools.h" #include "threading.h" #include "frame.h" #include "base64.h" // https://stackoverflow.com/questions/47635545/why-webrtc-chose-rtp-max-packet-size-to-1200-bytes #define RTP_DATAGRAM_SIZE 1200 typedef struct { uint32_t ssrc; uint16_t seq; uint8_t datagram[RTP_DATAGRAM_SIZE]; frame_s *sps; // Actually not a frame, just a bytes storage frame_s *pps; pthread_mutex_t mutex; } rtp_s; typedef void (*rtp_callback_f)(const uint8_t *datagram, size_t size); rtp_s *rtp_init(void); void rtp_destroy(rtp_s *rtp); char *rtp_make_sdp(rtp_s *rtp); void rtp_wrap_h264(rtp_s *rtp, const frame_s *frame, rtp_callback_f callback);