244 lines
9.8 KiB
C
244 lines
9.8 KiB
C
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/***************************************************************************************
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*
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* IMPORTANT: READ BEFORE DOWNLOADING, COPYING, INSTALLING OR USING.
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*
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* By downloading, copying, installing or using the software you agree to this license.
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* If you do not agree to this license, do not download, install,
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* copy or use the software.
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*
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* Copyright (C) 2014-2024, Happytimesoft Corporation, all rights reserved.
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*
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* Redistribution and use in binary forms, with or without modification, are permitted.
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*
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* Unless required by applicable law or agreed to in writing, software distributed
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* under the License is distributed on an "AS IS" BASIS, WITHOUT WARRANTIES OR
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* CONDITIONS OF ANY KIND, either express or implied. See the License for the specific
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* language governing permissions and limitations under the License.
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*
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****************************************************************************************/
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#ifndef RTSP_RCUA_H
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#define RTSP_RCUA_H
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#include "rtsp_parse.h"
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#include "http.h"
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#ifdef BACKCHANNEL
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#include "audio_capture.h"
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#endif
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#ifdef OVER_WEBSOCKET
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#include "rtsp_ws.h"
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#endif
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#define AV_TYPE_VIDEO 0
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#define AV_TYPE_AUDIO 1
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#define AV_TYPE_METADATA 2
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#define AV_TYPE_BACKCHANNEL 3
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#define AV_MAX_CHS 4
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#define AV_VIDEO_CH AV_TYPE_VIDEO
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#define AV_AUDIO_CH AV_TYPE_AUDIO
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#define AV_METADATA_CH AV_TYPE_METADATA
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#define AV_BACK_CH AV_TYPE_BACKCHANNEL
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/**
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* RTSP over websocket needs to reserve some space
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* in the header of the send buffer
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*/
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#define RTSP_SND_RESV_SIZE 32
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typedef enum rtsp_client_states
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{
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RCS_NULL = 0,
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RCS_OPTIONS,
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RCS_DESCRIBE,
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RCS_INIT_V,
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RCS_INIT_A,
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RCS_INIT_M,
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RCS_INIT_BC,
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RCS_READY,
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RCS_PLAYING,
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RCS_RECORDING,
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} RCSTATE;
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typedef struct rtsp_client_media_channel
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{
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int disabled; // not setup this channel
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SOCKET udp_fd; // udp socket
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SOCKET rtcp_fd; // rtcp udp socket
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char ctl[128]; // control string
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uint16 setup; // whether the media channel already be setuped
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uint16 r_port; // remote udp port
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uint16 l_port; // local udp port
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uint16 interleaved; // rtp channel values
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char destination[32]; // multicast address
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int cap_count; // Local number of capabilities
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uint8 cap[MAX_AVN]; // Local capability
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char cap_desc[MAX_AVN][MAX_AVDESCLEN];
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} RCMCH;
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typedef struct rtsp_client_user_agent
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{
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uint32 used_flag : 1; // used flag
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uint32 rtp_tcp : 1; // rtp over tcp
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uint32 mcast_flag : 1; // use rtp multicast, set by user
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uint32 rtp_mcast : 1; // use rtp multicast, set by stack
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uint32 rtp_tx : 1; // rtp sending flag
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uint32 need_auth : 1; // need auth flag
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uint32 auth_mode : 2; // 0 - baisc; 1 - digest
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uint32 gp_cmd : 1; // is support get_parameter command
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uint32 backchannel : 1; // audio backchannle flag
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uint32 send_bc_data: 1; // audio backchannel data sending flag
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uint32 replay : 1; // replay flag
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uint32 over_http : 1; // rtsp over http flag
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uint32 over_ws : 1; // rtsp over websocket flag
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uint32 auth_retry : 3; // auth retry numbers
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uint32 reserved : 15;
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int state; // state, RCSTATE
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SOCKET fd; // socket handler
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uint32 keepalive_time; // keepalive time
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char ripstr[128]; // remote ip
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uint16 rport; // rtsp server port
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uint32 cseq; // seq no
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char sid[64]; // Session ID
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char uri[256]; // rtsp://221.10.50.195:554/cctv.sdp
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char cbase[256]; // Content-Base: rtsp://221.10.50.195:554/broadcast.sdp/
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char user_agent[64]; // user agent string
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int session_timeout; // session timeout value
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int play_start; // a=range:npt=0-20.735, unit is millisecond
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int play_end; // a=range:npt=0-20.735, unit is millisecond
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uint32 video_init_ts; // video init timestampe
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uint32 audio_init_ts; // audio init timestampe
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int seek_pos; // seek pos
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int media_start; // media start time, unit is millisecond
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char rcv_buf[2052]; // receive buffer
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int rcv_dlen; // receive data length
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int rtp_t_len; // rtp payload total length
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int rtp_rcv_len; // rtp payload receive length
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char * rtp_rcv_buf; // rtp payload receive buffer
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RCMCH channels[AV_MAX_CHS]; // media channels
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HD_AUTH_INFO auth_info; // auth information
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int video_codec; // video codec
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int audio_codec; // audio codec
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int sample_rate; // audio sample rate
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int audio_channels; // audio channels
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int bit_per_sample; // audio bit per sample
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uint8 * audio_spec; // audio special data, for AAC etc.
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uint32 audio_spec_len; // audio special data length
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#ifdef BACKCHANNEL
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int bc_audio_device; // back channel audio device index, start from 0
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int bc_audio_codec; // back channel audio codec
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int bc_sample_rate; // back channel sample rate
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int bc_channels; // back channel channel numbers
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int bc_bit_per_sample; // back channel bit per sample
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UA_RTP_INFO bc_rtp_info; // back channel audio rtp info
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CAudioCapture * audio_captrue; // audio capture
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#endif
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#ifdef REPLAY
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uint32 scale_flag : 1;
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uint32 rate_control_flag : 1;
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uint32 immediate_flag : 1;
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uint32 frame_flag : 1;
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uint32 frame_interval_flag : 1;
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uint32 range_flag : 1;
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uint32 replay_reserved : 26;
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int scale; // scale info, when not set the rata control flag, the scale is valid.
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// It shall be either 100.0 or -100.0, to indicate forward or reverse playback respectively.
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// If it is not present, forward playback is assumed
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// 100 means 1.0, Divide by 100 when using
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int rate_control; // rate control flag,
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// 1-the stream is delivered in real time using standard RTP timing mechanisms
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// 0-the stream is delivered as fast as possible, using only the flow control provided by the transport to limit the delivery rate
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int immediate; // 1 - immediately start playing from the new location, cancelling any existing PLAY command.
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// The first packet sent from the new location shall have the D (discontinuity) bit set in its RTP extension header.
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int frame; // 0 - all frames
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// 1 - I-frame and P-frame
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// 2 - I-frame
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int frame_interval; // I-frame interval, unit is milliseconds
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time_t replay_start; // replay start time
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time_t replay_end; // replay end time
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#endif
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#ifdef OVER_HTTP
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uint16 http_port; // rtsp over http port
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HTTPREQ rtsp_send; // rtsp over http get connection
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HTTPREQ rtsp_recv; // rtsp over http post connection
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#endif
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#ifdef OVER_WEBSOCKET
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uint16 ws_port; // rtsp over websocket port
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HTTPREQ ws_http; // rtsp over websocket connection
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WSMSG ws_msg; // rtsp over websocket message handler
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#endif
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} RCUA;
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#ifdef __cplusplus
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extern "C" {
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#endif
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/*************************************************************************/
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HRTSP_MSG * rcua_build_describe(RCUA * p_rua);
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HRTSP_MSG * rcua_build_setup(RCUA * p_rua,int type);
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HRTSP_MSG * rcua_build_play(RCUA * p_rua);
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HRTSP_MSG * rcua_build_pause(RCUA * p_rua);
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HRTSP_MSG * rcua_build_teardown(RCUA * p_rua);
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HRTSP_MSG * rcua_build_get_parameter(RCUA * p_rua);
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HRTSP_MSG * rcua_build_options(RCUA * p_rua);
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/*************************************************************************/
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BOOL rcua_get_media_info(RCUA * p_rua, HRTSP_MSG * rx_msg);
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BOOL rcua_get_sdp_video_desc(RCUA * p_rua, const char * key, int * pt, char * p_sdp, int max_len);
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BOOL rcua_get_sdp_audio_desc(RCUA * p_rua, const char * key, int * pt, char * p_sdp, int max_len);
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BOOL rcua_get_sdp_h264_desc(RCUA * p_rua, int * pt, char * p_sdp, int max_len);
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BOOL rcua_get_sdp_h264_params(RCUA * p_rua, int * pt, char * p_sps_pps, int max_len);
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BOOL rcua_get_sdp_h265_desc(RCUA * p_rua, int * pt, char * p_sdp, int max_len);
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BOOL rcua_get_sdp_h265_params(RCUA * p_rua, int * pt, BOOL * donfield, char * p_vps, int vps_len, char * p_sps, int sps_len, char * p_pps, int pps_len);
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BOOL rcua_get_sdp_mp4_desc(RCUA * p_rua, int * pt, char * p_sdp, int max_len);
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BOOL rcua_get_sdp_mp4_params(RCUA * p_rua, int * pt, char * p_cfg, int max_len);
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BOOL rcua_get_sdp_aac_desc(RCUA * p_rua, int * pt, char * p_sdp, int max_len);
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BOOL rcua_get_sdp_aac_params(RCUA * p_rua, int *pt, int *sizelength, int *indexlength, int *indexdeltalength, char * p_cfg, int max_len);
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SOCKET rcua_init_udp_connection(uint16 port);
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SOCKET rcua_init_mc_connection(uint16 port, char * destination);
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/*************************************************************************/
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void rcua_send_rtsp_msg(RCUA * p_rua,HRTSP_MSG * tx_msg);
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#define rcua_send_free_rtsp_msg(p_rua,tx_msg) \
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do { \
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rcua_send_rtsp_msg(p_rua,tx_msg); \
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rtsp_free_msg(tx_msg); \
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} while(0)
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#ifdef __cplusplus
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}
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#endif
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#endif // RTSP_RCUA_H
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