16 #include <arpa/inet.h> 18 #include <machine/endian.h> 19 #define __BYTE_ORDER BYTE_ORDER 20 #define __BIG_ENDIAN BIG_ENDIAN 21 #define __LITTLE_ENDIAN LITTLE_ENDIAN 29 #define RTP_HEADER_SIZE 12 34 #if __BYTE_ORDER == __BIG_ENDIAN 41 #elif __BYTE_ORDER == __LITTLE_ENDIAN 71 #define JANUS_RTP_EXTMAP_AUDIO_LEVEL "urn:ietf:params:rtp-hdrext:ssrc-audio-level" 73 #define JANUS_RTP_EXTMAP_TOFFSET "urn:ietf:params:rtp-hdrext:toffset" 75 #define JANUS_RTP_EXTMAP_ABS_SEND_TIME "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time" 77 #define JANUS_RTP_EXTMAP_VIDEO_ORIENTATION "urn:3gpp:video-orientation" 79 #define JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01" 81 #define JANUS_RTP_EXTMAP_PLAYOUT_DELAY "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay" 83 #define JANUS_RTP_EXTMAP_RTP_STREAM_ID "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id" 123 gboolean *c, gboolean *f, gboolean *r1, gboolean *r0);
133 uint16_t *min_delay, uint16_t *max_delay);
143 char *sdes_item,
int sdes_len);
152 uint16_t *transSeqNum);
156 uint32_t a_last_ssrc, a_last_ts, a_base_ts, a_base_ts_prev, a_prev_ts, a_target_ts, a_start_ts,
157 v_last_ssrc, v_last_ts, v_base_ts, v_base_ts_prev, v_prev_ts,
v_target_ts, v_start_ts;
158 uint16_t a_last_seq, a_prev_seq, a_base_seq, a_base_seq_prev,
159 v_last_seq,
v_prev_seq, v_base_seq, v_base_seq_prev;
160 gboolean a_seq_reset, a_new_ssrc,
164 gint32 a_prev_delay, a_active_delay, a_ts_offset,
166 gint64 a_last_time, a_reference_time, a_start_time,
181 #define RTP_AUDIO_SKEW_TH_MS 160 182 #define RTP_VIDEO_SKEW_TH_MS 160 183 #define SKEW_DETECTION_WAIT_TIME_SECS 10
gint64 v_start_time
Definition: rtp.h:166
gint64 created
Definition: rtp.h:60
int janus_videocodec_pt(janus_videocodec vcodec)
Definition: rtp.c:703
int janus_rtp_skew_compensate_video(janus_rtp_header *header, janus_rtp_switching_context *context, gint64 now)
Use the context info to compensate for video source skew, if needed.
Definition: rtp.c:341
int janus_rtp_header_extension_parse_video_orientation(char *buf, int len, int id, gboolean *c, gboolean *f, gboolean *r1, gboolean *r0)
Helper to parse a video-orientation RTP extension (http://www.3gpp.org/ftp/Specs/html-info/26114.htm)
Definition: rtp.c:169
janus_videocodec
Definition: rtp.h:211
struct janus_rtp_header_extension janus_rtp_header_extension
RTP extension.
char * janus_rtp_payload(char *buf, int len, int *plen)
Helper to quickly access the RTP payload, skipping header and extensions.
Definition: rtp.c:19
const char * janus_videocodec_name(janus_videocodec vcodec)
Definition: rtp.c:676
struct rtp_header rtp_header
RTP Header (http://tools.ietf.org/html/rfc3550#section-5.1)
int janus_rtp_header_extension_parse_transport_wide_cc(char *buf, int len, int id, uint16_t *transSeqNum)
Helper to parse a rtp-stream-id RTP extension (https://tools.ietf.org/html/draft-ietf-avtext-rid-09) ...
Definition: rtp.c:225
janus_audiocodec
Definition: rtp.h:198
RTP packet.
Definition: rtp.h:57
void janus_rtp_header_update(janus_rtp_header *header, janus_rtp_switching_context *context, gboolean video, int step)
Use the context info to update the RTP header of a packet, if needed.
Definition: rtp.c:433
int janus_rtp_skew_compensate_audio(janus_rtp_header *header, janus_rtp_switching_context *context, gint64 now)
Use the context info to compensate for audio source skew, if needed.
Definition: rtp.c:247
janus_videocodec janus_videocodec_from_name(const char *name)
Definition: rtp.c:691
RTP context, in order to make sure SSRC changes result in coherent seq/ts increases.
Definition: rtp.h:155
gint32 v_ts_offset
Definition: rtp.h:164
gint64 last_retransmit
Definition: rtp.h:61
struct janus_rtp_packet janus_rtp_packet
RTP packet.
int janus_rtp_header_extension_parse_rtp_stream_id(char *buf, int len, int id, char *sdes_item, int sdes_len)
Helper to parse a rtp-stream-id RTP extension (https://tools.ietf.org/html/draft-ietf-avtext-rid-09) ...
Definition: rtp.c:207
janus_audiocodec janus_audiocodec_from_name(const char *name)
Definition: rtp.c:636
struct janus_rtp_switching_context janus_rtp_switching_context
RTP context, in order to make sure SSRC changes result in coherent seq/ts increases.
gboolean v_seq_reset
Definition: rtp.h:160
uint32_t v_target_ts
Definition: rtp.h:156
int janus_audiocodec_pt(janus_audiocodec acodec)
Definition: rtp.c:654
gint16 v_seq_offset
Definition: rtp.h:162
rtp_header janus_rtp_header
Definition: rtp.h:54
char * data
Definition: rtp.h:58
const char * janus_rtp_header_extension_get_from_id(const char *sdp, int id)
Ugly and dirty helper to quickly get the RTP extension namespace associated with an id (extmap) in an...
Definition: rtp.c:68
uint16_t v_prev_seq
Definition: rtp.h:158
int janus_rtp_header_extension_get_id(const char *sdp, const char *extension)
Ugly and dirty helper to quickly get the id associated with an RTP extension (extmap) in an SDP...
Definition: rtp.c:40
void janus_rtp_switching_context_reset(janus_rtp_switching_context *context)
Set (or reset) the context fields to their default values.
Definition: rtp.c:240
int janus_rtp_header_extension_parse_audio_level(char *buf, int len, int id, int *level)
Helper to parse a ssrc-audio-level RTP extension (https://tools.ietf.org/html/rfc6464) ...
Definition: rtp.c:156
gint length
Definition: rtp.h:59
const char * janus_audiocodec_name(janus_audiocodec acodec)
Definition: rtp.c:615
int janus_rtp_header_extension_parse_playout_delay(char *buf, int len, int id, uint16_t *min_delay, uint16_t *max_delay)
Helper to parse a playout-delay RTP extension (https://webrtc.org/experiments/rtp-hdrext/playout-dela...
Definition: rtp.c:191