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" 128 gboolean *c, gboolean *f, gboolean *r1, gboolean *r0);
138 uint16_t *min_delay, uint16_t *max_delay);
148 char *sdes_item,
int sdes_len);
157 uint16_t *transSeqNum);
161 uint32_t a_last_ssrc, a_last_ts, a_base_ts, a_base_ts_prev, a_prev_ts, a_target_ts, a_start_ts,
162 v_last_ssrc, v_last_ts, v_base_ts, v_base_ts_prev, v_prev_ts,
v_target_ts, v_start_ts;
163 uint16_t a_last_seq, a_prev_seq, a_base_seq, a_base_seq_prev,
164 v_last_seq,
v_prev_seq, v_base_seq, v_base_seq_prev;
165 gboolean a_seq_reset, a_new_ssrc,
169 gint32 a_prev_delay, a_active_delay, a_ts_offset,
171 gint64 a_last_time, a_reference_time, a_start_time, a_evaluating_start_time,
172 v_last_time, v_reference_time,
v_start_time, v_evaluating_start_time;
186 #define RTP_AUDIO_SKEW_TH_MS 120 187 #define RTP_VIDEO_SKEW_TH_MS 120 188 #define SKEW_DETECTION_WAIT_TIME_SECS 10 gboolean changed_temporal
Whether the temporal layer has changed after processing a packet.
Definition: rtp.h:242
int templayer_target
As above, but to handle transitions (e.g., wait for keyframe)
Definition: rtp.h:236
gint64 v_start_time
Definition: rtp.h:171
gint64 created
Definition: rtp.h:60
gboolean janus_rtp_simulcasting_context_process_rtp(janus_rtp_simulcasting_context *context, char *buf, int len, uint32_t *ssrcs, janus_videocodec vcodec, janus_rtp_switching_context *sc)
Process an RTP packet, and decide whether this should be relayed or not, updating the context accordi...
Definition: rtp.c:792
gint64 last_relayed
When we relayed the last packet (used to detect when substreams become unavailable) ...
Definition: rtp.h:238
int janus_videocodec_pt(janus_videocodec vcodec)
Definition: rtp.c:767
int templayer
Which simulcast temporal layer we should forward back.
Definition: rtp.h:234
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:383
gboolean changed_substream
Whether the substream has changed after processing a packet.
Definition: rtp.h:240
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:186
janus_videocodec
Definition: rtp.h:216
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:26
const char * janus_videocodec_name(janus_videocodec vcodec)
Definition: rtp.c:740
gboolean janus_is_rtp(char *buf, guint len)
Helper method to demultiplex RTP from other protocols.
Definition: rtp.c:19
struct rtp_header rtp_header
RTP Header (http://tools.ietf.org/html/rfc3550#section-5.1)
struct janus_rtp_simulcasting_context janus_rtp_simulcasting_context
Helper struct for processing and tracking simulcast streams.
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:245
janus_audiocodec
Definition: rtp.h:203
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:497
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:267
janus_videocodec janus_videocodec_from_name(const char *name)
Definition: rtp.c:755
RTP context, in order to make sure SSRC changes result in coherent seq/ts increases.
Definition: rtp.h:160
gboolean need_pli
Whether we need to send the user a keyframe request (PLI)
Definition: rtp.h:244
int substream
Which simulcast substream we should forward back.
Definition: rtp.h:230
gint32 v_ts_offset
Definition: rtp.h:169
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:224
janus_audiocodec janus_audiocodec_from_name(const char *name)
Definition: rtp.c:700
int substream_target
As above, but to handle transitions (e.g., wait for keyframe, or get this if available) ...
Definition: rtp.h:232
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:165
uint32_t v_target_ts
Definition: rtp.h:161
int janus_audiocodec_pt(janus_audiocodec acodec)
Definition: rtp.c:718
gint16 v_seq_offset
Definition: rtp.h:167
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:80
uint16_t v_prev_seq
Definition: rtp.h:163
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:52
Helper struct for processing and tracking simulcast streams.
Definition: rtp.h:228
void janus_rtp_switching_context_reset(janus_rtp_switching_context *context)
Set (or reset) the context fields to their default values.
Definition: rtp.c:260
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:173
gint length
Definition: rtp.h:59
const char * janus_audiocodec_name(janus_audiocodec acodec)
Definition: rtp.c:679
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:208
void janus_rtp_simulcasting_context_reset(janus_rtp_simulcasting_context *context)
Set (or reset) the context fields to their default values.
Definition: rtp.c:783