3 * @brief RTP session handling
5 /*****************************************************************************
6 * Copyright © 2008 Rémi Denis-Courmont
8 * This library is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU General Public License
10 * as published by the Free Software Foundation; either version 2.0
11 * of the License, or (at your option) any later version.
13 * This library is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
16 * GNU General Public License for more details.
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with this library; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
21 ****************************************************************************/
32 #include <vlc_demux.h>
36 typedef struct rtp_source_t rtp_source_t;
38 /** State for a RTP session: */
48 rtp_source_create (demux_t *, const rtp_session_t *, uint32_t, uint16_t);
50 rtp_source_destroy (demux_t *, const rtp_session_t *, rtp_source_t *);
52 static void rtp_decode (demux_t *, const rtp_session_t *, rtp_source_t *);
55 * Creates a new RTP session.
58 rtp_session_create (demux_t *demux)
60 rtp_session_t *session = malloc (sizeof (*session));
75 * Destroys an RTP session.
77 void rtp_session_destroy (demux_t *demux, rtp_session_t *session)
79 for (unsigned i = 0; i < session->srcc; i++)
80 rtp_source_destroy (demux, session, session->srcv[i]);
88 static void *no_init (demux_t *demux)
94 static void no_destroy (demux_t *demux, void *opaque)
96 (void)demux; (void)opaque;
99 static void no_decode (demux_t *demux, void *opaque, block_t *block)
101 (void)demux; (void)opaque;
102 block_Release (block);
106 * Adds a payload type to an RTP session.
108 int rtp_add_type (demux_t *demux, rtp_session_t *ses, const rtp_pt_t *pt)
112 msg_Err (demux, "cannot change RTP payload formats during session");
116 rtp_pt_t *ppt = realloc (ses->ptv, (ses->ptc + 1) * sizeof (rtp_pt_t));
123 ppt->init = pt->init ? pt->init : no_init;
124 ppt->destroy = pt->destroy ? pt->destroy : no_destroy;
125 ppt->decode = pt->decode ? pt->decode : no_decode;
126 ppt->frequency = pt->frequency;
127 ppt->number = pt->number;
128 msg_Dbg (demux, "added payload type %"PRIu8" (f = %"PRIu32" Hz)",
129 ppt->number, ppt->frequency);
131 assert (ppt->frequency > 0); /* SIGFPE! */
136 /** State for an RTP source */
140 uint32_t jitter; /* interarrival delay jitter estimate */
141 mtime_t last_rx; /* last received packet local timestamp */
142 uint32_t last_ts; /* last received packet RTP timestamp */
144 uint16_t bad_seq; /* tentatively next expected sequence for resync */
145 uint16_t max_seq; /* next expected sequence */
147 uint16_t last_seq; /* sequence of the last dequeued packet */
148 block_t *blocks; /* re-ordered blocks queue */
149 void *opaque[0]; /* Per-source private payload data */
153 * Initializes a new RTP source within an RTP session.
155 static rtp_source_t *
156 rtp_source_create (demux_t *demux, const rtp_session_t *session,
157 uint32_t ssrc, uint16_t init_seq)
159 rtp_source_t *source;
161 source = malloc (sizeof (*source) + (sizeof (void *) * session->ptc));
167 source->max_seq = source->bad_seq = init_seq;
168 source->last_seq = init_seq - 1;
169 source->blocks = NULL;
171 /* Initializes all payload */
172 for (unsigned i = 0; i < session->ptc; i++)
173 source->opaque[i] = session->ptv[i].init (demux);
175 msg_Dbg (demux, "added RTP source (%08x)", ssrc);
181 * Destroys an RTP source and its associated streams.
184 rtp_source_destroy (demux_t *demux, const rtp_session_t *session,
185 rtp_source_t *source)
187 msg_Dbg (demux, "removing RTP source (%08x)", source->ssrc);
189 for (unsigned i = 0; i < session->ptc; i++)
190 session->ptv[i].destroy (demux, source->opaque[i]);
191 block_ChainRelease (source->blocks);
195 static inline uint8_t rtp_ptype (const block_t *block)
197 return block->p_buffer[1] & 0x7F;
200 static inline uint16_t rtp_seq (const block_t *block)
202 assert (block->i_buffer >= 4);
203 return GetWBE (block->p_buffer + 2);
206 static inline uint32_t rtp_timestamp (const block_t *block)
208 assert (block->i_buffer >= 12);
209 return GetDWBE (block->p_buffer + 4);
212 static const struct rtp_pt_t *
213 rtp_find_ptype (const rtp_session_t *session, rtp_source_t *source,
214 const block_t *block, void **pt_data)
216 uint8_t ptype = rtp_ptype (block);
218 for (unsigned i = 0; i < session->ptc; i++)
220 if (session->ptv[i].number == ptype)
223 *pt_data = source->opaque[i];
224 return &session->ptv[i];
231 * Receives an RTP packet and queues it.
232 * @param demux VLC demux object
233 * @param session RTP session receiving the packet
234 * @param block RTP packet including the RTP header
237 rtp_receive (demux_t *demux, rtp_session_t *session, block_t *block)
239 demux_sys_t *p_sys = demux->p_sys;
241 /* RTP header sanity checks (see RFC 3550) */
242 if (block->i_buffer < 12)
244 if ((block->p_buffer[0] >> 6 ) != 2) /* RTP version number */
247 /* Remove padding if present */
248 if (block->p_buffer[0] & 0x20)
250 uint8_t padding = block->p_buffer[block->i_buffer - 1];
251 if ((padding == 0) || (block->i_buffer < (12u + padding)))
252 goto drop; /* illegal value */
254 block->i_buffer -= padding;
257 mtime_t now = mdate ();
258 rtp_source_t *src = NULL;
259 const uint16_t seq = GetWBE (block->p_buffer + 2);
260 const uint32_t ssrc = GetDWBE (block->p_buffer + 8);
262 /* In most case, we know this source already */
263 for (unsigned i = 0, max = session->srcc; i < max; i++)
265 rtp_source_t *tmp = session->srcv[i];
266 if (tmp->ssrc == ssrc)
272 /* RTP source garbage collection */
273 if ((tmp->last_rx + (p_sys->timeout * CLOCK_FREQ)) < now)
275 rtp_source_destroy (demux, session, tmp);
276 if (--session->srcc > 0)
277 session->srcv[i] = session->srcv[session->srcc - 1];
284 if (session->srcc >= p_sys->max_src)
286 msg_Warn (demux, "too many RTP sessions");
291 tab = realloc (session->srcv, (session->srcc + 1) * sizeof (*tab));
296 src = rtp_source_create (demux, session, ssrc, seq);
300 tab[session->srcc++] = src;
301 /* Cannot compute jitter yet */
305 const rtp_pt_t *pt = rtp_find_ptype (session, src, block, NULL);
309 /* Recompute jitter estimate.
310 * That is computed from the RTP timestamps and the system clock.
311 * It is independent of RTP sequence. */
312 uint32_t freq = pt->frequency;
313 uint32_t ts = rtp_timestamp (block);
314 int64_t d = ((now - src->last_rx) * freq) / CLOCK_FREQ;
315 d -= ts - src->last_ts;
317 src->jitter += ((d - src->jitter) + 8) >> 4;
321 src->last_ts = rtp_timestamp (block);
323 /* Be optimistic for the first packet. Certain codec, such as Vorbis
324 * do not like loosing the first packet(s), so we cannot just wait
325 * for proper sequence synchronization. And we don't want to assume that
326 * the sender starts at seq=0 either. */
327 if (src->blocks == NULL)
328 src->max_seq = seq - p_sys->max_dropout;
330 /* Check sequence number */
331 /* NOTE: the sequence number is per-source,
332 * but is independent from the payload type. */
333 uint16_t delta_seq = seq - (src->max_seq + 1);
334 if ((delta_seq < 0x8000) ? (delta_seq > p_sys->max_dropout)
335 : ((65535 - delta_seq) > p_sys->max_misorder))
337 msg_Dbg (demux, "sequence discontinuity (got: %u, expected: %u)",
338 seq, (src->max_seq + 1) & 0xffff);
339 if (seq == ((src->bad_seq + 1) & 0xffff))
341 src->max_seq = src->bad_seq = seq;
342 msg_Warn (demux, "sequence resynchronized");
343 block_ChainRelease (src->blocks);
353 if (delta_seq < 0x8000)
356 /* Queues the block in sequence order,
357 * hence there is a single queue for all payload types. */
358 block_t **pp = &src->blocks;
359 for (block_t *prev = *pp; prev != NULL; prev = *pp)
361 int16_t delta_seq = seq - rtp_seq (prev);
365 goto drop; /* duplicate */
371 rtp_decode (demux, session, src);
375 block_Release (block);
380 rtp_decode (demux_t *demux, const rtp_session_t *session, rtp_source_t *src)
382 block_t *block = src->blocks;
384 /* Buffer underflow? */
385 if (!block || !block->p_next || !block->p_next->p_next)
387 /* TODO: use time rather than packet counts for buffer measurement */
388 src->blocks = block->p_next;
389 block->p_next = NULL;
391 /* Discontinuity detection */
392 if (((src->last_seq + 1) & 0xffff) != rtp_seq (block))
393 block->i_flags |= BLOCK_FLAG_DISCONTINUITY;
394 src->last_seq = rtp_seq (block);
396 /* Match the payload type */
398 const rtp_pt_t *pt = rtp_find_ptype (session, src, block, &pt_data);
401 msg_Dbg (demux, "ignoring unknown payload (%"PRIu8")",
406 /* Computes the PTS from the RTP timestamp and payload RTP frequency.
407 * DTS is unknown. Also, while the clock frequency depends on the payload
408 * format, a single source MUST only use payloads of a chosen frequency.
409 * Otherwise it would be impossible to compute consistent timestamps. */
410 /* FIXME: handle timestamp wrap properly */
411 /* TODO: sync multiple sources sanely... */
412 const uint32_t timestamp = rtp_timestamp (block);
413 block->i_pts = UINT64_C(1) * CLOCK_FREQ * timestamp / pt->frequency;
416 size_t skip = 12u + (block->p_buffer[0] & 0x0F) * 4;
418 /* Extension header (ignored for now) */
419 if (block->p_buffer[0] & 0x10)
422 if (block->i_buffer < skip)
425 skip += 4 * GetWBE (block->p_buffer + skip - 2);
428 if (block->i_buffer < skip)
431 block->p_buffer += skip;
432 block->i_buffer -= skip;
434 pt->decode (demux, pt_data, block);
438 block_Release (block);