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 uint16_t rtp_seq (const block_t *block)
197 assert (block->i_buffer >= 4);
198 return GetWBE (block->p_buffer + 2);
201 static inline uint32_t rtp_timestamp (const block_t *block)
203 assert (block->i_buffer >= 12);
204 return GetDWBE (block->p_buffer + 4);
207 static const struct rtp_pt_t *
208 rtp_find_ptype (const rtp_session_t *session, rtp_source_t *source,
209 const block_t *block, void **pt_data)
211 uint8_t ptype = rtp_ptype (block);
213 for (unsigned i = 0; i < session->ptc; i++)
215 if (session->ptv[i].number == ptype)
218 *pt_data = source->opaque[i];
219 return &session->ptv[i];
226 * Receives an RTP packet and queues it.
227 * @param demux VLC demux object
228 * @param session RTP session receiving the packet
229 * @param block RTP packet including the RTP header
232 rtp_queue (demux_t *demux, rtp_session_t *session, block_t *block)
234 demux_sys_t *p_sys = demux->p_sys;
236 /* RTP header sanity checks (see RFC 3550) */
237 if (block->i_buffer < 12)
239 if ((block->p_buffer[0] >> 6 ) != 2) /* RTP version number */
242 /* Remove padding if present */
243 if (block->p_buffer[0] & 0x20)
245 uint8_t padding = block->p_buffer[block->i_buffer - 1];
246 if ((padding == 0) || (block->i_buffer < (12u + padding)))
247 goto drop; /* illegal value */
249 block->i_buffer -= padding;
252 mtime_t now = mdate ();
253 rtp_source_t *src = NULL;
254 const uint16_t seq = GetWBE (block->p_buffer + 2);
255 const uint32_t ssrc = GetDWBE (block->p_buffer + 8);
257 /* In most case, we know this source already */
258 for (unsigned i = 0, max = session->srcc; i < max; i++)
260 rtp_source_t *tmp = session->srcv[i];
261 if (tmp->ssrc == ssrc)
267 /* RTP source garbage collection */
268 if ((tmp->last_rx + (p_sys->timeout * CLOCK_FREQ)) < now)
270 rtp_source_destroy (demux, session, tmp);
271 if (--session->srcc > 0)
272 session->srcv[i] = session->srcv[session->srcc - 1];
279 if (session->srcc >= p_sys->max_src)
281 msg_Warn (demux, "too many RTP sessions");
286 tab = realloc (session->srcv, (session->srcc + 1) * sizeof (*tab));
291 src = rtp_source_create (demux, session, ssrc, seq);
295 tab[session->srcc++] = src;
296 /* Cannot compute jitter yet */
300 const rtp_pt_t *pt = rtp_find_ptype (session, src, block, NULL);
304 /* Recompute jitter estimate.
305 * That is computed from the RTP timestamps and the system clock.
306 * It is independent of RTP sequence. */
307 uint32_t freq = pt->frequency;
308 uint32_t ts = rtp_timestamp (block);
309 int64_t d = ((now - src->last_rx) * freq) / CLOCK_FREQ;
310 d -= ts - src->last_ts;
312 src->jitter += ((d - src->jitter) + 8) >> 4;
316 src->last_ts = rtp_timestamp (block);
318 /* Be optimistic for the first packet. Certain codec, such as Vorbis
319 * do not like loosing the first packet(s), so we cannot just wait
320 * for proper sequence synchronization. And we don't want to assume that
321 * the sender starts at seq=0 either. */
322 if (src->blocks == NULL)
323 src->max_seq = seq - p_sys->max_dropout;
325 /* Check sequence number */
326 /* NOTE: the sequence number is per-source,
327 * but is independent from the payload type. */
328 uint16_t delta_seq = seq - (src->max_seq + 1);
329 if ((delta_seq < 0x8000) ? (delta_seq > p_sys->max_dropout)
330 : ((65535 - delta_seq) > p_sys->max_misorder))
332 msg_Dbg (demux, "sequence discontinuity (got: %u, expected: %u)",
333 seq, (src->max_seq + 1) & 0xffff);
334 if (seq == ((src->bad_seq + 1) & 0xffff))
336 src->max_seq = src->bad_seq = seq;
337 msg_Warn (demux, "sequence resynchronized");
338 block_ChainRelease (src->blocks);
348 if (delta_seq < 0x8000)
351 /* Queues the block in sequence order,
352 * hence there is a single queue for all payload types. */
353 block_t **pp = &src->blocks;
354 for (block_t *prev = *pp; prev != NULL; prev = *pp)
356 int16_t delta_seq = seq - rtp_seq (prev);
360 goto drop; /* duplicate */
366 /*rtp_decode (demux, session, src);*/
370 block_Release (block);
375 rtp_decode (demux_t *demux, const rtp_session_t *session, rtp_source_t *src)
377 block_t *block = src->blocks;
380 src->blocks = block->p_next;
381 block->p_next = NULL;
383 /* Discontinuity detection */
384 uint16_t delta_seq = rtp_seq (block) - (src->last_seq + 1);
387 if (delta_seq >= 0x8000)
388 { /* Unrecoverable if later packets have already been dequeued */
389 msg_Warn (demux, "ignoring late packet (sequence: %u)",
393 block->i_flags |= BLOCK_FLAG_DISCONTINUITY;
395 src->last_seq = rtp_seq (block);
397 /* Match the payload type */
399 const rtp_pt_t *pt = rtp_find_ptype (session, src, block, &pt_data);
402 msg_Dbg (demux, "ignoring unknown payload (%"PRIu8")",
407 /* Computes the PTS from the RTP timestamp and payload RTP frequency.
408 * DTS is unknown. Also, while the clock frequency depends on the payload
409 * format, a single source MUST only use payloads of a chosen frequency.
410 * Otherwise it would be impossible to compute consistent timestamps. */
411 /* FIXME: handle timestamp wrap properly */
412 /* TODO: inter-medias/sessions sync (using RTCP-SR) */
413 const uint32_t timestamp = rtp_timestamp (block);
414 block->i_pts = UINT64_C(1) * CLOCK_FREQ * timestamp / pt->frequency;
417 size_t skip = 12u + (block->p_buffer[0] & 0x0F) * 4;
419 /* Extension header (ignored for now) */
420 if (block->p_buffer[0] & 0x10)
423 if (block->i_buffer < skip)
426 skip += 4 * GetWBE (block->p_buffer + skip - 2);
429 if (block->i_buffer < skip)
432 block->p_buffer += skip;
433 block->i_buffer -= skip;
435 pt->decode (demux, pt_data, block);
439 block_Release (block);
443 bool rtp_dequeue (demux_t *demux, const rtp_session_t *session,
444 mtime_t *restrict deadlinep)
446 mtime_t now = mdate ();
447 bool pending = false;
449 for (unsigned i = 0, max = session->srcc; i < max; i++)
451 rtp_source_t *src = session->srcv[i];
454 /* Because of IP packet delay variation (IPDV), we need to guesstimate
455 * how long to wait for a missing packet in the RTP sequence
456 * (see RFC3393 for background on IPDV).
458 * This situation occurs if a packet got lost, or if the network has
459 * re-ordered packets. Unfortunately, the MSL is 2 minutes, orders of
460 * magnitude too long for multimedia. We need a tradeoff.
461 * If we underestimated IPDV, we may have to discard valid but late
462 * packets. If we overestimate it, we will either cause too much
463 * delay, or worse, underflow our downstream buffers, as we wait for
464 * definitely a lost packets.
466 * The rest of the "de-jitter buffer" work is done by the interval
467 * LibVLC E/S-out clock synchronization. Here, we need to bother about
468 * re-ordering packets, as decoders can't cope with mis-ordered data.
470 while (((block = src->blocks)) != NULL)
473 if (rtp_seq (block) == ((src->last_seq + 1) & 0xffff))
474 { /* Next block ready, no need to wait */
475 rtp_decode (demux, session, src);
479 /* Wait for 3 times the inter-arrival delay variance (about 99.7%
480 * match for random gaussian jitter). Additionnaly, we implicitly
481 * wait for misordering times the packetization time.
483 mtime_t deadline = src->last_rx;
484 const rtp_pt_t *pt = rtp_find_ptype (session, src, block, NULL);
486 deadline += UINT64_C(3) * CLOCK_FREQ * src->jitter
491 rtp_decode (demux, session, src);
494 if (*deadlinep > deadline)
495 *deadlinep = deadline;
496 pending = true; /* packet pending in buffer */