3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 /* needed for gethostname() */
23 #define _XOPEN_SOURCE 600
25 #include "libavcodec/get_bits.h"
33 #include "rtpdec_amr.h"
34 #include "rtpdec_asf.h"
35 #include "rtpdec_h263.h"
36 #include "rtpdec_h264.h"
37 #include "rtpdec_vorbis.h"
38 #include "rtpdec_theora.h"
42 /* TODO: - add RTCP statistics reporting (should be optional).
44 - add support for h263/mpeg4 packetized output : IDEA: send a
45 buffer to 'rtp_write_packet' contains all the packets for ONE
46 frame. Each packet should have a four byte header containing
47 the length in big endian format (same trick as
48 'url_open_dyn_packet_buf')
51 /* statistics functions */
52 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
54 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
55 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
57 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
59 handler->next= RTPFirstDynamicPayloadHandler;
60 RTPFirstDynamicPayloadHandler= handler;
63 void av_register_rtp_dynamic_payload_handlers(void)
65 ff_register_dynamic_payload_handler(&mp4v_es_handler);
66 ff_register_dynamic_payload_handler(&mpeg4_generic_handler);
67 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
68 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
69 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
70 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
71 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
72 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
73 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
75 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
76 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
79 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
83 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
84 s->last_rtcp_timestamp = AV_RB32(buf + 16);
88 #define RTP_SEQ_MOD (1<<16)
91 * called on parse open packet
93 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
95 memset(s, 0, sizeof(RTPStatistics));
96 s->max_seq= base_sequence;
101 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
103 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
108 s->bad_seq= RTP_SEQ_MOD + 1;
110 s->expected_prior= 0;
111 s->received_prior= 0;
117 * returns 1 if we should handle this packet.
119 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
121 uint16_t udelta= seq - s->max_seq;
122 const int MAX_DROPOUT= 3000;
123 const int MAX_MISORDER = 100;
124 const int MIN_SEQUENTIAL = 2;
126 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
129 if(seq==s->max_seq + 1) {
132 if(s->probation==0) {
133 rtp_init_sequence(s, seq);
138 s->probation= MIN_SEQUENTIAL - 1;
141 } else if (udelta < MAX_DROPOUT) {
142 // in order, with permissible gap
143 if(seq < s->max_seq) {
144 //sequence number wrapped; count antother 64k cycles
145 s->cycles += RTP_SEQ_MOD;
148 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
149 // sequence made a large jump...
150 if(seq==s->bad_seq) {
151 // two sequential packets-- assume that the other side restarted without telling us; just resync.
152 rtp_init_sequence(s, seq);
154 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
158 // duplicate or reordered packet...
166 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
167 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
168 * never change. I left this in in case someone else can see a way. (rdm)
170 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
172 uint32_t transit= arrival_timestamp - sent_timestamp;
175 d= FFABS(transit - s->transit);
176 s->jitter += d - ((s->jitter + 8)>>4);
180 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
186 RTPStatistics *stats= &s->statistics;
188 uint32_t extended_max;
189 uint32_t expected_interval;
190 uint32_t received_interval;
191 uint32_t lost_interval;
194 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
196 if (!s->rtp_ctx || (count < 1))
199 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
200 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
201 s->octet_count += count;
202 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
204 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
207 s->last_octet_count = s->octet_count;
209 if (url_open_dyn_buf(&pb) < 0)
213 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
215 put_be16(pb, 7); /* length in words - 1 */
216 put_be32(pb, s->ssrc); // our own SSRC
217 put_be32(pb, s->ssrc); // XXX: should be the server's here!
218 // some placeholders we should really fill...
220 extended_max= stats->cycles + stats->max_seq;
221 expected= extended_max - stats->base_seq + 1;
222 lost= expected - stats->received;
223 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
224 expected_interval= expected - stats->expected_prior;
225 stats->expected_prior= expected;
226 received_interval= stats->received - stats->received_prior;
227 stats->received_prior= stats->received;
228 lost_interval= expected_interval - received_interval;
229 if (expected_interval==0 || lost_interval<=0) fraction= 0;
230 else fraction = (lost_interval<<8)/expected_interval;
232 fraction= (fraction<<24) | lost;
234 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
235 put_be32(pb, extended_max); /* max sequence received */
236 put_be32(pb, stats->jitter>>4); /* jitter */
238 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
240 put_be32(pb, 0); /* last SR timestamp */
241 put_be32(pb, 0); /* delay since last SR */
243 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
244 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
246 put_be32(pb, middle_32_bits); /* last SR timestamp */
247 put_be32(pb, delay_since_last); /* delay since last SR */
251 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
253 len = strlen(s->hostname);
254 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
255 put_be32(pb, s->ssrc);
258 put_buffer(pb, s->hostname, len);
260 for (len = (6 + len) % 4; len % 4; len++) {
264 put_flush_packet(pb);
265 len = url_close_dyn_buf(pb, &buf);
266 if ((len > 0) && buf) {
268 dprintf(s->ic, "sending %d bytes of RR\n", len);
269 result= url_write(s->rtp_ctx, buf, len);
270 dprintf(s->ic, "result from url_write: %d\n", result);
276 void rtp_send_punch_packets(URLContext* rtp_handle)
282 /* Send a small RTP packet */
283 if (url_open_dyn_buf(&pb) < 0)
286 put_byte(pb, (RTP_VERSION << 6));
287 put_byte(pb, 0); /* Payload type */
288 put_be16(pb, 0); /* Seq */
289 put_be32(pb, 0); /* Timestamp */
290 put_be32(pb, 0); /* SSRC */
292 put_flush_packet(pb);
293 len = url_close_dyn_buf(pb, &buf);
294 if ((len > 0) && buf)
295 url_write(rtp_handle, buf, len);
298 /* Send a minimal RTCP RR */
299 if (url_open_dyn_buf(&pb) < 0)
302 put_byte(pb, (RTP_VERSION << 6));
303 put_byte(pb, 201); /* receiver report */
304 put_be16(pb, 1); /* length in words - 1 */
305 put_be32(pb, 0); /* our own SSRC */
307 put_flush_packet(pb);
308 len = url_close_dyn_buf(pb, &buf);
309 if ((len > 0) && buf)
310 url_write(rtp_handle, buf, len);
316 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
317 * MPEG2TS streams to indicate that they should be demuxed inside the
318 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
319 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
321 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data)
325 s = av_mallocz(sizeof(RTPDemuxContext));
328 s->payload_type = payload_type;
329 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
332 s->rtp_payload_data = rtp_payload_data;
333 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
334 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
335 s->ts = ff_mpegts_parse_open(s->ic);
341 av_set_pts_info(st, 32, 1, 90000);
342 switch(st->codec->codec_id) {
343 case CODEC_ID_MPEG1VIDEO:
344 case CODEC_ID_MPEG2VIDEO:
350 st->need_parsing = AVSTREAM_PARSE_FULL;
353 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
354 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
359 // needed to send back RTCP RR in RTSP sessions
361 gethostname(s->hostname, sizeof(s->hostname));
366 rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
367 RTPDynamicProtocolHandler *handler)
369 s->dynamic_protocol_context = ctx;
370 s->parse_packet = handler->parse_packet;
373 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
375 int au_headers_length, au_header_size, i;
376 GetBitContext getbitcontext;
377 RTPPayloadData *infos;
379 infos = s->rtp_payload_data;
384 /* decode the first 2 bytes where the AUHeader sections are stored
386 au_headers_length = AV_RB16(buf);
388 if (au_headers_length > RTP_MAX_PACKET_LENGTH)
391 infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
393 /* skip AU headers length section (2 bytes) */
396 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
398 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
399 au_header_size = infos->sizelength + infos->indexlength;
400 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
403 infos->nb_au_headers = au_headers_length / au_header_size;
404 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
406 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
407 In my test, the FAAD decoder does not behave correctly when sending each AU one by one
408 but does when sending the whole as one big packet... */
409 infos->au_headers[0].size = 0;
410 infos->au_headers[0].index = 0;
411 for (i = 0; i < infos->nb_au_headers; ++i) {
412 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
413 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
416 infos->nb_au_headers = 1;
422 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
424 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
426 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
430 /* compute pts from timestamp with received ntp_time */
431 delta_timestamp = timestamp - s->last_rtcp_timestamp;
432 /* convert to the PTS timebase */
433 addend = av_rescale(s->last_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);
434 pkt->pts = addend + delta_timestamp;
439 * Parse an RTP or RTCP packet directly sent as a buffer.
440 * @param s RTP parse context.
441 * @param pkt returned packet
442 * @param buf input buffer or NULL to read the next packets
443 * @param len buffer len
444 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
445 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
447 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
448 const uint8_t *buf, int len)
450 unsigned int ssrc, h;
451 int payload_type, seq, ret, flags = 0;
457 /* return the next packets, if any */
458 if(s->st && s->parse_packet) {
459 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
460 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
461 s->st, pkt, ×tamp, NULL, 0, flags);
462 finalize_packet(s, pkt, timestamp);
465 // TODO: Move to a dynamic packet handler (like above)
466 if (s->read_buf_index >= s->read_buf_size)
468 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
469 s->read_buf_size - s->read_buf_index);
472 s->read_buf_index += ret;
473 if (s->read_buf_index < s->read_buf_size)
483 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
485 if (buf[1] >= 200 && buf[1] <= 204) {
486 rtcp_parse_packet(s, buf, len);
489 payload_type = buf[1] & 0x7f;
491 flags |= RTP_FLAG_MARKER;
492 seq = AV_RB16(buf + 2);
493 timestamp = AV_RB32(buf + 4);
494 ssrc = AV_RB32(buf + 8);
495 /* store the ssrc in the RTPDemuxContext */
498 /* NOTE: we can handle only one payload type */
499 if (s->payload_type != payload_type)
503 // only do something with this if all the rtp checks pass...
504 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
506 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
507 payload_type, seq, ((s->seq + 1) & 0xffff));
516 /* specific MPEG2TS demux support */
517 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
521 s->read_buf_size = len - ret;
522 memcpy(s->buf, buf + ret, s->read_buf_size);
523 s->read_buf_index = 0;
527 } else if (s->parse_packet) {
528 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
529 s->st, pkt, ×tamp, buf, len, flags);
531 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
532 switch(st->codec->codec_id) {
535 /* better than nothing: skip mpeg audio RTP header */
541 av_new_packet(pkt, len);
542 memcpy(pkt->data, buf, len);
544 case CODEC_ID_MPEG1VIDEO:
545 case CODEC_ID_MPEG2VIDEO:
546 /* better than nothing: skip mpeg video RTP header */
559 av_new_packet(pkt, len);
560 memcpy(pkt->data, buf, len);
562 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles
564 // TODO: Put this into a dynamic packet handler...
566 if (rtp_parse_mp4_au(s, buf))
569 RTPPayloadData *infos = s->rtp_payload_data;
572 buf += infos->au_headers_length_bytes + 2;
573 len -= infos->au_headers_length_bytes + 2;
575 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
577 av_new_packet(pkt, infos->au_headers[0].size);
578 memcpy(pkt->data, buf, infos->au_headers[0].size);
579 buf += infos->au_headers[0].size;
580 len -= infos->au_headers[0].size;
582 s->read_buf_size = len;
586 av_new_packet(pkt, len);
587 memcpy(pkt->data, buf, len);
591 pkt->stream_index = st->index;
594 // now perform timestamp things....
595 finalize_packet(s, pkt, timestamp);
600 void rtp_parse_close(RTPDemuxContext *s)
602 // TODO: fold this into the protocol specific data fields.
603 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
604 ff_mpegts_parse_close(s->ts);