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/bitstream.h"
32 #include "rtp_internal.h"
37 /* TODO: - add RTCP statistics reporting (should be optional).
39 - add support for h263/mpeg4 packetized output : IDEA: send a
40 buffer to 'rtp_write_packet' contains all the packets for ONE
41 frame. Each packet should have a four byte header containing
42 the length in big endian format (same trick as
43 'url_open_dyn_packet_buf')
46 /* statistics functions */
47 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
49 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
50 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
52 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
54 handler->next= RTPFirstDynamicPayloadHandler;
55 RTPFirstDynamicPayloadHandler= handler;
58 void av_register_rtp_dynamic_payload_handlers(void)
60 ff_register_dynamic_payload_handler(&mp4v_es_handler);
61 ff_register_dynamic_payload_handler(&mpeg4_generic_handler);
62 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
65 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
69 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
70 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
71 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
72 s->last_rtcp_timestamp = AV_RB32(buf + 16);
76 #define RTP_SEQ_MOD (1<<16)
79 * called on parse open packet
81 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
83 memset(s, 0, sizeof(RTPStatistics));
84 s->max_seq= base_sequence;
89 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
91 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
96 s->bad_seq= RTP_SEQ_MOD + 1;
105 * returns 1 if we should handle this packet.
107 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
109 uint16_t udelta= seq - s->max_seq;
110 const int MAX_DROPOUT= 3000;
111 const int MAX_MISORDER = 100;
112 const int MIN_SEQUENTIAL = 2;
114 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
117 if(seq==s->max_seq + 1) {
120 if(s->probation==0) {
121 rtp_init_sequence(s, seq);
126 s->probation= MIN_SEQUENTIAL - 1;
129 } else if (udelta < MAX_DROPOUT) {
130 // in order, with permissible gap
131 if(seq < s->max_seq) {
132 //sequence number wrapped; count antother 64k cycles
133 s->cycles += RTP_SEQ_MOD;
136 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
137 // sequence made a large jump...
138 if(seq==s->bad_seq) {
139 // two sequential packets-- assume that the other side restarted without telling us; just resync.
140 rtp_init_sequence(s, seq);
142 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
146 // duplicate or reordered packet...
154 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
155 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
156 * never change. I left this in in case someone else can see a way. (rdm)
158 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
160 uint32_t transit= arrival_timestamp - sent_timestamp;
163 d= FFABS(transit - s->transit);
164 s->jitter += d - ((s->jitter + 8)>>4);
168 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
174 RTPStatistics *stats= &s->statistics;
176 uint32_t extended_max;
177 uint32_t expected_interval;
178 uint32_t received_interval;
179 uint32_t lost_interval;
182 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
184 if (!s->rtp_ctx || (count < 1))
187 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
188 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
189 s->octet_count += count;
190 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
192 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
195 s->last_octet_count = s->octet_count;
197 if (url_open_dyn_buf(&pb) < 0)
201 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
203 put_be16(pb, 7); /* length in words - 1 */
204 put_be32(pb, s->ssrc); // our own SSRC
205 put_be32(pb, s->ssrc); // XXX: should be the server's here!
206 // some placeholders we should really fill...
208 extended_max= stats->cycles + stats->max_seq;
209 expected= extended_max - stats->base_seq + 1;
210 lost= expected - stats->received;
211 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
212 expected_interval= expected - stats->expected_prior;
213 stats->expected_prior= expected;
214 received_interval= stats->received - stats->received_prior;
215 stats->received_prior= stats->received;
216 lost_interval= expected_interval - received_interval;
217 if (expected_interval==0 || lost_interval<=0) fraction= 0;
218 else fraction = (lost_interval<<8)/expected_interval;
220 fraction= (fraction<<24) | lost;
222 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
223 put_be32(pb, extended_max); /* max sequence received */
224 put_be32(pb, stats->jitter>>4); /* jitter */
226 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
228 put_be32(pb, 0); /* last SR timestamp */
229 put_be32(pb, 0); /* delay since last SR */
231 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
232 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
234 put_be32(pb, middle_32_bits); /* last SR timestamp */
235 put_be32(pb, delay_since_last); /* delay since last SR */
239 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
241 len = strlen(s->hostname);
242 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
243 put_be32(pb, s->ssrc);
246 put_buffer(pb, s->hostname, len);
248 for (len = (6 + len) % 4; len % 4; len++) {
252 put_flush_packet(pb);
253 len = url_close_dyn_buf(pb, &buf);
254 if ((len > 0) && buf) {
256 dprintf(s->ic, "sending %d bytes of RR\n", len);
257 result= url_write(s->rtp_ctx, buf, len);
258 dprintf(s->ic, "result from url_write: %d\n", result);
265 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
266 * MPEG2TS streams to indicate that they should be demuxed inside the
267 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
268 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
270 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
274 s = av_mallocz(sizeof(RTPDemuxContext));
277 s->payload_type = payload_type;
278 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
279 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
282 s->rtp_payload_data = rtp_payload_data;
283 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
284 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
285 s->ts = mpegts_parse_open(s->ic);
291 av_set_pts_info(st, 32, 1, 90000);
292 switch(st->codec->codec_id) {
293 case CODEC_ID_MPEG1VIDEO:
294 case CODEC_ID_MPEG2VIDEO:
299 st->need_parsing = AVSTREAM_PARSE_FULL;
302 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
303 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
308 // needed to send back RTCP RR in RTSP sessions
310 gethostname(s->hostname, sizeof(s->hostname));
314 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
316 int au_headers_length, au_header_size, i;
317 GetBitContext getbitcontext;
318 rtp_payload_data_t *infos;
320 infos = s->rtp_payload_data;
325 /* decode the first 2 bytes where the AUHeader sections are stored
327 au_headers_length = AV_RB16(buf);
329 if (au_headers_length > RTP_MAX_PACKET_LENGTH)
332 infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
334 /* skip AU headers length section (2 bytes) */
337 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
339 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
340 au_header_size = infos->sizelength + infos->indexlength;
341 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
344 infos->nb_au_headers = au_headers_length / au_header_size;
345 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
347 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
348 In my test, the FAAD decoder does not behave correctly when sending each AU one by one
349 but does when sending the whole as one big packet... */
350 infos->au_headers[0].size = 0;
351 infos->au_headers[0].index = 0;
352 for (i = 0; i < infos->nb_au_headers; ++i) {
353 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
354 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
357 infos->nb_au_headers = 1;
363 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
365 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
367 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
371 /* compute pts from timestamp with received ntp_time */
372 delta_timestamp = timestamp - s->last_rtcp_timestamp;
373 /* convert to the PTS timebase */
374 addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);
375 pkt->pts = addend + delta_timestamp;
377 pkt->stream_index = s->st->index;
381 * Parse an RTP or RTCP packet directly sent as a buffer.
382 * @param s RTP parse context.
383 * @param pkt returned packet
384 * @param buf input buffer or NULL to read the next packets
385 * @param len buffer len
386 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
387 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
389 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
390 const uint8_t *buf, int len)
392 unsigned int ssrc, h;
393 int payload_type, seq, ret, flags = 0;
399 /* return the next packets, if any */
400 if(s->st && s->parse_packet) {
401 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
402 rv= s->parse_packet(s->dynamic_protocol_context,
403 s->st, pkt, ×tamp, NULL, 0, flags);
404 finalize_packet(s, pkt, timestamp);
407 // TODO: Move to a dynamic packet handler (like above)
408 if (s->read_buf_index >= s->read_buf_size)
410 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
411 s->read_buf_size - s->read_buf_index);
414 s->read_buf_index += ret;
415 if (s->read_buf_index < s->read_buf_size)
425 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
427 if (buf[1] >= 200 && buf[1] <= 204) {
428 rtcp_parse_packet(s, buf, len);
431 payload_type = buf[1] & 0x7f;
432 seq = AV_RB16(buf + 2);
433 timestamp = AV_RB32(buf + 4);
434 ssrc = AV_RB32(buf + 8);
435 /* store the ssrc in the RTPDemuxContext */
438 /* NOTE: we can handle only one payload type */
439 if (s->payload_type != payload_type)
443 // only do something with this if all the rtp checks pass...
444 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
446 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
447 payload_type, seq, ((s->seq + 1) & 0xffff));
456 /* specific MPEG2TS demux support */
457 ret = mpegts_parse_packet(s->ts, pkt, buf, len);
461 s->read_buf_size = len - ret;
462 memcpy(s->buf, buf + ret, s->read_buf_size);
463 s->read_buf_index = 0;
466 } else if (s->parse_packet) {
467 rv = s->parse_packet(s->dynamic_protocol_context,
468 s->st, pkt, ×tamp, buf, len, flags);
470 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
471 switch(st->codec->codec_id) {
473 /* better than nothing: skip mpeg audio RTP header */
479 av_new_packet(pkt, len);
480 memcpy(pkt->data, buf, len);
482 case CODEC_ID_MPEG1VIDEO:
483 case CODEC_ID_MPEG2VIDEO:
484 /* better than nothing: skip mpeg video RTP header */
497 av_new_packet(pkt, len);
498 memcpy(pkt->data, buf, len);
500 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles
502 // TODO: Put this into a dynamic packet handler...
504 if (rtp_parse_mp4_au(s, buf))
507 rtp_payload_data_t *infos = s->rtp_payload_data;
510 buf += infos->au_headers_length_bytes + 2;
511 len -= infos->au_headers_length_bytes + 2;
513 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
515 av_new_packet(pkt, infos->au_headers[0].size);
516 memcpy(pkt->data, buf, infos->au_headers[0].size);
517 buf += infos->au_headers[0].size;
518 len -= infos->au_headers[0].size;
520 s->read_buf_size = len;
524 av_new_packet(pkt, len);
525 memcpy(pkt->data, buf, len);
529 // now perform timestamp things....
530 finalize_packet(s, pkt, timestamp);
535 void rtp_parse_close(RTPDemuxContext *s)
537 // TODO: fold this into the protocol specific data fields.
538 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
539 mpegts_parse_close(s->ts);