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"
35 #include "rtp_vorbis.h"
36 #include "rtpdec_h263.h"
40 /* TODO: - add RTCP statistics reporting (should be optional).
42 - add support for h263/mpeg4 packetized output : IDEA: send a
43 buffer to 'rtp_write_packet' contains all the packets for ONE
44 frame. Each packet should have a four byte header containing
45 the length in big endian format (same trick as
46 'url_open_dyn_packet_buf')
49 /* statistics functions */
50 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
52 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
53 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
55 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
57 handler->next= RTPFirstDynamicPayloadHandler;
58 RTPFirstDynamicPayloadHandler= handler;
61 void av_register_rtp_dynamic_payload_handlers(void)
63 ff_register_dynamic_payload_handler(&mp4v_es_handler);
64 ff_register_dynamic_payload_handler(&mpeg4_generic_handler);
65 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
66 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
67 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
68 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
70 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
71 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
74 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
78 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
79 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
80 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
81 s->last_rtcp_timestamp = AV_RB32(buf + 16);
85 #define RTP_SEQ_MOD (1<<16)
88 * called on parse open packet
90 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
92 memset(s, 0, sizeof(RTPStatistics));
93 s->max_seq= base_sequence;
98 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
100 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
105 s->bad_seq= RTP_SEQ_MOD + 1;
107 s->expected_prior= 0;
108 s->received_prior= 0;
114 * returns 1 if we should handle this packet.
116 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
118 uint16_t udelta= seq - s->max_seq;
119 const int MAX_DROPOUT= 3000;
120 const int MAX_MISORDER = 100;
121 const int MIN_SEQUENTIAL = 2;
123 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
126 if(seq==s->max_seq + 1) {
129 if(s->probation==0) {
130 rtp_init_sequence(s, seq);
135 s->probation= MIN_SEQUENTIAL - 1;
138 } else if (udelta < MAX_DROPOUT) {
139 // in order, with permissible gap
140 if(seq < s->max_seq) {
141 //sequence number wrapped; count antother 64k cycles
142 s->cycles += RTP_SEQ_MOD;
145 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
146 // sequence made a large jump...
147 if(seq==s->bad_seq) {
148 // two sequential packets-- assume that the other side restarted without telling us; just resync.
149 rtp_init_sequence(s, seq);
151 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
155 // duplicate or reordered packet...
163 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
164 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
165 * never change. I left this in in case someone else can see a way. (rdm)
167 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
169 uint32_t transit= arrival_timestamp - sent_timestamp;
172 d= FFABS(transit - s->transit);
173 s->jitter += d - ((s->jitter + 8)>>4);
177 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
183 RTPStatistics *stats= &s->statistics;
185 uint32_t extended_max;
186 uint32_t expected_interval;
187 uint32_t received_interval;
188 uint32_t lost_interval;
191 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
193 if (!s->rtp_ctx || (count < 1))
196 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
197 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
198 s->octet_count += count;
199 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
201 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
204 s->last_octet_count = s->octet_count;
206 if (url_open_dyn_buf(&pb) < 0)
210 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
212 put_be16(pb, 7); /* length in words - 1 */
213 put_be32(pb, s->ssrc); // our own SSRC
214 put_be32(pb, s->ssrc); // XXX: should be the server's here!
215 // some placeholders we should really fill...
217 extended_max= stats->cycles + stats->max_seq;
218 expected= extended_max - stats->base_seq + 1;
219 lost= expected - stats->received;
220 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
221 expected_interval= expected - stats->expected_prior;
222 stats->expected_prior= expected;
223 received_interval= stats->received - stats->received_prior;
224 stats->received_prior= stats->received;
225 lost_interval= expected_interval - received_interval;
226 if (expected_interval==0 || lost_interval<=0) fraction= 0;
227 else fraction = (lost_interval<<8)/expected_interval;
229 fraction= (fraction<<24) | lost;
231 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
232 put_be32(pb, extended_max); /* max sequence received */
233 put_be32(pb, stats->jitter>>4); /* jitter */
235 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
237 put_be32(pb, 0); /* last SR timestamp */
238 put_be32(pb, 0); /* delay since last SR */
240 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
241 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
243 put_be32(pb, middle_32_bits); /* last SR timestamp */
244 put_be32(pb, delay_since_last); /* delay since last SR */
248 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
250 len = strlen(s->hostname);
251 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
252 put_be32(pb, s->ssrc);
255 put_buffer(pb, s->hostname, len);
257 for (len = (6 + len) % 4; len % 4; len++) {
261 put_flush_packet(pb);
262 len = url_close_dyn_buf(pb, &buf);
263 if ((len > 0) && buf) {
265 dprintf(s->ic, "sending %d bytes of RR\n", len);
266 result= url_write(s->rtp_ctx, buf, len);
267 dprintf(s->ic, "result from url_write: %d\n", result);
274 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
275 * MPEG2TS streams to indicate that they should be demuxed inside the
276 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
277 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
279 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data)
283 s = av_mallocz(sizeof(RTPDemuxContext));
286 s->payload_type = payload_type;
287 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
288 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
291 s->rtp_payload_data = rtp_payload_data;
292 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
293 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
294 s->ts = mpegts_parse_open(s->ic);
300 av_set_pts_info(st, 32, 1, 90000);
301 switch(st->codec->codec_id) {
302 case CODEC_ID_MPEG1VIDEO:
303 case CODEC_ID_MPEG2VIDEO:
309 st->need_parsing = AVSTREAM_PARSE_FULL;
312 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
313 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
318 // needed to send back RTCP RR in RTSP sessions
320 gethostname(s->hostname, sizeof(s->hostname));
325 rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
326 RTPDynamicProtocolHandler *handler)
328 s->dynamic_protocol_context = ctx;
329 s->parse_packet = handler->parse_packet;
332 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
334 int au_headers_length, au_header_size, i;
335 GetBitContext getbitcontext;
336 RTPPayloadData *infos;
338 infos = s->rtp_payload_data;
343 /* decode the first 2 bytes where the AUHeader sections are stored
345 au_headers_length = AV_RB16(buf);
347 if (au_headers_length > RTP_MAX_PACKET_LENGTH)
350 infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
352 /* skip AU headers length section (2 bytes) */
355 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
357 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
358 au_header_size = infos->sizelength + infos->indexlength;
359 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
362 infos->nb_au_headers = au_headers_length / au_header_size;
363 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
365 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
366 In my test, the FAAD decoder does not behave correctly when sending each AU one by one
367 but does when sending the whole as one big packet... */
368 infos->au_headers[0].size = 0;
369 infos->au_headers[0].index = 0;
370 for (i = 0; i < infos->nb_au_headers; ++i) {
371 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
372 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
375 infos->nb_au_headers = 1;
381 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
383 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
385 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
389 /* compute pts from timestamp with received ntp_time */
390 delta_timestamp = timestamp - s->last_rtcp_timestamp;
391 /* convert to the PTS timebase */
392 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);
393 pkt->pts = addend + delta_timestamp;
398 * Parse an RTP or RTCP packet directly sent as a buffer.
399 * @param s RTP parse context.
400 * @param pkt returned packet
401 * @param buf input buffer or NULL to read the next packets
402 * @param len buffer len
403 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
404 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
406 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
407 const uint8_t *buf, int len)
409 unsigned int ssrc, h;
410 int payload_type, seq, ret, flags = 0;
416 /* return the next packets, if any */
417 if(s->st && s->parse_packet) {
418 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
419 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
420 s->st, pkt, ×tamp, NULL, 0, flags);
421 finalize_packet(s, pkt, timestamp);
424 // TODO: Move to a dynamic packet handler (like above)
425 if (s->read_buf_index >= s->read_buf_size)
427 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
428 s->read_buf_size - s->read_buf_index);
431 s->read_buf_index += ret;
432 if (s->read_buf_index < s->read_buf_size)
442 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
444 if (buf[1] >= 200 && buf[1] <= 204) {
445 rtcp_parse_packet(s, buf, len);
448 payload_type = buf[1] & 0x7f;
450 flags |= RTP_FLAG_MARKER;
451 seq = AV_RB16(buf + 2);
452 timestamp = AV_RB32(buf + 4);
453 ssrc = AV_RB32(buf + 8);
454 /* store the ssrc in the RTPDemuxContext */
457 /* NOTE: we can handle only one payload type */
458 if (s->payload_type != payload_type)
462 // only do something with this if all the rtp checks pass...
463 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
465 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
466 payload_type, seq, ((s->seq + 1) & 0xffff));
475 /* specific MPEG2TS demux support */
476 ret = mpegts_parse_packet(s->ts, pkt, buf, len);
480 s->read_buf_size = len - ret;
481 memcpy(s->buf, buf + ret, s->read_buf_size);
482 s->read_buf_index = 0;
486 } else if (s->parse_packet) {
487 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
488 s->st, pkt, ×tamp, buf, len, flags);
490 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
491 switch(st->codec->codec_id) {
494 /* better than nothing: skip mpeg audio RTP header */
500 av_new_packet(pkt, len);
501 memcpy(pkt->data, buf, len);
503 case CODEC_ID_MPEG1VIDEO:
504 case CODEC_ID_MPEG2VIDEO:
505 /* better than nothing: skip mpeg video RTP header */
518 av_new_packet(pkt, len);
519 memcpy(pkt->data, buf, len);
521 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles
523 // TODO: Put this into a dynamic packet handler...
525 if (rtp_parse_mp4_au(s, buf))
528 RTPPayloadData *infos = s->rtp_payload_data;
531 buf += infos->au_headers_length_bytes + 2;
532 len -= infos->au_headers_length_bytes + 2;
534 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
536 av_new_packet(pkt, infos->au_headers[0].size);
537 memcpy(pkt->data, buf, infos->au_headers[0].size);
538 buf += infos->au_headers[0].size;
539 len -= infos->au_headers[0].size;
541 s->read_buf_size = len;
545 av_new_packet(pkt, len);
546 memcpy(pkt->data, buf, len);
550 pkt->stream_index = st->index;
553 // now perform timestamp things....
554 finalize_packet(s, pkt, timestamp);
559 void rtp_parse_close(RTPDemuxContext *s)
561 // TODO: fold this into the protocol specific data fields.
562 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
563 mpegts_parse_close(s->ts);