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_xiph.h"
41 /* TODO: - add RTCP statistics reporting (should be optional).
43 - add support for h263/mpeg4 packetized output : IDEA: send a
44 buffer to 'rtp_write_packet' contains all the packets for ONE
45 frame. Each packet should have a four byte header containing
46 the length in big endian format (same trick as
47 'url_open_dyn_packet_buf')
50 /* statistics functions */
51 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
53 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", AVMEDIA_TYPE_VIDEO, CODEC_ID_MPEG4};
54 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", AVMEDIA_TYPE_AUDIO, CODEC_ID_AAC};
56 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
58 handler->next= RTPFirstDynamicPayloadHandler;
59 RTPFirstDynamicPayloadHandler= handler;
62 void av_register_rtp_dynamic_payload_handlers(void)
64 ff_register_dynamic_payload_handler(&mp4v_es_handler);
65 ff_register_dynamic_payload_handler(&mpeg4_generic_handler);
66 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
67 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
68 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
69 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
70 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
71 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
72 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
74 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
75 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
78 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
82 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
83 s->last_rtcp_timestamp = AV_RB32(buf + 16);
87 #define RTP_SEQ_MOD (1<<16)
90 * called on parse open packet
92 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
94 memset(s, 0, sizeof(RTPStatistics));
95 s->max_seq= base_sequence;
100 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
102 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
107 s->bad_seq= RTP_SEQ_MOD + 1;
109 s->expected_prior= 0;
110 s->received_prior= 0;
116 * returns 1 if we should handle this packet.
118 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
120 uint16_t udelta= seq - s->max_seq;
121 const int MAX_DROPOUT= 3000;
122 const int MAX_MISORDER = 100;
123 const int MIN_SEQUENTIAL = 2;
125 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
128 if(seq==s->max_seq + 1) {
131 if(s->probation==0) {
132 rtp_init_sequence(s, seq);
137 s->probation= MIN_SEQUENTIAL - 1;
140 } else if (udelta < MAX_DROPOUT) {
141 // in order, with permissible gap
142 if(seq < s->max_seq) {
143 //sequence number wrapped; count antother 64k cycles
144 s->cycles += RTP_SEQ_MOD;
147 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
148 // sequence made a large jump...
149 if(seq==s->bad_seq) {
150 // two sequential packets-- assume that the other side restarted without telling us; just resync.
151 rtp_init_sequence(s, seq);
153 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
157 // duplicate or reordered packet...
165 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
166 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
167 * never change. I left this in in case someone else can see a way. (rdm)
169 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
171 uint32_t transit= arrival_timestamp - sent_timestamp;
174 d= FFABS(transit - s->transit);
175 s->jitter += d - ((s->jitter + 8)>>4);
179 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
185 RTPStatistics *stats= &s->statistics;
187 uint32_t extended_max;
188 uint32_t expected_interval;
189 uint32_t received_interval;
190 uint32_t lost_interval;
193 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
195 if (!s->rtp_ctx || (count < 1))
198 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
199 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
200 s->octet_count += count;
201 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
203 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
206 s->last_octet_count = s->octet_count;
208 if (url_open_dyn_buf(&pb) < 0)
212 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
214 put_be16(pb, 7); /* length in words - 1 */
215 put_be32(pb, s->ssrc); // our own SSRC
216 put_be32(pb, s->ssrc); // XXX: should be the server's here!
217 // some placeholders we should really fill...
219 extended_max= stats->cycles + stats->max_seq;
220 expected= extended_max - stats->base_seq + 1;
221 lost= expected - stats->received;
222 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
223 expected_interval= expected - stats->expected_prior;
224 stats->expected_prior= expected;
225 received_interval= stats->received - stats->received_prior;
226 stats->received_prior= stats->received;
227 lost_interval= expected_interval - received_interval;
228 if (expected_interval==0 || lost_interval<=0) fraction= 0;
229 else fraction = (lost_interval<<8)/expected_interval;
231 fraction= (fraction<<24) | lost;
233 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
234 put_be32(pb, extended_max); /* max sequence received */
235 put_be32(pb, stats->jitter>>4); /* jitter */
237 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
239 put_be32(pb, 0); /* last SR timestamp */
240 put_be32(pb, 0); /* delay since last SR */
242 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
243 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
245 put_be32(pb, middle_32_bits); /* last SR timestamp */
246 put_be32(pb, delay_since_last); /* delay since last SR */
250 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
252 len = strlen(s->hostname);
253 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
254 put_be32(pb, s->ssrc);
257 put_buffer(pb, s->hostname, len);
259 for (len = (6 + len) % 4; len % 4; len++) {
263 put_flush_packet(pb);
264 len = url_close_dyn_buf(pb, &buf);
265 if ((len > 0) && buf) {
267 dprintf(s->ic, "sending %d bytes of RR\n", len);
268 result= url_write(s->rtp_ctx, buf, len);
269 dprintf(s->ic, "result from url_write: %d\n", result);
275 void rtp_send_punch_packets(URLContext* rtp_handle)
281 /* Send a small RTP packet */
282 if (url_open_dyn_buf(&pb) < 0)
285 put_byte(pb, (RTP_VERSION << 6));
286 put_byte(pb, 0); /* Payload type */
287 put_be16(pb, 0); /* Seq */
288 put_be32(pb, 0); /* Timestamp */
289 put_be32(pb, 0); /* SSRC */
291 put_flush_packet(pb);
292 len = url_close_dyn_buf(pb, &buf);
293 if ((len > 0) && buf)
294 url_write(rtp_handle, buf, len);
297 /* Send a minimal RTCP RR */
298 if (url_open_dyn_buf(&pb) < 0)
301 put_byte(pb, (RTP_VERSION << 6));
302 put_byte(pb, 201); /* receiver report */
303 put_be16(pb, 1); /* length in words - 1 */
304 put_be32(pb, 0); /* our own SSRC */
306 put_flush_packet(pb);
307 len = url_close_dyn_buf(pb, &buf);
308 if ((len > 0) && buf)
309 url_write(rtp_handle, buf, len);
315 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
316 * MPEG2TS streams to indicate that they should be demuxed inside the
317 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
318 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
320 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data)
324 s = av_mallocz(sizeof(RTPDemuxContext));
327 s->payload_type = payload_type;
328 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
331 s->rtp_payload_data = rtp_payload_data;
332 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
333 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
334 s->ts = ff_mpegts_parse_open(s->ic);
340 av_set_pts_info(st, 32, 1, 90000);
341 switch(st->codec->codec_id) {
342 case CODEC_ID_MPEG1VIDEO:
343 case CODEC_ID_MPEG2VIDEO:
349 st->need_parsing = AVSTREAM_PARSE_FULL;
352 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
353 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
358 // needed to send back RTCP RR in RTSP sessions
360 gethostname(s->hostname, sizeof(s->hostname));
365 rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
366 RTPDynamicProtocolHandler *handler)
368 s->dynamic_protocol_context = ctx;
369 s->parse_packet = handler->parse_packet;
372 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
374 int au_headers_length, au_header_size, i;
375 GetBitContext getbitcontext;
376 RTPPayloadData *infos;
378 infos = s->rtp_payload_data;
383 /* decode the first 2 bytes where the AUHeader sections are stored
385 au_headers_length = AV_RB16(buf);
387 if (au_headers_length > RTP_MAX_PACKET_LENGTH)
390 infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
392 /* skip AU headers length section (2 bytes) */
395 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
397 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
398 au_header_size = infos->sizelength + infos->indexlength;
399 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
402 infos->nb_au_headers = au_headers_length / au_header_size;
403 if (!infos->au_headers || infos->au_headers_allocated < infos->nb_au_headers) {
404 av_free(infos->au_headers);
405 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
406 infos->au_headers_allocated = infos->nb_au_headers;
409 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
410 In my test, the FAAD decoder does not behave correctly when sending each AU one by one
411 but does when sending the whole as one big packet... */
412 infos->au_headers[0].size = 0;
413 infos->au_headers[0].index = 0;
414 for (i = 0; i < infos->nb_au_headers; ++i) {
415 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
416 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
419 infos->nb_au_headers = 1;
425 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
427 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
429 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
433 /* compute pts from timestamp with received ntp_time */
434 delta_timestamp = timestamp - s->last_rtcp_timestamp;
435 /* convert to the PTS timebase */
436 addend = av_rescale(s->last_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);
437 pkt->pts = addend + delta_timestamp;
442 * Parse an RTP or RTCP packet directly sent as a buffer.
443 * @param s RTP parse context.
444 * @param pkt returned packet
445 * @param buf input buffer or NULL to read the next packets
446 * @param len buffer len
447 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
448 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
450 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
451 const uint8_t *buf, int len)
453 unsigned int ssrc, h;
454 int payload_type, seq, ret, flags = 0;
460 /* return the next packets, if any */
461 if(s->st && s->parse_packet) {
462 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
463 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
464 s->st, pkt, ×tamp, NULL, 0, flags);
465 finalize_packet(s, pkt, timestamp);
468 // TODO: Move to a dynamic packet handler (like above)
469 if (s->read_buf_index >= s->read_buf_size)
471 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
472 s->read_buf_size - s->read_buf_index);
475 s->read_buf_index += ret;
476 if (s->read_buf_index < s->read_buf_size)
486 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
488 if (buf[1] >= 200 && buf[1] <= 204) {
489 rtcp_parse_packet(s, buf, len);
492 payload_type = buf[1] & 0x7f;
494 flags |= RTP_FLAG_MARKER;
495 seq = AV_RB16(buf + 2);
496 timestamp = AV_RB32(buf + 4);
497 ssrc = AV_RB32(buf + 8);
498 /* store the ssrc in the RTPDemuxContext */
501 /* NOTE: we can handle only one payload type */
502 if (s->payload_type != payload_type)
506 // only do something with this if all the rtp checks pass...
507 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
509 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
510 payload_type, seq, ((s->seq + 1) & 0xffff));
519 /* specific MPEG2TS demux support */
520 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
524 s->read_buf_size = len - ret;
525 memcpy(s->buf, buf + ret, s->read_buf_size);
526 s->read_buf_index = 0;
530 } else if (s->parse_packet) {
531 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
532 s->st, pkt, ×tamp, buf, len, flags);
534 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
535 switch(st->codec->codec_id) {
538 /* better than nothing: skip mpeg audio RTP header */
544 av_new_packet(pkt, len);
545 memcpy(pkt->data, buf, len);
547 case CODEC_ID_MPEG1VIDEO:
548 case CODEC_ID_MPEG2VIDEO:
549 /* better than nothing: skip mpeg video RTP header */
562 av_new_packet(pkt, len);
563 memcpy(pkt->data, buf, len);
565 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles
567 // TODO: Put this into a dynamic packet handler...
569 if (rtp_parse_mp4_au(s, buf))
572 RTPPayloadData *infos = s->rtp_payload_data;
575 buf += infos->au_headers_length_bytes + 2;
576 len -= infos->au_headers_length_bytes + 2;
578 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
580 av_new_packet(pkt, infos->au_headers[0].size);
581 memcpy(pkt->data, buf, infos->au_headers[0].size);
582 buf += infos->au_headers[0].size;
583 len -= infos->au_headers[0].size;
585 s->read_buf_size = len;
589 av_new_packet(pkt, len);
590 memcpy(pkt->data, buf, len);
594 pkt->stream_index = st->index;
597 // now perform timestamp things....
598 finalize_packet(s, pkt, timestamp);
603 void rtp_parse_close(RTPDemuxContext *s)
605 // TODO: fold this into the protocol specific data fields.
606 av_free(s->rtp_payload_data->mode);
607 av_free(s->rtp_payload_data->au_headers);
608 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
609 ff_mpegts_parse_close(s->ts);