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 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
84 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
85 s->last_rtcp_timestamp = AV_RB32(buf + 16);
89 #define RTP_SEQ_MOD (1<<16)
92 * called on parse open packet
94 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
96 memset(s, 0, sizeof(RTPStatistics));
97 s->max_seq= base_sequence;
102 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
104 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
109 s->bad_seq= RTP_SEQ_MOD + 1;
111 s->expected_prior= 0;
112 s->received_prior= 0;
118 * returns 1 if we should handle this packet.
120 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
122 uint16_t udelta= seq - s->max_seq;
123 const int MAX_DROPOUT= 3000;
124 const int MAX_MISORDER = 100;
125 const int MIN_SEQUENTIAL = 2;
127 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
130 if(seq==s->max_seq + 1) {
133 if(s->probation==0) {
134 rtp_init_sequence(s, seq);
139 s->probation= MIN_SEQUENTIAL - 1;
142 } else if (udelta < MAX_DROPOUT) {
143 // in order, with permissible gap
144 if(seq < s->max_seq) {
145 //sequence number wrapped; count antother 64k cycles
146 s->cycles += RTP_SEQ_MOD;
149 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
150 // sequence made a large jump...
151 if(seq==s->bad_seq) {
152 // two sequential packets-- assume that the other side restarted without telling us; just resync.
153 rtp_init_sequence(s, seq);
155 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
159 // duplicate or reordered packet...
167 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
168 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
169 * never change. I left this in in case someone else can see a way. (rdm)
171 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
173 uint32_t transit= arrival_timestamp - sent_timestamp;
176 d= FFABS(transit - s->transit);
177 s->jitter += d - ((s->jitter + 8)>>4);
181 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
187 RTPStatistics *stats= &s->statistics;
189 uint32_t extended_max;
190 uint32_t expected_interval;
191 uint32_t received_interval;
192 uint32_t lost_interval;
195 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
197 if (!s->rtp_ctx || (count < 1))
200 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
201 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
202 s->octet_count += count;
203 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
205 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
208 s->last_octet_count = s->octet_count;
210 if (url_open_dyn_buf(&pb) < 0)
214 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
216 put_be16(pb, 7); /* length in words - 1 */
217 put_be32(pb, s->ssrc); // our own SSRC
218 put_be32(pb, s->ssrc); // XXX: should be the server's here!
219 // some placeholders we should really fill...
221 extended_max= stats->cycles + stats->max_seq;
222 expected= extended_max - stats->base_seq + 1;
223 lost= expected - stats->received;
224 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
225 expected_interval= expected - stats->expected_prior;
226 stats->expected_prior= expected;
227 received_interval= stats->received - stats->received_prior;
228 stats->received_prior= stats->received;
229 lost_interval= expected_interval - received_interval;
230 if (expected_interval==0 || lost_interval<=0) fraction= 0;
231 else fraction = (lost_interval<<8)/expected_interval;
233 fraction= (fraction<<24) | lost;
235 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
236 put_be32(pb, extended_max); /* max sequence received */
237 put_be32(pb, stats->jitter>>4); /* jitter */
239 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
241 put_be32(pb, 0); /* last SR timestamp */
242 put_be32(pb, 0); /* delay since last SR */
244 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
245 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
247 put_be32(pb, middle_32_bits); /* last SR timestamp */
248 put_be32(pb, delay_since_last); /* delay since last SR */
252 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
254 len = strlen(s->hostname);
255 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
256 put_be32(pb, s->ssrc);
259 put_buffer(pb, s->hostname, len);
261 for (len = (6 + len) % 4; len % 4; len++) {
265 put_flush_packet(pb);
266 len = url_close_dyn_buf(pb, &buf);
267 if ((len > 0) && buf) {
269 dprintf(s->ic, "sending %d bytes of RR\n", len);
270 result= url_write(s->rtp_ctx, buf, len);
271 dprintf(s->ic, "result from url_write: %d\n", result);
277 void rtp_send_punch_packets(URLContext* rtp_handle)
283 /* Send a small RTP packet */
284 if (url_open_dyn_buf(&pb) < 0)
287 put_byte(pb, (RTP_VERSION << 6));
288 put_byte(pb, 0); /* Payload type */
289 put_be16(pb, 0); /* Seq */
290 put_be32(pb, 0); /* Timestamp */
291 put_be32(pb, 0); /* SSRC */
293 put_flush_packet(pb);
294 len = url_close_dyn_buf(pb, &buf);
295 if ((len > 0) && buf)
296 url_write(rtp_handle, buf, len);
299 /* Send a minimal RTCP RR */
300 if (url_open_dyn_buf(&pb) < 0)
303 put_byte(pb, (RTP_VERSION << 6));
304 put_byte(pb, 201); /* receiver report */
305 put_be16(pb, 1); /* length in words - 1 */
306 put_be32(pb, 0); /* our own SSRC */
308 put_flush_packet(pb);
309 len = url_close_dyn_buf(pb, &buf);
310 if ((len > 0) && buf)
311 url_write(rtp_handle, buf, len);
317 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
318 * MPEG2TS streams to indicate that they should be demuxed inside the
319 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
320 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
322 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data)
326 s = av_mallocz(sizeof(RTPDemuxContext));
329 s->payload_type = payload_type;
330 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
331 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
334 s->rtp_payload_data = rtp_payload_data;
335 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
336 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
337 s->ts = ff_mpegts_parse_open(s->ic);
343 av_set_pts_info(st, 32, 1, 90000);
344 switch(st->codec->codec_id) {
345 case CODEC_ID_MPEG1VIDEO:
346 case CODEC_ID_MPEG2VIDEO:
352 st->need_parsing = AVSTREAM_PARSE_FULL;
355 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
356 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
361 // needed to send back RTCP RR in RTSP sessions
363 gethostname(s->hostname, sizeof(s->hostname));
368 rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
369 RTPDynamicProtocolHandler *handler)
371 s->dynamic_protocol_context = ctx;
372 s->parse_packet = handler->parse_packet;
375 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
377 int au_headers_length, au_header_size, i;
378 GetBitContext getbitcontext;
379 RTPPayloadData *infos;
381 infos = s->rtp_payload_data;
386 /* decode the first 2 bytes where the AUHeader sections are stored
388 au_headers_length = AV_RB16(buf);
390 if (au_headers_length > RTP_MAX_PACKET_LENGTH)
393 infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
395 /* skip AU headers length section (2 bytes) */
398 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
400 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
401 au_header_size = infos->sizelength + infos->indexlength;
402 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
405 infos->nb_au_headers = au_headers_length / au_header_size;
406 if (!infos->au_headers || infos->au_headers_allocated < infos->nb_au_headers) {
407 av_free(infos->au_headers);
408 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
409 infos->au_headers_allocated = infos->nb_au_headers;
412 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
413 In my test, the FAAD decoder does not behave correctly when sending each AU one by one
414 but does when sending the whole as one big packet... */
415 infos->au_headers[0].size = 0;
416 infos->au_headers[0].index = 0;
417 for (i = 0; i < infos->nb_au_headers; ++i) {
418 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
419 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
422 infos->nb_au_headers = 1;
428 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
430 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
432 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
436 /* compute pts from timestamp with received ntp_time */
437 delta_timestamp = timestamp - s->last_rtcp_timestamp;
438 /* convert to the PTS timebase */
439 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);
440 pkt->pts = s->range_start_offset + addend + delta_timestamp;
445 * Parse an RTP or RTCP packet directly sent as a buffer.
446 * @param s RTP parse context.
447 * @param pkt returned packet
448 * @param buf input buffer or NULL to read the next packets
449 * @param len buffer len
450 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
451 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
453 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
454 const uint8_t *buf, int len)
456 unsigned int ssrc, h;
457 int payload_type, seq, ret, flags = 0;
463 /* return the next packets, if any */
464 if(s->st && s->parse_packet) {
465 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
466 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
467 s->st, pkt, ×tamp, NULL, 0, flags);
468 finalize_packet(s, pkt, timestamp);
471 // TODO: Move to a dynamic packet handler (like above)
472 if (s->read_buf_index >= s->read_buf_size)
474 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
475 s->read_buf_size - s->read_buf_index);
478 s->read_buf_index += ret;
479 if (s->read_buf_index < s->read_buf_size)
489 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
491 if (buf[1] >= 200 && buf[1] <= 204) {
492 rtcp_parse_packet(s, buf, len);
495 payload_type = buf[1] & 0x7f;
497 flags |= RTP_FLAG_MARKER;
498 seq = AV_RB16(buf + 2);
499 timestamp = AV_RB32(buf + 4);
500 ssrc = AV_RB32(buf + 8);
501 /* store the ssrc in the RTPDemuxContext */
504 /* NOTE: we can handle only one payload type */
505 if (s->payload_type != payload_type)
509 // only do something with this if all the rtp checks pass...
510 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
512 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
513 payload_type, seq, ((s->seq + 1) & 0xffff));
522 /* specific MPEG2TS demux support */
523 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
527 s->read_buf_size = len - ret;
528 memcpy(s->buf, buf + ret, s->read_buf_size);
529 s->read_buf_index = 0;
533 } else if (s->parse_packet) {
534 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
535 s->st, pkt, ×tamp, buf, len, flags);
537 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
538 switch(st->codec->codec_id) {
541 /* better than nothing: skip mpeg audio RTP header */
547 av_new_packet(pkt, len);
548 memcpy(pkt->data, buf, len);
550 case CODEC_ID_MPEG1VIDEO:
551 case CODEC_ID_MPEG2VIDEO:
552 /* better than nothing: skip mpeg video RTP header */
565 av_new_packet(pkt, len);
566 memcpy(pkt->data, buf, len);
568 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles
570 // TODO: Put this into a dynamic packet handler...
572 if (rtp_parse_mp4_au(s, buf))
575 RTPPayloadData *infos = s->rtp_payload_data;
578 buf += infos->au_headers_length_bytes + 2;
579 len -= infos->au_headers_length_bytes + 2;
581 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
583 av_new_packet(pkt, infos->au_headers[0].size);
584 memcpy(pkt->data, buf, infos->au_headers[0].size);
585 buf += infos->au_headers[0].size;
586 len -= infos->au_headers[0].size;
588 s->read_buf_size = len;
592 av_new_packet(pkt, len);
593 memcpy(pkt->data, buf, len);
597 pkt->stream_index = st->index;
600 // now perform timestamp things....
601 finalize_packet(s, pkt, timestamp);
606 void rtp_parse_close(RTPDemuxContext *s)
608 // TODO: fold this into the protocol specific data fields.
609 av_free(s->rtp_payload_data->mode);
610 av_free(s->rtp_payload_data->au_headers);
611 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
612 ff_mpegts_parse_close(s->ts);