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"
38 /* TODO: - add RTCP statistics reporting (should be optional).
40 - add support for h263/mpeg4 packetized output : IDEA: send a
41 buffer to 'rtp_write_packet' contains all the packets for ONE
42 frame. Each packet should have a four byte header containing
43 the length in big endian format (same trick as
44 'url_open_dyn_packet_buf')
47 /* statistics functions */
48 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
50 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
51 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
53 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
55 handler->next= RTPFirstDynamicPayloadHandler;
56 RTPFirstDynamicPayloadHandler= handler;
59 void av_register_rtp_dynamic_payload_handlers(void)
61 ff_register_dynamic_payload_handler(&mp4v_es_handler);
62 ff_register_dynamic_payload_handler(&mpeg4_generic_handler);
63 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
65 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
66 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
69 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
73 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
74 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
75 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
76 s->last_rtcp_timestamp = AV_RB32(buf + 16);
80 #define RTP_SEQ_MOD (1<<16)
83 * called on parse open packet
85 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
87 memset(s, 0, sizeof(RTPStatistics));
88 s->max_seq= base_sequence;
93 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
95 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
100 s->bad_seq= RTP_SEQ_MOD + 1;
102 s->expected_prior= 0;
103 s->received_prior= 0;
109 * returns 1 if we should handle this packet.
111 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
113 uint16_t udelta= seq - s->max_seq;
114 const int MAX_DROPOUT= 3000;
115 const int MAX_MISORDER = 100;
116 const int MIN_SEQUENTIAL = 2;
118 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
121 if(seq==s->max_seq + 1) {
124 if(s->probation==0) {
125 rtp_init_sequence(s, seq);
130 s->probation= MIN_SEQUENTIAL - 1;
133 } else if (udelta < MAX_DROPOUT) {
134 // in order, with permissible gap
135 if(seq < s->max_seq) {
136 //sequence number wrapped; count antother 64k cycles
137 s->cycles += RTP_SEQ_MOD;
140 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
141 // sequence made a large jump...
142 if(seq==s->bad_seq) {
143 // two sequential packets-- assume that the other side restarted without telling us; just resync.
144 rtp_init_sequence(s, seq);
146 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
150 // duplicate or reordered packet...
158 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
159 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
160 * never change. I left this in in case someone else can see a way. (rdm)
162 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
164 uint32_t transit= arrival_timestamp - sent_timestamp;
167 d= FFABS(transit - s->transit);
168 s->jitter += d - ((s->jitter + 8)>>4);
172 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
178 RTPStatistics *stats= &s->statistics;
180 uint32_t extended_max;
181 uint32_t expected_interval;
182 uint32_t received_interval;
183 uint32_t lost_interval;
186 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
188 if (!s->rtp_ctx || (count < 1))
191 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
192 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
193 s->octet_count += count;
194 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
196 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
199 s->last_octet_count = s->octet_count;
201 if (url_open_dyn_buf(&pb) < 0)
205 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
207 put_be16(pb, 7); /* length in words - 1 */
208 put_be32(pb, s->ssrc); // our own SSRC
209 put_be32(pb, s->ssrc); // XXX: should be the server's here!
210 // some placeholders we should really fill...
212 extended_max= stats->cycles + stats->max_seq;
213 expected= extended_max - stats->base_seq + 1;
214 lost= expected - stats->received;
215 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
216 expected_interval= expected - stats->expected_prior;
217 stats->expected_prior= expected;
218 received_interval= stats->received - stats->received_prior;
219 stats->received_prior= stats->received;
220 lost_interval= expected_interval - received_interval;
221 if (expected_interval==0 || lost_interval<=0) fraction= 0;
222 else fraction = (lost_interval<<8)/expected_interval;
224 fraction= (fraction<<24) | lost;
226 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
227 put_be32(pb, extended_max); /* max sequence received */
228 put_be32(pb, stats->jitter>>4); /* jitter */
230 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
232 put_be32(pb, 0); /* last SR timestamp */
233 put_be32(pb, 0); /* delay since last SR */
235 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
236 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
238 put_be32(pb, middle_32_bits); /* last SR timestamp */
239 put_be32(pb, delay_since_last); /* delay since last SR */
243 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
245 len = strlen(s->hostname);
246 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
247 put_be32(pb, s->ssrc);
250 put_buffer(pb, s->hostname, len);
252 for (len = (6 + len) % 4; len % 4; len++) {
256 put_flush_packet(pb);
257 len = url_close_dyn_buf(pb, &buf);
258 if ((len > 0) && buf) {
260 dprintf(s->ic, "sending %d bytes of RR\n", len);
261 result= url_write(s->rtp_ctx, buf, len);
262 dprintf(s->ic, "result from url_write: %d\n", result);
269 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
270 * MPEG2TS streams to indicate that they should be demuxed inside the
271 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
272 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
274 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data)
278 s = av_mallocz(sizeof(RTPDemuxContext));
281 s->payload_type = payload_type;
282 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
283 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
286 s->rtp_payload_data = rtp_payload_data;
287 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
288 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
289 s->ts = mpegts_parse_open(s->ic);
295 av_set_pts_info(st, 32, 1, 90000);
296 switch(st->codec->codec_id) {
297 case CODEC_ID_MPEG1VIDEO:
298 case CODEC_ID_MPEG2VIDEO:
303 st->need_parsing = AVSTREAM_PARSE_FULL;
306 if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
307 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
312 // needed to send back RTCP RR in RTSP sessions
314 gethostname(s->hostname, sizeof(s->hostname));
319 rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
320 RTPDynamicProtocolHandler *handler)
322 s->dynamic_protocol_context = ctx;
323 s->parse_packet = handler->parse_packet;
326 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
328 int au_headers_length, au_header_size, i;
329 GetBitContext getbitcontext;
330 RTPPayloadData *infos;
332 infos = s->rtp_payload_data;
337 /* decode the first 2 bytes where the AUHeader sections are stored
339 au_headers_length = AV_RB16(buf);
341 if (au_headers_length > RTP_MAX_PACKET_LENGTH)
344 infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
346 /* skip AU headers length section (2 bytes) */
349 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
351 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
352 au_header_size = infos->sizelength + infos->indexlength;
353 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
356 infos->nb_au_headers = au_headers_length / au_header_size;
357 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
359 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
360 In my test, the FAAD decoder does not behave correctly when sending each AU one by one
361 but does when sending the whole as one big packet... */
362 infos->au_headers[0].size = 0;
363 infos->au_headers[0].index = 0;
364 for (i = 0; i < infos->nb_au_headers; ++i) {
365 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
366 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
369 infos->nb_au_headers = 1;
375 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
377 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
379 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
383 /* compute pts from timestamp with received ntp_time */
384 delta_timestamp = timestamp - s->last_rtcp_timestamp;
385 /* convert to the PTS timebase */
386 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);
387 pkt->pts = addend + delta_timestamp;
392 * Parse an RTP or RTCP packet directly sent as a buffer.
393 * @param s RTP parse context.
394 * @param pkt returned packet
395 * @param buf input buffer or NULL to read the next packets
396 * @param len buffer len
397 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
398 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
400 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
401 const uint8_t *buf, int len)
403 unsigned int ssrc, h;
404 int payload_type, seq, ret, flags = 0;
410 /* return the next packets, if any */
411 if(s->st && s->parse_packet) {
412 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
413 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
414 s->st, pkt, ×tamp, NULL, 0, flags);
415 finalize_packet(s, pkt, timestamp);
418 // TODO: Move to a dynamic packet handler (like above)
419 if (s->read_buf_index >= s->read_buf_size)
421 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
422 s->read_buf_size - s->read_buf_index);
425 s->read_buf_index += ret;
426 if (s->read_buf_index < s->read_buf_size)
436 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
438 if (buf[1] >= 200 && buf[1] <= 204) {
439 rtcp_parse_packet(s, buf, len);
442 payload_type = buf[1] & 0x7f;
444 flags |= RTP_FLAG_MARKER;
445 seq = AV_RB16(buf + 2);
446 timestamp = AV_RB32(buf + 4);
447 ssrc = AV_RB32(buf + 8);
448 /* store the ssrc in the RTPDemuxContext */
451 /* NOTE: we can handle only one payload type */
452 if (s->payload_type != payload_type)
456 // only do something with this if all the rtp checks pass...
457 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
459 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
460 payload_type, seq, ((s->seq + 1) & 0xffff));
469 /* specific MPEG2TS demux support */
470 ret = mpegts_parse_packet(s->ts, pkt, buf, len);
474 s->read_buf_size = len - ret;
475 memcpy(s->buf, buf + ret, s->read_buf_size);
476 s->read_buf_index = 0;
480 } else if (s->parse_packet) {
481 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
482 s->st, pkt, ×tamp, buf, len, flags);
484 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
485 switch(st->codec->codec_id) {
487 /* better than nothing: skip mpeg audio RTP header */
493 av_new_packet(pkt, len);
494 memcpy(pkt->data, buf, len);
496 case CODEC_ID_MPEG1VIDEO:
497 case CODEC_ID_MPEG2VIDEO:
498 /* better than nothing: skip mpeg video RTP header */
511 av_new_packet(pkt, len);
512 memcpy(pkt->data, buf, len);
514 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles
516 // TODO: Put this into a dynamic packet handler...
518 if (rtp_parse_mp4_au(s, buf))
521 RTPPayloadData *infos = s->rtp_payload_data;
524 buf += infos->au_headers_length_bytes + 2;
525 len -= infos->au_headers_length_bytes + 2;
527 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
529 av_new_packet(pkt, infos->au_headers[0].size);
530 memcpy(pkt->data, buf, infos->au_headers[0].size);
531 buf += infos->au_headers[0].size;
532 len -= infos->au_headers[0].size;
534 s->read_buf_size = len;
538 av_new_packet(pkt, len);
539 memcpy(pkt->data, buf, len);
543 pkt->stream_index = st->index;
546 // now perform timestamp things....
547 finalize_packet(s, pkt, timestamp);
552 void rtp_parse_close(RTPDemuxContext *s)
554 // TODO: fold this into the protocol specific data fields.
555 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
556 mpegts_parse_close(s->ts);