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
23 #include "bitstream.h"
28 #include "rtp_internal.h"
33 /* TODO: - add RTCP statistics reporting (should be optional).
35 - add support for h263/mpeg4 packetized output : IDEA: send a
36 buffer to 'rtp_write_packet' contains all the packets for ONE
37 frame. Each packet should have a four byte header containing
38 the length in big endian format (same trick as
39 'url_open_dyn_packet_buf')
42 /* statistics functions */
43 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
45 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
46 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
48 static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
50 handler->next= RTPFirstDynamicPayloadHandler;
51 RTPFirstDynamicPayloadHandler= handler;
54 void av_register_rtp_dynamic_payload_handlers(void)
56 register_dynamic_payload_handler(&mp4v_es_handler);
57 register_dynamic_payload_handler(&mpeg4_generic_handler);
58 register_dynamic_payload_handler(&ff_h264_dynamic_handler);
61 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
65 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
66 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
67 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
68 s->last_rtcp_timestamp = AV_RB32(buf + 16);
72 #define RTP_SEQ_MOD (1<<16)
75 * called on parse open packet
77 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
79 memset(s, 0, sizeof(RTPStatistics));
80 s->max_seq= base_sequence;
85 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
87 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
92 s->bad_seq= RTP_SEQ_MOD + 1;
101 * returns 1 if we should handle this packet.
103 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
105 uint16_t udelta= seq - s->max_seq;
106 const int MAX_DROPOUT= 3000;
107 const int MAX_MISORDER = 100;
108 const int MIN_SEQUENTIAL = 2;
110 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
113 if(seq==s->max_seq + 1) {
116 if(s->probation==0) {
117 rtp_init_sequence(s, seq);
122 s->probation= MIN_SEQUENTIAL - 1;
125 } else if (udelta < MAX_DROPOUT) {
126 // in order, with permissible gap
127 if(seq < s->max_seq) {
128 //sequence number wrapped; count antother 64k cycles
129 s->cycles += RTP_SEQ_MOD;
132 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
133 // sequence made a large jump...
134 if(seq==s->bad_seq) {
135 // two sequential packets-- assume that the other side restarted without telling us; just resync.
136 rtp_init_sequence(s, seq);
138 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
142 // duplicate or reordered packet...
150 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
151 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
152 * never change. I left this in in case someone else can see a way. (rdm)
154 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
156 uint32_t transit= arrival_timestamp - sent_timestamp;
159 d= FFABS(transit - s->transit);
160 s->jitter += d - ((s->jitter + 8)>>4);
164 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
170 RTPStatistics *stats= &s->statistics;
172 uint32_t extended_max;
173 uint32_t expected_interval;
174 uint32_t received_interval;
175 uint32_t lost_interval;
178 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
180 if (!s->rtp_ctx || (count < 1))
183 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
184 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
185 s->octet_count += count;
186 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
188 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
191 s->last_octet_count = s->octet_count;
193 if (url_open_dyn_buf(&pb) < 0)
197 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
199 put_be16(pb, 7); /* length in words - 1 */
200 put_be32(pb, s->ssrc); // our own SSRC
201 put_be32(pb, s->ssrc); // XXX: should be the server's here!
202 // some placeholders we should really fill...
204 extended_max= stats->cycles + stats->max_seq;
205 expected= extended_max - stats->base_seq + 1;
206 lost= expected - stats->received;
207 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
208 expected_interval= expected - stats->expected_prior;
209 stats->expected_prior= expected;
210 received_interval= stats->received - stats->received_prior;
211 stats->received_prior= stats->received;
212 lost_interval= expected_interval - received_interval;
213 if (expected_interval==0 || lost_interval<=0) fraction= 0;
214 else fraction = (lost_interval<<8)/expected_interval;
216 fraction= (fraction<<24) | lost;
218 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
219 put_be32(pb, extended_max); /* max sequence received */
220 put_be32(pb, stats->jitter>>4); /* jitter */
222 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
224 put_be32(pb, 0); /* last SR timestamp */
225 put_be32(pb, 0); /* delay since last SR */
227 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
228 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
230 put_be32(pb, middle_32_bits); /* last SR timestamp */
231 put_be32(pb, delay_since_last); /* delay since last SR */
235 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
237 len = strlen(s->hostname);
238 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
239 put_be32(pb, s->ssrc);
242 put_buffer(pb, s->hostname, len);
244 for (len = (6 + len) % 4; len % 4; len++) {
248 put_flush_packet(pb);
249 len = url_close_dyn_buf(pb, &buf);
250 if ((len > 0) && buf) {
253 printf("sending %d bytes of RR\n", len);
255 result= url_write(s->rtp_ctx, buf, len);
257 printf("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 switch(st->codec->codec_id) {
292 case CODEC_ID_MPEG1VIDEO:
293 case CODEC_ID_MPEG2VIDEO:
298 st->need_parsing = AVSTREAM_PARSE_FULL;
304 // needed to send back RTCP RR in RTSP sessions
306 gethostname(s->hostname, sizeof(s->hostname));
310 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
312 int au_headers_length, au_header_size, i;
313 GetBitContext getbitcontext;
314 rtp_payload_data_t *infos;
316 infos = s->rtp_payload_data;
321 /* decode the first 2 bytes where are stored the AUHeader sections
323 au_headers_length = AV_RB16(buf);
325 if (au_headers_length > RTP_MAX_PACKET_LENGTH)
328 infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
330 /* skip AU headers length section (2 bytes) */
333 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
335 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
336 au_header_size = infos->sizelength + infos->indexlength;
337 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
340 infos->nb_au_headers = au_headers_length / au_header_size;
341 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
343 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
344 In my test, the FAAD decoder does not behave correctly when sending each AU one by one
345 but does when sending the whole as one big packet... */
346 infos->au_headers[0].size = 0;
347 infos->au_headers[0].index = 0;
348 for (i = 0; i < infos->nb_au_headers; ++i) {
349 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
350 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
353 infos->nb_au_headers = 1;
359 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
361 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
363 switch(s->st->codec->codec_id) {
365 case CODEC_ID_MPEG1VIDEO:
366 case CODEC_ID_MPEG2VIDEO:
367 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
371 /* XXX: is it really necessary to unify the timestamp base ? */
372 /* compute pts from timestamp with received ntp_time */
373 delta_timestamp = timestamp - s->last_rtcp_timestamp;
374 /* convert to 90 kHz without overflow */
375 addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
376 addend = (addend * 5625) >> 14;
377 pkt->pts = addend + delta_timestamp;
383 pkt->pts = timestamp;
386 /* no timestamp info yet */
389 pkt->stream_index = s->st->index;
393 * Parse an RTP or RTCP packet directly sent as a buffer.
394 * @param s RTP parse context.
395 * @param pkt returned packet
396 * @param buf input buffer or NULL to read the next packets
397 * @param len buffer len
398 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
399 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
401 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
402 const uint8_t *buf, int len)
404 unsigned int ssrc, h;
405 int payload_type, seq, ret;
411 /* return the next packets, if any */
412 if(s->st && s->parse_packet) {
413 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
414 rv= s->parse_packet(s, pkt, ×tamp, NULL, 0);
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;
443 seq = AV_RB16(buf + 2);
444 timestamp = AV_RB32(buf + 4);
445 ssrc = AV_RB32(buf + 8);
446 /* store the ssrc in the RTPDemuxContext */
449 /* NOTE: we can handle only one payload type */
450 if (s->payload_type != payload_type)
454 // only do something with this if all the rtp checks pass...
455 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
457 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
458 payload_type, seq, ((s->seq + 1) & 0xffff));
467 /* specific MPEG2TS demux support */
468 ret = mpegts_parse_packet(s->ts, pkt, buf, len);
472 s->read_buf_size = len - ret;
473 memcpy(s->buf, buf + ret, s->read_buf_size);
474 s->read_buf_index = 0;
477 } else if (s->parse_packet) {
478 rv = s->parse_packet(s, pkt, ×tamp, buf, len);
480 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
481 switch(st->codec->codec_id) {
483 /* better than nothing: skip mpeg audio RTP header */
489 av_new_packet(pkt, len);
490 memcpy(pkt->data, buf, len);
492 case CODEC_ID_MPEG1VIDEO:
493 case CODEC_ID_MPEG2VIDEO:
494 /* better than nothing: skip mpeg video RTP header */
507 av_new_packet(pkt, len);
508 memcpy(pkt->data, buf, len);
510 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles
512 // TODO: Put this into a dynamic packet handler...
514 if (rtp_parse_mp4_au(s, buf))
517 rtp_payload_data_t *infos = s->rtp_payload_data;
520 buf += infos->au_headers_length_bytes + 2;
521 len -= infos->au_headers_length_bytes + 2;
523 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
525 av_new_packet(pkt, infos->au_headers[0].size);
526 memcpy(pkt->data, buf, infos->au_headers[0].size);
527 buf += infos->au_headers[0].size;
528 len -= infos->au_headers[0].size;
530 s->read_buf_size = len;
534 av_new_packet(pkt, len);
535 memcpy(pkt->data, buf, len);
539 // now perform timestamp things....
540 finalize_packet(s, pkt, timestamp);
545 void rtp_parse_close(RTPDemuxContext *s)
547 // TODO: fold this into the protocol specific data fields.
548 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
549 mpegts_parse_close(s->ts);