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_mpeg4.h"
38 #include "rtpdec_xiph.h"
42 /* TODO: - add RTCP statistics reporting (should be optional).
44 - add support for h263/mpeg4 packetized output : IDEA: send a
45 buffer to 'rtp_write_packet' contains all the packets for ONE
46 frame. Each packet should have a four byte header containing
47 the length in big endian format (same trick as
48 'url_open_dyn_packet_buf')
51 /* statistics functions */
52 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
54 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
56 handler->next= RTPFirstDynamicPayloadHandler;
57 RTPFirstDynamicPayloadHandler= handler;
60 void av_register_rtp_dynamic_payload_handlers(void)
62 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
63 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
64 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
65 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
66 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
67 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
68 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
69 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
70 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
72 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
73 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
76 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
80 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
81 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
82 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
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)
319 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type)
323 s = av_mallocz(sizeof(RTPDemuxContext));
326 s->payload_type = payload_type;
327 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
328 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
331 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
332 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
333 s->ts = ff_mpegts_parse_open(s->ic);
339 av_set_pts_info(st, 32, 1, 90000);
340 switch(st->codec->codec_id) {
341 case CODEC_ID_MPEG1VIDEO:
342 case CODEC_ID_MPEG2VIDEO:
348 st->need_parsing = AVSTREAM_PARSE_FULL;
351 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
352 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
357 // needed to send back RTCP RR in RTSP sessions
359 gethostname(s->hostname, sizeof(s->hostname));
364 rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
365 RTPDynamicProtocolHandler *handler)
367 s->dynamic_protocol_context = ctx;
368 s->parse_packet = handler->parse_packet;
372 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
374 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
376 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
380 /* compute pts from timestamp with received ntp_time */
381 delta_timestamp = timestamp - s->last_rtcp_timestamp;
382 /* convert to the PTS timebase */
383 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);
384 pkt->pts = s->range_start_offset + addend + delta_timestamp;
389 * Parse an RTP or RTCP packet directly sent as a buffer.
390 * @param s RTP parse context.
391 * @param pkt returned packet
392 * @param buf input buffer or NULL to read the next packets
393 * @param len buffer len
394 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
395 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
397 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
398 const uint8_t *buf, int len)
400 unsigned int ssrc, h;
401 int payload_type, seq, ret, flags = 0;
407 /* return the next packets, if any */
408 if(s->st && s->parse_packet) {
409 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
410 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
411 s->st, pkt, ×tamp, NULL, 0, flags);
412 finalize_packet(s, pkt, timestamp);
415 // TODO: Move to a dynamic packet handler (like above)
416 if (s->read_buf_index >= s->read_buf_size)
418 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
419 s->read_buf_size - s->read_buf_index);
422 s->read_buf_index += ret;
423 if (s->read_buf_index < s->read_buf_size)
433 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
435 if (buf[1] >= 200 && buf[1] <= 204) {
436 rtcp_parse_packet(s, buf, len);
439 payload_type = buf[1] & 0x7f;
441 flags |= RTP_FLAG_MARKER;
442 seq = AV_RB16(buf + 2);
443 timestamp = AV_RB32(buf + 4);
444 ssrc = AV_RB32(buf + 8);
445 /* store the ssrc in the RTPDemuxContext */
448 /* NOTE: we can handle only one payload type */
449 if (s->payload_type != payload_type)
453 // only do something with this if all the rtp checks pass...
454 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
456 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
457 payload_type, seq, ((s->seq + 1) & 0xffff));
466 /* specific MPEG2TS demux support */
467 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
471 s->read_buf_size = len - ret;
472 memcpy(s->buf, buf + ret, s->read_buf_size);
473 s->read_buf_index = 0;
477 } else if (s->parse_packet) {
478 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
479 s->st, pkt, ×tamp, buf, len, flags);
481 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
482 switch(st->codec->codec_id) {
485 /* better than nothing: skip mpeg audio RTP header */
491 av_new_packet(pkt, len);
492 memcpy(pkt->data, buf, len);
494 case CODEC_ID_MPEG1VIDEO:
495 case CODEC_ID_MPEG2VIDEO:
496 /* better than nothing: skip mpeg video RTP header */
509 av_new_packet(pkt, len);
510 memcpy(pkt->data, buf, len);
513 av_new_packet(pkt, len);
514 memcpy(pkt->data, buf, len);
518 pkt->stream_index = st->index;
521 // now perform timestamp things....
522 finalize_packet(s, pkt, timestamp);
527 void rtp_parse_close(RTPDemuxContext *s)
529 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
530 ff_mpegts_parse_close(s->ts);
535 int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
536 int (*parse_fmtp)(AVStream *stream,
537 PayloadContext *data,
538 char *attr, char *value))
544 // remove protocol identifier
545 while (*p && *p == ' ') p++; // strip spaces
546 while (*p && *p != ' ') p++; // eat protocol identifier
547 while (*p && *p == ' ') p++; // strip trailing spaces
549 while (ff_rtsp_next_attr_and_value(&p,
551 value, sizeof(value))) {
553 res = parse_fmtp(stream, data, attr, value);