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_formats.h"
37 /* TODO: - add RTCP statistics reporting (should be optional).
39 - add support for h263/mpeg4 packetized output : IDEA: send a
40 buffer to 'rtp_write_packet' contains all the packets for ONE
41 frame. Each packet should have a four byte header containing
42 the length in big endian format (same trick as
43 'url_open_dyn_packet_buf')
46 /* statistics functions */
47 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
49 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
51 handler->next= RTPFirstDynamicPayloadHandler;
52 RTPFirstDynamicPayloadHandler= handler;
55 void av_register_rtp_dynamic_payload_handlers(void)
57 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
58 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
59 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
60 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
61 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
62 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
63 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
64 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
65 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
66 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
67 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
69 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
70 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
73 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
77 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
78 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
79 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
80 s->last_rtcp_timestamp = AV_RB32(buf + 16);
84 #define RTP_SEQ_MOD (1<<16)
87 * called on parse open packet
89 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
91 memset(s, 0, sizeof(RTPStatistics));
92 s->max_seq= base_sequence;
97 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
99 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
104 s->bad_seq= RTP_SEQ_MOD + 1;
106 s->expected_prior= 0;
107 s->received_prior= 0;
113 * returns 1 if we should handle this packet.
115 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
117 uint16_t udelta= seq - s->max_seq;
118 const int MAX_DROPOUT= 3000;
119 const int MAX_MISORDER = 100;
120 const int MIN_SEQUENTIAL = 2;
122 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
125 if(seq==s->max_seq + 1) {
128 if(s->probation==0) {
129 rtp_init_sequence(s, seq);
134 s->probation= MIN_SEQUENTIAL - 1;
137 } else if (udelta < MAX_DROPOUT) {
138 // in order, with permissible gap
139 if(seq < s->max_seq) {
140 //sequence number wrapped; count antother 64k cycles
141 s->cycles += RTP_SEQ_MOD;
144 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
145 // sequence made a large jump...
146 if(seq==s->bad_seq) {
147 // two sequential packets-- assume that the other side restarted without telling us; just resync.
148 rtp_init_sequence(s, seq);
150 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
154 // duplicate or reordered packet...
162 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
163 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
164 * never change. I left this in in case someone else can see a way. (rdm)
166 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
168 uint32_t transit= arrival_timestamp - sent_timestamp;
171 d= FFABS(transit - s->transit);
172 s->jitter += d - ((s->jitter + 8)>>4);
176 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
182 RTPStatistics *stats= &s->statistics;
184 uint32_t extended_max;
185 uint32_t expected_interval;
186 uint32_t received_interval;
187 uint32_t lost_interval;
190 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
192 if (!s->rtp_ctx || (count < 1))
195 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
196 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
197 s->octet_count += count;
198 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
200 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
203 s->last_octet_count = s->octet_count;
205 if (url_open_dyn_buf(&pb) < 0)
209 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
211 put_be16(pb, 7); /* length in words - 1 */
212 put_be32(pb, s->ssrc); // our own SSRC
213 put_be32(pb, s->ssrc); // XXX: should be the server's here!
214 // some placeholders we should really fill...
216 extended_max= stats->cycles + stats->max_seq;
217 expected= extended_max - stats->base_seq + 1;
218 lost= expected - stats->received;
219 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
220 expected_interval= expected - stats->expected_prior;
221 stats->expected_prior= expected;
222 received_interval= stats->received - stats->received_prior;
223 stats->received_prior= stats->received;
224 lost_interval= expected_interval - received_interval;
225 if (expected_interval==0 || lost_interval<=0) fraction= 0;
226 else fraction = (lost_interval<<8)/expected_interval;
228 fraction= (fraction<<24) | lost;
230 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
231 put_be32(pb, extended_max); /* max sequence received */
232 put_be32(pb, stats->jitter>>4); /* jitter */
234 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
236 put_be32(pb, 0); /* last SR timestamp */
237 put_be32(pb, 0); /* delay since last SR */
239 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
240 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
242 put_be32(pb, middle_32_bits); /* last SR timestamp */
243 put_be32(pb, delay_since_last); /* delay since last SR */
247 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
249 len = strlen(s->hostname);
250 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
251 put_be32(pb, s->ssrc);
254 put_buffer(pb, s->hostname, len);
256 for (len = (6 + len) % 4; len % 4; len++) {
260 put_flush_packet(pb);
261 len = url_close_dyn_buf(pb, &buf);
262 if ((len > 0) && buf) {
264 dprintf(s->ic, "sending %d bytes of RR\n", len);
265 result= url_write(s->rtp_ctx, buf, len);
266 dprintf(s->ic, "result from url_write: %d\n", result);
272 void rtp_send_punch_packets(URLContext* rtp_handle)
278 /* Send a small RTP packet */
279 if (url_open_dyn_buf(&pb) < 0)
282 put_byte(pb, (RTP_VERSION << 6));
283 put_byte(pb, 0); /* Payload type */
284 put_be16(pb, 0); /* Seq */
285 put_be32(pb, 0); /* Timestamp */
286 put_be32(pb, 0); /* SSRC */
288 put_flush_packet(pb);
289 len = url_close_dyn_buf(pb, &buf);
290 if ((len > 0) && buf)
291 url_write(rtp_handle, buf, len);
294 /* Send a minimal RTCP RR */
295 if (url_open_dyn_buf(&pb) < 0)
298 put_byte(pb, (RTP_VERSION << 6));
299 put_byte(pb, 201); /* receiver report */
300 put_be16(pb, 1); /* length in words - 1 */
301 put_be32(pb, 0); /* our own SSRC */
303 put_flush_packet(pb);
304 len = url_close_dyn_buf(pb, &buf);
305 if ((len > 0) && buf)
306 url_write(rtp_handle, buf, len);
312 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
313 * MPEG2TS streams to indicate that they should be demuxed inside the
314 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
316 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type)
320 s = av_mallocz(sizeof(RTPDemuxContext));
323 s->payload_type = payload_type;
324 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
325 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
328 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
329 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
330 s->ts = ff_mpegts_parse_open(s->ic);
336 av_set_pts_info(st, 32, 1, 90000);
337 switch(st->codec->codec_id) {
338 case CODEC_ID_MPEG1VIDEO:
339 case CODEC_ID_MPEG2VIDEO:
345 st->need_parsing = AVSTREAM_PARSE_FULL;
348 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
349 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
354 // needed to send back RTCP RR in RTSP sessions
356 gethostname(s->hostname, sizeof(s->hostname));
361 rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
362 RTPDynamicProtocolHandler *handler)
364 s->dynamic_protocol_context = ctx;
365 s->parse_packet = handler->parse_packet;
369 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
371 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
373 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
377 /* compute pts from timestamp with received ntp_time */
378 delta_timestamp = timestamp - s->last_rtcp_timestamp;
379 /* convert to the PTS timebase */
380 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);
381 pkt->pts = s->range_start_offset + addend + delta_timestamp;
386 * Parse an RTP or RTCP packet directly sent as a buffer.
387 * @param s RTP parse context.
388 * @param pkt returned packet
389 * @param buf input buffer or NULL to read the next packets
390 * @param len buffer len
391 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
392 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
394 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
395 const uint8_t *buf, int len)
397 unsigned int ssrc, h;
398 int payload_type, seq, ret, flags = 0;
404 /* return the next packets, if any */
405 if(s->st && s->parse_packet) {
406 /* timestamp should be overwritten by parse_packet, if not,
407 * the packet is left with pts == AV_NOPTS_VALUE */
408 timestamp = RTP_NOTS_VALUE;
409 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
410 s->st, pkt, ×tamp, NULL, 0, flags);
411 finalize_packet(s, pkt, timestamp);
414 // TODO: Move to a dynamic packet handler (like above)
415 if (s->read_buf_index >= s->read_buf_size)
417 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
418 s->read_buf_size - s->read_buf_index);
421 s->read_buf_index += ret;
422 if (s->read_buf_index < s->read_buf_size)
432 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
434 if (buf[1] >= 200 && buf[1] <= 204) {
435 rtcp_parse_packet(s, buf, len);
438 payload_type = buf[1] & 0x7f;
440 flags |= RTP_FLAG_MARKER;
441 seq = AV_RB16(buf + 2);
442 timestamp = AV_RB32(buf + 4);
443 ssrc = AV_RB32(buf + 8);
444 /* store the ssrc in the RTPDemuxContext */
447 /* NOTE: we can handle only one payload type */
448 if (s->payload_type != payload_type)
452 // only do something with this if all the rtp checks pass...
453 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
455 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
456 payload_type, seq, ((s->seq + 1) & 0xffff));
465 /* specific MPEG2TS demux support */
466 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
470 s->read_buf_size = len - ret;
471 memcpy(s->buf, buf + ret, s->read_buf_size);
472 s->read_buf_index = 0;
476 } else if (s->parse_packet) {
477 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
478 s->st, pkt, ×tamp, buf, len, flags);
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) {
484 /* better than nothing: skip mpeg audio RTP header */
490 av_new_packet(pkt, len);
491 memcpy(pkt->data, buf, len);
493 case CODEC_ID_MPEG1VIDEO:
494 case CODEC_ID_MPEG2VIDEO:
495 /* better than nothing: skip mpeg video RTP header */
508 av_new_packet(pkt, len);
509 memcpy(pkt->data, buf, len);
512 av_new_packet(pkt, len);
513 memcpy(pkt->data, buf, len);
517 pkt->stream_index = st->index;
520 // now perform timestamp things....
521 finalize_packet(s, pkt, timestamp);
526 void rtp_parse_close(RTPDemuxContext *s)
528 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
529 ff_mpegts_parse_close(s->ts);
534 int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
535 int (*parse_fmtp)(AVStream *stream,
536 PayloadContext *data,
537 char *attr, char *value))
542 int value_size = strlen(p) + 1;
544 if (!(value = av_malloc(value_size))) {
545 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
546 return AVERROR(ENOMEM);
549 // remove protocol identifier
550 while (*p && *p == ' ') p++; // strip spaces
551 while (*p && *p != ' ') p++; // eat protocol identifier
552 while (*p && *p == ' ') p++; // strip trailing spaces
554 while (ff_rtsp_next_attr_and_value(&p,
556 value, value_size)) {
558 res = parse_fmtp(stream, data, attr, value);
559 if (res < 0 && res != AVERROR_PATCHWELCOME) {