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"
34 #include "rtpdec_formats.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 static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
48 .enc_name = "X-MP3-draft-00",
49 .codec_type = AVMEDIA_TYPE_AUDIO,
50 .codec_id = CODEC_ID_MP3ADU,
53 /* statistics functions */
54 static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
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(&ff_mp4v_es_dynamic_handler);
65 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_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);
73 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
74 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
75 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
76 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
77 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
78 ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);
80 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
81 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
83 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
84 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
85 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
86 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
89 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
90 enum AVMediaType codec_type)
92 RTPDynamicProtocolHandler *handler;
93 for (handler = RTPFirstDynamicPayloadHandler;
94 handler; handler = handler->next)
95 if (!strcasecmp(name, handler->enc_name) &&
96 codec_type == handler->codec_type)
101 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
102 enum AVMediaType codec_type)
104 RTPDynamicProtocolHandler *handler;
105 for (handler = RTPFirstDynamicPayloadHandler;
106 handler; handler = handler->next)
107 if (handler->static_payload_id && handler->static_payload_id == id &&
108 codec_type == handler->codec_type)
113 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
120 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
121 return AVERROR_INVALIDDATA;
123 payload_len = (AV_RB16(buf + 2) + 1) * 4;
125 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
126 s->last_rtcp_timestamp = AV_RB32(buf + 16);
127 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
128 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
129 if (!s->base_timestamp)
130 s->base_timestamp = s->last_rtcp_timestamp;
131 s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
146 #define RTP_SEQ_MOD (1<<16)
149 * called on parse open packet
151 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
153 memset(s, 0, sizeof(RTPStatistics));
154 s->max_seq= base_sequence;
159 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
161 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
166 s->bad_seq= RTP_SEQ_MOD + 1;
168 s->expected_prior= 0;
169 s->received_prior= 0;
175 * returns 1 if we should handle this packet.
177 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
179 uint16_t udelta= seq - s->max_seq;
180 const int MAX_DROPOUT= 3000;
181 const int MAX_MISORDER = 100;
182 const int MIN_SEQUENTIAL = 2;
184 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
187 if(seq==s->max_seq + 1) {
190 if(s->probation==0) {
191 rtp_init_sequence(s, seq);
196 s->probation= MIN_SEQUENTIAL - 1;
199 } else if (udelta < MAX_DROPOUT) {
200 // in order, with permissible gap
201 if(seq < s->max_seq) {
202 //sequence number wrapped; count antother 64k cycles
203 s->cycles += RTP_SEQ_MOD;
206 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
207 // sequence made a large jump...
208 if(seq==s->bad_seq) {
209 // two sequential packets-- assume that the other side restarted without telling us; just resync.
210 rtp_init_sequence(s, seq);
212 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
216 // duplicate or reordered packet...
224 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
225 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
226 * never change. I left this in in case someone else can see a way. (rdm)
228 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
230 uint32_t transit= arrival_timestamp - sent_timestamp;
233 d= FFABS(transit - s->transit);
234 s->jitter += d - ((s->jitter + 8)>>4);
238 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
244 RTPStatistics *stats= &s->statistics;
246 uint32_t extended_max;
247 uint32_t expected_interval;
248 uint32_t received_interval;
249 uint32_t lost_interval;
252 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
254 if (!s->rtp_ctx || (count < 1))
257 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
258 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
259 s->octet_count += count;
260 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
262 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
265 s->last_octet_count = s->octet_count;
267 if (url_open_dyn_buf(&pb) < 0)
271 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
272 put_byte(pb, RTCP_RR);
273 put_be16(pb, 7); /* length in words - 1 */
274 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
275 put_be32(pb, s->ssrc + 1);
276 put_be32(pb, s->ssrc); // server SSRC
277 // some placeholders we should really fill...
279 extended_max= stats->cycles + stats->max_seq;
280 expected= extended_max - stats->base_seq + 1;
281 lost= expected - stats->received;
282 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
283 expected_interval= expected - stats->expected_prior;
284 stats->expected_prior= expected;
285 received_interval= stats->received - stats->received_prior;
286 stats->received_prior= stats->received;
287 lost_interval= expected_interval - received_interval;
288 if (expected_interval==0 || lost_interval<=0) fraction= 0;
289 else fraction = (lost_interval<<8)/expected_interval;
291 fraction= (fraction<<24) | lost;
293 put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
294 put_be32(pb, extended_max); /* max sequence received */
295 put_be32(pb, stats->jitter>>4); /* jitter */
297 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
299 put_be32(pb, 0); /* last SR timestamp */
300 put_be32(pb, 0); /* delay since last SR */
302 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
303 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
305 put_be32(pb, middle_32_bits); /* last SR timestamp */
306 put_be32(pb, delay_since_last); /* delay since last SR */
310 put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
311 put_byte(pb, RTCP_SDES);
312 len = strlen(s->hostname);
313 put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
314 put_be32(pb, s->ssrc);
317 put_buffer(pb, s->hostname, len);
319 for (len = (6 + len) % 4; len % 4; len++) {
323 put_flush_packet(pb);
324 len = url_close_dyn_buf(pb, &buf);
325 if ((len > 0) && buf) {
327 av_dlog(s->ic, "sending %d bytes of RR\n", len);
328 result= url_write(s->rtp_ctx, buf, len);
329 av_dlog(s->ic, "result from url_write: %d\n", result);
335 void rtp_send_punch_packets(URLContext* rtp_handle)
341 /* Send a small RTP packet */
342 if (url_open_dyn_buf(&pb) < 0)
345 put_byte(pb, (RTP_VERSION << 6));
346 put_byte(pb, 0); /* Payload type */
347 put_be16(pb, 0); /* Seq */
348 put_be32(pb, 0); /* Timestamp */
349 put_be32(pb, 0); /* SSRC */
351 put_flush_packet(pb);
352 len = url_close_dyn_buf(pb, &buf);
353 if ((len > 0) && buf)
354 url_write(rtp_handle, buf, len);
357 /* Send a minimal RTCP RR */
358 if (url_open_dyn_buf(&pb) < 0)
361 put_byte(pb, (RTP_VERSION << 6));
362 put_byte(pb, RTCP_RR); /* receiver report */
363 put_be16(pb, 1); /* length in words - 1 */
364 put_be32(pb, 0); /* our own SSRC */
366 put_flush_packet(pb);
367 len = url_close_dyn_buf(pb, &buf);
368 if ((len > 0) && buf)
369 url_write(rtp_handle, buf, len);
375 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
376 * MPEG2TS streams to indicate that they should be demuxed inside the
377 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
379 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
383 s = av_mallocz(sizeof(RTPDemuxContext));
386 s->payload_type = payload_type;
387 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
388 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
391 s->queue_size = queue_size;
392 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
393 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
394 s->ts = ff_mpegts_parse_open(s->ic);
400 switch(st->codec->codec_id) {
401 case CODEC_ID_MPEG1VIDEO:
402 case CODEC_ID_MPEG2VIDEO:
408 st->need_parsing = AVSTREAM_PARSE_FULL;
410 case CODEC_ID_ADPCM_G722:
411 /* According to RFC 3551, the stream clock rate is 8000
412 * even if the sample rate is 16000. */
413 if (st->codec->sample_rate == 8000)
414 st->codec->sample_rate = 16000;
420 // needed to send back RTCP RR in RTSP sessions
422 gethostname(s->hostname, sizeof(s->hostname));
427 rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
428 RTPDynamicProtocolHandler *handler)
430 s->dynamic_protocol_context = ctx;
431 s->parse_packet = handler->parse_packet;
435 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
437 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
439 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
440 return; /* Timestamp already set by depacketizer */
441 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
445 /* compute pts from timestamp with received ntp_time */
446 delta_timestamp = timestamp - s->last_rtcp_timestamp;
447 /* convert to the PTS timebase */
448 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);
449 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
453 if (timestamp == RTP_NOTS_VALUE)
455 if (!s->base_timestamp)
456 s->base_timestamp = timestamp;
457 pkt->pts = s->range_start_offset + timestamp - s->base_timestamp;
460 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
461 const uint8_t *buf, int len)
463 unsigned int ssrc, h;
464 int payload_type, seq, ret, flags = 0;
471 payload_type = buf[1] & 0x7f;
473 flags |= RTP_FLAG_MARKER;
474 seq = AV_RB16(buf + 2);
475 timestamp = AV_RB32(buf + 4);
476 ssrc = AV_RB32(buf + 8);
477 /* store the ssrc in the RTPDemuxContext */
480 /* NOTE: we can handle only one payload type */
481 if (s->payload_type != payload_type)
485 // only do something with this if all the rtp checks pass...
486 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
488 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
489 payload_type, seq, ((s->seq + 1) & 0xffff));
494 int padding = buf[len - 1];
495 if (len >= 12 + padding)
503 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
507 /* calculate the header extension length (stored as number
508 * of 32-bit words) */
509 ext = (AV_RB16(buf + 2) + 1) << 2;
513 // skip past RTP header extension
519 /* specific MPEG2TS demux support */
520 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
521 /* The only error that can be returned from ff_mpegts_parse_packet
522 * is "no more data to return from the provided buffer", so return
523 * AVERROR(EAGAIN) for all errors */
525 return AVERROR(EAGAIN);
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);
569 av_new_packet(pkt, len);
570 memcpy(pkt->data, buf, len);
574 pkt->stream_index = st->index;
577 // now perform timestamp things....
578 finalize_packet(s, pkt, timestamp);
583 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
586 RTPPacket *next = s->queue->next;
587 av_free(s->queue->buf);
596 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
598 uint16_t seq = AV_RB16(buf + 2);
599 RTPPacket *cur = s->queue, *prev = NULL, *packet;
601 /* Find the correct place in the queue to insert the packet */
603 int16_t diff = seq - cur->seq;
610 packet = av_mallocz(sizeof(*packet));
613 packet->recvtime = av_gettime();
625 static int has_next_packet(RTPDemuxContext *s)
627 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
630 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
632 return s->queue ? s->queue->recvtime : 0;
635 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
640 if (s->queue_len <= 0)
643 if (!has_next_packet(s))
644 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
645 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
647 /* Parse the first packet in the queue, and dequeue it */
648 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
649 next = s->queue->next;
650 av_free(s->queue->buf);
657 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
658 uint8_t **bufptr, int len)
660 uint8_t* buf = bufptr ? *bufptr : NULL;
666 /* If parsing of the previous packet actually returned 0 or an error,
667 * there's nothing more to be parsed from that packet, but we may have
668 * indicated that we can return the next enqueued packet. */
669 if (s->prev_ret <= 0)
670 return rtp_parse_queued_packet(s, pkt);
671 /* return the next packets, if any */
672 if(s->st && s->parse_packet) {
673 /* timestamp should be overwritten by parse_packet, if not,
674 * the packet is left with pts == AV_NOPTS_VALUE */
675 timestamp = RTP_NOTS_VALUE;
676 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
677 s->st, pkt, ×tamp, NULL, 0, flags);
678 finalize_packet(s, pkt, timestamp);
681 // TODO: Move to a dynamic packet handler (like above)
682 if (s->read_buf_index >= s->read_buf_size)
683 return AVERROR(EAGAIN);
684 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
685 s->read_buf_size - s->read_buf_index);
687 return AVERROR(EAGAIN);
688 s->read_buf_index += ret;
689 if (s->read_buf_index < s->read_buf_size)
699 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
701 if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
702 return rtcp_parse_packet(s, buf, len);
705 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
706 /* First packet, or no reordering */
707 return rtp_parse_packet_internal(s, pkt, buf, len);
709 uint16_t seq = AV_RB16(buf + 2);
710 int16_t diff = seq - s->seq;
712 /* Packet older than the previously emitted one, drop */
713 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
714 "RTP: dropping old packet received too late\n");
716 } else if (diff <= 1) {
718 rv = rtp_parse_packet_internal(s, pkt, buf, len);
721 /* Still missing some packet, enqueue this one. */
722 enqueue_packet(s, buf, len);
724 /* Return the first enqueued packet if the queue is full,
725 * even if we're missing something */
726 if (s->queue_len >= s->queue_size)
727 return rtp_parse_queued_packet(s, pkt);
734 * Parse an RTP or RTCP packet directly sent as a buffer.
735 * @param s RTP parse context.
736 * @param pkt returned packet
737 * @param bufptr pointer to the input buffer or NULL to read the next packets
738 * @param len buffer len
739 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
740 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
742 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
743 uint8_t **bufptr, int len)
745 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
747 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
748 rv = rtp_parse_queued_packet(s, pkt);
749 return rv ? rv : has_next_packet(s);
752 void rtp_parse_close(RTPDemuxContext *s)
754 ff_rtp_reset_packet_queue(s);
755 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
756 ff_mpegts_parse_close(s->ts);
761 int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
762 int (*parse_fmtp)(AVStream *stream,
763 PayloadContext *data,
764 char *attr, char *value))
769 int value_size = strlen(p) + 1;
771 if (!(value = av_malloc(value_size))) {
772 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
773 return AVERROR(ENOMEM);
776 // remove protocol identifier
777 while (*p && *p == ' ') p++; // strip spaces
778 while (*p && *p != ' ') p++; // eat protocol identifier
779 while (*p && *p == ' ') p++; // strip trailing spaces
781 while (ff_rtsp_next_attr_and_value(&p,
783 value, value_size)) {
785 res = parse_fmtp(stream, data, attr, value);
786 if (res < 0 && res != AVERROR_PATCHWELCOME) {