3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of Libav.
7 * Libav 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 * Libav 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 Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/mathematics.h"
23 #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 'ffio_open_dyn_packet_buf')
46 static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
47 .enc_name = "X-MP3-draft-00",
48 .codec_type = AVMEDIA_TYPE_AUDIO,
49 .codec_id = CODEC_ID_MP3ADU,
52 /* statistics functions */
53 static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
55 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
57 handler->next= RTPFirstDynamicPayloadHandler;
58 RTPFirstDynamicPayloadHandler= handler;
61 void av_register_rtp_dynamic_payload_handlers(void)
63 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
64 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
65 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
66 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
67 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
68 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
69 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
70 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
71 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
72 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
73 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
74 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
75 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
76 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
77 ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);
79 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
80 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
82 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
83 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
84 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
85 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
88 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
89 enum AVMediaType codec_type)
91 RTPDynamicProtocolHandler *handler;
92 for (handler = RTPFirstDynamicPayloadHandler;
93 handler; handler = handler->next)
94 if (!strcasecmp(name, handler->enc_name) &&
95 codec_type == handler->codec_type)
100 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
101 enum AVMediaType codec_type)
103 RTPDynamicProtocolHandler *handler;
104 for (handler = RTPFirstDynamicPayloadHandler;
105 handler; handler = handler->next)
106 if (handler->static_payload_id && handler->static_payload_id == id &&
107 codec_type == handler->codec_type)
112 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
119 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
120 return AVERROR_INVALIDDATA;
122 payload_len = (AV_RB16(buf + 2) + 1) * 4;
124 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
125 s->last_rtcp_timestamp = AV_RB32(buf + 16);
126 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
127 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
128 if (!s->base_timestamp)
129 s->base_timestamp = s->last_rtcp_timestamp;
130 s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
145 #define RTP_SEQ_MOD (1<<16)
148 * called on parse open packet
150 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
152 memset(s, 0, sizeof(RTPStatistics));
153 s->max_seq= base_sequence;
158 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
160 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
165 s->bad_seq= RTP_SEQ_MOD + 1;
167 s->expected_prior= 0;
168 s->received_prior= 0;
174 * returns 1 if we should handle this packet.
176 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
178 uint16_t udelta= seq - s->max_seq;
179 const int MAX_DROPOUT= 3000;
180 const int MAX_MISORDER = 100;
181 const int MIN_SEQUENTIAL = 2;
183 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
186 if(seq==s->max_seq + 1) {
189 if(s->probation==0) {
190 rtp_init_sequence(s, seq);
195 s->probation= MIN_SEQUENTIAL - 1;
198 } else if (udelta < MAX_DROPOUT) {
199 // in order, with permissible gap
200 if(seq < s->max_seq) {
201 //sequence number wrapped; count antother 64k cycles
202 s->cycles += RTP_SEQ_MOD;
205 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
206 // sequence made a large jump...
207 if(seq==s->bad_seq) {
208 // two sequential packets-- assume that the other side restarted without telling us; just resync.
209 rtp_init_sequence(s, seq);
211 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
215 // duplicate or reordered packet...
223 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
224 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
225 * never change. I left this in in case someone else can see a way. (rdm)
227 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
229 uint32_t transit= arrival_timestamp - sent_timestamp;
232 d= FFABS(transit - s->transit);
233 s->jitter += d - ((s->jitter + 8)>>4);
237 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
243 RTPStatistics *stats= &s->statistics;
245 uint32_t extended_max;
246 uint32_t expected_interval;
247 uint32_t received_interval;
248 uint32_t lost_interval;
251 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
253 if (!s->rtp_ctx || (count < 1))
256 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
257 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
258 s->octet_count += count;
259 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
261 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
264 s->last_octet_count = s->octet_count;
266 if (avio_open_dyn_buf(&pb) < 0)
270 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
271 avio_w8(pb, RTCP_RR);
272 avio_wb16(pb, 7); /* length in words - 1 */
273 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
274 avio_wb32(pb, s->ssrc + 1);
275 avio_wb32(pb, s->ssrc); // server SSRC
276 // some placeholders we should really fill...
278 extended_max= stats->cycles + stats->max_seq;
279 expected= extended_max - stats->base_seq + 1;
280 lost= expected - stats->received;
281 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
282 expected_interval= expected - stats->expected_prior;
283 stats->expected_prior= expected;
284 received_interval= stats->received - stats->received_prior;
285 stats->received_prior= stats->received;
286 lost_interval= expected_interval - received_interval;
287 if (expected_interval==0 || lost_interval<=0) fraction= 0;
288 else fraction = (lost_interval<<8)/expected_interval;
290 fraction= (fraction<<24) | lost;
292 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
293 avio_wb32(pb, extended_max); /* max sequence received */
294 avio_wb32(pb, stats->jitter>>4); /* jitter */
296 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
298 avio_wb32(pb, 0); /* last SR timestamp */
299 avio_wb32(pb, 0); /* delay since last SR */
301 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
302 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
304 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
305 avio_wb32(pb, delay_since_last); /* delay since last SR */
309 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
310 avio_w8(pb, RTCP_SDES);
311 len = strlen(s->hostname);
312 avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
313 avio_wb32(pb, s->ssrc);
316 avio_write(pb, s->hostname, len);
318 for (len = (6 + len) % 4; len % 4; len++) {
323 len = avio_close_dyn_buf(pb, &buf);
324 if ((len > 0) && buf) {
325 int av_unused result;
326 av_dlog(s->ic, "sending %d bytes of RR\n", len);
327 result= ffurl_write(s->rtp_ctx, buf, len);
328 av_dlog(s->ic, "result from ffurl_write: %d\n", result);
334 void rtp_send_punch_packets(URLContext* rtp_handle)
340 /* Send a small RTP packet */
341 if (avio_open_dyn_buf(&pb) < 0)
344 avio_w8(pb, (RTP_VERSION << 6));
345 avio_w8(pb, 0); /* Payload type */
346 avio_wb16(pb, 0); /* Seq */
347 avio_wb32(pb, 0); /* Timestamp */
348 avio_wb32(pb, 0); /* SSRC */
351 len = avio_close_dyn_buf(pb, &buf);
352 if ((len > 0) && buf)
353 ffurl_write(rtp_handle, buf, len);
356 /* Send a minimal RTCP RR */
357 if (avio_open_dyn_buf(&pb) < 0)
360 avio_w8(pb, (RTP_VERSION << 6));
361 avio_w8(pb, RTCP_RR); /* receiver report */
362 avio_wb16(pb, 1); /* length in words - 1 */
363 avio_wb32(pb, 0); /* our own SSRC */
366 len = avio_close_dyn_buf(pb, &buf);
367 if ((len > 0) && buf)
368 ffurl_write(rtp_handle, buf, len);
374 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
375 * MPEG2TS streams to indicate that they should be demuxed inside the
376 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
378 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
382 s = av_mallocz(sizeof(RTPDemuxContext));
385 s->payload_type = payload_type;
386 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
387 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
390 s->queue_size = queue_size;
391 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
392 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
393 s->ts = ff_mpegts_parse_open(s->ic);
399 switch(st->codec->codec_id) {
400 case CODEC_ID_MPEG1VIDEO:
401 case CODEC_ID_MPEG2VIDEO:
407 st->need_parsing = AVSTREAM_PARSE_FULL;
409 case CODEC_ID_ADPCM_G722:
410 /* According to RFC 3551, the stream clock rate is 8000
411 * even if the sample rate is 16000. */
412 if (st->codec->sample_rate == 8000)
413 st->codec->sample_rate = 16000;
419 // needed to send back RTCP RR in RTSP sessions
421 gethostname(s->hostname, sizeof(s->hostname));
426 rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
427 RTPDynamicProtocolHandler *handler)
429 s->dynamic_protocol_context = ctx;
430 s->parse_packet = handler->parse_packet;
434 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
436 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
438 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
439 return; /* Timestamp already set by depacketizer */
440 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
444 /* compute pts from timestamp with received ntp_time */
445 delta_timestamp = timestamp - s->last_rtcp_timestamp;
446 /* convert to the PTS timebase */
447 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);
448 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
452 if (timestamp == RTP_NOTS_VALUE)
454 if (!s->base_timestamp)
455 s->base_timestamp = timestamp;
456 pkt->pts = s->range_start_offset + timestamp - s->base_timestamp;
459 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
460 const uint8_t *buf, int len)
462 unsigned int ssrc, h;
463 int payload_type, seq, ret, flags = 0;
470 payload_type = buf[1] & 0x7f;
472 flags |= RTP_FLAG_MARKER;
473 seq = AV_RB16(buf + 2);
474 timestamp = AV_RB32(buf + 4);
475 ssrc = AV_RB32(buf + 8);
476 /* store the ssrc in the RTPDemuxContext */
479 /* NOTE: we can handle only one payload type */
480 if (s->payload_type != payload_type)
484 // only do something with this if all the rtp checks pass...
485 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
487 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
488 payload_type, seq, ((s->seq + 1) & 0xffff));
493 int padding = buf[len - 1];
494 if (len >= 12 + padding)
502 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
506 /* calculate the header extension length (stored as number
507 * of 32-bit words) */
508 ext = (AV_RB16(buf + 2) + 1) << 2;
512 // skip past RTP header extension
518 /* specific MPEG2TS demux support */
519 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
520 /* The only error that can be returned from ff_mpegts_parse_packet
521 * is "no more data to return from the provided buffer", so return
522 * AVERROR(EAGAIN) for all errors */
524 return AVERROR(EAGAIN);
526 s->read_buf_size = len - ret;
527 memcpy(s->buf, buf + ret, s->read_buf_size);
528 s->read_buf_index = 0;
532 } else if (s->parse_packet) {
533 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
534 s->st, pkt, ×tamp, buf, len, flags);
536 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
537 switch(st->codec->codec_id) {
540 /* better than nothing: skip mpeg audio RTP header */
546 av_new_packet(pkt, len);
547 memcpy(pkt->data, buf, len);
549 case CODEC_ID_MPEG1VIDEO:
550 case CODEC_ID_MPEG2VIDEO:
551 /* better than nothing: skip mpeg video RTP header */
564 av_new_packet(pkt, len);
565 memcpy(pkt->data, buf, len);
568 av_new_packet(pkt, len);
569 memcpy(pkt->data, buf, len);
573 pkt->stream_index = st->index;
576 // now perform timestamp things....
577 finalize_packet(s, pkt, timestamp);
582 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
585 RTPPacket *next = s->queue->next;
586 av_free(s->queue->buf);
595 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
597 uint16_t seq = AV_RB16(buf + 2);
598 RTPPacket *cur = s->queue, *prev = NULL, *packet;
600 /* Find the correct place in the queue to insert the packet */
602 int16_t diff = seq - cur->seq;
609 packet = av_mallocz(sizeof(*packet));
612 packet->recvtime = av_gettime();
624 static int has_next_packet(RTPDemuxContext *s)
626 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
629 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
631 return s->queue ? s->queue->recvtime : 0;
634 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
639 if (s->queue_len <= 0)
642 if (!has_next_packet(s))
643 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
644 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
646 /* Parse the first packet in the queue, and dequeue it */
647 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
648 next = s->queue->next;
649 av_free(s->queue->buf);
656 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
657 uint8_t **bufptr, int len)
659 uint8_t* buf = bufptr ? *bufptr : NULL;
665 /* If parsing of the previous packet actually returned 0 or an error,
666 * there's nothing more to be parsed from that packet, but we may have
667 * indicated that we can return the next enqueued packet. */
668 if (s->prev_ret <= 0)
669 return rtp_parse_queued_packet(s, pkt);
670 /* return the next packets, if any */
671 if(s->st && s->parse_packet) {
672 /* timestamp should be overwritten by parse_packet, if not,
673 * the packet is left with pts == AV_NOPTS_VALUE */
674 timestamp = RTP_NOTS_VALUE;
675 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
676 s->st, pkt, ×tamp, NULL, 0, flags);
677 finalize_packet(s, pkt, timestamp);
680 // TODO: Move to a dynamic packet handler (like above)
681 if (s->read_buf_index >= s->read_buf_size)
682 return AVERROR(EAGAIN);
683 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
684 s->read_buf_size - s->read_buf_index);
686 return AVERROR(EAGAIN);
687 s->read_buf_index += ret;
688 if (s->read_buf_index < s->read_buf_size)
698 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
700 if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
701 return rtcp_parse_packet(s, buf, len);
704 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
705 /* First packet, or no reordering */
706 return rtp_parse_packet_internal(s, pkt, buf, len);
708 uint16_t seq = AV_RB16(buf + 2);
709 int16_t diff = seq - s->seq;
711 /* Packet older than the previously emitted one, drop */
712 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
713 "RTP: dropping old packet received too late\n");
715 } else if (diff <= 1) {
717 rv = rtp_parse_packet_internal(s, pkt, buf, len);
720 /* Still missing some packet, enqueue this one. */
721 enqueue_packet(s, buf, len);
723 /* Return the first enqueued packet if the queue is full,
724 * even if we're missing something */
725 if (s->queue_len >= s->queue_size)
726 return rtp_parse_queued_packet(s, pkt);
733 * Parse an RTP or RTCP packet directly sent as a buffer.
734 * @param s RTP parse context.
735 * @param pkt returned packet
736 * @param bufptr pointer to the input buffer or NULL to read the next packets
737 * @param len buffer len
738 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
739 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
741 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
742 uint8_t **bufptr, int len)
744 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
746 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
747 rv = rtp_parse_queued_packet(s, pkt);
748 return rv ? rv : has_next_packet(s);
751 void rtp_parse_close(RTPDemuxContext *s)
753 ff_rtp_reset_packet_queue(s);
754 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
755 ff_mpegts_parse_close(s->ts);
760 int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
761 int (*parse_fmtp)(AVStream *stream,
762 PayloadContext *data,
763 char *attr, char *value))
768 int value_size = strlen(p) + 1;
770 if (!(value = av_malloc(value_size))) {
771 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
772 return AVERROR(ENOMEM);
775 // remove protocol identifier
776 while (*p && *p == ' ') p++; // strip spaces
777 while (*p && *p != ' ') p++; // eat protocol identifier
778 while (*p && *p == ' ') p++; // strip trailing spaces
780 while (ff_rtsp_next_attr_and_value(&p,
782 value, value_size)) {
784 res = parse_fmtp(stream, data, attr, value);
785 if (res < 0 && res != AVERROR_PATCHWELCOME) {