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 /* needed for gethostname() */
23 #define _XOPEN_SOURCE 600
25 #include "libavcodec/get_bits.h"
35 #include "rtpdec_formats.h"
39 /* TODO: - add RTCP statistics reporting (should be optional).
41 - add support for h263/mpeg4 packetized output : IDEA: send a
42 buffer to 'rtp_write_packet' contains all the packets for ONE
43 frame. Each packet should have a four byte header containing
44 the length in big endian format (same trick as
45 'ffio_open_dyn_packet_buf')
48 static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
49 .enc_name = "X-MP3-draft-00",
50 .codec_type = AVMEDIA_TYPE_AUDIO,
51 .codec_id = CODEC_ID_MP3ADU,
54 /* statistics functions */
55 static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
57 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
59 handler->next= RTPFirstDynamicPayloadHandler;
60 RTPFirstDynamicPayloadHandler= handler;
63 void av_register_rtp_dynamic_payload_handlers(void)
65 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
66 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
67 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
68 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
69 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
70 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
71 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
72 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
73 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
74 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
75 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
76 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
77 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
78 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
79 ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);
81 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
82 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
84 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
85 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
86 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
87 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
90 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
91 enum AVMediaType codec_type)
93 RTPDynamicProtocolHandler *handler;
94 for (handler = RTPFirstDynamicPayloadHandler;
95 handler; handler = handler->next)
96 if (!strcasecmp(name, handler->enc_name) &&
97 codec_type == handler->codec_type)
102 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
103 enum AVMediaType codec_type)
105 RTPDynamicProtocolHandler *handler;
106 for (handler = RTPFirstDynamicPayloadHandler;
107 handler; handler = handler->next)
108 if (handler->static_payload_id && handler->static_payload_id == id &&
109 codec_type == handler->codec_type)
114 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
121 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
122 return AVERROR_INVALIDDATA;
124 payload_len = (AV_RB16(buf + 2) + 1) * 4;
126 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
127 s->last_rtcp_timestamp = AV_RB32(buf + 16);
128 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
129 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
130 if (!s->base_timestamp)
131 s->base_timestamp = s->last_rtcp_timestamp;
132 s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
147 #define RTP_SEQ_MOD (1<<16)
150 * called on parse open packet
152 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
154 memset(s, 0, sizeof(RTPStatistics));
155 s->max_seq= base_sequence;
160 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
162 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
167 s->bad_seq= RTP_SEQ_MOD + 1;
169 s->expected_prior= 0;
170 s->received_prior= 0;
176 * returns 1 if we should handle this packet.
178 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
180 uint16_t udelta= seq - s->max_seq;
181 const int MAX_DROPOUT= 3000;
182 const int MAX_MISORDER = 100;
183 const int MIN_SEQUENTIAL = 2;
185 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
188 if(seq==s->max_seq + 1) {
191 if(s->probation==0) {
192 rtp_init_sequence(s, seq);
197 s->probation= MIN_SEQUENTIAL - 1;
200 } else if (udelta < MAX_DROPOUT) {
201 // in order, with permissible gap
202 if(seq < s->max_seq) {
203 //sequence number wrapped; count antother 64k cycles
204 s->cycles += RTP_SEQ_MOD;
207 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
208 // sequence made a large jump...
209 if(seq==s->bad_seq) {
210 // two sequential packets-- assume that the other side restarted without telling us; just resync.
211 rtp_init_sequence(s, seq);
213 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
217 // duplicate or reordered packet...
225 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
226 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
227 * never change. I left this in in case someone else can see a way. (rdm)
229 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
231 uint32_t transit= arrival_timestamp - sent_timestamp;
234 d= FFABS(transit - s->transit);
235 s->jitter += d - ((s->jitter + 8)>>4);
239 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
245 RTPStatistics *stats= &s->statistics;
247 uint32_t extended_max;
248 uint32_t expected_interval;
249 uint32_t received_interval;
250 uint32_t lost_interval;
253 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
255 if (!s->rtp_ctx || (count < 1))
258 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
259 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
260 s->octet_count += count;
261 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
263 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
266 s->last_octet_count = s->octet_count;
268 if (avio_open_dyn_buf(&pb) < 0)
272 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
273 avio_w8(pb, RTCP_RR);
274 avio_wb16(pb, 7); /* length in words - 1 */
275 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
276 avio_wb32(pb, s->ssrc + 1);
277 avio_wb32(pb, s->ssrc); // server SSRC
278 // some placeholders we should really fill...
280 extended_max= stats->cycles + stats->max_seq;
281 expected= extended_max - stats->base_seq + 1;
282 lost= expected - stats->received;
283 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
284 expected_interval= expected - stats->expected_prior;
285 stats->expected_prior= expected;
286 received_interval= stats->received - stats->received_prior;
287 stats->received_prior= stats->received;
288 lost_interval= expected_interval - received_interval;
289 if (expected_interval==0 || lost_interval<=0) fraction= 0;
290 else fraction = (lost_interval<<8)/expected_interval;
292 fraction= (fraction<<24) | lost;
294 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
295 avio_wb32(pb, extended_max); /* max sequence received */
296 avio_wb32(pb, stats->jitter>>4); /* jitter */
298 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
300 avio_wb32(pb, 0); /* last SR timestamp */
301 avio_wb32(pb, 0); /* delay since last SR */
303 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
304 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
306 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
307 avio_wb32(pb, delay_since_last); /* delay since last SR */
311 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
312 avio_w8(pb, RTCP_SDES);
313 len = strlen(s->hostname);
314 avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
315 avio_wb32(pb, s->ssrc);
318 avio_write(pb, s->hostname, len);
320 for (len = (6 + len) % 4; len % 4; len++) {
325 len = avio_close_dyn_buf(pb, &buf);
326 if ((len > 0) && buf) {
328 av_dlog(s->ic, "sending %d bytes of RR\n", len);
329 result= ffurl_write(s->rtp_ctx, buf, len);
330 av_dlog(s->ic, "result from ffurl_write: %d\n", result);
336 void rtp_send_punch_packets(URLContext* rtp_handle)
342 /* Send a small RTP packet */
343 if (avio_open_dyn_buf(&pb) < 0)
346 avio_w8(pb, (RTP_VERSION << 6));
347 avio_w8(pb, 0); /* Payload type */
348 avio_wb16(pb, 0); /* Seq */
349 avio_wb32(pb, 0); /* Timestamp */
350 avio_wb32(pb, 0); /* SSRC */
353 len = avio_close_dyn_buf(pb, &buf);
354 if ((len > 0) && buf)
355 ffurl_write(rtp_handle, buf, len);
358 /* Send a minimal RTCP RR */
359 if (avio_open_dyn_buf(&pb) < 0)
362 avio_w8(pb, (RTP_VERSION << 6));
363 avio_w8(pb, RTCP_RR); /* receiver report */
364 avio_wb16(pb, 1); /* length in words - 1 */
365 avio_wb32(pb, 0); /* our own SSRC */
368 len = avio_close_dyn_buf(pb, &buf);
369 if ((len > 0) && buf)
370 ffurl_write(rtp_handle, buf, len);
376 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
377 * MPEG2TS streams to indicate that they should be demuxed inside the
378 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
380 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
384 s = av_mallocz(sizeof(RTPDemuxContext));
387 s->payload_type = payload_type;
388 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
389 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
392 s->queue_size = queue_size;
393 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
394 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
395 s->ts = ff_mpegts_parse_open(s->ic);
401 switch(st->codec->codec_id) {
402 case CODEC_ID_MPEG1VIDEO:
403 case CODEC_ID_MPEG2VIDEO:
409 st->need_parsing = AVSTREAM_PARSE_FULL;
411 case CODEC_ID_ADPCM_G722:
412 /* According to RFC 3551, the stream clock rate is 8000
413 * even if the sample rate is 16000. */
414 if (st->codec->sample_rate == 8000)
415 st->codec->sample_rate = 16000;
421 // needed to send back RTCP RR in RTSP sessions
423 gethostname(s->hostname, sizeof(s->hostname));
428 rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
429 RTPDynamicProtocolHandler *handler)
431 s->dynamic_protocol_context = ctx;
432 s->parse_packet = handler->parse_packet;
436 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
438 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
440 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
441 return; /* Timestamp already set by depacketizer */
442 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
446 /* compute pts from timestamp with received ntp_time */
447 delta_timestamp = timestamp - s->last_rtcp_timestamp;
448 /* convert to the PTS timebase */
449 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);
450 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
454 if (timestamp == RTP_NOTS_VALUE)
456 if (!s->base_timestamp)
457 s->base_timestamp = timestamp;
458 pkt->pts = s->range_start_offset + timestamp - s->base_timestamp;
461 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
462 const uint8_t *buf, int len)
464 unsigned int ssrc, h;
465 int payload_type, seq, ret, flags = 0;
472 payload_type = buf[1] & 0x7f;
474 flags |= RTP_FLAG_MARKER;
475 seq = AV_RB16(buf + 2);
476 timestamp = AV_RB32(buf + 4);
477 ssrc = AV_RB32(buf + 8);
478 /* store the ssrc in the RTPDemuxContext */
481 /* NOTE: we can handle only one payload type */
482 if (s->payload_type != payload_type)
486 // only do something with this if all the rtp checks pass...
487 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
489 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
490 payload_type, seq, ((s->seq + 1) & 0xffff));
495 int padding = buf[len - 1];
496 if (len >= 12 + padding)
504 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
508 /* calculate the header extension length (stored as number
509 * of 32-bit words) */
510 ext = (AV_RB16(buf + 2) + 1) << 2;
514 // skip past RTP header extension
520 /* specific MPEG2TS demux support */
521 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
522 /* The only error that can be returned from ff_mpegts_parse_packet
523 * is "no more data to return from the provided buffer", so return
524 * AVERROR(EAGAIN) for all errors */
526 return AVERROR(EAGAIN);
528 s->read_buf_size = len - ret;
529 memcpy(s->buf, buf + ret, s->read_buf_size);
530 s->read_buf_index = 0;
534 } else if (s->parse_packet) {
535 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
536 s->st, pkt, ×tamp, buf, len, flags);
538 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
539 switch(st->codec->codec_id) {
542 /* better than nothing: skip mpeg audio RTP header */
548 av_new_packet(pkt, len);
549 memcpy(pkt->data, buf, len);
551 case CODEC_ID_MPEG1VIDEO:
552 case CODEC_ID_MPEG2VIDEO:
553 /* better than nothing: skip mpeg video RTP header */
566 av_new_packet(pkt, len);
567 memcpy(pkt->data, buf, len);
570 av_new_packet(pkt, len);
571 memcpy(pkt->data, buf, len);
575 pkt->stream_index = st->index;
578 // now perform timestamp things....
579 finalize_packet(s, pkt, timestamp);
584 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
587 RTPPacket *next = s->queue->next;
588 av_free(s->queue->buf);
597 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
599 uint16_t seq = AV_RB16(buf + 2);
600 RTPPacket *cur = s->queue, *prev = NULL, *packet;
602 /* Find the correct place in the queue to insert the packet */
604 int16_t diff = seq - cur->seq;
611 packet = av_mallocz(sizeof(*packet));
614 packet->recvtime = av_gettime();
626 static int has_next_packet(RTPDemuxContext *s)
628 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
631 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
633 return s->queue ? s->queue->recvtime : 0;
636 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
641 if (s->queue_len <= 0)
644 if (!has_next_packet(s))
645 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
646 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
648 /* Parse the first packet in the queue, and dequeue it */
649 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
650 next = s->queue->next;
651 av_free(s->queue->buf);
658 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
659 uint8_t **bufptr, int len)
661 uint8_t* buf = bufptr ? *bufptr : NULL;
667 /* If parsing of the previous packet actually returned 0 or an error,
668 * there's nothing more to be parsed from that packet, but we may have
669 * indicated that we can return the next enqueued packet. */
670 if (s->prev_ret <= 0)
671 return rtp_parse_queued_packet(s, pkt);
672 /* return the next packets, if any */
673 if(s->st && s->parse_packet) {
674 /* timestamp should be overwritten by parse_packet, if not,
675 * the packet is left with pts == AV_NOPTS_VALUE */
676 timestamp = RTP_NOTS_VALUE;
677 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
678 s->st, pkt, ×tamp, NULL, 0, flags);
679 finalize_packet(s, pkt, timestamp);
682 // TODO: Move to a dynamic packet handler (like above)
683 if (s->read_buf_index >= s->read_buf_size)
684 return AVERROR(EAGAIN);
685 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
686 s->read_buf_size - s->read_buf_index);
688 return AVERROR(EAGAIN);
689 s->read_buf_index += ret;
690 if (s->read_buf_index < s->read_buf_size)
700 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
702 if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
703 return rtcp_parse_packet(s, buf, len);
706 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
707 /* First packet, or no reordering */
708 return rtp_parse_packet_internal(s, pkt, buf, len);
710 uint16_t seq = AV_RB16(buf + 2);
711 int16_t diff = seq - s->seq;
713 /* Packet older than the previously emitted one, drop */
714 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
715 "RTP: dropping old packet received too late\n");
717 } else if (diff <= 1) {
719 rv = rtp_parse_packet_internal(s, pkt, buf, len);
722 /* Still missing some packet, enqueue this one. */
723 enqueue_packet(s, buf, len);
725 /* Return the first enqueued packet if the queue is full,
726 * even if we're missing something */
727 if (s->queue_len >= s->queue_size)
728 return rtp_parse_queued_packet(s, pkt);
735 * Parse an RTP or RTCP packet directly sent as a buffer.
736 * @param s RTP parse context.
737 * @param pkt returned packet
738 * @param bufptr pointer to the input buffer or NULL to read the next packets
739 * @param len buffer len
740 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
741 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
743 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
744 uint8_t **bufptr, int len)
746 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
748 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
749 rv = rtp_parse_queued_packet(s, pkt);
750 return rv ? rv : has_next_packet(s);
753 void rtp_parse_close(RTPDemuxContext *s)
755 ff_rtp_reset_packet_queue(s);
756 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
757 ff_mpegts_parse_close(s->ts);
762 int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
763 int (*parse_fmtp)(AVStream *stream,
764 PayloadContext *data,
765 char *attr, char *value))
770 int value_size = strlen(p) + 1;
772 if (!(value = av_malloc(value_size))) {
773 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
774 return AVERROR(ENOMEM);
777 // remove protocol identifier
778 while (*p && *p == ' ') p++; // strip spaces
779 while (*p && *p != ' ') p++; // eat protocol identifier
780 while (*p && *p == ' ') p++; // strip trailing spaces
782 while (ff_rtsp_next_attr_and_value(&p,
784 value, value_size)) {
786 res = parse_fmtp(stream, data, attr, value);
787 if (res < 0 && res != AVERROR_PATCHWELCOME) {