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 "libavutil/avstring.h"
24 #include "libavutil/time.h"
25 #include "libavcodec/get_bits.h"
31 #include "rtpdec_formats.h"
33 #define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */
35 static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
36 .enc_name = "X-MP3-draft-00",
37 .codec_type = AVMEDIA_TYPE_AUDIO,
38 .codec_id = AV_CODEC_ID_MP3ADU,
41 static RTPDynamicProtocolHandler speex_dynamic_handler = {
43 .codec_type = AVMEDIA_TYPE_AUDIO,
44 .codec_id = AV_CODEC_ID_SPEEX,
47 static RTPDynamicProtocolHandler opus_dynamic_handler = {
49 .codec_type = AVMEDIA_TYPE_AUDIO,
50 .codec_id = AV_CODEC_ID_OPUS,
53 /* statistics functions */
54 static RTPDynamicProtocolHandler *rtp_first_dynamic_payload_handler = NULL;
56 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
58 handler->next = rtp_first_dynamic_payload_handler;
59 rtp_first_dynamic_payload_handler = 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_h263_rfc2190_dynamic_handler);
71 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
72 ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
73 ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler);
74 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
75 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
76 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
77 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
78 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
79 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
80 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
81 ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
82 ff_register_dynamic_payload_handler(&speex_dynamic_handler);
83 ff_register_dynamic_payload_handler(&opus_dynamic_handler);
85 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
86 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
88 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
89 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
90 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
91 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
93 ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
94 ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
95 ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
96 ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
99 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
100 enum AVMediaType codec_type)
102 RTPDynamicProtocolHandler *handler;
103 for (handler = rtp_first_dynamic_payload_handler;
104 handler; handler = handler->next)
105 if (!av_strcasecmp(name, handler->enc_name) &&
106 codec_type == handler->codec_type)
111 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
112 enum AVMediaType codec_type)
114 RTPDynamicProtocolHandler *handler;
115 for (handler = rtp_first_dynamic_payload_handler;
116 handler; handler = handler->next)
117 if (handler->static_payload_id && handler->static_payload_id == id &&
118 codec_type == handler->codec_type)
123 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
128 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
132 if (payload_len < 20) {
133 av_log(NULL, AV_LOG_ERROR,
134 "Invalid length for RTCP SR packet\n");
135 return AVERROR_INVALIDDATA;
138 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
139 s->last_rtcp_timestamp = AV_RB32(buf + 16);
140 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
141 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
142 if (!s->base_timestamp)
143 s->base_timestamp = s->last_rtcp_timestamp;
144 s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
158 #define RTP_SEQ_MOD (1 << 16)
160 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
162 memset(s, 0, sizeof(RTPStatistics));
163 s->max_seq = base_sequence;
168 * Called whenever there is a large jump in sequence numbers,
169 * or when they get out of probation...
171 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
175 s->base_seq = seq - 1;
176 s->bad_seq = RTP_SEQ_MOD + 1;
178 s->expected_prior = 0;
179 s->received_prior = 0;
184 /* Returns 1 if we should handle this packet. */
185 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
187 uint16_t udelta = seq - s->max_seq;
188 const int MAX_DROPOUT = 3000;
189 const int MAX_MISORDER = 100;
190 const int MIN_SEQUENTIAL = 2;
192 /* source not valid until MIN_SEQUENTIAL packets with sequence
193 * seq. numbers have been received */
195 if (seq == s->max_seq + 1) {
198 if (s->probation == 0) {
199 rtp_init_sequence(s, seq);
204 s->probation = MIN_SEQUENTIAL - 1;
207 } else if (udelta < MAX_DROPOUT) {
208 // in order, with permissible gap
209 if (seq < s->max_seq) {
210 // sequence number wrapped; count another 64k cycles
211 s->cycles += RTP_SEQ_MOD;
214 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
215 // sequence made a large jump...
216 if (seq == s->bad_seq) {
217 /* two sequential packets -- assume that the other side
218 * restarted without telling us; just resync. */
219 rtp_init_sequence(s, seq);
221 s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
225 // duplicate or reordered packet...
231 int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
232 AVIOContext *avio, int count)
238 RTPStatistics *stats = &s->statistics;
240 uint32_t extended_max;
241 uint32_t expected_interval;
242 uint32_t received_interval;
243 uint32_t lost_interval;
246 uint64_t ntp_time = s->last_rtcp_ntp_time; // TODO: Get local ntp time?
248 if ((!fd && !avio) || (count < 1))
251 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
252 /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
253 s->octet_count += count;
254 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
256 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
259 s->last_octet_count = s->octet_count;
263 else if (avio_open_dyn_buf(&pb) < 0)
267 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
268 avio_w8(pb, RTCP_RR);
269 avio_wb16(pb, 7); /* length in words - 1 */
270 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
271 avio_wb32(pb, s->ssrc + 1);
272 avio_wb32(pb, s->ssrc); // server SSRC
273 // some placeholders we should really fill...
275 extended_max = stats->cycles + stats->max_seq;
276 expected = extended_max - stats->base_seq + 1;
277 lost = expected - stats->received;
278 lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
279 expected_interval = expected - stats->expected_prior;
280 stats->expected_prior = expected;
281 received_interval = stats->received - stats->received_prior;
282 stats->received_prior = stats->received;
283 lost_interval = expected_interval - received_interval;
284 if (expected_interval == 0 || lost_interval <= 0)
287 fraction = (lost_interval << 8) / expected_interval;
289 fraction = (fraction << 24) | lost;
291 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
292 avio_wb32(pb, extended_max); /* max sequence received */
293 avio_wb32(pb, stats->jitter >> 4); /* jitter */
295 if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) {
296 avio_wb32(pb, 0); /* last SR timestamp */
297 avio_wb32(pb, 0); /* delay since last SR */
299 uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
300 uint32_t delay_since_last = ntp_time - s->last_rtcp_ntp_time;
302 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
303 avio_wb32(pb, delay_since_last); /* delay since last SR */
307 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
308 avio_w8(pb, RTCP_SDES);
309 len = strlen(s->hostname);
310 avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
311 avio_wb32(pb, s->ssrc + 1);
314 avio_write(pb, s->hostname, len);
315 avio_w8(pb, 0); /* END */
317 for (len = (7 + 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(fd, buf, len);
328 av_dlog(s->ic, "result from ffurl_write: %d\n", result);
334 void ff_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);
372 static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
373 uint16_t *missing_mask)
376 uint16_t next_seq = s->seq + 1;
377 RTPPacket *pkt = s->queue;
379 if (!pkt || pkt->seq == next_seq)
383 for (i = 1; i <= 16; i++) {
384 uint16_t missing_seq = next_seq + i;
386 int16_t diff = pkt->seq - missing_seq;
393 if (pkt->seq == missing_seq)
395 *missing_mask |= 1 << (i - 1);
398 *first_missing = next_seq;
402 int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
405 int len, need_keyframe, missing_packets;
409 uint16_t first_missing, missing_mask;
414 need_keyframe = s->handler && s->handler->need_keyframe &&
415 s->handler->need_keyframe(s->dynamic_protocol_context);
416 missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
418 if (!need_keyframe && !missing_packets)
421 /* Send new feedback if enough time has elapsed since the last
422 * feedback packet. */
425 if (s->last_feedback_time &&
426 (now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL)
428 s->last_feedback_time = now;
432 else if (avio_open_dyn_buf(&pb) < 0)
436 avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
437 avio_w8(pb, RTCP_PSFB);
438 avio_wb16(pb, 2); /* length in words - 1 */
439 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
440 avio_wb32(pb, s->ssrc + 1);
441 avio_wb32(pb, s->ssrc); // server SSRC
444 if (missing_packets) {
445 avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
446 avio_w8(pb, RTCP_RTPFB);
447 avio_wb16(pb, 3); /* length in words - 1 */
448 avio_wb32(pb, s->ssrc + 1);
449 avio_wb32(pb, s->ssrc); // server SSRC
451 avio_wb16(pb, first_missing);
452 avio_wb16(pb, missing_mask);
458 len = avio_close_dyn_buf(pb, &buf);
459 if (len > 0 && buf) {
460 ffurl_write(fd, buf, len);
467 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
468 * MPEG2-TS streams to indicate that they should be demuxed inside the
469 * rtp demux (otherwise AV_CODEC_ID_MPEG2TS packets are returned)
471 RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
472 int payload_type, int queue_size)
476 s = av_mallocz(sizeof(RTPDemuxContext));
479 s->payload_type = payload_type;
480 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
481 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
484 s->queue_size = queue_size;
485 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
486 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
487 s->ts = ff_mpegts_parse_open(s->ic);
493 switch (st->codec->codec_id) {
494 case AV_CODEC_ID_MPEG1VIDEO:
495 case AV_CODEC_ID_MPEG2VIDEO:
496 case AV_CODEC_ID_MP2:
497 case AV_CODEC_ID_MP3:
498 case AV_CODEC_ID_MPEG4:
499 case AV_CODEC_ID_H263:
500 case AV_CODEC_ID_H264:
501 st->need_parsing = AVSTREAM_PARSE_FULL;
503 case AV_CODEC_ID_VORBIS:
504 st->need_parsing = AVSTREAM_PARSE_HEADERS;
506 case AV_CODEC_ID_ADPCM_G722:
507 /* According to RFC 3551, the stream clock rate is 8000
508 * even if the sample rate is 16000. */
509 if (st->codec->sample_rate == 8000)
510 st->codec->sample_rate = 16000;
516 // needed to send back RTCP RR in RTSP sessions
517 gethostname(s->hostname, sizeof(s->hostname));
521 void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
522 RTPDynamicProtocolHandler *handler)
524 s->dynamic_protocol_context = ctx;
525 s->handler = handler;
529 * This was the second switch in rtp_parse packet.
530 * Normalizes time, if required, sets stream_index, etc.
532 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
534 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
535 return; /* Timestamp already set by depacketizer */
536 if (timestamp == RTP_NOTS_VALUE)
539 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
543 /* compute pts from timestamp with received ntp_time */
544 delta_timestamp = timestamp - s->last_rtcp_timestamp;
545 /* convert to the PTS timebase */
546 addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time,
547 s->st->time_base.den,
548 (uint64_t) s->st->time_base.num << 32);
549 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
554 if (!s->base_timestamp)
555 s->base_timestamp = timestamp;
556 /* assume that the difference is INT32_MIN < x < INT32_MAX,
557 * but allow the first timestamp to exceed INT32_MAX */
559 s->unwrapped_timestamp += timestamp;
561 s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
562 s->timestamp = timestamp;
563 pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
567 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
568 const uint8_t *buf, int len)
570 unsigned int ssrc, h;
571 int payload_type, seq, ret, flags = 0;
578 payload_type = buf[1] & 0x7f;
580 flags |= RTP_FLAG_MARKER;
581 seq = AV_RB16(buf + 2);
582 timestamp = AV_RB32(buf + 4);
583 ssrc = AV_RB32(buf + 8);
584 /* store the ssrc in the RTPDemuxContext */
587 /* NOTE: we can handle only one payload type */
588 if (s->payload_type != payload_type)
592 // only do something with this if all the rtp checks pass...
593 if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
594 av_log(st ? st->codec : NULL, AV_LOG_ERROR,
595 "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
596 payload_type, seq, ((s->seq + 1) & 0xffff));
601 int padding = buf[len - 1];
602 if (len >= 12 + padding)
610 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
614 /* calculate the header extension length (stored as number
615 * of 32-bit words) */
616 ext = (AV_RB16(buf + 2) + 1) << 2;
620 // skip past RTP header extension
626 /* specific MPEG2-TS demux support */
627 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
628 /* The only error that can be returned from ff_mpegts_parse_packet
629 * is "no more data to return from the provided buffer", so return
630 * AVERROR(EAGAIN) for all errors */
632 return AVERROR(EAGAIN);
634 s->read_buf_size = FFMIN(len - ret, sizeof(s->buf));
635 memcpy(s->buf, buf + ret, s->read_buf_size);
636 s->read_buf_index = 0;
640 } else if (s->handler && s->handler->parse_packet) {
641 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
642 s->st, pkt, ×tamp, buf, len, seq,
645 /* At this point, the RTP header has been stripped;
646 * This is ASSUMING that there is only 1 CSRC, which isn't wise. */
647 switch (st->codec->codec_id) {
648 case AV_CODEC_ID_MP2:
649 case AV_CODEC_ID_MP3:
650 /* better than nothing: skip MPEG audio RTP header */
656 av_new_packet(pkt, len);
657 memcpy(pkt->data, buf, len);
659 case AV_CODEC_ID_MPEG1VIDEO:
660 case AV_CODEC_ID_MPEG2VIDEO:
661 /* better than nothing: skip MPEG video RTP header */
674 av_new_packet(pkt, len);
675 memcpy(pkt->data, buf, len);
678 av_new_packet(pkt, len);
679 memcpy(pkt->data, buf, len);
683 pkt->stream_index = st->index;
686 // now perform timestamp things....
687 finalize_packet(s, pkt, timestamp);
692 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
695 RTPPacket *next = s->queue->next;
696 av_free(s->queue->buf);
705 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
707 uint16_t seq = AV_RB16(buf + 2);
708 RTPPacket *cur = s->queue, *prev = NULL, *packet;
710 /* Find the correct place in the queue to insert the packet */
712 int16_t diff = seq - cur->seq;
719 packet = av_mallocz(sizeof(*packet));
722 packet->recvtime = av_gettime();
734 static int has_next_packet(RTPDemuxContext *s)
736 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
739 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
741 return s->queue ? s->queue->recvtime : 0;
744 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
749 if (s->queue_len <= 0)
752 if (!has_next_packet(s))
753 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
754 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
756 /* Parse the first packet in the queue, and dequeue it */
757 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
758 next = s->queue->next;
759 av_free(s->queue->buf);
766 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
767 uint8_t **bufptr, int len)
769 uint8_t *buf = bufptr ? *bufptr : NULL;
775 /* If parsing of the previous packet actually returned 0 or an error,
776 * there's nothing more to be parsed from that packet, but we may have
777 * indicated that we can return the next enqueued packet. */
778 if (s->prev_ret <= 0)
779 return rtp_parse_queued_packet(s, pkt);
780 /* return the next packets, if any */
781 if (s->st && s->handler && s->handler->parse_packet) {
782 /* timestamp should be overwritten by parse_packet, if not,
783 * the packet is left with pts == AV_NOPTS_VALUE */
784 timestamp = RTP_NOTS_VALUE;
785 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
786 s->st, pkt, ×tamp, NULL, 0, 0,
788 finalize_packet(s, pkt, timestamp);
791 // TODO: Move to a dynamic packet handler (like above)
792 if (s->read_buf_index >= s->read_buf_size)
793 return AVERROR(EAGAIN);
794 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
795 s->read_buf_size - s->read_buf_index);
797 return AVERROR(EAGAIN);
798 s->read_buf_index += ret;
799 if (s->read_buf_index < s->read_buf_size)
809 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
811 if (RTP_PT_IS_RTCP(buf[1])) {
812 return rtcp_parse_packet(s, buf, len);
815 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
816 /* First packet, or no reordering */
817 return rtp_parse_packet_internal(s, pkt, buf, len);
819 uint16_t seq = AV_RB16(buf + 2);
820 int16_t diff = seq - s->seq;
822 /* Packet older than the previously emitted one, drop */
823 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
824 "RTP: dropping old packet received too late\n");
826 } else if (diff <= 1) {
828 rv = rtp_parse_packet_internal(s, pkt, buf, len);
831 /* Still missing some packet, enqueue this one. */
832 enqueue_packet(s, buf, len);
834 /* Return the first enqueued packet if the queue is full,
835 * even if we're missing something */
836 if (s->queue_len >= s->queue_size)
837 return rtp_parse_queued_packet(s, pkt);
844 * Parse an RTP or RTCP packet directly sent as a buffer.
845 * @param s RTP parse context.
846 * @param pkt returned packet
847 * @param bufptr pointer to the input buffer or NULL to read the next packets
848 * @param len buffer len
849 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
850 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
852 int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
853 uint8_t **bufptr, int len)
855 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
857 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
858 rv = rtp_parse_queued_packet(s, pkt);
859 return rv ? rv : has_next_packet(s);
862 void ff_rtp_parse_close(RTPDemuxContext *s)
864 ff_rtp_reset_packet_queue(s);
865 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
866 ff_mpegts_parse_close(s->ts);
871 int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
872 int (*parse_fmtp)(AVStream *stream,
873 PayloadContext *data,
874 char *attr, char *value))
879 int value_size = strlen(p) + 1;
881 if (!(value = av_malloc(value_size))) {
882 av_log(NULL, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
883 return AVERROR(ENOMEM);
886 // remove protocol identifier
887 while (*p && *p == ' ')
889 while (*p && *p != ' ')
890 p++; // eat protocol identifier
891 while (*p && *p == ' ')
892 p++; // strip trailing spaces
894 while (ff_rtsp_next_attr_and_value(&p,
896 value, value_size)) {
897 res = parse_fmtp(stream, data, attr, value);
898 if (res < 0 && res != AVERROR_PATCHWELCOME) {
907 int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
911 pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
912 pkt->stream_index = stream_idx;
913 pkt->destruct = av_destruct_packet;