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 #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 gsm_dynamic_handler = {
37 .codec_type = AVMEDIA_TYPE_AUDIO,
38 .codec_id = AV_CODEC_ID_GSM,
41 static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
42 .enc_name = "X-MP3-draft-00",
43 .codec_type = AVMEDIA_TYPE_AUDIO,
44 .codec_id = AV_CODEC_ID_MP3ADU,
47 static RTPDynamicProtocolHandler speex_dynamic_handler = {
49 .codec_type = AVMEDIA_TYPE_AUDIO,
50 .codec_id = AV_CODEC_ID_SPEEX,
53 static RTPDynamicProtocolHandler opus_dynamic_handler = {
55 .codec_type = AVMEDIA_TYPE_AUDIO,
56 .codec_id = AV_CODEC_ID_OPUS,
59 static RTPDynamicProtocolHandler t140_dynamic_handler = { /* RFC 4103 */
61 .codec_type = AVMEDIA_TYPE_SUBTITLE,
62 .codec_id = AV_CODEC_ID_TEXT,
65 static RTPDynamicProtocolHandler *rtp_first_dynamic_payload_handler = NULL;
67 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
69 handler->next = rtp_first_dynamic_payload_handler;
70 rtp_first_dynamic_payload_handler = handler;
73 void ff_register_rtp_dynamic_payload_handlers(void)
75 ff_register_dynamic_payload_handler(&ff_ac3_dynamic_handler);
76 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
77 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
78 ff_register_dynamic_payload_handler(&ff_dv_dynamic_handler);
79 ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
80 ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
81 ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
82 ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
83 ff_register_dynamic_payload_handler(&ff_h261_dynamic_handler);
84 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
85 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
86 ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
87 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
88 ff_register_dynamic_payload_handler(&ff_hevc_dynamic_handler);
89 ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
90 ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler);
91 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
92 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
93 ff_register_dynamic_payload_handler(&ff_mpeg_audio_dynamic_handler);
94 ff_register_dynamic_payload_handler(&ff_mpeg_audio_robust_dynamic_handler);
95 ff_register_dynamic_payload_handler(&ff_mpeg_video_dynamic_handler);
96 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
97 ff_register_dynamic_payload_handler(&ff_mpegts_dynamic_handler);
98 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
99 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
100 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
101 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
102 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
103 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
104 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
105 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
106 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
107 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
108 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
109 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
110 ff_register_dynamic_payload_handler(&ff_vp9_dynamic_handler);
111 ff_register_dynamic_payload_handler(&gsm_dynamic_handler);
112 ff_register_dynamic_payload_handler(&opus_dynamic_handler);
113 ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
114 ff_register_dynamic_payload_handler(&speex_dynamic_handler);
115 ff_register_dynamic_payload_handler(&t140_dynamic_handler);
118 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
119 enum AVMediaType codec_type)
121 RTPDynamicProtocolHandler *handler;
122 for (handler = rtp_first_dynamic_payload_handler;
123 handler; handler = handler->next)
124 if (handler->enc_name &&
125 !av_strcasecmp(name, handler->enc_name) &&
126 codec_type == handler->codec_type)
131 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
132 enum AVMediaType codec_type)
134 RTPDynamicProtocolHandler *handler;
135 for (handler = rtp_first_dynamic_payload_handler;
136 handler; handler = handler->next)
137 if (handler->static_payload_id && handler->static_payload_id == id &&
138 codec_type == handler->codec_type)
143 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
148 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
152 if (payload_len < 20) {
153 av_log(s->ic, AV_LOG_ERROR, "Invalid RTCP SR packet length\n");
154 return AVERROR_INVALIDDATA;
157 s->last_rtcp_reception_time = av_gettime_relative();
158 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
159 s->last_rtcp_timestamp = AV_RB32(buf + 16);
160 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
161 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
162 if (!s->base_timestamp)
163 s->base_timestamp = s->last_rtcp_timestamp;
164 s->rtcp_ts_offset = (int32_t)(s->last_rtcp_timestamp - s->base_timestamp);
178 #define RTP_SEQ_MOD (1 << 16)
180 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
182 memset(s, 0, sizeof(RTPStatistics));
183 s->max_seq = base_sequence;
188 * Called whenever there is a large jump in sequence numbers,
189 * or when they get out of probation...
191 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
195 s->base_seq = seq - 1;
196 s->bad_seq = RTP_SEQ_MOD + 1;
198 s->expected_prior = 0;
199 s->received_prior = 0;
204 /* Returns 1 if we should handle this packet. */
205 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
207 uint16_t udelta = seq - s->max_seq;
208 const int MAX_DROPOUT = 3000;
209 const int MAX_MISORDER = 100;
210 const int MIN_SEQUENTIAL = 2;
212 /* source not valid until MIN_SEQUENTIAL packets with sequence
213 * seq. numbers have been received */
215 if (seq == s->max_seq + 1) {
218 if (s->probation == 0) {
219 rtp_init_sequence(s, seq);
224 s->probation = MIN_SEQUENTIAL - 1;
227 } else if (udelta < MAX_DROPOUT) {
228 // in order, with permissible gap
229 if (seq < s->max_seq) {
230 // sequence number wrapped; count another 64k cycles
231 s->cycles += RTP_SEQ_MOD;
234 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
235 // sequence made a large jump...
236 if (seq == s->bad_seq) {
237 /* two sequential packets -- assume that the other side
238 * restarted without telling us; just resync. */
239 rtp_init_sequence(s, seq);
241 s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
245 // duplicate or reordered packet...
251 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp,
252 uint32_t arrival_timestamp)
254 // Most of this is pretty straight from RFC 3550 appendix A.8
255 uint32_t transit = arrival_timestamp - sent_timestamp;
256 uint32_t prev_transit = s->transit;
257 int32_t d = transit - prev_transit;
258 // Doing the FFABS() call directly on the "transit - prev_transit"
259 // expression doesn't work, since it's an unsigned expression. Doing the
260 // transit calculation in unsigned is desired though, since it most
261 // probably will need to wrap around.
263 s->transit = transit;
266 s->jitter += d - (int32_t) ((s->jitter + 8) >> 4);
269 int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
270 AVIOContext *avio, int count)
276 RTPStatistics *stats = &s->statistics;
278 uint32_t extended_max;
279 uint32_t expected_interval;
280 uint32_t received_interval;
281 int32_t lost_interval;
285 if ((!fd && !avio) || (count < 1))
288 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
289 /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
290 s->octet_count += count;
291 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
293 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
296 s->last_octet_count = s->octet_count;
300 else if (avio_open_dyn_buf(&pb) < 0)
304 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
305 avio_w8(pb, RTCP_RR);
306 avio_wb16(pb, 7); /* length in words - 1 */
307 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
308 avio_wb32(pb, s->ssrc + 1);
309 avio_wb32(pb, s->ssrc); // server SSRC
310 // some placeholders we should really fill...
312 extended_max = stats->cycles + stats->max_seq;
313 expected = extended_max - stats->base_seq;
314 lost = expected - stats->received;
315 lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
316 expected_interval = expected - stats->expected_prior;
317 stats->expected_prior = expected;
318 received_interval = stats->received - stats->received_prior;
319 stats->received_prior = stats->received;
320 lost_interval = expected_interval - received_interval;
321 if (expected_interval == 0 || lost_interval <= 0)
324 fraction = (lost_interval << 8) / expected_interval;
326 fraction = (fraction << 24) | lost;
328 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
329 avio_wb32(pb, extended_max); /* max sequence received */
330 avio_wb32(pb, stats->jitter >> 4); /* jitter */
332 if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) {
333 avio_wb32(pb, 0); /* last SR timestamp */
334 avio_wb32(pb, 0); /* delay since last SR */
336 uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
337 uint32_t delay_since_last = av_rescale(av_gettime_relative() - s->last_rtcp_reception_time,
338 65536, AV_TIME_BASE);
340 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
341 avio_wb32(pb, delay_since_last); /* delay since last SR */
345 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
346 avio_w8(pb, RTCP_SDES);
347 len = strlen(s->hostname);
348 avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
349 avio_wb32(pb, s->ssrc + 1);
352 avio_write(pb, s->hostname, len);
353 avio_w8(pb, 0); /* END */
355 for (len = (7 + len) % 4; len % 4; len++)
361 len = avio_close_dyn_buf(pb, &buf);
362 if ((len > 0) && buf) {
363 int av_unused result;
364 av_log(s->ic, AV_LOG_TRACE, "sending %d bytes of RR\n", len);
365 result = ffurl_write(fd, buf, len);
366 av_log(s->ic, AV_LOG_TRACE, "result from ffurl_write: %d\n", result);
372 void ff_rtp_send_punch_packets(URLContext *rtp_handle)
378 /* Send a small RTP packet */
379 if (avio_open_dyn_buf(&pb) < 0)
382 avio_w8(pb, (RTP_VERSION << 6));
383 avio_w8(pb, 0); /* Payload type */
384 avio_wb16(pb, 0); /* Seq */
385 avio_wb32(pb, 0); /* Timestamp */
386 avio_wb32(pb, 0); /* SSRC */
389 len = avio_close_dyn_buf(pb, &buf);
390 if ((len > 0) && buf)
391 ffurl_write(rtp_handle, buf, len);
394 /* Send a minimal RTCP RR */
395 if (avio_open_dyn_buf(&pb) < 0)
398 avio_w8(pb, (RTP_VERSION << 6));
399 avio_w8(pb, RTCP_RR); /* receiver report */
400 avio_wb16(pb, 1); /* length in words - 1 */
401 avio_wb32(pb, 0); /* our own SSRC */
404 len = avio_close_dyn_buf(pb, &buf);
405 if ((len > 0) && buf)
406 ffurl_write(rtp_handle, buf, len);
410 static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
411 uint16_t *missing_mask)
414 uint16_t next_seq = s->seq + 1;
415 RTPPacket *pkt = s->queue;
417 if (!pkt || pkt->seq == next_seq)
421 for (i = 1; i <= 16; i++) {
422 uint16_t missing_seq = next_seq + i;
424 int16_t diff = pkt->seq - missing_seq;
431 if (pkt->seq == missing_seq)
433 *missing_mask |= 1 << (i - 1);
436 *first_missing = next_seq;
440 int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
443 int len, need_keyframe, missing_packets;
447 uint16_t first_missing = 0, missing_mask = 0;
452 need_keyframe = s->handler && s->handler->need_keyframe &&
453 s->handler->need_keyframe(s->dynamic_protocol_context);
454 missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
456 if (!need_keyframe && !missing_packets)
459 /* Send new feedback if enough time has elapsed since the last
460 * feedback packet. */
462 now = av_gettime_relative();
463 if (s->last_feedback_time &&
464 (now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL)
466 s->last_feedback_time = now;
470 else if (avio_open_dyn_buf(&pb) < 0)
474 avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
475 avio_w8(pb, RTCP_PSFB);
476 avio_wb16(pb, 2); /* length in words - 1 */
477 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
478 avio_wb32(pb, s->ssrc + 1);
479 avio_wb32(pb, s->ssrc); // server SSRC
482 if (missing_packets) {
483 avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
484 avio_w8(pb, RTCP_RTPFB);
485 avio_wb16(pb, 3); /* length in words - 1 */
486 avio_wb32(pb, s->ssrc + 1);
487 avio_wb32(pb, s->ssrc); // server SSRC
489 avio_wb16(pb, first_missing);
490 avio_wb16(pb, missing_mask);
496 len = avio_close_dyn_buf(pb, &buf);
497 if (len > 0 && buf) {
498 ffurl_write(fd, buf, len);
505 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
508 RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
509 int payload_type, int queue_size)
513 s = av_mallocz(sizeof(RTPDemuxContext));
516 s->payload_type = payload_type;
517 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
518 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
521 s->queue_size = queue_size;
523 av_log(s->ic, AV_LOG_VERBOSE, "setting jitter buffer size to %d\n",
526 rtp_init_statistics(&s->statistics, 0);
528 switch (st->codec->codec_id) {
529 case AV_CODEC_ID_ADPCM_G722:
530 /* According to RFC 3551, the stream clock rate is 8000
531 * even if the sample rate is 16000. */
532 if (st->codec->sample_rate == 8000)
533 st->codec->sample_rate = 16000;
539 // needed to send back RTCP RR in RTSP sessions
540 gethostname(s->hostname, sizeof(s->hostname));
544 void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
545 RTPDynamicProtocolHandler *handler)
547 s->dynamic_protocol_context = ctx;
548 s->handler = handler;
551 void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
554 if (!ff_srtp_set_crypto(&s->srtp, suite, params))
559 * This was the second switch in rtp_parse packet.
560 * Normalizes time, if required, sets stream_index, etc.
562 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
564 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
565 return; /* Timestamp already set by depacketizer */
566 if (timestamp == RTP_NOTS_VALUE)
569 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
573 /* compute pts from timestamp with received ntp_time */
574 delta_timestamp = timestamp - s->last_rtcp_timestamp;
575 /* convert to the PTS timebase */
576 addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time,
577 s->st->time_base.den,
578 (uint64_t) s->st->time_base.num << 32);
579 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
584 if (!s->base_timestamp)
585 s->base_timestamp = timestamp;
586 /* assume that the difference is INT32_MIN < x < INT32_MAX,
587 * but allow the first timestamp to exceed INT32_MAX */
589 s->unwrapped_timestamp += timestamp;
591 s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
592 s->timestamp = timestamp;
593 pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
597 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
598 const uint8_t *buf, int len)
601 int payload_type, seq, flags = 0;
607 csrc = buf[0] & 0x0f;
609 payload_type = buf[1] & 0x7f;
611 flags |= RTP_FLAG_MARKER;
612 seq = AV_RB16(buf + 2);
613 timestamp = AV_RB32(buf + 4);
614 ssrc = AV_RB32(buf + 8);
615 /* store the ssrc in the RTPDemuxContext */
618 /* NOTE: we can handle only one payload type */
619 if (s->payload_type != payload_type)
623 // only do something with this if all the rtp checks pass...
624 if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
625 av_log(s->ic, AV_LOG_ERROR,
626 "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
627 payload_type, seq, ((s->seq + 1) & 0xffff));
632 int padding = buf[len - 1];
633 if (len >= 12 + padding)
644 return AVERROR_INVALIDDATA;
646 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
650 /* calculate the header extension length (stored as number
651 * of 32-bit words) */
652 ext = (AV_RB16(buf + 2) + 1) << 2;
656 // skip past RTP header extension
661 if (s->handler && s->handler->parse_packet) {
662 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
663 s->st, pkt, ×tamp, buf, len, seq,
666 if ((rv = av_new_packet(pkt, len)) < 0)
668 memcpy(pkt->data, buf, len);
669 pkt->stream_index = st->index;
671 return AVERROR(EINVAL);
674 // now perform timestamp things....
675 finalize_packet(s, pkt, timestamp);
680 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
683 RTPPacket *next = s->queue->next;
684 av_freep(&s->queue->buf);
693 static int enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
695 uint16_t seq = AV_RB16(buf + 2);
696 RTPPacket **cur = &s->queue, *packet;
698 /* Find the correct place in the queue to insert the packet */
700 int16_t diff = seq - (*cur)->seq;
706 packet = av_mallocz(sizeof(*packet));
708 return AVERROR(ENOMEM);
709 packet->recvtime = av_gettime_relative();
720 static int has_next_packet(RTPDemuxContext *s)
722 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
725 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
727 return s->queue ? s->queue->recvtime : 0;
730 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
735 if (s->queue_len <= 0)
738 if (!has_next_packet(s))
739 av_log(s->ic, AV_LOG_WARNING,
740 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
742 /* Parse the first packet in the queue, and dequeue it */
743 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
744 next = s->queue->next;
745 av_freep(&s->queue->buf);
752 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
753 uint8_t **bufptr, int len)
755 uint8_t *buf = bufptr ? *bufptr : NULL;
761 /* If parsing of the previous packet actually returned 0 or an error,
762 * there's nothing more to be parsed from that packet, but we may have
763 * indicated that we can return the next enqueued packet. */
764 if (s->prev_ret <= 0)
765 return rtp_parse_queued_packet(s, pkt);
766 /* return the next packets, if any */
767 if (s->handler && s->handler->parse_packet) {
768 /* timestamp should be overwritten by parse_packet, if not,
769 * the packet is left with pts == AV_NOPTS_VALUE */
770 timestamp = RTP_NOTS_VALUE;
771 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
772 s->st, pkt, ×tamp, NULL, 0, 0,
774 finalize_packet(s, pkt, timestamp);
782 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
784 if (RTP_PT_IS_RTCP(buf[1])) {
785 return rtcp_parse_packet(s, buf, len);
789 int64_t received = av_gettime_relative();
790 uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
792 timestamp = AV_RB32(buf + 4);
793 // Calculate the jitter immediately, before queueing the packet
794 // into the reordering queue.
795 rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
798 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
799 /* First packet, or no reordering */
800 return rtp_parse_packet_internal(s, pkt, buf, len);
802 uint16_t seq = AV_RB16(buf + 2);
803 int16_t diff = seq - s->seq;
805 /* Packet older than the previously emitted one, drop */
806 av_log(s->ic, AV_LOG_WARNING,
807 "RTP: dropping old packet received too late\n");
809 } else if (diff <= 1) {
811 rv = rtp_parse_packet_internal(s, pkt, buf, len);
814 /* Still missing some packet, enqueue this one. */
815 rv = enqueue_packet(s, buf, len);
819 /* Return the first enqueued packet if the queue is full,
820 * even if we're missing something */
821 if (s->queue_len >= s->queue_size) {
822 av_log(s->ic, AV_LOG_WARNING, "jitter buffer full\n");
823 return rtp_parse_queued_packet(s, pkt);
831 * Parse an RTP or RTCP packet directly sent as a buffer.
832 * @param s RTP parse context.
833 * @param pkt returned packet
834 * @param bufptr pointer to the input buffer or NULL to read the next packets
835 * @param len buffer len
836 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
837 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
839 int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
840 uint8_t **bufptr, int len)
843 if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
845 rv = rtp_parse_one_packet(s, pkt, bufptr, len);
847 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
848 rv = rtp_parse_queued_packet(s, pkt);
849 return rv ? rv : has_next_packet(s);
852 void ff_rtp_parse_close(RTPDemuxContext *s)
854 ff_rtp_reset_packet_queue(s);
855 ff_srtp_free(&s->srtp);
859 int ff_parse_fmtp(AVFormatContext *s,
860 AVStream *stream, PayloadContext *data, const char *p,
861 int (*parse_fmtp)(AVFormatContext *s,
863 PayloadContext *data,
864 const char *attr, const char *value))
869 int value_size = strlen(p) + 1;
871 if (!(value = av_malloc(value_size))) {
872 av_log(s, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
873 return AVERROR(ENOMEM);
876 // remove protocol identifier
877 while (*p && *p == ' ')
879 while (*p && *p != ' ')
880 p++; // eat protocol identifier
881 while (*p && *p == ' ')
882 p++; // strip trailing spaces
884 while (ff_rtsp_next_attr_and_value(&p,
886 value, value_size)) {
887 res = parse_fmtp(s, stream, data, attr, value);
888 if (res < 0 && res != AVERROR_PATCHWELCOME) {
897 int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
902 pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
903 pkt->stream_index = stream_idx;
905 if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) {
906 av_freep(&pkt->data);