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/intreadwrite.h"
25 #include "libavutil/time.h"
32 #include "rtpdec_formats.h"
34 #define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */
36 static RTPDynamicProtocolHandler l24_dynamic_handler = {
38 .codec_type = AVMEDIA_TYPE_AUDIO,
39 .codec_id = AV_CODEC_ID_PCM_S24BE,
42 static RTPDynamicProtocolHandler gsm_dynamic_handler = {
44 .codec_type = AVMEDIA_TYPE_AUDIO,
45 .codec_id = AV_CODEC_ID_GSM,
48 static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
49 .enc_name = "X-MP3-draft-00",
50 .codec_type = AVMEDIA_TYPE_AUDIO,
51 .codec_id = AV_CODEC_ID_MP3ADU,
54 static RTPDynamicProtocolHandler speex_dynamic_handler = {
56 .codec_type = AVMEDIA_TYPE_AUDIO,
57 .codec_id = AV_CODEC_ID_SPEEX,
60 static RTPDynamicProtocolHandler opus_dynamic_handler = {
62 .codec_type = AVMEDIA_TYPE_AUDIO,
63 .codec_id = AV_CODEC_ID_OPUS,
66 static RTPDynamicProtocolHandler t140_dynamic_handler = { /* RFC 4103 */
68 .codec_type = AVMEDIA_TYPE_SUBTITLE,
69 .codec_id = AV_CODEC_ID_TEXT,
72 static RTPDynamicProtocolHandler *rtp_first_dynamic_payload_handler = NULL;
74 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
76 handler->next = rtp_first_dynamic_payload_handler;
77 rtp_first_dynamic_payload_handler = handler;
80 void ff_register_rtp_dynamic_payload_handlers(void)
82 ff_register_dynamic_payload_handler(&ff_ac3_dynamic_handler);
83 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
84 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
85 ff_register_dynamic_payload_handler(&ff_dv_dynamic_handler);
86 ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
87 ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
88 ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
89 ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
90 ff_register_dynamic_payload_handler(&ff_g726le_16_dynamic_handler);
91 ff_register_dynamic_payload_handler(&ff_g726le_24_dynamic_handler);
92 ff_register_dynamic_payload_handler(&ff_g726le_32_dynamic_handler);
93 ff_register_dynamic_payload_handler(&ff_g726le_40_dynamic_handler);
94 ff_register_dynamic_payload_handler(&ff_h261_dynamic_handler);
95 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
96 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
97 ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
98 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
99 ff_register_dynamic_payload_handler(&ff_hevc_dynamic_handler);
100 ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
101 ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler);
102 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
103 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
104 ff_register_dynamic_payload_handler(&ff_mpeg_audio_dynamic_handler);
105 ff_register_dynamic_payload_handler(&ff_mpeg_audio_robust_dynamic_handler);
106 ff_register_dynamic_payload_handler(&ff_mpeg_video_dynamic_handler);
107 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
108 ff_register_dynamic_payload_handler(&ff_mpegts_dynamic_handler);
109 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
110 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
111 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
112 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
113 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
114 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
115 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
116 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
117 ff_register_dynamic_payload_handler(&ff_rfc4175_rtp_handler);
118 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
119 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
120 ff_register_dynamic_payload_handler(&ff_vc2hq_dynamic_handler);
121 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
122 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
123 ff_register_dynamic_payload_handler(&ff_vp9_dynamic_handler);
124 ff_register_dynamic_payload_handler(&gsm_dynamic_handler);
125 ff_register_dynamic_payload_handler(&l24_dynamic_handler);
126 ff_register_dynamic_payload_handler(&opus_dynamic_handler);
127 ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
128 ff_register_dynamic_payload_handler(&speex_dynamic_handler);
129 ff_register_dynamic_payload_handler(&t140_dynamic_handler);
132 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
133 enum AVMediaType codec_type)
135 RTPDynamicProtocolHandler *handler;
136 for (handler = rtp_first_dynamic_payload_handler;
137 handler; handler = handler->next)
138 if (handler->enc_name &&
139 !av_strcasecmp(name, handler->enc_name) &&
140 codec_type == handler->codec_type)
145 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
146 enum AVMediaType codec_type)
148 RTPDynamicProtocolHandler *handler;
149 for (handler = rtp_first_dynamic_payload_handler;
150 handler; handler = handler->next)
151 if (handler->static_payload_id && handler->static_payload_id == id &&
152 codec_type == handler->codec_type)
157 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
162 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
166 if (payload_len < 20) {
167 av_log(s->ic, AV_LOG_ERROR, "Invalid RTCP SR packet length\n");
168 return AVERROR_INVALIDDATA;
171 s->last_rtcp_reception_time = av_gettime_relative();
172 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
173 s->last_rtcp_timestamp = AV_RB32(buf + 16);
174 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
175 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
176 if (!s->base_timestamp)
177 s->base_timestamp = s->last_rtcp_timestamp;
178 s->rtcp_ts_offset = (int32_t)(s->last_rtcp_timestamp - s->base_timestamp);
192 #define RTP_SEQ_MOD (1 << 16)
194 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
196 memset(s, 0, sizeof(RTPStatistics));
197 s->max_seq = base_sequence;
202 * Called whenever there is a large jump in sequence numbers,
203 * or when they get out of probation...
205 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
209 s->base_seq = seq - 1;
210 s->bad_seq = RTP_SEQ_MOD + 1;
212 s->expected_prior = 0;
213 s->received_prior = 0;
218 /* Returns 1 if we should handle this packet. */
219 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
221 uint16_t udelta = seq - s->max_seq;
222 const int MAX_DROPOUT = 3000;
223 const int MAX_MISORDER = 100;
224 const int MIN_SEQUENTIAL = 2;
226 /* source not valid until MIN_SEQUENTIAL packets with sequence
227 * seq. numbers have been received */
229 if (seq == s->max_seq + 1) {
232 if (s->probation == 0) {
233 rtp_init_sequence(s, seq);
238 s->probation = MIN_SEQUENTIAL - 1;
241 } else if (udelta < MAX_DROPOUT) {
242 // in order, with permissible gap
243 if (seq < s->max_seq) {
244 // sequence number wrapped; count another 64k cycles
245 s->cycles += RTP_SEQ_MOD;
248 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
249 // sequence made a large jump...
250 if (seq == s->bad_seq) {
251 /* two sequential packets -- assume that the other side
252 * restarted without telling us; just resync. */
253 rtp_init_sequence(s, seq);
255 s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
259 // duplicate or reordered packet...
265 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp,
266 uint32_t arrival_timestamp)
268 // Most of this is pretty straight from RFC 3550 appendix A.8
269 uint32_t transit = arrival_timestamp - sent_timestamp;
270 uint32_t prev_transit = s->transit;
271 int32_t d = transit - prev_transit;
272 // Doing the FFABS() call directly on the "transit - prev_transit"
273 // expression doesn't work, since it's an unsigned expression. Doing the
274 // transit calculation in unsigned is desired though, since it most
275 // probably will need to wrap around.
277 s->transit = transit;
280 s->jitter += d - (int32_t) ((s->jitter + 8) >> 4);
283 int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
284 AVIOContext *avio, int count)
290 RTPStatistics *stats = &s->statistics;
292 uint32_t extended_max;
293 uint32_t expected_interval;
294 uint32_t received_interval;
295 int32_t lost_interval;
299 if ((!fd && !avio) || (count < 1))
302 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
303 /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
304 s->octet_count += count;
305 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
307 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
310 s->last_octet_count = s->octet_count;
314 else if (avio_open_dyn_buf(&pb) < 0)
318 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
319 avio_w8(pb, RTCP_RR);
320 avio_wb16(pb, 7); /* length in words - 1 */
321 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
322 avio_wb32(pb, s->ssrc + 1);
323 avio_wb32(pb, s->ssrc); // server SSRC
324 // some placeholders we should really fill...
326 extended_max = stats->cycles + stats->max_seq;
327 expected = extended_max - stats->base_seq;
328 lost = expected - stats->received;
329 lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
330 expected_interval = expected - stats->expected_prior;
331 stats->expected_prior = expected;
332 received_interval = stats->received - stats->received_prior;
333 stats->received_prior = stats->received;
334 lost_interval = expected_interval - received_interval;
335 if (expected_interval == 0 || lost_interval <= 0)
338 fraction = (lost_interval << 8) / expected_interval;
340 fraction = (fraction << 24) | lost;
342 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
343 avio_wb32(pb, extended_max); /* max sequence received */
344 avio_wb32(pb, stats->jitter >> 4); /* jitter */
346 if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) {
347 avio_wb32(pb, 0); /* last SR timestamp */
348 avio_wb32(pb, 0); /* delay since last SR */
350 uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
351 uint32_t delay_since_last = av_rescale(av_gettime_relative() - s->last_rtcp_reception_time,
352 65536, AV_TIME_BASE);
354 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
355 avio_wb32(pb, delay_since_last); /* delay since last SR */
359 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
360 avio_w8(pb, RTCP_SDES);
361 len = strlen(s->hostname);
362 avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
363 avio_wb32(pb, s->ssrc + 1);
366 avio_write(pb, s->hostname, len);
367 avio_w8(pb, 0); /* END */
369 for (len = (7 + len) % 4; len % 4; len++)
375 len = avio_close_dyn_buf(pb, &buf);
376 if ((len > 0) && buf) {
377 int av_unused result;
378 av_log(s->ic, AV_LOG_TRACE, "sending %d bytes of RR\n", len);
379 result = ffurl_write(fd, buf, len);
380 av_log(s->ic, AV_LOG_TRACE, "result from ffurl_write: %d\n", result);
386 void ff_rtp_send_punch_packets(URLContext *rtp_handle)
392 /* Send a small RTP packet */
393 if (avio_open_dyn_buf(&pb) < 0)
396 avio_w8(pb, (RTP_VERSION << 6));
397 avio_w8(pb, 0); /* Payload type */
398 avio_wb16(pb, 0); /* Seq */
399 avio_wb32(pb, 0); /* Timestamp */
400 avio_wb32(pb, 0); /* SSRC */
403 len = avio_close_dyn_buf(pb, &buf);
404 if ((len > 0) && buf)
405 ffurl_write(rtp_handle, buf, len);
408 /* Send a minimal RTCP RR */
409 if (avio_open_dyn_buf(&pb) < 0)
412 avio_w8(pb, (RTP_VERSION << 6));
413 avio_w8(pb, RTCP_RR); /* receiver report */
414 avio_wb16(pb, 1); /* length in words - 1 */
415 avio_wb32(pb, 0); /* our own SSRC */
418 len = avio_close_dyn_buf(pb, &buf);
419 if ((len > 0) && buf)
420 ffurl_write(rtp_handle, buf, len);
424 static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
425 uint16_t *missing_mask)
428 uint16_t next_seq = s->seq + 1;
429 RTPPacket *pkt = s->queue;
431 if (!pkt || pkt->seq == next_seq)
435 for (i = 1; i <= 16; i++) {
436 uint16_t missing_seq = next_seq + i;
438 int16_t diff = pkt->seq - missing_seq;
445 if (pkt->seq == missing_seq)
447 *missing_mask |= 1 << (i - 1);
450 *first_missing = next_seq;
454 int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
457 int len, need_keyframe, missing_packets;
461 uint16_t first_missing = 0, missing_mask = 0;
466 need_keyframe = s->handler && s->handler->need_keyframe &&
467 s->handler->need_keyframe(s->dynamic_protocol_context);
468 missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
470 if (!need_keyframe && !missing_packets)
473 /* Send new feedback if enough time has elapsed since the last
474 * feedback packet. */
476 now = av_gettime_relative();
477 if (s->last_feedback_time &&
478 (now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL)
480 s->last_feedback_time = now;
484 else if (avio_open_dyn_buf(&pb) < 0)
488 avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
489 avio_w8(pb, RTCP_PSFB);
490 avio_wb16(pb, 2); /* length in words - 1 */
491 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
492 avio_wb32(pb, s->ssrc + 1);
493 avio_wb32(pb, s->ssrc); // server SSRC
496 if (missing_packets) {
497 avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
498 avio_w8(pb, RTCP_RTPFB);
499 avio_wb16(pb, 3); /* length in words - 1 */
500 avio_wb32(pb, s->ssrc + 1);
501 avio_wb32(pb, s->ssrc); // server SSRC
503 avio_wb16(pb, first_missing);
504 avio_wb16(pb, missing_mask);
510 len = avio_close_dyn_buf(pb, &buf);
511 if (len > 0 && buf) {
512 ffurl_write(fd, buf, len);
519 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
522 RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
523 int payload_type, int queue_size)
527 s = av_mallocz(sizeof(RTPDemuxContext));
530 s->payload_type = payload_type;
531 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
532 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
535 s->queue_size = queue_size;
537 av_log(s->ic, AV_LOG_VERBOSE, "setting jitter buffer size to %d\n",
540 rtp_init_statistics(&s->statistics, 0);
542 switch (st->codecpar->codec_id) {
543 case AV_CODEC_ID_ADPCM_G722:
544 /* According to RFC 3551, the stream clock rate is 8000
545 * even if the sample rate is 16000. */
546 if (st->codecpar->sample_rate == 8000)
547 st->codecpar->sample_rate = 16000;
553 // needed to send back RTCP RR in RTSP sessions
554 gethostname(s->hostname, sizeof(s->hostname));
558 void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
559 RTPDynamicProtocolHandler *handler)
561 s->dynamic_protocol_context = ctx;
562 s->handler = handler;
565 void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
568 if (!ff_srtp_set_crypto(&s->srtp, suite, params))
573 * This was the second switch in rtp_parse packet.
574 * Normalizes time, if required, sets stream_index, etc.
576 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
578 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
579 return; /* Timestamp already set by depacketizer */
580 if (timestamp == RTP_NOTS_VALUE)
583 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
587 /* compute pts from timestamp with received ntp_time */
588 delta_timestamp = timestamp - s->last_rtcp_timestamp;
589 /* convert to the PTS timebase */
590 addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time,
591 s->st->time_base.den,
592 (uint64_t) s->st->time_base.num << 32);
593 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
598 if (!s->base_timestamp)
599 s->base_timestamp = timestamp;
600 /* assume that the difference is INT32_MIN < x < INT32_MAX,
601 * but allow the first timestamp to exceed INT32_MAX */
603 s->unwrapped_timestamp += timestamp;
605 s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
606 s->timestamp = timestamp;
607 pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
611 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
612 const uint8_t *buf, int len)
615 int payload_type, seq, flags = 0;
621 csrc = buf[0] & 0x0f;
623 payload_type = buf[1] & 0x7f;
625 flags |= RTP_FLAG_MARKER;
626 seq = AV_RB16(buf + 2);
627 timestamp = AV_RB32(buf + 4);
628 ssrc = AV_RB32(buf + 8);
629 /* store the ssrc in the RTPDemuxContext */
632 /* NOTE: we can handle only one payload type */
633 if (s->payload_type != payload_type)
637 // only do something with this if all the rtp checks pass...
638 if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
639 av_log(s->ic, AV_LOG_ERROR,
640 "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
641 payload_type, seq, ((s->seq + 1) & 0xffff));
646 int padding = buf[len - 1];
647 if (len >= 12 + padding)
658 return AVERROR_INVALIDDATA;
660 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
664 /* calculate the header extension length (stored as number
665 * of 32-bit words) */
666 ext = (AV_RB16(buf + 2) + 1) << 2;
670 // skip past RTP header extension
675 if (s->handler && s->handler->parse_packet) {
676 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
677 s->st, pkt, ×tamp, buf, len, seq,
680 if ((rv = av_new_packet(pkt, len)) < 0)
682 memcpy(pkt->data, buf, len);
683 pkt->stream_index = st->index;
685 return AVERROR(EINVAL);
688 // now perform timestamp things....
689 finalize_packet(s, pkt, timestamp);
694 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
697 RTPPacket *next = s->queue->next;
698 av_freep(&s->queue->buf);
707 static int enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
709 uint16_t seq = AV_RB16(buf + 2);
710 RTPPacket **cur = &s->queue, *packet;
712 /* Find the correct place in the queue to insert the packet */
714 int16_t diff = seq - (*cur)->seq;
720 packet = av_mallocz(sizeof(*packet));
722 return AVERROR(ENOMEM);
723 packet->recvtime = av_gettime_relative();
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->ic, 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_freep(&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->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);
796 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
798 if (RTP_PT_IS_RTCP(buf[1])) {
799 return rtcp_parse_packet(s, buf, len);
803 int64_t received = av_gettime_relative();
804 uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
806 timestamp = AV_RB32(buf + 4);
807 // Calculate the jitter immediately, before queueing the packet
808 // into the reordering queue.
809 rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
812 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
813 /* First packet, or no reordering */
814 return rtp_parse_packet_internal(s, pkt, buf, len);
816 uint16_t seq = AV_RB16(buf + 2);
817 int16_t diff = seq - s->seq;
819 /* Packet older than the previously emitted one, drop */
820 av_log(s->ic, AV_LOG_WARNING,
821 "RTP: dropping old packet received too late\n");
823 } else if (diff <= 1) {
825 rv = rtp_parse_packet_internal(s, pkt, buf, len);
828 /* Still missing some packet, enqueue this one. */
829 rv = enqueue_packet(s, buf, len);
833 /* Return the first enqueued packet if the queue is full,
834 * even if we're missing something */
835 if (s->queue_len >= s->queue_size) {
836 av_log(s->ic, AV_LOG_WARNING, "jitter buffer full\n");
837 return rtp_parse_queued_packet(s, pkt);
845 * Parse an RTP or RTCP packet directly sent as a buffer.
846 * @param s RTP parse context.
847 * @param pkt returned packet
848 * @param bufptr pointer to the input buffer or NULL to read the next packets
849 * @param len buffer len
850 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
851 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
853 int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
854 uint8_t **bufptr, int len)
857 if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
859 rv = rtp_parse_one_packet(s, pkt, bufptr, len);
861 while (rv < 0 && has_next_packet(s))
862 rv = rtp_parse_queued_packet(s, pkt);
863 return rv ? rv : has_next_packet(s);
866 void ff_rtp_parse_close(RTPDemuxContext *s)
868 ff_rtp_reset_packet_queue(s);
869 ff_srtp_free(&s->srtp);
873 int ff_parse_fmtp(AVFormatContext *s,
874 AVStream *stream, PayloadContext *data, const char *p,
875 int (*parse_fmtp)(AVFormatContext *s,
877 PayloadContext *data,
878 const char *attr, const char *value))
883 int value_size = strlen(p) + 1;
885 if (!(value = av_malloc(value_size))) {
886 av_log(s, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
887 return AVERROR(ENOMEM);
890 // remove protocol identifier
891 while (*p && *p == ' ')
893 while (*p && *p != ' ')
894 p++; // eat protocol identifier
895 while (*p && *p == ' ')
896 p++; // strip trailing spaces
898 while (ff_rtsp_next_attr_and_value(&p,
900 value, value_size)) {
901 res = parse_fmtp(s, stream, data, attr, value);
902 if (res < 0 && res != AVERROR_PATCHWELCOME) {
911 int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
916 pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
917 pkt->stream_index = stream_idx;
919 if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) {
920 av_freep(&pkt->data);