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 ff_t140_dynamic_handler = {
61 .codec_type = AVMEDIA_TYPE_SUBTITLE,
62 .codec_id = AV_CODEC_ID_SUBRIP,
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_amr_nb_dynamic_handler);
76 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
77 ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
78 ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
79 ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
80 ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
81 ff_register_dynamic_payload_handler(&ff_h261_dynamic_handler);
82 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
83 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
84 ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
85 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
86 ff_register_dynamic_payload_handler(&ff_hevc_dynamic_handler);
87 ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
88 ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler);
89 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
90 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
91 ff_register_dynamic_payload_handler(&ff_mpeg_audio_dynamic_handler);
92 ff_register_dynamic_payload_handler(&ff_mpeg_video_dynamic_handler);
93 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
94 ff_register_dynamic_payload_handler(&ff_mpegts_dynamic_handler);
95 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
96 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
97 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
98 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
99 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
100 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
101 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
102 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
103 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
104 ff_register_dynamic_payload_handler(&ff_t140_dynamic_handler);
105 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
106 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
107 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
108 ff_register_dynamic_payload_handler(&gsm_dynamic_handler);
109 ff_register_dynamic_payload_handler(&opus_dynamic_handler);
110 ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
111 ff_register_dynamic_payload_handler(&speex_dynamic_handler);
114 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
115 enum AVMediaType codec_type)
117 RTPDynamicProtocolHandler *handler;
118 for (handler = rtp_first_dynamic_payload_handler;
119 handler; handler = handler->next)
120 if (!av_strcasecmp(name, handler->enc_name) &&
121 codec_type == handler->codec_type)
126 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
127 enum AVMediaType codec_type)
129 RTPDynamicProtocolHandler *handler;
130 for (handler = rtp_first_dynamic_payload_handler;
131 handler; handler = handler->next)
132 if (handler->static_payload_id && handler->static_payload_id == id &&
133 codec_type == handler->codec_type)
138 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
143 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
147 if (payload_len < 20) {
148 av_log(NULL, AV_LOG_ERROR,
149 "Invalid length for RTCP SR packet\n");
150 return AVERROR_INVALIDDATA;
153 s->last_rtcp_reception_time = av_gettime_relative();
154 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
155 s->last_rtcp_timestamp = AV_RB32(buf + 16);
156 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
157 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
158 if (!s->base_timestamp)
159 s->base_timestamp = s->last_rtcp_timestamp;
160 s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
174 #define RTP_SEQ_MOD (1 << 16)
176 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
178 memset(s, 0, sizeof(RTPStatistics));
179 s->max_seq = base_sequence;
184 * Called whenever there is a large jump in sequence numbers,
185 * or when they get out of probation...
187 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
191 s->base_seq = seq - 1;
192 s->bad_seq = RTP_SEQ_MOD + 1;
194 s->expected_prior = 0;
195 s->received_prior = 0;
200 /* Returns 1 if we should handle this packet. */
201 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
203 uint16_t udelta = seq - s->max_seq;
204 const int MAX_DROPOUT = 3000;
205 const int MAX_MISORDER = 100;
206 const int MIN_SEQUENTIAL = 2;
208 /* source not valid until MIN_SEQUENTIAL packets with sequence
209 * seq. numbers have been received */
211 if (seq == s->max_seq + 1) {
214 if (s->probation == 0) {
215 rtp_init_sequence(s, seq);
220 s->probation = MIN_SEQUENTIAL - 1;
223 } else if (udelta < MAX_DROPOUT) {
224 // in order, with permissible gap
225 if (seq < s->max_seq) {
226 // sequence number wrapped; count another 64k cycles
227 s->cycles += RTP_SEQ_MOD;
230 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
231 // sequence made a large jump...
232 if (seq == s->bad_seq) {
233 /* two sequential packets -- assume that the other side
234 * restarted without telling us; just resync. */
235 rtp_init_sequence(s, seq);
237 s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
241 // duplicate or reordered packet...
247 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp,
248 uint32_t arrival_timestamp)
250 // Most of this is pretty straight from RFC 3550 appendix A.8
251 uint32_t transit = arrival_timestamp - sent_timestamp;
252 uint32_t prev_transit = s->transit;
253 int32_t d = transit - prev_transit;
254 // Doing the FFABS() call directly on the "transit - prev_transit"
255 // expression doesn't work, since it's an unsigned expression. Doing the
256 // transit calculation in unsigned is desired though, since it most
257 // probably will need to wrap around.
259 s->transit = transit;
262 s->jitter += d - (int32_t) ((s->jitter + 8) >> 4);
265 int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
266 AVIOContext *avio, int count)
272 RTPStatistics *stats = &s->statistics;
274 uint32_t extended_max;
275 uint32_t expected_interval;
276 uint32_t received_interval;
277 int32_t lost_interval;
281 if ((!fd && !avio) || (count < 1))
284 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
285 /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
286 s->octet_count += count;
287 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
289 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
292 s->last_octet_count = s->octet_count;
296 else if (avio_open_dyn_buf(&pb) < 0)
300 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
301 avio_w8(pb, RTCP_RR);
302 avio_wb16(pb, 7); /* length in words - 1 */
303 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
304 avio_wb32(pb, s->ssrc + 1);
305 avio_wb32(pb, s->ssrc); // server SSRC
306 // some placeholders we should really fill...
308 extended_max = stats->cycles + stats->max_seq;
309 expected = extended_max - stats->base_seq;
310 lost = expected - stats->received;
311 lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
312 expected_interval = expected - stats->expected_prior;
313 stats->expected_prior = expected;
314 received_interval = stats->received - stats->received_prior;
315 stats->received_prior = stats->received;
316 lost_interval = expected_interval - received_interval;
317 if (expected_interval == 0 || lost_interval <= 0)
320 fraction = (lost_interval << 8) / expected_interval;
322 fraction = (fraction << 24) | lost;
324 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
325 avio_wb32(pb, extended_max); /* max sequence received */
326 avio_wb32(pb, stats->jitter >> 4); /* jitter */
328 if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) {
329 avio_wb32(pb, 0); /* last SR timestamp */
330 avio_wb32(pb, 0); /* delay since last SR */
332 uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
333 uint32_t delay_since_last = av_rescale(av_gettime_relative() - s->last_rtcp_reception_time,
334 65536, AV_TIME_BASE);
336 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
337 avio_wb32(pb, delay_since_last); /* delay since last SR */
341 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
342 avio_w8(pb, RTCP_SDES);
343 len = strlen(s->hostname);
344 avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
345 avio_wb32(pb, s->ssrc + 1);
348 avio_write(pb, s->hostname, len);
349 avio_w8(pb, 0); /* END */
351 for (len = (7 + len) % 4; len % 4; len++)
357 len = avio_close_dyn_buf(pb, &buf);
358 if ((len > 0) && buf) {
359 int av_unused result;
360 av_dlog(s->ic, "sending %d bytes of RR\n", len);
361 result = ffurl_write(fd, buf, len);
362 av_dlog(s->ic, "result from ffurl_write: %d\n", result);
368 void ff_rtp_send_punch_packets(URLContext *rtp_handle)
374 /* Send a small RTP packet */
375 if (avio_open_dyn_buf(&pb) < 0)
378 avio_w8(pb, (RTP_VERSION << 6));
379 avio_w8(pb, 0); /* Payload type */
380 avio_wb16(pb, 0); /* Seq */
381 avio_wb32(pb, 0); /* Timestamp */
382 avio_wb32(pb, 0); /* SSRC */
385 len = avio_close_dyn_buf(pb, &buf);
386 if ((len > 0) && buf)
387 ffurl_write(rtp_handle, buf, len);
390 /* Send a minimal RTCP RR */
391 if (avio_open_dyn_buf(&pb) < 0)
394 avio_w8(pb, (RTP_VERSION << 6));
395 avio_w8(pb, RTCP_RR); /* receiver report */
396 avio_wb16(pb, 1); /* length in words - 1 */
397 avio_wb32(pb, 0); /* our own SSRC */
400 len = avio_close_dyn_buf(pb, &buf);
401 if ((len > 0) && buf)
402 ffurl_write(rtp_handle, buf, len);
406 static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
407 uint16_t *missing_mask)
410 uint16_t next_seq = s->seq + 1;
411 RTPPacket *pkt = s->queue;
413 if (!pkt || pkt->seq == next_seq)
417 for (i = 1; i <= 16; i++) {
418 uint16_t missing_seq = next_seq + i;
420 int16_t diff = pkt->seq - missing_seq;
427 if (pkt->seq == missing_seq)
429 *missing_mask |= 1 << (i - 1);
432 *first_missing = next_seq;
436 int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
439 int len, need_keyframe, missing_packets;
443 uint16_t first_missing = 0, missing_mask = 0;
448 need_keyframe = s->handler && s->handler->need_keyframe &&
449 s->handler->need_keyframe(s->dynamic_protocol_context);
450 missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
452 if (!need_keyframe && !missing_packets)
455 /* Send new feedback if enough time has elapsed since the last
456 * feedback packet. */
458 now = av_gettime_relative();
459 if (s->last_feedback_time &&
460 (now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL)
462 s->last_feedback_time = now;
466 else if (avio_open_dyn_buf(&pb) < 0)
470 avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
471 avio_w8(pb, RTCP_PSFB);
472 avio_wb16(pb, 2); /* length in words - 1 */
473 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
474 avio_wb32(pb, s->ssrc + 1);
475 avio_wb32(pb, s->ssrc); // server SSRC
478 if (missing_packets) {
479 avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
480 avio_w8(pb, RTCP_RTPFB);
481 avio_wb16(pb, 3); /* length in words - 1 */
482 avio_wb32(pb, s->ssrc + 1);
483 avio_wb32(pb, s->ssrc); // server SSRC
485 avio_wb16(pb, first_missing);
486 avio_wb16(pb, missing_mask);
492 len = avio_close_dyn_buf(pb, &buf);
493 if (len > 0 && buf) {
494 ffurl_write(fd, buf, len);
501 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
504 RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
505 int payload_type, int queue_size)
509 s = av_mallocz(sizeof(RTPDemuxContext));
512 s->payload_type = payload_type;
513 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
514 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
517 s->queue_size = queue_size;
518 rtp_init_statistics(&s->statistics, 0);
520 switch (st->codec->codec_id) {
521 case AV_CODEC_ID_ADPCM_G722:
522 /* According to RFC 3551, the stream clock rate is 8000
523 * even if the sample rate is 16000. */
524 if (st->codec->sample_rate == 8000)
525 st->codec->sample_rate = 16000;
531 // needed to send back RTCP RR in RTSP sessions
532 gethostname(s->hostname, sizeof(s->hostname));
536 void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
537 RTPDynamicProtocolHandler *handler)
539 s->dynamic_protocol_context = ctx;
540 s->handler = handler;
543 void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
546 if (!ff_srtp_set_crypto(&s->srtp, suite, params))
551 * This was the second switch in rtp_parse packet.
552 * Normalizes time, if required, sets stream_index, etc.
554 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
556 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
557 return; /* Timestamp already set by depacketizer */
558 if (timestamp == RTP_NOTS_VALUE)
561 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
565 /* compute pts from timestamp with received ntp_time */
566 delta_timestamp = timestamp - s->last_rtcp_timestamp;
567 /* convert to the PTS timebase */
568 addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time,
569 s->st->time_base.den,
570 (uint64_t) s->st->time_base.num << 32);
571 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
576 if (!s->base_timestamp)
577 s->base_timestamp = timestamp;
578 /* assume that the difference is INT32_MIN < x < INT32_MAX,
579 * but allow the first timestamp to exceed INT32_MAX */
581 s->unwrapped_timestamp += timestamp;
583 s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
584 s->timestamp = timestamp;
585 pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
589 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
590 const uint8_t *buf, int len)
593 int payload_type, seq, flags = 0;
599 csrc = buf[0] & 0x0f;
601 payload_type = buf[1] & 0x7f;
603 flags |= RTP_FLAG_MARKER;
604 seq = AV_RB16(buf + 2);
605 timestamp = AV_RB32(buf + 4);
606 ssrc = AV_RB32(buf + 8);
607 /* store the ssrc in the RTPDemuxContext */
610 /* NOTE: we can handle only one payload type */
611 if (s->payload_type != payload_type)
615 // only do something with this if all the rtp checks pass...
616 if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
617 av_log(st ? st->codec : NULL, AV_LOG_ERROR,
618 "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
619 payload_type, seq, ((s->seq + 1) & 0xffff));
624 int padding = buf[len - 1];
625 if (len >= 12 + padding)
636 return AVERROR_INVALIDDATA;
638 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
642 /* calculate the header extension length (stored as number
643 * of 32-bit words) */
644 ext = (AV_RB16(buf + 2) + 1) << 2;
648 // skip past RTP header extension
653 if (s->handler && s->handler->parse_packet) {
654 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
655 s->st, pkt, ×tamp, buf, len, seq,
658 if ((rv = av_new_packet(pkt, len)) < 0)
660 memcpy(pkt->data, buf, len);
661 pkt->stream_index = st->index;
663 return AVERROR(EINVAL);
666 // now perform timestamp things....
667 finalize_packet(s, pkt, timestamp);
672 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
675 RTPPacket *next = s->queue->next;
676 av_freep(&s->queue->buf);
685 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
687 uint16_t seq = AV_RB16(buf + 2);
688 RTPPacket **cur = &s->queue, *packet;
690 /* Find the correct place in the queue to insert the packet */
692 int16_t diff = seq - (*cur)->seq;
698 packet = av_mallocz(sizeof(*packet));
701 packet->recvtime = av_gettime_relative();
710 static int has_next_packet(RTPDemuxContext *s)
712 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
715 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
717 return s->queue ? s->queue->recvtime : 0;
720 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
725 if (s->queue_len <= 0)
728 if (!has_next_packet(s))
729 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
730 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
732 /* Parse the first packet in the queue, and dequeue it */
733 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
734 next = s->queue->next;
735 av_freep(&s->queue->buf);
742 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
743 uint8_t **bufptr, int len)
745 uint8_t *buf = bufptr ? *bufptr : NULL;
751 /* If parsing of the previous packet actually returned 0 or an error,
752 * there's nothing more to be parsed from that packet, but we may have
753 * indicated that we can return the next enqueued packet. */
754 if (s->prev_ret <= 0)
755 return rtp_parse_queued_packet(s, pkt);
756 /* return the next packets, if any */
757 if (s->handler && s->handler->parse_packet) {
758 /* timestamp should be overwritten by parse_packet, if not,
759 * the packet is left with pts == AV_NOPTS_VALUE */
760 timestamp = RTP_NOTS_VALUE;
761 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
762 s->st, pkt, ×tamp, NULL, 0, 0,
764 finalize_packet(s, pkt, timestamp);
772 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
774 if (RTP_PT_IS_RTCP(buf[1])) {
775 return rtcp_parse_packet(s, buf, len);
779 int64_t received = av_gettime_relative();
780 uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
782 timestamp = AV_RB32(buf + 4);
783 // Calculate the jitter immediately, before queueing the packet
784 // into the reordering queue.
785 rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
788 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
789 /* First packet, or no reordering */
790 return rtp_parse_packet_internal(s, pkt, buf, len);
792 uint16_t seq = AV_RB16(buf + 2);
793 int16_t diff = seq - s->seq;
795 /* Packet older than the previously emitted one, drop */
796 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
797 "RTP: dropping old packet received too late\n");
799 } else if (diff <= 1) {
801 rv = rtp_parse_packet_internal(s, pkt, buf, len);
804 /* Still missing some packet, enqueue this one. */
805 enqueue_packet(s, buf, len);
807 /* Return the first enqueued packet if the queue is full,
808 * even if we're missing something */
809 if (s->queue_len >= s->queue_size)
810 return rtp_parse_queued_packet(s, pkt);
817 * Parse an RTP or RTCP packet directly sent as a buffer.
818 * @param s RTP parse context.
819 * @param pkt returned packet
820 * @param bufptr pointer to the input buffer or NULL to read the next packets
821 * @param len buffer len
822 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
823 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
825 int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
826 uint8_t **bufptr, int len)
829 if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
831 rv = rtp_parse_one_packet(s, pkt, bufptr, len);
833 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
834 rv = rtp_parse_queued_packet(s, pkt);
835 return rv ? rv : has_next_packet(s);
838 void ff_rtp_parse_close(RTPDemuxContext *s)
840 ff_rtp_reset_packet_queue(s);
841 ff_srtp_free(&s->srtp);
845 int ff_parse_fmtp(AVFormatContext *s,
846 AVStream *stream, PayloadContext *data, const char *p,
847 int (*parse_fmtp)(AVFormatContext *s,
849 PayloadContext *data,
850 char *attr, char *value))
855 int value_size = strlen(p) + 1;
857 if (!(value = av_malloc(value_size))) {
858 av_log(NULL, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
859 return AVERROR(ENOMEM);
862 // remove protocol identifier
863 while (*p && *p == ' ')
865 while (*p && *p != ' ')
866 p++; // eat protocol identifier
867 while (*p && *p == ' ')
868 p++; // strip trailing spaces
870 while (ff_rtsp_next_attr_and_value(&p,
872 value, value_size)) {
873 res = parse_fmtp(s, stream, data, attr, value);
874 if (res < 0 && res != AVERROR_PATCHWELCOME) {
883 int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
888 pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
889 pkt->stream_index = stream_idx;
891 if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) {
892 av_freep(&pkt->data);