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 static RTPDynamicProtocolHandler *rtp_first_dynamic_payload_handler = NULL;
55 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
57 handler->next = rtp_first_dynamic_payload_handler;
58 rtp_first_dynamic_payload_handler = handler;
61 void ff_register_rtp_dynamic_payload_handlers(void)
63 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
64 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
65 ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
66 ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
67 ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
68 ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
69 ff_register_dynamic_payload_handler(&ff_h261_dynamic_handler);
70 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
71 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
72 ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
73 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
74 ff_register_dynamic_payload_handler(&ff_hevc_dynamic_handler);
75 ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
76 ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler);
77 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
78 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
79 ff_register_dynamic_payload_handler(&ff_mpeg_audio_dynamic_handler);
80 ff_register_dynamic_payload_handler(&ff_mpeg_video_dynamic_handler);
81 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
82 ff_register_dynamic_payload_handler(&ff_mpegts_dynamic_handler);
83 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
84 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
85 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
86 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
87 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
88 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
89 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
90 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
91 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
92 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
93 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
94 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
95 ff_register_dynamic_payload_handler(&opus_dynamic_handler);
96 ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
97 ff_register_dynamic_payload_handler(&speex_dynamic_handler);
100 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
101 enum AVMediaType codec_type)
103 RTPDynamicProtocolHandler *handler;
104 for (handler = rtp_first_dynamic_payload_handler;
105 handler; handler = handler->next)
106 if (!av_strcasecmp(name, handler->enc_name) &&
107 codec_type == handler->codec_type)
112 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
113 enum AVMediaType codec_type)
115 RTPDynamicProtocolHandler *handler;
116 for (handler = rtp_first_dynamic_payload_handler;
117 handler; handler = handler->next)
118 if (handler->static_payload_id && handler->static_payload_id == id &&
119 codec_type == handler->codec_type)
124 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
129 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
133 if (payload_len < 20) {
134 av_log(NULL, AV_LOG_ERROR,
135 "Invalid length for RTCP SR packet\n");
136 return AVERROR_INVALIDDATA;
139 s->last_rtcp_reception_time = av_gettime_relative();
140 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
141 s->last_rtcp_timestamp = AV_RB32(buf + 16);
142 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
143 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
144 if (!s->base_timestamp)
145 s->base_timestamp = s->last_rtcp_timestamp;
146 s->rtcp_ts_offset = (int32_t)(s->last_rtcp_timestamp - s->base_timestamp);
160 #define RTP_SEQ_MOD (1 << 16)
162 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
164 memset(s, 0, sizeof(RTPStatistics));
165 s->max_seq = base_sequence;
170 * Called whenever there is a large jump in sequence numbers,
171 * or when they get out of probation...
173 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
177 s->base_seq = seq - 1;
178 s->bad_seq = RTP_SEQ_MOD + 1;
180 s->expected_prior = 0;
181 s->received_prior = 0;
186 /* Returns 1 if we should handle this packet. */
187 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
189 uint16_t udelta = seq - s->max_seq;
190 const int MAX_DROPOUT = 3000;
191 const int MAX_MISORDER = 100;
192 const int MIN_SEQUENTIAL = 2;
194 /* source not valid until MIN_SEQUENTIAL packets with sequence
195 * seq. numbers have been received */
197 if (seq == s->max_seq + 1) {
200 if (s->probation == 0) {
201 rtp_init_sequence(s, seq);
206 s->probation = MIN_SEQUENTIAL - 1;
209 } else if (udelta < MAX_DROPOUT) {
210 // in order, with permissible gap
211 if (seq < s->max_seq) {
212 // sequence number wrapped; count another 64k cycles
213 s->cycles += RTP_SEQ_MOD;
216 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
217 // sequence made a large jump...
218 if (seq == s->bad_seq) {
219 /* two sequential packets -- assume that the other side
220 * restarted without telling us; just resync. */
221 rtp_init_sequence(s, seq);
223 s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
227 // duplicate or reordered packet...
233 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp,
234 uint32_t arrival_timestamp)
236 // Most of this is pretty straight from RFC 3550 appendix A.8
237 uint32_t transit = arrival_timestamp - sent_timestamp;
238 uint32_t prev_transit = s->transit;
239 int32_t d = transit - prev_transit;
240 // Doing the FFABS() call directly on the "transit - prev_transit"
241 // expression doesn't work, since it's an unsigned expression. Doing the
242 // transit calculation in unsigned is desired though, since it most
243 // probably will need to wrap around.
245 s->transit = transit;
248 s->jitter += d - (int32_t) ((s->jitter + 8) >> 4);
251 int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
252 AVIOContext *avio, int count)
258 RTPStatistics *stats = &s->statistics;
260 uint32_t extended_max;
261 uint32_t expected_interval;
262 uint32_t received_interval;
263 int32_t lost_interval;
267 if ((!fd && !avio) || (count < 1))
270 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
271 /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
272 s->octet_count += count;
273 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
275 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
278 s->last_octet_count = s->octet_count;
282 else if (avio_open_dyn_buf(&pb) < 0)
286 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
287 avio_w8(pb, RTCP_RR);
288 avio_wb16(pb, 7); /* length in words - 1 */
289 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
290 avio_wb32(pb, s->ssrc + 1);
291 avio_wb32(pb, s->ssrc); // server SSRC
292 // some placeholders we should really fill...
294 extended_max = stats->cycles + stats->max_seq;
295 expected = extended_max - stats->base_seq;
296 lost = expected - stats->received;
297 lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
298 expected_interval = expected - stats->expected_prior;
299 stats->expected_prior = expected;
300 received_interval = stats->received - stats->received_prior;
301 stats->received_prior = stats->received;
302 lost_interval = expected_interval - received_interval;
303 if (expected_interval == 0 || lost_interval <= 0)
306 fraction = (lost_interval << 8) / expected_interval;
308 fraction = (fraction << 24) | lost;
310 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
311 avio_wb32(pb, extended_max); /* max sequence received */
312 avio_wb32(pb, stats->jitter >> 4); /* jitter */
314 if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) {
315 avio_wb32(pb, 0); /* last SR timestamp */
316 avio_wb32(pb, 0); /* delay since last SR */
318 uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
319 uint32_t delay_since_last = av_rescale(av_gettime_relative() - s->last_rtcp_reception_time,
320 65536, AV_TIME_BASE);
322 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
323 avio_wb32(pb, delay_since_last); /* delay since last SR */
327 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
328 avio_w8(pb, RTCP_SDES);
329 len = strlen(s->hostname);
330 avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
331 avio_wb32(pb, s->ssrc + 1);
334 avio_write(pb, s->hostname, len);
335 avio_w8(pb, 0); /* END */
337 for (len = (7 + len) % 4; len % 4; len++)
343 len = avio_close_dyn_buf(pb, &buf);
344 if ((len > 0) && buf) {
345 int av_unused result;
346 av_dlog(s->ic, "sending %d bytes of RR\n", len);
347 result = ffurl_write(fd, buf, len);
348 av_dlog(s->ic, "result from ffurl_write: %d\n", result);
354 void ff_rtp_send_punch_packets(URLContext *rtp_handle)
360 /* Send a small RTP packet */
361 if (avio_open_dyn_buf(&pb) < 0)
364 avio_w8(pb, (RTP_VERSION << 6));
365 avio_w8(pb, 0); /* Payload type */
366 avio_wb16(pb, 0); /* Seq */
367 avio_wb32(pb, 0); /* Timestamp */
368 avio_wb32(pb, 0); /* SSRC */
371 len = avio_close_dyn_buf(pb, &buf);
372 if ((len > 0) && buf)
373 ffurl_write(rtp_handle, buf, len);
376 /* Send a minimal RTCP RR */
377 if (avio_open_dyn_buf(&pb) < 0)
380 avio_w8(pb, (RTP_VERSION << 6));
381 avio_w8(pb, RTCP_RR); /* receiver report */
382 avio_wb16(pb, 1); /* length in words - 1 */
383 avio_wb32(pb, 0); /* our own SSRC */
386 len = avio_close_dyn_buf(pb, &buf);
387 if ((len > 0) && buf)
388 ffurl_write(rtp_handle, buf, len);
392 static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
393 uint16_t *missing_mask)
396 uint16_t next_seq = s->seq + 1;
397 RTPPacket *pkt = s->queue;
399 if (!pkt || pkt->seq == next_seq)
403 for (i = 1; i <= 16; i++) {
404 uint16_t missing_seq = next_seq + i;
406 int16_t diff = pkt->seq - missing_seq;
413 if (pkt->seq == missing_seq)
415 *missing_mask |= 1 << (i - 1);
418 *first_missing = next_seq;
422 int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
425 int len, need_keyframe, missing_packets;
429 uint16_t first_missing = 0, missing_mask = 0;
434 need_keyframe = s->handler && s->handler->need_keyframe &&
435 s->handler->need_keyframe(s->dynamic_protocol_context);
436 missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
438 if (!need_keyframe && !missing_packets)
441 /* Send new feedback if enough time has elapsed since the last
442 * feedback packet. */
444 now = av_gettime_relative();
445 if (s->last_feedback_time &&
446 (now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL)
448 s->last_feedback_time = now;
452 else if (avio_open_dyn_buf(&pb) < 0)
456 avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
457 avio_w8(pb, RTCP_PSFB);
458 avio_wb16(pb, 2); /* length in words - 1 */
459 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
460 avio_wb32(pb, s->ssrc + 1);
461 avio_wb32(pb, s->ssrc); // server SSRC
464 if (missing_packets) {
465 avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
466 avio_w8(pb, RTCP_RTPFB);
467 avio_wb16(pb, 3); /* length in words - 1 */
468 avio_wb32(pb, s->ssrc + 1);
469 avio_wb32(pb, s->ssrc); // server SSRC
471 avio_wb16(pb, first_missing);
472 avio_wb16(pb, missing_mask);
478 len = avio_close_dyn_buf(pb, &buf);
479 if (len > 0 && buf) {
480 ffurl_write(fd, buf, len);
487 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
490 RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
491 int payload_type, int queue_size)
495 s = av_mallocz(sizeof(RTPDemuxContext));
498 s->payload_type = payload_type;
499 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
500 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
503 s->queue_size = queue_size;
504 rtp_init_statistics(&s->statistics, 0);
506 switch (st->codec->codec_id) {
507 case AV_CODEC_ID_ADPCM_G722:
508 /* According to RFC 3551, the stream clock rate is 8000
509 * even if the sample rate is 16000. */
510 if (st->codec->sample_rate == 8000)
511 st->codec->sample_rate = 16000;
517 // needed to send back RTCP RR in RTSP sessions
518 gethostname(s->hostname, sizeof(s->hostname));
522 void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
523 RTPDynamicProtocolHandler *handler)
525 s->dynamic_protocol_context = ctx;
526 s->handler = handler;
529 void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
532 if (!ff_srtp_set_crypto(&s->srtp, suite, params))
537 * This was the second switch in rtp_parse packet.
538 * Normalizes time, if required, sets stream_index, etc.
540 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
542 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
543 return; /* Timestamp already set by depacketizer */
544 if (timestamp == RTP_NOTS_VALUE)
547 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
551 /* compute pts from timestamp with received ntp_time */
552 delta_timestamp = timestamp - s->last_rtcp_timestamp;
553 /* convert to the PTS timebase */
554 addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time,
555 s->st->time_base.den,
556 (uint64_t) s->st->time_base.num << 32);
557 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
562 if (!s->base_timestamp)
563 s->base_timestamp = timestamp;
564 /* assume that the difference is INT32_MIN < x < INT32_MAX,
565 * but allow the first timestamp to exceed INT32_MAX */
567 s->unwrapped_timestamp += timestamp;
569 s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
570 s->timestamp = timestamp;
571 pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
575 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
576 const uint8_t *buf, int len)
579 int payload_type, seq, flags = 0;
585 csrc = buf[0] & 0x0f;
587 payload_type = buf[1] & 0x7f;
589 flags |= RTP_FLAG_MARKER;
590 seq = AV_RB16(buf + 2);
591 timestamp = AV_RB32(buf + 4);
592 ssrc = AV_RB32(buf + 8);
593 /* store the ssrc in the RTPDemuxContext */
596 /* NOTE: we can handle only one payload type */
597 if (s->payload_type != payload_type)
601 // only do something with this if all the rtp checks pass...
602 if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
603 av_log(st ? st->codec : NULL, AV_LOG_ERROR,
604 "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
605 payload_type, seq, ((s->seq + 1) & 0xffff));
610 int padding = buf[len - 1];
611 if (len >= 12 + padding)
622 return AVERROR_INVALIDDATA;
624 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
628 /* calculate the header extension length (stored as number
629 * of 32-bit words) */
630 ext = (AV_RB16(buf + 2) + 1) << 2;
634 // skip past RTP header extension
639 if (s->handler && s->handler->parse_packet) {
640 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
641 s->st, pkt, ×tamp, buf, len, seq,
644 if ((rv = av_new_packet(pkt, len)) < 0)
646 memcpy(pkt->data, buf, len);
647 pkt->stream_index = st->index;
649 return AVERROR(EINVAL);
652 // now perform timestamp things....
653 finalize_packet(s, pkt, timestamp);
658 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
661 RTPPacket *next = s->queue->next;
662 av_free(s->queue->buf);
671 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
673 uint16_t seq = AV_RB16(buf + 2);
674 RTPPacket **cur = &s->queue, *packet;
676 /* Find the correct place in the queue to insert the packet */
678 int16_t diff = seq - (*cur)->seq;
684 packet = av_mallocz(sizeof(*packet));
687 packet->recvtime = av_gettime_relative();
696 static int has_next_packet(RTPDemuxContext *s)
698 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
701 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
703 return s->queue ? s->queue->recvtime : 0;
706 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
711 if (s->queue_len <= 0)
714 if (!has_next_packet(s))
715 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
716 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
718 /* Parse the first packet in the queue, and dequeue it */
719 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
720 next = s->queue->next;
721 av_free(s->queue->buf);
728 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
729 uint8_t **bufptr, int len)
731 uint8_t *buf = bufptr ? *bufptr : NULL;
737 /* If parsing of the previous packet actually returned 0 or an error,
738 * there's nothing more to be parsed from that packet, but we may have
739 * indicated that we can return the next enqueued packet. */
740 if (s->prev_ret <= 0)
741 return rtp_parse_queued_packet(s, pkt);
742 /* return the next packets, if any */
743 if (s->handler && s->handler->parse_packet) {
744 /* timestamp should be overwritten by parse_packet, if not,
745 * the packet is left with pts == AV_NOPTS_VALUE */
746 timestamp = RTP_NOTS_VALUE;
747 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
748 s->st, pkt, ×tamp, NULL, 0, 0,
750 finalize_packet(s, pkt, timestamp);
758 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
760 if (RTP_PT_IS_RTCP(buf[1])) {
761 return rtcp_parse_packet(s, buf, len);
765 int64_t received = av_gettime_relative();
766 uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
768 timestamp = AV_RB32(buf + 4);
769 // Calculate the jitter immediately, before queueing the packet
770 // into the reordering queue.
771 rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
774 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
775 /* First packet, or no reordering */
776 return rtp_parse_packet_internal(s, pkt, buf, len);
778 uint16_t seq = AV_RB16(buf + 2);
779 int16_t diff = seq - s->seq;
781 /* Packet older than the previously emitted one, drop */
782 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
783 "RTP: dropping old packet received too late\n");
785 } else if (diff <= 1) {
787 rv = rtp_parse_packet_internal(s, pkt, buf, len);
790 /* Still missing some packet, enqueue this one. */
791 enqueue_packet(s, buf, len);
793 /* Return the first enqueued packet if the queue is full,
794 * even if we're missing something */
795 if (s->queue_len >= s->queue_size)
796 return rtp_parse_queued_packet(s, pkt);
803 * Parse an RTP or RTCP packet directly sent as a buffer.
804 * @param s RTP parse context.
805 * @param pkt returned packet
806 * @param bufptr pointer to the input buffer or NULL to read the next packets
807 * @param len buffer len
808 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
809 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
811 int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
812 uint8_t **bufptr, int len)
815 if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
817 rv = rtp_parse_one_packet(s, pkt, bufptr, len);
819 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
820 rv = rtp_parse_queued_packet(s, pkt);
821 return rv ? rv : has_next_packet(s);
824 void ff_rtp_parse_close(RTPDemuxContext *s)
826 ff_rtp_reset_packet_queue(s);
827 ff_srtp_free(&s->srtp);
831 int ff_parse_fmtp(AVFormatContext *s,
832 AVStream *stream, PayloadContext *data, const char *p,
833 int (*parse_fmtp)(AVFormatContext *s,
835 PayloadContext *data,
836 char *attr, char *value))
841 int value_size = strlen(p) + 1;
843 if (!(value = av_malloc(value_size))) {
844 av_log(NULL, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
845 return AVERROR(ENOMEM);
848 // remove protocol identifier
849 while (*p && *p == ' ')
851 while (*p && *p != ' ')
852 p++; // eat protocol identifier
853 while (*p && *p == ' ')
854 p++; // strip trailing spaces
856 while (ff_rtsp_next_attr_and_value(&p,
858 value, value_size)) {
859 res = parse_fmtp(s, stream, data, attr, value);
860 if (res < 0 && res != AVERROR_PATCHWELCOME) {
869 int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
874 pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
875 pkt->stream_index = stream_idx;
877 if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) {
878 av_freep(&pkt->data);