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"
33 #include "rtpdec_formats.h"
37 /* TODO: - add RTCP statistics reporting (should be optional).
39 - add support for h263/mpeg4 packetized output : IDEA: send a
40 buffer to 'rtp_write_packet' contains all the packets for ONE
41 frame. Each packet should have a four byte header containing
42 the length in big endian format (same trick as
43 'ffio_open_dyn_packet_buf')
46 static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
47 .enc_name = "X-MP3-draft-00",
48 .codec_type = AVMEDIA_TYPE_AUDIO,
49 .codec_id = AV_CODEC_ID_MP3ADU,
52 static RTPDynamicProtocolHandler speex_dynamic_handler = {
54 .codec_type = AVMEDIA_TYPE_AUDIO,
55 .codec_id = AV_CODEC_ID_SPEEX,
58 /* statistics functions */
59 static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
61 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
63 handler->next= RTPFirstDynamicPayloadHandler;
64 RTPFirstDynamicPayloadHandler= handler;
67 void av_register_rtp_dynamic_payload_handlers(void)
69 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
70 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
71 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
72 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
73 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
74 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
75 ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
76 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
77 ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
78 ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler);
79 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
80 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
81 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
82 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
83 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
84 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
85 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
86 ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
87 ff_register_dynamic_payload_handler(&speex_dynamic_handler);
89 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
90 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
92 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
93 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
94 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
95 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
97 ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
98 ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
99 ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
100 ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
103 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
104 enum AVMediaType codec_type)
106 RTPDynamicProtocolHandler *handler;
107 for (handler = RTPFirstDynamicPayloadHandler;
108 handler; handler = handler->next)
109 if (!av_strcasecmp(name, handler->enc_name) &&
110 codec_type == handler->codec_type)
115 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
116 enum AVMediaType codec_type)
118 RTPDynamicProtocolHandler *handler;
119 for (handler = RTPFirstDynamicPayloadHandler;
120 handler; handler = handler->next)
121 if (handler->static_payload_id && handler->static_payload_id == id &&
122 codec_type == handler->codec_type)
127 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
131 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
135 if (payload_len < 20) {
136 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
137 return AVERROR_INVALIDDATA;
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 = s->last_rtcp_timestamp - s->base_timestamp;
160 #define RTP_SEQ_MOD (1<<16)
163 * called on parse open packet
165 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
167 memset(s, 0, sizeof(RTPStatistics));
168 s->max_seq= base_sequence;
173 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
175 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
180 s->bad_seq= RTP_SEQ_MOD + 1;
182 s->expected_prior= 0;
183 s->received_prior= 0;
189 * returns 1 if we should handle this packet.
191 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
193 uint16_t udelta= seq - s->max_seq;
194 const int MAX_DROPOUT= 3000;
195 const int MAX_MISORDER = 100;
196 const int MIN_SEQUENTIAL = 2;
198 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
201 if(seq==s->max_seq + 1) {
204 if(s->probation==0) {
205 rtp_init_sequence(s, seq);
210 s->probation= MIN_SEQUENTIAL - 1;
213 } else if (udelta < MAX_DROPOUT) {
214 // in order, with permissible gap
215 if(seq < s->max_seq) {
216 //sequence number wrapped; count antother 64k cycles
217 s->cycles += RTP_SEQ_MOD;
220 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
221 // sequence made a large jump...
222 if(seq==s->bad_seq) {
223 // two sequential packets-- assume that the other side restarted without telling us; just resync.
224 rtp_init_sequence(s, seq);
226 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
230 // duplicate or reordered packet...
236 int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
242 RTPStatistics *stats= &s->statistics;
244 uint32_t extended_max;
245 uint32_t expected_interval;
246 uint32_t received_interval;
247 uint32_t lost_interval;
250 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
252 if (!s->rtp_ctx || (count < 1))
255 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
256 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
257 s->octet_count += count;
258 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
260 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
263 s->last_octet_count = s->octet_count;
265 if (avio_open_dyn_buf(&pb) < 0)
269 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
270 avio_w8(pb, RTCP_RR);
271 avio_wb16(pb, 7); /* length in words - 1 */
272 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
273 avio_wb32(pb, s->ssrc + 1);
274 avio_wb32(pb, s->ssrc); // server SSRC
275 // some placeholders we should really fill...
277 extended_max= stats->cycles + stats->max_seq;
278 expected= extended_max - stats->base_seq + 1;
279 lost= expected - stats->received;
280 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
281 expected_interval= expected - stats->expected_prior;
282 stats->expected_prior= expected;
283 received_interval= stats->received - stats->received_prior;
284 stats->received_prior= stats->received;
285 lost_interval= expected_interval - received_interval;
286 if (expected_interval==0 || lost_interval<=0) fraction= 0;
287 else 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)
297 avio_wb32(pb, 0); /* last SR timestamp */
298 avio_wb32(pb, 0); /* delay since last SR */
300 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
301 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
303 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
304 avio_wb32(pb, delay_since_last); /* delay since last SR */
308 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
309 avio_w8(pb, RTCP_SDES);
310 len = strlen(s->hostname);
311 avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
312 avio_wb32(pb, s->ssrc + 1);
315 avio_write(pb, s->hostname, len);
317 for (len = (6 + len) % 4; len % 4; len++) {
322 len = avio_close_dyn_buf(pb, &buf);
323 if ((len > 0) && buf) {
324 int av_unused result;
325 av_dlog(s->ic, "sending %d bytes of RR\n", len);
326 result= ffurl_write(s->rtp_ctx, buf, len);
327 av_dlog(s->ic, "result from ffurl_write: %d\n", result);
333 void ff_rtp_send_punch_packets(URLContext* rtp_handle)
339 /* Send a small RTP packet */
340 if (avio_open_dyn_buf(&pb) < 0)
343 avio_w8(pb, (RTP_VERSION << 6));
344 avio_w8(pb, 0); /* Payload type */
345 avio_wb16(pb, 0); /* Seq */
346 avio_wb32(pb, 0); /* Timestamp */
347 avio_wb32(pb, 0); /* SSRC */
350 len = avio_close_dyn_buf(pb, &buf);
351 if ((len > 0) && buf)
352 ffurl_write(rtp_handle, buf, len);
355 /* Send a minimal RTCP RR */
356 if (avio_open_dyn_buf(&pb) < 0)
359 avio_w8(pb, (RTP_VERSION << 6));
360 avio_w8(pb, RTCP_RR); /* receiver report */
361 avio_wb16(pb, 1); /* length in words - 1 */
362 avio_wb32(pb, 0); /* our own SSRC */
365 len = avio_close_dyn_buf(pb, &buf);
366 if ((len > 0) && buf)
367 ffurl_write(rtp_handle, buf, len);
373 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
374 * MPEG2TS streams to indicate that they should be demuxed inside the
375 * rtp demux (otherwise AV_CODEC_ID_MPEG2TS packets are returned)
377 RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
381 s = av_mallocz(sizeof(RTPDemuxContext));
384 s->payload_type = payload_type;
385 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
386 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
389 s->queue_size = queue_size;
390 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
391 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
392 s->ts = ff_mpegts_parse_open(s->ic);
398 switch(st->codec->codec_id) {
399 case AV_CODEC_ID_MPEG1VIDEO:
400 case AV_CODEC_ID_MPEG2VIDEO:
401 case AV_CODEC_ID_MP2:
402 case AV_CODEC_ID_MP3:
403 case AV_CODEC_ID_MPEG4:
404 case AV_CODEC_ID_H263:
405 case AV_CODEC_ID_H264:
406 st->need_parsing = AVSTREAM_PARSE_FULL;
408 case AV_CODEC_ID_VORBIS:
409 st->need_parsing = AVSTREAM_PARSE_HEADERS;
411 case AV_CODEC_ID_ADPCM_G722:
412 /* According to RFC 3551, the stream clock rate is 8000
413 * even if the sample rate is 16000. */
414 if (st->codec->sample_rate == 8000)
415 st->codec->sample_rate = 16000;
421 // needed to send back RTCP RR in RTSP sessions
423 gethostname(s->hostname, sizeof(s->hostname));
428 ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
429 RTPDynamicProtocolHandler *handler)
431 s->dynamic_protocol_context = ctx;
432 s->parse_packet = handler->parse_packet;
436 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
438 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
440 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
441 return; /* Timestamp already set by depacketizer */
442 if (timestamp == RTP_NOTS_VALUE)
445 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
449 /* compute pts from timestamp with received ntp_time */
450 delta_timestamp = timestamp - s->last_rtcp_timestamp;
451 /* convert to the PTS timebase */
452 addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);
453 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
458 if (!s->base_timestamp)
459 s->base_timestamp = timestamp;
460 /* assume that the difference is INT32_MIN < x < INT32_MAX, but allow the first timestamp to exceed INT32_MAX */
462 s->unwrapped_timestamp += timestamp;
464 s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
465 s->timestamp = timestamp;
466 pkt->pts = s->unwrapped_timestamp + s->range_start_offset - s->base_timestamp;
469 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
470 const uint8_t *buf, int len)
472 unsigned int ssrc, h;
473 int payload_type, seq, ret, flags = 0;
480 payload_type = buf[1] & 0x7f;
482 flags |= RTP_FLAG_MARKER;
483 seq = AV_RB16(buf + 2);
484 timestamp = AV_RB32(buf + 4);
485 ssrc = AV_RB32(buf + 8);
486 /* store the ssrc in the RTPDemuxContext */
489 /* NOTE: we can handle only one payload type */
490 if (s->payload_type != payload_type)
494 // only do something with this if all the rtp checks pass...
495 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
497 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
498 payload_type, seq, ((s->seq + 1) & 0xffff));
503 int padding = buf[len - 1];
504 if (len >= 12 + padding)
512 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
516 /* calculate the header extension length (stored as number
517 * of 32-bit words) */
518 ext = (AV_RB16(buf + 2) + 1) << 2;
522 // skip past RTP header extension
528 /* specific MPEG2TS demux support */
529 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
530 /* The only error that can be returned from ff_mpegts_parse_packet
531 * is "no more data to return from the provided buffer", so return
532 * AVERROR(EAGAIN) for all errors */
534 return AVERROR(EAGAIN);
536 s->read_buf_size = len - ret;
537 memcpy(s->buf, buf + ret, s->read_buf_size);
538 s->read_buf_index = 0;
542 } else if (s->parse_packet) {
543 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
544 s->st, pkt, ×tamp, buf, len, flags);
546 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
547 switch(st->codec->codec_id) {
548 case AV_CODEC_ID_MP2:
549 case AV_CODEC_ID_MP3:
550 /* better than nothing: skip mpeg audio RTP header */
556 av_new_packet(pkt, len);
557 memcpy(pkt->data, buf, len);
559 case AV_CODEC_ID_MPEG1VIDEO:
560 case AV_CODEC_ID_MPEG2VIDEO:
561 /* better than nothing: skip mpeg video RTP header */
574 av_new_packet(pkt, len);
575 memcpy(pkt->data, buf, len);
578 av_new_packet(pkt, len);
579 memcpy(pkt->data, buf, len);
583 pkt->stream_index = st->index;
586 // now perform timestamp things....
587 finalize_packet(s, pkt, timestamp);
592 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
595 RTPPacket *next = s->queue->next;
596 av_free(s->queue->buf);
605 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
607 uint16_t seq = AV_RB16(buf + 2);
608 RTPPacket *cur = s->queue, *prev = NULL, *packet;
610 /* Find the correct place in the queue to insert the packet */
612 int16_t diff = seq - cur->seq;
619 packet = av_mallocz(sizeof(*packet));
622 packet->recvtime = av_gettime();
634 static int has_next_packet(RTPDemuxContext *s)
636 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
639 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
641 return s->queue ? s->queue->recvtime : 0;
644 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
649 if (s->queue_len <= 0)
652 if (!has_next_packet(s))
653 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
654 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
656 /* Parse the first packet in the queue, and dequeue it */
657 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
658 next = s->queue->next;
659 av_free(s->queue->buf);
666 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
667 uint8_t **bufptr, int len)
669 uint8_t* buf = bufptr ? *bufptr : NULL;
675 /* If parsing of the previous packet actually returned 0 or an error,
676 * there's nothing more to be parsed from that packet, but we may have
677 * indicated that we can return the next enqueued packet. */
678 if (s->prev_ret <= 0)
679 return rtp_parse_queued_packet(s, pkt);
680 /* return the next packets, if any */
681 if(s->st && s->parse_packet) {
682 /* timestamp should be overwritten by parse_packet, if not,
683 * the packet is left with pts == AV_NOPTS_VALUE */
684 timestamp = RTP_NOTS_VALUE;
685 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
686 s->st, pkt, ×tamp, NULL, 0, flags);
687 finalize_packet(s, pkt, timestamp);
690 // TODO: Move to a dynamic packet handler (like above)
691 if (s->read_buf_index >= s->read_buf_size)
692 return AVERROR(EAGAIN);
693 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
694 s->read_buf_size - s->read_buf_index);
696 return AVERROR(EAGAIN);
697 s->read_buf_index += ret;
698 if (s->read_buf_index < s->read_buf_size)
708 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
710 if (RTP_PT_IS_RTCP(buf[1])) {
711 return rtcp_parse_packet(s, buf, len);
714 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
715 /* First packet, or no reordering */
716 return rtp_parse_packet_internal(s, pkt, buf, len);
718 uint16_t seq = AV_RB16(buf + 2);
719 int16_t diff = seq - s->seq;
721 /* Packet older than the previously emitted one, drop */
722 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
723 "RTP: dropping old packet received too late\n");
725 } else if (diff <= 1) {
727 rv = rtp_parse_packet_internal(s, pkt, buf, len);
730 /* Still missing some packet, enqueue this one. */
731 enqueue_packet(s, buf, len);
733 /* Return the first enqueued packet if the queue is full,
734 * even if we're missing something */
735 if (s->queue_len >= s->queue_size)
736 return rtp_parse_queued_packet(s, pkt);
743 * Parse an RTP or RTCP packet directly sent as a buffer.
744 * @param s RTP parse context.
745 * @param pkt returned packet
746 * @param bufptr pointer to the input buffer or NULL to read the next packets
747 * @param len buffer len
748 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
749 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
751 int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
752 uint8_t **bufptr, int len)
754 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
756 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
757 rv = rtp_parse_queued_packet(s, pkt);
758 return rv ? rv : has_next_packet(s);
761 void ff_rtp_parse_close(RTPDemuxContext *s)
763 ff_rtp_reset_packet_queue(s);
764 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
765 ff_mpegts_parse_close(s->ts);
770 int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
771 int (*parse_fmtp)(AVStream *stream,
772 PayloadContext *data,
773 char *attr, char *value))
778 int value_size = strlen(p) + 1;
780 if (!(value = av_malloc(value_size))) {
781 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
782 return AVERROR(ENOMEM);
785 // remove protocol identifier
786 while (*p && *p == ' ') p++; // strip spaces
787 while (*p && *p != ' ') p++; // eat protocol identifier
788 while (*p && *p == ' ') p++; // strip trailing spaces
790 while (ff_rtsp_next_attr_and_value(&p,
792 value, value_size)) {
794 res = parse_fmtp(stream, data, attr, value);
795 if (res < 0 && res != AVERROR_PATCHWELCOME) {