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"
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 static RTPDynamicProtocolHandler opus_dynamic_handler = {
60 .codec_type = AVMEDIA_TYPE_AUDIO,
61 .codec_id = AV_CODEC_ID_OPUS,
64 /* statistics functions */
65 static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
67 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
69 handler->next= RTPFirstDynamicPayloadHandler;
70 RTPFirstDynamicPayloadHandler= handler;
73 void av_register_rtp_dynamic_payload_handlers(void)
75 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
76 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
77 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
78 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
79 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
80 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
81 ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
82 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
83 ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
84 ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler);
85 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
86 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
87 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
88 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
89 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
90 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
91 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
92 ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
93 ff_register_dynamic_payload_handler(&speex_dynamic_handler);
94 ff_register_dynamic_payload_handler(&opus_dynamic_handler);
96 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
97 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_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);
104 ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
105 ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
106 ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
107 ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
110 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
111 enum AVMediaType codec_type)
113 RTPDynamicProtocolHandler *handler;
114 for (handler = RTPFirstDynamicPayloadHandler;
115 handler; handler = handler->next)
116 if (!av_strcasecmp(name, handler->enc_name) &&
117 codec_type == handler->codec_type)
122 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
123 enum AVMediaType codec_type)
125 RTPDynamicProtocolHandler *handler;
126 for (handler = RTPFirstDynamicPayloadHandler;
127 handler; handler = handler->next)
128 if (handler->static_payload_id && handler->static_payload_id == id &&
129 codec_type == handler->codec_type)
134 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
138 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
142 if (payload_len < 20) {
143 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
144 return AVERROR_INVALIDDATA;
147 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
148 s->last_rtcp_timestamp = AV_RB32(buf + 16);
149 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
150 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
151 if (!s->base_timestamp)
152 s->base_timestamp = s->last_rtcp_timestamp;
153 s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
167 #define RTP_SEQ_MOD (1<<16)
169 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
171 memset(s, 0, sizeof(RTPStatistics));
172 s->max_seq = base_sequence;
177 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
179 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
183 s->base_seq = seq - 1;
184 s->bad_seq = RTP_SEQ_MOD + 1;
186 s->expected_prior = 0;
187 s->received_prior = 0;
193 * returns 1 if we should handle this packet.
195 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
197 uint16_t udelta = seq - s->max_seq;
198 const int MAX_DROPOUT = 3000;
199 const int MAX_MISORDER = 100;
200 const int MIN_SEQUENTIAL = 2;
202 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
204 if (seq == s->max_seq + 1) {
207 if (s->probation == 0) {
208 rtp_init_sequence(s, seq);
213 s->probation = MIN_SEQUENTIAL - 1;
216 } else if (udelta < MAX_DROPOUT) {
217 // in order, with permissible gap
218 if (seq < s->max_seq) {
219 // sequence number wrapped; count another 64k cycles
220 s->cycles += RTP_SEQ_MOD;
223 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
224 // sequence made a large jump...
225 if (seq == s->bad_seq) {
226 // two sequential packets-- assume that the other side restarted without telling us; just resync.
227 rtp_init_sequence(s, seq);
229 s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
233 // duplicate or reordered packet...
239 int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
245 RTPStatistics *stats = &s->statistics;
247 uint32_t extended_max;
248 uint32_t expected_interval;
249 uint32_t received_interval;
250 uint32_t lost_interval;
253 uint64_t ntp_time = s->last_rtcp_ntp_time; // TODO: Get local ntp time?
255 if (!s->rtp_ctx || (count < 1))
258 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
259 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
260 s->octet_count += count;
261 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
263 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
266 s->last_octet_count = s->octet_count;
268 if (avio_open_dyn_buf(&pb) < 0)
272 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
273 avio_w8(pb, RTCP_RR);
274 avio_wb16(pb, 7); /* length in words - 1 */
275 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
276 avio_wb32(pb, s->ssrc + 1);
277 avio_wb32(pb, s->ssrc); // server SSRC
278 // some placeholders we should really fill...
280 extended_max = stats->cycles + stats->max_seq;
281 expected = extended_max - stats->base_seq + 1;
282 lost = expected - stats->received;
283 lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
284 expected_interval = expected - stats->expected_prior;
285 stats->expected_prior = expected;
286 received_interval = stats->received - stats->received_prior;
287 stats->received_prior = stats->received;
288 lost_interval = expected_interval - received_interval;
289 if (expected_interval == 0 || lost_interval <= 0)
292 fraction = (lost_interval << 8) / expected_interval;
294 fraction = (fraction << 24) | lost;
296 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
297 avio_wb32(pb, extended_max); /* max sequence received */
298 avio_wb32(pb, stats->jitter >> 4); /* jitter */
300 if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) {
301 avio_wb32(pb, 0); /* last SR timestamp */
302 avio_wb32(pb, 0); /* delay since last SR */
304 uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
305 uint32_t delay_since_last = ntp_time - s->last_rtcp_ntp_time;
307 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
308 avio_wb32(pb, delay_since_last); /* delay since last SR */
312 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
313 avio_w8(pb, RTCP_SDES);
314 len = strlen(s->hostname);
315 avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
316 avio_wb32(pb, s->ssrc + 1);
319 avio_write(pb, s->hostname, len);
321 for (len = (6 + len) % 4; len % 4; len++) {
326 len = avio_close_dyn_buf(pb, &buf);
327 if ((len > 0) && buf) {
328 int av_unused result;
329 av_dlog(s->ic, "sending %d bytes of RR\n", len);
330 result= ffurl_write(s->rtp_ctx, buf, len);
331 av_dlog(s->ic, "result from ffurl_write: %d\n", result);
337 void ff_rtp_send_punch_packets(URLContext* rtp_handle)
343 /* Send a small RTP packet */
344 if (avio_open_dyn_buf(&pb) < 0)
347 avio_w8(pb, (RTP_VERSION << 6));
348 avio_w8(pb, 0); /* Payload type */
349 avio_wb16(pb, 0); /* Seq */
350 avio_wb32(pb, 0); /* Timestamp */
351 avio_wb32(pb, 0); /* SSRC */
354 len = avio_close_dyn_buf(pb, &buf);
355 if ((len > 0) && buf)
356 ffurl_write(rtp_handle, buf, len);
359 /* Send a minimal RTCP RR */
360 if (avio_open_dyn_buf(&pb) < 0)
363 avio_w8(pb, (RTP_VERSION << 6));
364 avio_w8(pb, RTCP_RR); /* receiver report */
365 avio_wb16(pb, 1); /* length in words - 1 */
366 avio_wb32(pb, 0); /* our own SSRC */
369 len = avio_close_dyn_buf(pb, &buf);
370 if ((len > 0) && buf)
371 ffurl_write(rtp_handle, buf, len);
377 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
378 * MPEG2TS streams to indicate that they should be demuxed inside the
379 * rtp demux (otherwise AV_CODEC_ID_MPEG2TS packets are returned)
381 RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
385 s = av_mallocz(sizeof(RTPDemuxContext));
388 s->payload_type = payload_type;
389 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
390 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
393 s->queue_size = queue_size;
394 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
395 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
396 s->ts = ff_mpegts_parse_open(s->ic);
402 switch(st->codec->codec_id) {
403 case AV_CODEC_ID_MPEG1VIDEO:
404 case AV_CODEC_ID_MPEG2VIDEO:
405 case AV_CODEC_ID_MP2:
406 case AV_CODEC_ID_MP3:
407 case AV_CODEC_ID_MPEG4:
408 case AV_CODEC_ID_H263:
409 case AV_CODEC_ID_H264:
410 st->need_parsing = AVSTREAM_PARSE_FULL;
412 case AV_CODEC_ID_VORBIS:
413 st->need_parsing = AVSTREAM_PARSE_HEADERS;
415 case AV_CODEC_ID_ADPCM_G722:
416 /* According to RFC 3551, the stream clock rate is 8000
417 * even if the sample rate is 16000. */
418 if (st->codec->sample_rate == 8000)
419 st->codec->sample_rate = 16000;
425 // needed to send back RTCP RR in RTSP sessions
427 gethostname(s->hostname, sizeof(s->hostname));
431 void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
432 RTPDynamicProtocolHandler *handler)
434 s->dynamic_protocol_context = ctx;
435 s->parse_packet = handler->parse_packet;
439 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
441 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
443 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
444 return; /* Timestamp already set by depacketizer */
445 if (timestamp == RTP_NOTS_VALUE)
448 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
452 /* compute pts from timestamp with received ntp_time */
453 delta_timestamp = timestamp - s->last_rtcp_timestamp;
454 /* convert to the PTS timebase */
455 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);
456 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
461 if (!s->base_timestamp)
462 s->base_timestamp = timestamp;
463 /* assume that the difference is INT32_MIN < x < INT32_MAX, but allow the first timestamp to exceed INT32_MAX */
465 s->unwrapped_timestamp += timestamp;
467 s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
468 s->timestamp = timestamp;
469 pkt->pts = s->unwrapped_timestamp + s->range_start_offset - s->base_timestamp;
472 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
473 const uint8_t *buf, int len)
475 unsigned int ssrc, h;
476 int payload_type, seq, ret, flags = 0;
483 payload_type = buf[1] & 0x7f;
485 flags |= RTP_FLAG_MARKER;
486 seq = AV_RB16(buf + 2);
487 timestamp = AV_RB32(buf + 4);
488 ssrc = AV_RB32(buf + 8);
489 /* store the ssrc in the RTPDemuxContext */
492 /* NOTE: we can handle only one payload type */
493 if (s->payload_type != payload_type)
497 // only do something with this if all the rtp checks pass...
498 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
500 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
501 payload_type, seq, ((s->seq + 1) & 0xffff));
506 int padding = buf[len - 1];
507 if (len >= 12 + padding)
515 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
519 /* calculate the header extension length (stored as number
520 * of 32-bit words) */
521 ext = (AV_RB16(buf + 2) + 1) << 2;
525 // skip past RTP header extension
531 /* specific MPEG2TS demux support */
532 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
533 /* The only error that can be returned from ff_mpegts_parse_packet
534 * is "no more data to return from the provided buffer", so return
535 * AVERROR(EAGAIN) for all errors */
537 return AVERROR(EAGAIN);
539 s->read_buf_size = len - ret;
540 memcpy(s->buf, buf + ret, s->read_buf_size);
541 s->read_buf_index = 0;
545 } else if (s->parse_packet) {
546 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
547 s->st, pkt, ×tamp, buf, len, flags);
549 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
550 switch(st->codec->codec_id) {
551 case AV_CODEC_ID_MP2:
552 case AV_CODEC_ID_MP3:
553 /* better than nothing: skip mpeg audio RTP header */
559 if (av_new_packet(pkt, len) < 0)
560 return AVERROR(ENOMEM);
561 memcpy(pkt->data, buf, len);
563 case AV_CODEC_ID_MPEG1VIDEO:
564 case AV_CODEC_ID_MPEG2VIDEO:
565 /* better than nothing: skip mpeg video RTP header */
578 if (av_new_packet(pkt, len) < 0)
579 return AVERROR(ENOMEM);
580 memcpy(pkt->data, buf, len);
583 if (av_new_packet(pkt, len) < 0)
584 return AVERROR(ENOMEM);
585 memcpy(pkt->data, buf, len);
589 pkt->stream_index = st->index;
592 // now perform timestamp things....
593 finalize_packet(s, pkt, timestamp);
598 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
601 RTPPacket *next = s->queue->next;
602 av_free(s->queue->buf);
611 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
613 uint16_t seq = AV_RB16(buf + 2);
614 RTPPacket *cur = s->queue, *prev = NULL, *packet;
616 /* Find the correct place in the queue to insert the packet */
618 int16_t diff = seq - cur->seq;
625 packet = av_mallocz(sizeof(*packet));
628 packet->recvtime = av_gettime();
640 static int has_next_packet(RTPDemuxContext *s)
642 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
645 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
647 return s->queue ? s->queue->recvtime : 0;
650 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
655 if (s->queue_len <= 0)
658 if (!has_next_packet(s))
659 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
660 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
662 /* Parse the first packet in the queue, and dequeue it */
663 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
664 next = s->queue->next;
665 av_free(s->queue->buf);
672 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
673 uint8_t **bufptr, int len)
675 uint8_t* buf = bufptr ? *bufptr : NULL;
681 /* If parsing of the previous packet actually returned 0 or an error,
682 * there's nothing more to be parsed from that packet, but we may have
683 * indicated that we can return the next enqueued packet. */
684 if (s->prev_ret <= 0)
685 return rtp_parse_queued_packet(s, pkt);
686 /* return the next packets, if any */
687 if(s->st && s->parse_packet) {
688 /* timestamp should be overwritten by parse_packet, if not,
689 * the packet is left with pts == AV_NOPTS_VALUE */
690 timestamp = RTP_NOTS_VALUE;
691 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
692 s->st, pkt, ×tamp, NULL, 0, flags);
693 finalize_packet(s, pkt, timestamp);
696 // TODO: Move to a dynamic packet handler (like above)
697 if (s->read_buf_index >= s->read_buf_size)
698 return AVERROR(EAGAIN);
699 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
700 s->read_buf_size - s->read_buf_index);
702 return AVERROR(EAGAIN);
703 s->read_buf_index += ret;
704 if (s->read_buf_index < s->read_buf_size)
714 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
716 if (RTP_PT_IS_RTCP(buf[1])) {
717 return rtcp_parse_packet(s, buf, len);
720 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
721 /* First packet, or no reordering */
722 return rtp_parse_packet_internal(s, pkt, buf, len);
724 uint16_t seq = AV_RB16(buf + 2);
725 int16_t diff = seq - s->seq;
727 /* Packet older than the previously emitted one, drop */
728 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
729 "RTP: dropping old packet received too late\n");
731 } else if (diff <= 1) {
733 rv = rtp_parse_packet_internal(s, pkt, buf, len);
736 /* Still missing some packet, enqueue this one. */
737 enqueue_packet(s, buf, len);
739 /* Return the first enqueued packet if the queue is full,
740 * even if we're missing something */
741 if (s->queue_len >= s->queue_size)
742 return rtp_parse_queued_packet(s, pkt);
749 * Parse an RTP or RTCP packet directly sent as a buffer.
750 * @param s RTP parse context.
751 * @param pkt returned packet
752 * @param bufptr pointer to the input buffer or NULL to read the next packets
753 * @param len buffer len
754 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
755 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
757 int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
758 uint8_t **bufptr, int len)
760 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
762 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
763 rv = rtp_parse_queued_packet(s, pkt);
764 return rv ? rv : has_next_packet(s);
767 void ff_rtp_parse_close(RTPDemuxContext *s)
769 ff_rtp_reset_packet_queue(s);
770 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
771 ff_mpegts_parse_close(s->ts);
776 int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
777 int (*parse_fmtp)(AVStream *stream,
778 PayloadContext *data,
779 char *attr, char *value))
784 int value_size = strlen(p) + 1;
786 if (!(value = av_malloc(value_size))) {
787 av_log(NULL, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
788 return AVERROR(ENOMEM);
791 // remove protocol identifier
792 while (*p && *p == ' ') p++; // strip spaces
793 while (*p && *p != ' ') p++; // eat protocol identifier
794 while (*p && *p == ' ') p++; // strip trailing spaces
796 while (ff_rtsp_next_attr_and_value(&p,
798 value, value_size)) {
800 res = parse_fmtp(stream, data, attr, value);
801 if (res < 0 && res != AVERROR_PATCHWELCOME) {