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)
170 * called on parse open packet
172 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
174 memset(s, 0, sizeof(RTPStatistics));
175 s->max_seq= base_sequence;
180 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
182 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
187 s->bad_seq= RTP_SEQ_MOD + 1;
189 s->expected_prior= 0;
190 s->received_prior= 0;
196 * returns 1 if we should handle this packet.
198 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
200 uint16_t udelta= seq - s->max_seq;
201 const int MAX_DROPOUT= 3000;
202 const int MAX_MISORDER = 100;
203 const int MIN_SEQUENTIAL = 2;
205 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
208 if(seq==s->max_seq + 1) {
211 if(s->probation==0) {
212 rtp_init_sequence(s, seq);
217 s->probation= MIN_SEQUENTIAL - 1;
220 } else if (udelta < MAX_DROPOUT) {
221 // in order, with permissible gap
222 if(seq < s->max_seq) {
223 //sequence number wrapped; count antother 64k cycles
224 s->cycles += RTP_SEQ_MOD;
227 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
228 // sequence made a large jump...
229 if(seq==s->bad_seq) {
230 // two sequential packets-- assume that the other side restarted without telling us; just resync.
231 rtp_init_sequence(s, seq);
233 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
237 // duplicate or reordered packet...
243 int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
249 RTPStatistics *stats= &s->statistics;
251 uint32_t extended_max;
252 uint32_t expected_interval;
253 uint32_t received_interval;
254 uint32_t lost_interval;
257 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
259 if (!s->rtp_ctx || (count < 1))
262 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
263 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
264 s->octet_count += count;
265 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
267 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
270 s->last_octet_count = s->octet_count;
272 if (avio_open_dyn_buf(&pb) < 0)
276 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
277 avio_w8(pb, RTCP_RR);
278 avio_wb16(pb, 7); /* length in words - 1 */
279 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
280 avio_wb32(pb, s->ssrc + 1);
281 avio_wb32(pb, s->ssrc); // server SSRC
282 // some placeholders we should really fill...
284 extended_max= stats->cycles + stats->max_seq;
285 expected= extended_max - stats->base_seq + 1;
286 lost= expected - stats->received;
287 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
288 expected_interval= expected - stats->expected_prior;
289 stats->expected_prior= expected;
290 received_interval= stats->received - stats->received_prior;
291 stats->received_prior= stats->received;
292 lost_interval= expected_interval - received_interval;
293 if (expected_interval==0 || lost_interval<=0) fraction= 0;
294 else fraction = (lost_interval<<8)/expected_interval;
296 fraction= (fraction<<24) | lost;
298 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
299 avio_wb32(pb, extended_max); /* max sequence received */
300 avio_wb32(pb, stats->jitter>>4); /* jitter */
302 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
304 avio_wb32(pb, 0); /* last SR timestamp */
305 avio_wb32(pb, 0); /* delay since last SR */
307 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
308 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
310 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
311 avio_wb32(pb, delay_since_last); /* delay since last SR */
315 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
316 avio_w8(pb, RTCP_SDES);
317 len = strlen(s->hostname);
318 avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
319 avio_wb32(pb, s->ssrc + 1);
322 avio_write(pb, s->hostname, len);
324 for (len = (6 + len) % 4; len % 4; len++) {
329 len = avio_close_dyn_buf(pb, &buf);
330 if ((len > 0) && buf) {
331 int av_unused result;
332 av_dlog(s->ic, "sending %d bytes of RR\n", len);
333 result= ffurl_write(s->rtp_ctx, buf, len);
334 av_dlog(s->ic, "result from ffurl_write: %d\n", result);
340 void ff_rtp_send_punch_packets(URLContext* rtp_handle)
346 /* Send a small RTP packet */
347 if (avio_open_dyn_buf(&pb) < 0)
350 avio_w8(pb, (RTP_VERSION << 6));
351 avio_w8(pb, 0); /* Payload type */
352 avio_wb16(pb, 0); /* Seq */
353 avio_wb32(pb, 0); /* Timestamp */
354 avio_wb32(pb, 0); /* SSRC */
357 len = avio_close_dyn_buf(pb, &buf);
358 if ((len > 0) && buf)
359 ffurl_write(rtp_handle, buf, len);
362 /* Send a minimal RTCP RR */
363 if (avio_open_dyn_buf(&pb) < 0)
366 avio_w8(pb, (RTP_VERSION << 6));
367 avio_w8(pb, RTCP_RR); /* receiver report */
368 avio_wb16(pb, 1); /* length in words - 1 */
369 avio_wb32(pb, 0); /* our own SSRC */
372 len = avio_close_dyn_buf(pb, &buf);
373 if ((len > 0) && buf)
374 ffurl_write(rtp_handle, buf, len);
380 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
381 * MPEG2TS streams to indicate that they should be demuxed inside the
382 * rtp demux (otherwise AV_CODEC_ID_MPEG2TS packets are returned)
384 RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
388 s = av_mallocz(sizeof(RTPDemuxContext));
391 s->payload_type = payload_type;
392 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
393 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
396 s->queue_size = queue_size;
397 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
398 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
399 s->ts = ff_mpegts_parse_open(s->ic);
405 switch(st->codec->codec_id) {
406 case AV_CODEC_ID_MPEG1VIDEO:
407 case AV_CODEC_ID_MPEG2VIDEO:
408 case AV_CODEC_ID_MP2:
409 case AV_CODEC_ID_MP3:
410 case AV_CODEC_ID_MPEG4:
411 case AV_CODEC_ID_H263:
412 case AV_CODEC_ID_H264:
413 st->need_parsing = AVSTREAM_PARSE_FULL;
415 case AV_CODEC_ID_VORBIS:
416 st->need_parsing = AVSTREAM_PARSE_HEADERS;
418 case AV_CODEC_ID_ADPCM_G722:
419 /* According to RFC 3551, the stream clock rate is 8000
420 * even if the sample rate is 16000. */
421 if (st->codec->sample_rate == 8000)
422 st->codec->sample_rate = 16000;
428 // needed to send back RTCP RR in RTSP sessions
430 gethostname(s->hostname, sizeof(s->hostname));
435 ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
436 RTPDynamicProtocolHandler *handler)
438 s->dynamic_protocol_context = ctx;
439 s->parse_packet = handler->parse_packet;
443 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
445 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
447 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
448 return; /* Timestamp already set by depacketizer */
449 if (timestamp == RTP_NOTS_VALUE)
452 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
456 /* compute pts from timestamp with received ntp_time */
457 delta_timestamp = timestamp - s->last_rtcp_timestamp;
458 /* convert to the PTS timebase */
459 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);
460 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
465 if (!s->base_timestamp)
466 s->base_timestamp = timestamp;
467 /* assume that the difference is INT32_MIN < x < INT32_MAX, but allow the first timestamp to exceed INT32_MAX */
469 s->unwrapped_timestamp += timestamp;
471 s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
472 s->timestamp = timestamp;
473 pkt->pts = s->unwrapped_timestamp + s->range_start_offset - s->base_timestamp;
476 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
477 const uint8_t *buf, int len)
479 unsigned int ssrc, h;
480 int payload_type, seq, ret, flags = 0;
487 payload_type = buf[1] & 0x7f;
489 flags |= RTP_FLAG_MARKER;
490 seq = AV_RB16(buf + 2);
491 timestamp = AV_RB32(buf + 4);
492 ssrc = AV_RB32(buf + 8);
493 /* store the ssrc in the RTPDemuxContext */
496 /* NOTE: we can handle only one payload type */
497 if (s->payload_type != payload_type)
501 // only do something with this if all the rtp checks pass...
502 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
504 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
505 payload_type, seq, ((s->seq + 1) & 0xffff));
510 int padding = buf[len - 1];
511 if (len >= 12 + padding)
519 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
523 /* calculate the header extension length (stored as number
524 * of 32-bit words) */
525 ext = (AV_RB16(buf + 2) + 1) << 2;
529 // skip past RTP header extension
535 /* specific MPEG2TS demux support */
536 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
537 /* The only error that can be returned from ff_mpegts_parse_packet
538 * is "no more data to return from the provided buffer", so return
539 * AVERROR(EAGAIN) for all errors */
541 return AVERROR(EAGAIN);
543 s->read_buf_size = len - ret;
544 memcpy(s->buf, buf + ret, s->read_buf_size);
545 s->read_buf_index = 0;
549 } else if (s->parse_packet) {
550 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
551 s->st, pkt, ×tamp, buf, len, flags);
553 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
554 switch(st->codec->codec_id) {
555 case AV_CODEC_ID_MP2:
556 case AV_CODEC_ID_MP3:
557 /* better than nothing: skip mpeg audio RTP header */
563 if (av_new_packet(pkt, len) < 0)
564 return AVERROR(ENOMEM);
565 memcpy(pkt->data, buf, len);
567 case AV_CODEC_ID_MPEG1VIDEO:
568 case AV_CODEC_ID_MPEG2VIDEO:
569 /* better than nothing: skip mpeg video RTP header */
582 if (av_new_packet(pkt, len) < 0)
583 return AVERROR(ENOMEM);
584 memcpy(pkt->data, buf, len);
587 if (av_new_packet(pkt, len) < 0)
588 return AVERROR(ENOMEM);
589 memcpy(pkt->data, buf, len);
593 pkt->stream_index = st->index;
596 // now perform timestamp things....
597 finalize_packet(s, pkt, timestamp);
602 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
605 RTPPacket *next = s->queue->next;
606 av_free(s->queue->buf);
615 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
617 uint16_t seq = AV_RB16(buf + 2);
618 RTPPacket *cur = s->queue, *prev = NULL, *packet;
620 /* Find the correct place in the queue to insert the packet */
622 int16_t diff = seq - cur->seq;
629 packet = av_mallocz(sizeof(*packet));
632 packet->recvtime = av_gettime();
644 static int has_next_packet(RTPDemuxContext *s)
646 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
649 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
651 return s->queue ? s->queue->recvtime : 0;
654 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
659 if (s->queue_len <= 0)
662 if (!has_next_packet(s))
663 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
664 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
666 /* Parse the first packet in the queue, and dequeue it */
667 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
668 next = s->queue->next;
669 av_free(s->queue->buf);
676 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
677 uint8_t **bufptr, int len)
679 uint8_t* buf = bufptr ? *bufptr : NULL;
685 /* If parsing of the previous packet actually returned 0 or an error,
686 * there's nothing more to be parsed from that packet, but we may have
687 * indicated that we can return the next enqueued packet. */
688 if (s->prev_ret <= 0)
689 return rtp_parse_queued_packet(s, pkt);
690 /* return the next packets, if any */
691 if(s->st && s->parse_packet) {
692 /* timestamp should be overwritten by parse_packet, if not,
693 * the packet is left with pts == AV_NOPTS_VALUE */
694 timestamp = RTP_NOTS_VALUE;
695 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
696 s->st, pkt, ×tamp, NULL, 0, flags);
697 finalize_packet(s, pkt, timestamp);
700 // TODO: Move to a dynamic packet handler (like above)
701 if (s->read_buf_index >= s->read_buf_size)
702 return AVERROR(EAGAIN);
703 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
704 s->read_buf_size - s->read_buf_index);
706 return AVERROR(EAGAIN);
707 s->read_buf_index += ret;
708 if (s->read_buf_index < s->read_buf_size)
718 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
720 if (RTP_PT_IS_RTCP(buf[1])) {
721 return rtcp_parse_packet(s, buf, len);
724 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
725 /* First packet, or no reordering */
726 return rtp_parse_packet_internal(s, pkt, buf, len);
728 uint16_t seq = AV_RB16(buf + 2);
729 int16_t diff = seq - s->seq;
731 /* Packet older than the previously emitted one, drop */
732 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
733 "RTP: dropping old packet received too late\n");
735 } else if (diff <= 1) {
737 rv = rtp_parse_packet_internal(s, pkt, buf, len);
740 /* Still missing some packet, enqueue this one. */
741 enqueue_packet(s, buf, len);
743 /* Return the first enqueued packet if the queue is full,
744 * even if we're missing something */
745 if (s->queue_len >= s->queue_size)
746 return rtp_parse_queued_packet(s, pkt);
753 * Parse an RTP or RTCP packet directly sent as a buffer.
754 * @param s RTP parse context.
755 * @param pkt returned packet
756 * @param bufptr pointer to the input buffer or NULL to read the next packets
757 * @param len buffer len
758 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
759 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
761 int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
762 uint8_t **bufptr, int len)
764 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
766 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
767 rv = rtp_parse_queued_packet(s, pkt);
768 return rv ? rv : has_next_packet(s);
771 void ff_rtp_parse_close(RTPDemuxContext *s)
773 ff_rtp_reset_packet_queue(s);
774 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
775 ff_mpegts_parse_close(s->ts);
780 int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
781 int (*parse_fmtp)(AVStream *stream,
782 PayloadContext *data,
783 char *attr, char *value))
788 int value_size = strlen(p) + 1;
790 if (!(value = av_malloc(value_size))) {
791 av_log(NULL, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
792 return AVERROR(ENOMEM);
795 // remove protocol identifier
796 while (*p && *p == ' ') p++; // strip spaces
797 while (*p && *p != ' ') p++; // eat protocol identifier
798 while (*p && *p == ' ') p++; // strip trailing spaces
800 while (ff_rtsp_next_attr_and_value(&p,
802 value, value_size)) {
804 res = parse_fmtp(stream, data, attr, value);
805 if (res < 0 && res != AVERROR_PATCHWELCOME) {