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 static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
34 .enc_name = "X-MP3-draft-00",
35 .codec_type = AVMEDIA_TYPE_AUDIO,
36 .codec_id = AV_CODEC_ID_MP3ADU,
39 static RTPDynamicProtocolHandler speex_dynamic_handler = {
41 .codec_type = AVMEDIA_TYPE_AUDIO,
42 .codec_id = AV_CODEC_ID_SPEEX,
45 static RTPDynamicProtocolHandler opus_dynamic_handler = {
47 .codec_type = AVMEDIA_TYPE_AUDIO,
48 .codec_id = AV_CODEC_ID_OPUS,
51 /* statistics functions */
52 static RTPDynamicProtocolHandler *rtp_first_dynamic_payload_handler = NULL;
54 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
56 handler->next = rtp_first_dynamic_payload_handler;
57 rtp_first_dynamic_payload_handler = handler;
60 void av_register_rtp_dynamic_payload_handlers(void)
62 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
63 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
64 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
65 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
66 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
67 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
68 ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
69 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
70 ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
71 ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler);
72 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
73 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
74 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
75 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
76 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
77 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
78 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
79 ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
80 ff_register_dynamic_payload_handler(&speex_dynamic_handler);
81 ff_register_dynamic_payload_handler(&opus_dynamic_handler);
83 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
84 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
86 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
87 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
88 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
89 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
91 ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
92 ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
93 ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
94 ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
97 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
98 enum AVMediaType codec_type)
100 RTPDynamicProtocolHandler *handler;
101 for (handler = rtp_first_dynamic_payload_handler;
102 handler; handler = handler->next)
103 if (!av_strcasecmp(name, handler->enc_name) &&
104 codec_type == handler->codec_type)
109 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
110 enum AVMediaType codec_type)
112 RTPDynamicProtocolHandler *handler;
113 for (handler = rtp_first_dynamic_payload_handler;
114 handler; handler = handler->next)
115 if (handler->static_payload_id && handler->static_payload_id == id &&
116 codec_type == handler->codec_type)
121 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
126 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
130 if (payload_len < 20) {
131 av_log(NULL, AV_LOG_ERROR,
132 "Invalid length for RTCP SR packet\n");
133 return AVERROR_INVALIDDATA;
136 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
137 s->last_rtcp_timestamp = AV_RB32(buf + 16);
138 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
139 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
140 if (!s->base_timestamp)
141 s->base_timestamp = s->last_rtcp_timestamp;
142 s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
156 #define RTP_SEQ_MOD (1 << 16)
158 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
160 memset(s, 0, sizeof(RTPStatistics));
161 s->max_seq = base_sequence;
166 * Called whenever there is a large jump in sequence numbers,
167 * or when they get out of probation...
169 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
173 s->base_seq = seq - 1;
174 s->bad_seq = RTP_SEQ_MOD + 1;
176 s->expected_prior = 0;
177 s->received_prior = 0;
182 /* Returns 1 if we should handle this packet. */
183 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
185 uint16_t udelta = seq - s->max_seq;
186 const int MAX_DROPOUT = 3000;
187 const int MAX_MISORDER = 100;
188 const int MIN_SEQUENTIAL = 2;
190 /* source not valid until MIN_SEQUENTIAL packets with sequence
191 * seq. numbers have been received */
193 if (seq == s->max_seq + 1) {
196 if (s->probation == 0) {
197 rtp_init_sequence(s, seq);
202 s->probation = MIN_SEQUENTIAL - 1;
205 } else if (udelta < MAX_DROPOUT) {
206 // in order, with permissible gap
207 if (seq < s->max_seq) {
208 // sequence number wrapped; count another 64k cycles
209 s->cycles += RTP_SEQ_MOD;
212 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
213 // sequence made a large jump...
214 if (seq == s->bad_seq) {
215 /* two sequential packets -- assume that the other side
216 * restarted without telling us; just resync. */
217 rtp_init_sequence(s, seq);
219 s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
223 // duplicate or reordered packet...
229 int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
235 RTPStatistics *stats = &s->statistics;
237 uint32_t extended_max;
238 uint32_t expected_interval;
239 uint32_t received_interval;
240 uint32_t lost_interval;
243 uint64_t ntp_time = s->last_rtcp_ntp_time; // TODO: Get local ntp time?
245 if (!s->rtp_ctx || (count < 1))
248 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
249 /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
250 s->octet_count += count;
251 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
253 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
256 s->last_octet_count = s->octet_count;
258 if (avio_open_dyn_buf(&pb) < 0)
262 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
263 avio_w8(pb, RTCP_RR);
264 avio_wb16(pb, 7); /* length in words - 1 */
265 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
266 avio_wb32(pb, s->ssrc + 1);
267 avio_wb32(pb, s->ssrc); // server SSRC
268 // some placeholders we should really fill...
270 extended_max = stats->cycles + stats->max_seq;
271 expected = extended_max - stats->base_seq + 1;
272 lost = expected - stats->received;
273 lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
274 expected_interval = expected - stats->expected_prior;
275 stats->expected_prior = expected;
276 received_interval = stats->received - stats->received_prior;
277 stats->received_prior = stats->received;
278 lost_interval = expected_interval - received_interval;
279 if (expected_interval == 0 || lost_interval <= 0)
282 fraction = (lost_interval << 8) / expected_interval;
284 fraction = (fraction << 24) | lost;
286 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
287 avio_wb32(pb, extended_max); /* max sequence received */
288 avio_wb32(pb, stats->jitter >> 4); /* jitter */
290 if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) {
291 avio_wb32(pb, 0); /* last SR timestamp */
292 avio_wb32(pb, 0); /* delay since last SR */
294 uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
295 uint32_t delay_since_last = ntp_time - s->last_rtcp_ntp_time;
297 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
298 avio_wb32(pb, delay_since_last); /* delay since last SR */
302 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
303 avio_w8(pb, RTCP_SDES);
304 len = strlen(s->hostname);
305 avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
306 avio_wb32(pb, s->ssrc + 1);
309 avio_write(pb, s->hostname, len);
311 for (len = (6 + len) % 4; len % 4; len++)
315 len = avio_close_dyn_buf(pb, &buf);
316 if ((len > 0) && buf) {
317 int av_unused result;
318 av_dlog(s->ic, "sending %d bytes of RR\n", len);
319 result = ffurl_write(s->rtp_ctx, buf, len);
320 av_dlog(s->ic, "result from ffurl_write: %d\n", result);
326 void ff_rtp_send_punch_packets(URLContext *rtp_handle)
332 /* Send a small RTP packet */
333 if (avio_open_dyn_buf(&pb) < 0)
336 avio_w8(pb, (RTP_VERSION << 6));
337 avio_w8(pb, 0); /* Payload type */
338 avio_wb16(pb, 0); /* Seq */
339 avio_wb32(pb, 0); /* Timestamp */
340 avio_wb32(pb, 0); /* SSRC */
343 len = avio_close_dyn_buf(pb, &buf);
344 if ((len > 0) && buf)
345 ffurl_write(rtp_handle, buf, len);
348 /* Send a minimal RTCP RR */
349 if (avio_open_dyn_buf(&pb) < 0)
352 avio_w8(pb, (RTP_VERSION << 6));
353 avio_w8(pb, RTCP_RR); /* receiver report */
354 avio_wb16(pb, 1); /* length in words - 1 */
355 avio_wb32(pb, 0); /* our own SSRC */
358 len = avio_close_dyn_buf(pb, &buf);
359 if ((len > 0) && buf)
360 ffurl_write(rtp_handle, buf, len);
365 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
366 * MPEG2-TS streams to indicate that they should be demuxed inside the
367 * rtp demux (otherwise AV_CODEC_ID_MPEG2TS packets are returned)
369 RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
370 URLContext *rtpc, int payload_type,
375 s = av_mallocz(sizeof(RTPDemuxContext));
378 s->payload_type = payload_type;
379 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
380 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
383 s->queue_size = queue_size;
384 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
385 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
386 s->ts = ff_mpegts_parse_open(s->ic);
392 switch (st->codec->codec_id) {
393 case AV_CODEC_ID_MPEG1VIDEO:
394 case AV_CODEC_ID_MPEG2VIDEO:
395 case AV_CODEC_ID_MP2:
396 case AV_CODEC_ID_MP3:
397 case AV_CODEC_ID_MPEG4:
398 case AV_CODEC_ID_H263:
399 case AV_CODEC_ID_H264:
400 st->need_parsing = AVSTREAM_PARSE_FULL;
402 case AV_CODEC_ID_VORBIS:
403 st->need_parsing = AVSTREAM_PARSE_HEADERS;
405 case AV_CODEC_ID_ADPCM_G722:
406 /* According to RFC 3551, the stream clock rate is 8000
407 * even if the sample rate is 16000. */
408 if (st->codec->sample_rate == 8000)
409 st->codec->sample_rate = 16000;
415 // needed to send back RTCP RR in RTSP sessions
417 gethostname(s->hostname, sizeof(s->hostname));
421 void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
422 RTPDynamicProtocolHandler *handler)
424 s->dynamic_protocol_context = ctx;
425 s->parse_packet = handler->parse_packet;
429 * This was the second switch in rtp_parse packet.
430 * Normalizes time, if required, sets stream_index, etc.
432 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
434 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
435 return; /* Timestamp already set by depacketizer */
436 if (timestamp == RTP_NOTS_VALUE)
439 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
443 /* compute pts from timestamp with received ntp_time */
444 delta_timestamp = timestamp - s->last_rtcp_timestamp;
445 /* convert to the PTS timebase */
446 addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time,
447 s->st->time_base.den,
448 (uint64_t) s->st->time_base.num << 32);
449 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
454 if (!s->base_timestamp)
455 s->base_timestamp = timestamp;
456 /* assume that the difference is INT32_MIN < x < INT32_MAX,
457 * but allow the first timestamp to exceed INT32_MAX */
459 s->unwrapped_timestamp += timestamp;
461 s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
462 s->timestamp = timestamp;
463 pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
467 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
468 const uint8_t *buf, int len)
470 unsigned int ssrc, h;
471 int payload_type, seq, ret, flags = 0;
478 payload_type = buf[1] & 0x7f;
480 flags |= RTP_FLAG_MARKER;
481 seq = AV_RB16(buf + 2);
482 timestamp = AV_RB32(buf + 4);
483 ssrc = AV_RB32(buf + 8);
484 /* store the ssrc in the RTPDemuxContext */
487 /* NOTE: we can handle only one payload type */
488 if (s->payload_type != payload_type)
492 // only do something with this if all the rtp checks pass...
493 if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
494 av_log(st ? st->codec : NULL, AV_LOG_ERROR,
495 "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
496 payload_type, seq, ((s->seq + 1) & 0xffff));
501 int padding = buf[len - 1];
502 if (len >= 12 + padding)
510 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
514 /* calculate the header extension length (stored as number
515 * of 32-bit words) */
516 ext = (AV_RB16(buf + 2) + 1) << 2;
520 // skip past RTP header extension
526 /* specific MPEG2-TS demux support */
527 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
528 /* The only error that can be returned from ff_mpegts_parse_packet
529 * is "no more data to return from the provided buffer", so return
530 * AVERROR(EAGAIN) for all errors */
532 return AVERROR(EAGAIN);
534 s->read_buf_size = FFMIN(len - ret, sizeof(s->buf));
535 memcpy(s->buf, buf + ret, s->read_buf_size);
536 s->read_buf_index = 0;
540 } else if (s->parse_packet) {
541 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
542 s->st, pkt, ×tamp, buf, len, flags);
544 /* At this point, the RTP header has been stripped;
545 * This is ASSUMING that there is only 1 CSRC, which isn't wise. */
546 switch (st->codec->codec_id) {
547 case AV_CODEC_ID_MP2:
548 case AV_CODEC_ID_MP3:
549 /* better than nothing: skip MPEG audio RTP header */
555 if (av_new_packet(pkt, len) < 0)
556 return AVERROR(ENOMEM);
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 if (av_new_packet(pkt, len) < 0)
575 return AVERROR(ENOMEM);
576 memcpy(pkt->data, buf, len);
579 if (av_new_packet(pkt, len) < 0)
580 return AVERROR(ENOMEM);
581 memcpy(pkt->data, buf, len);
585 pkt->stream_index = st->index;
588 // now perform timestamp things....
589 finalize_packet(s, pkt, timestamp);
594 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
597 RTPPacket *next = s->queue->next;
598 av_free(s->queue->buf);
607 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
609 uint16_t seq = AV_RB16(buf + 2);
610 RTPPacket *cur = s->queue, *prev = NULL, *packet;
612 /* Find the correct place in the queue to insert the packet */
614 int16_t diff = seq - cur->seq;
621 packet = av_mallocz(sizeof(*packet));
624 packet->recvtime = av_gettime();
636 static int has_next_packet(RTPDemuxContext *s)
638 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
641 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
643 return s->queue ? s->queue->recvtime : 0;
646 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
651 if (s->queue_len <= 0)
654 if (!has_next_packet(s))
655 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
656 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
658 /* Parse the first packet in the queue, and dequeue it */
659 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
660 next = s->queue->next;
661 av_free(s->queue->buf);
668 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
669 uint8_t **bufptr, int len)
671 uint8_t *buf = bufptr ? *bufptr : NULL;
677 /* If parsing of the previous packet actually returned 0 or an error,
678 * there's nothing more to be parsed from that packet, but we may have
679 * indicated that we can return the next enqueued packet. */
680 if (s->prev_ret <= 0)
681 return rtp_parse_queued_packet(s, pkt);
682 /* return the next packets, if any */
683 if (s->st && s->parse_packet) {
684 /* timestamp should be overwritten by parse_packet, if not,
685 * the packet is left with pts == AV_NOPTS_VALUE */
686 timestamp = RTP_NOTS_VALUE;
687 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
688 s->st, pkt, ×tamp, NULL, 0, flags);
689 finalize_packet(s, pkt, timestamp);
692 // TODO: Move to a dynamic packet handler (like above)
693 if (s->read_buf_index >= s->read_buf_size)
694 return AVERROR(EAGAIN);
695 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
696 s->read_buf_size - s->read_buf_index);
698 return AVERROR(EAGAIN);
699 s->read_buf_index += ret;
700 if (s->read_buf_index < s->read_buf_size)
710 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
712 if (RTP_PT_IS_RTCP(buf[1])) {
713 return rtcp_parse_packet(s, buf, len);
716 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
717 /* First packet, or no reordering */
718 return rtp_parse_packet_internal(s, pkt, buf, len);
720 uint16_t seq = AV_RB16(buf + 2);
721 int16_t diff = seq - s->seq;
723 /* Packet older than the previously emitted one, drop */
724 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
725 "RTP: dropping old packet received too late\n");
727 } else if (diff <= 1) {
729 rv = rtp_parse_packet_internal(s, pkt, buf, len);
732 /* Still missing some packet, enqueue this one. */
733 enqueue_packet(s, buf, len);
735 /* Return the first enqueued packet if the queue is full,
736 * even if we're missing something */
737 if (s->queue_len >= s->queue_size)
738 return rtp_parse_queued_packet(s, pkt);
745 * Parse an RTP or RTCP packet directly sent as a buffer.
746 * @param s RTP parse context.
747 * @param pkt returned packet
748 * @param bufptr pointer to the input buffer or NULL to read the next packets
749 * @param len buffer len
750 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
751 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
753 int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
754 uint8_t **bufptr, int len)
756 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
758 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
759 rv = rtp_parse_queued_packet(s, pkt);
760 return rv ? rv : has_next_packet(s);
763 void ff_rtp_parse_close(RTPDemuxContext *s)
765 ff_rtp_reset_packet_queue(s);
766 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
767 ff_mpegts_parse_close(s->ts);
772 int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
773 int (*parse_fmtp)(AVStream *stream,
774 PayloadContext *data,
775 char *attr, char *value))
780 int value_size = strlen(p) + 1;
782 if (!(value = av_malloc(value_size))) {
783 av_log(NULL, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
784 return AVERROR(ENOMEM);
787 // remove protocol identifier
788 while (*p && *p == ' ')
790 while (*p && *p != ' ')
791 p++; // eat protocol identifier
792 while (*p && *p == ' ')
793 p++; // strip trailing spaces
795 while (ff_rtsp_next_attr_and_value(&p,
797 value, value_size)) {
798 res = parse_fmtp(stream, data, attr, value);
799 if (res < 0 && res != AVERROR_PATCHWELCOME) {
808 int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
812 pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
813 pkt->stream_index = stream_idx;
814 pkt->destruct = av_destruct_packet;