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 #define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */
35 static RTPDynamicProtocolHandler gsm_dynamic_handler = {
37 .codec_type = AVMEDIA_TYPE_AUDIO,
38 .codec_id = AV_CODEC_ID_GSM,
41 static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
42 .enc_name = "X-MP3-draft-00",
43 .codec_type = AVMEDIA_TYPE_AUDIO,
44 .codec_id = AV_CODEC_ID_MP3ADU,
47 static RTPDynamicProtocolHandler speex_dynamic_handler = {
49 .codec_type = AVMEDIA_TYPE_AUDIO,
50 .codec_id = AV_CODEC_ID_SPEEX,
53 static RTPDynamicProtocolHandler opus_dynamic_handler = {
55 .codec_type = AVMEDIA_TYPE_AUDIO,
56 .codec_id = AV_CODEC_ID_OPUS,
59 static RTPDynamicProtocolHandler *rtp_first_dynamic_payload_handler = NULL;
61 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
63 handler->next = rtp_first_dynamic_payload_handler;
64 rtp_first_dynamic_payload_handler = handler;
67 void ff_register_rtp_dynamic_payload_handlers(void)
69 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
70 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
71 ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
72 ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
73 ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
74 ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
75 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
76 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
77 ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
78 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
79 ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
80 ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler);
81 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
82 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
83 ff_register_dynamic_payload_handler(&ff_mpeg_audio_dynamic_handler);
84 ff_register_dynamic_payload_handler(&ff_mpeg_video_dynamic_handler);
85 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
86 ff_register_dynamic_payload_handler(&ff_mpegts_dynamic_handler);
87 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
88 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
89 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
90 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
91 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
92 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
93 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
94 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
95 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
96 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
97 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
98 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
99 ff_register_dynamic_payload_handler(&gsm_dynamic_handler);
100 ff_register_dynamic_payload_handler(&opus_dynamic_handler);
101 ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
102 ff_register_dynamic_payload_handler(&speex_dynamic_handler);
105 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
106 enum AVMediaType codec_type)
108 RTPDynamicProtocolHandler *handler;
109 for (handler = rtp_first_dynamic_payload_handler;
110 handler; handler = handler->next)
111 if (!av_strcasecmp(name, handler->enc_name) &&
112 codec_type == handler->codec_type)
117 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
118 enum AVMediaType codec_type)
120 RTPDynamicProtocolHandler *handler;
121 for (handler = rtp_first_dynamic_payload_handler;
122 handler; handler = handler->next)
123 if (handler->static_payload_id && handler->static_payload_id == id &&
124 codec_type == handler->codec_type)
129 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
134 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
138 if (payload_len < 20) {
139 av_log(NULL, AV_LOG_ERROR,
140 "Invalid length for RTCP SR packet\n");
141 return AVERROR_INVALIDDATA;
144 s->last_rtcp_reception_time = av_gettime();
145 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
146 s->last_rtcp_timestamp = AV_RB32(buf + 16);
147 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
148 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
149 if (!s->base_timestamp)
150 s->base_timestamp = s->last_rtcp_timestamp;
151 s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
165 #define RTP_SEQ_MOD (1 << 16)
167 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
169 memset(s, 0, sizeof(RTPStatistics));
170 s->max_seq = base_sequence;
175 * Called whenever there is a large jump in sequence numbers,
176 * or when they get out of probation...
178 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
182 s->base_seq = seq - 1;
183 s->bad_seq = RTP_SEQ_MOD + 1;
185 s->expected_prior = 0;
186 s->received_prior = 0;
191 /* Returns 1 if we should handle this packet. */
192 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
194 uint16_t udelta = seq - s->max_seq;
195 const int MAX_DROPOUT = 3000;
196 const int MAX_MISORDER = 100;
197 const int MIN_SEQUENTIAL = 2;
199 /* source not valid until MIN_SEQUENTIAL packets with sequence
200 * seq. numbers have been received */
202 if (seq == s->max_seq + 1) {
205 if (s->probation == 0) {
206 rtp_init_sequence(s, seq);
211 s->probation = MIN_SEQUENTIAL - 1;
214 } else if (udelta < MAX_DROPOUT) {
215 // in order, with permissible gap
216 if (seq < s->max_seq) {
217 // sequence number wrapped; count another 64k cycles
218 s->cycles += RTP_SEQ_MOD;
221 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
222 // sequence made a large jump...
223 if (seq == s->bad_seq) {
224 /* two sequential packets -- assume that the other side
225 * restarted without telling us; just resync. */
226 rtp_init_sequence(s, seq);
228 s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
232 // duplicate or reordered packet...
238 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp,
239 uint32_t arrival_timestamp)
241 // Most of this is pretty straight from RFC 3550 appendix A.8
242 uint32_t transit = arrival_timestamp - sent_timestamp;
243 uint32_t prev_transit = s->transit;
244 int32_t d = transit - prev_transit;
245 // Doing the FFABS() call directly on the "transit - prev_transit"
246 // expression doesn't work, since it's an unsigned expression. Doing the
247 // transit calculation in unsigned is desired though, since it most
248 // probably will need to wrap around.
250 s->transit = transit;
253 s->jitter += d - (int32_t) ((s->jitter + 8) >> 4);
256 int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
257 AVIOContext *avio, int count)
263 RTPStatistics *stats = &s->statistics;
265 uint32_t extended_max;
266 uint32_t expected_interval;
267 uint32_t received_interval;
268 int32_t lost_interval;
272 if ((!fd && !avio) || (count < 1))
275 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
276 /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
277 s->octet_count += count;
278 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
280 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
283 s->last_octet_count = s->octet_count;
287 else if (avio_open_dyn_buf(&pb) < 0)
291 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
292 avio_w8(pb, RTCP_RR);
293 avio_wb16(pb, 7); /* length in words - 1 */
294 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
295 avio_wb32(pb, s->ssrc + 1);
296 avio_wb32(pb, s->ssrc); // server SSRC
297 // some placeholders we should really fill...
299 extended_max = stats->cycles + stats->max_seq;
300 expected = extended_max - stats->base_seq;
301 lost = expected - stats->received;
302 lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
303 expected_interval = expected - stats->expected_prior;
304 stats->expected_prior = expected;
305 received_interval = stats->received - stats->received_prior;
306 stats->received_prior = stats->received;
307 lost_interval = expected_interval - received_interval;
308 if (expected_interval == 0 || lost_interval <= 0)
311 fraction = (lost_interval << 8) / expected_interval;
313 fraction = (fraction << 24) | lost;
315 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
316 avio_wb32(pb, extended_max); /* max sequence received */
317 avio_wb32(pb, stats->jitter >> 4); /* jitter */
319 if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) {
320 avio_wb32(pb, 0); /* last SR timestamp */
321 avio_wb32(pb, 0); /* delay since last SR */
323 uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
324 uint32_t delay_since_last = av_rescale(av_gettime() - s->last_rtcp_reception_time,
325 65536, AV_TIME_BASE);
327 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
328 avio_wb32(pb, delay_since_last); /* delay since last SR */
332 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
333 avio_w8(pb, RTCP_SDES);
334 len = strlen(s->hostname);
335 avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
336 avio_wb32(pb, s->ssrc + 1);
339 avio_write(pb, s->hostname, len);
340 avio_w8(pb, 0); /* END */
342 for (len = (7 + len) % 4; len % 4; len++)
348 len = avio_close_dyn_buf(pb, &buf);
349 if ((len > 0) && buf) {
350 int av_unused result;
351 av_dlog(s->ic, "sending %d bytes of RR\n", len);
352 result = ffurl_write(fd, buf, len);
353 av_dlog(s->ic, "result from ffurl_write: %d\n", result);
359 void ff_rtp_send_punch_packets(URLContext *rtp_handle)
365 /* Send a small RTP packet */
366 if (avio_open_dyn_buf(&pb) < 0)
369 avio_w8(pb, (RTP_VERSION << 6));
370 avio_w8(pb, 0); /* Payload type */
371 avio_wb16(pb, 0); /* Seq */
372 avio_wb32(pb, 0); /* Timestamp */
373 avio_wb32(pb, 0); /* SSRC */
376 len = avio_close_dyn_buf(pb, &buf);
377 if ((len > 0) && buf)
378 ffurl_write(rtp_handle, buf, len);
381 /* Send a minimal RTCP RR */
382 if (avio_open_dyn_buf(&pb) < 0)
385 avio_w8(pb, (RTP_VERSION << 6));
386 avio_w8(pb, RTCP_RR); /* receiver report */
387 avio_wb16(pb, 1); /* length in words - 1 */
388 avio_wb32(pb, 0); /* our own SSRC */
391 len = avio_close_dyn_buf(pb, &buf);
392 if ((len > 0) && buf)
393 ffurl_write(rtp_handle, buf, len);
397 static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
398 uint16_t *missing_mask)
401 uint16_t next_seq = s->seq + 1;
402 RTPPacket *pkt = s->queue;
404 if (!pkt || pkt->seq == next_seq)
408 for (i = 1; i <= 16; i++) {
409 uint16_t missing_seq = next_seq + i;
411 int16_t diff = pkt->seq - missing_seq;
418 if (pkt->seq == missing_seq)
420 *missing_mask |= 1 << (i - 1);
423 *first_missing = next_seq;
427 int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
430 int len, need_keyframe, missing_packets;
434 uint16_t first_missing = 0, missing_mask = 0;
439 need_keyframe = s->handler && s->handler->need_keyframe &&
440 s->handler->need_keyframe(s->dynamic_protocol_context);
441 missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
443 if (!need_keyframe && !missing_packets)
446 /* Send new feedback if enough time has elapsed since the last
447 * feedback packet. */
450 if (s->last_feedback_time &&
451 (now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL)
453 s->last_feedback_time = now;
457 else if (avio_open_dyn_buf(&pb) < 0)
461 avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
462 avio_w8(pb, RTCP_PSFB);
463 avio_wb16(pb, 2); /* length in words - 1 */
464 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
465 avio_wb32(pb, s->ssrc + 1);
466 avio_wb32(pb, s->ssrc); // server SSRC
469 if (missing_packets) {
470 avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
471 avio_w8(pb, RTCP_RTPFB);
472 avio_wb16(pb, 3); /* length in words - 1 */
473 avio_wb32(pb, s->ssrc + 1);
474 avio_wb32(pb, s->ssrc); // server SSRC
476 avio_wb16(pb, first_missing);
477 avio_wb16(pb, missing_mask);
483 len = avio_close_dyn_buf(pb, &buf);
484 if (len > 0 && buf) {
485 ffurl_write(fd, buf, len);
492 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
495 RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
496 int payload_type, int queue_size)
500 s = av_mallocz(sizeof(RTPDemuxContext));
503 s->payload_type = payload_type;
504 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
505 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
508 s->queue_size = queue_size;
509 rtp_init_statistics(&s->statistics, 0);
511 switch (st->codec->codec_id) {
512 case AV_CODEC_ID_ADPCM_G722:
513 /* According to RFC 3551, the stream clock rate is 8000
514 * even if the sample rate is 16000. */
515 if (st->codec->sample_rate == 8000)
516 st->codec->sample_rate = 16000;
522 // needed to send back RTCP RR in RTSP sessions
523 gethostname(s->hostname, sizeof(s->hostname));
527 void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
528 RTPDynamicProtocolHandler *handler)
530 s->dynamic_protocol_context = ctx;
531 s->handler = handler;
534 void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
537 if (!ff_srtp_set_crypto(&s->srtp, suite, params))
542 * This was the second switch in rtp_parse packet.
543 * Normalizes time, if required, sets stream_index, etc.
545 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
547 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
548 return; /* Timestamp already set by depacketizer */
549 if (timestamp == RTP_NOTS_VALUE)
552 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
556 /* compute pts from timestamp with received ntp_time */
557 delta_timestamp = timestamp - s->last_rtcp_timestamp;
558 /* convert to the PTS timebase */
559 addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time,
560 s->st->time_base.den,
561 (uint64_t) s->st->time_base.num << 32);
562 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
567 if (!s->base_timestamp)
568 s->base_timestamp = timestamp;
569 /* assume that the difference is INT32_MIN < x < INT32_MAX,
570 * but allow the first timestamp to exceed INT32_MAX */
572 s->unwrapped_timestamp += timestamp;
574 s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
575 s->timestamp = timestamp;
576 pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
580 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
581 const uint8_t *buf, int len)
584 int payload_type, seq, flags = 0;
590 csrc = buf[0] & 0x0f;
592 payload_type = buf[1] & 0x7f;
594 flags |= RTP_FLAG_MARKER;
595 seq = AV_RB16(buf + 2);
596 timestamp = AV_RB32(buf + 4);
597 ssrc = AV_RB32(buf + 8);
598 /* store the ssrc in the RTPDemuxContext */
601 /* NOTE: we can handle only one payload type */
602 if (s->payload_type != payload_type)
606 // only do something with this if all the rtp checks pass...
607 if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
608 av_log(st ? st->codec : NULL, AV_LOG_ERROR,
609 "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
610 payload_type, seq, ((s->seq + 1) & 0xffff));
615 int padding = buf[len - 1];
616 if (len >= 12 + padding)
627 return AVERROR_INVALIDDATA;
629 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
633 /* calculate the header extension length (stored as number
634 * of 32-bit words) */
635 ext = (AV_RB16(buf + 2) + 1) << 2;
639 // skip past RTP header extension
644 if (s->handler && s->handler->parse_packet) {
645 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
646 s->st, pkt, ×tamp, buf, len, seq,
649 if ((rv = av_new_packet(pkt, len)) < 0)
651 memcpy(pkt->data, buf, len);
652 pkt->stream_index = st->index;
654 return AVERROR(EINVAL);
657 // now perform timestamp things....
658 finalize_packet(s, pkt, timestamp);
663 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
666 RTPPacket *next = s->queue->next;
667 av_free(s->queue->buf);
676 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
678 uint16_t seq = AV_RB16(buf + 2);
679 RTPPacket **cur = &s->queue, *packet;
681 /* Find the correct place in the queue to insert the packet */
683 int16_t diff = seq - (*cur)->seq;
689 packet = av_mallocz(sizeof(*packet));
692 packet->recvtime = av_gettime();
701 static int has_next_packet(RTPDemuxContext *s)
703 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
706 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
708 return s->queue ? s->queue->recvtime : 0;
711 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
716 if (s->queue_len <= 0)
719 if (!has_next_packet(s))
720 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
721 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
723 /* Parse the first packet in the queue, and dequeue it */
724 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
725 next = s->queue->next;
726 av_free(s->queue->buf);
733 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
734 uint8_t **bufptr, int len)
736 uint8_t *buf = bufptr ? *bufptr : NULL;
742 /* If parsing of the previous packet actually returned 0 or an error,
743 * there's nothing more to be parsed from that packet, but we may have
744 * indicated that we can return the next enqueued packet. */
745 if (s->prev_ret <= 0)
746 return rtp_parse_queued_packet(s, pkt);
747 /* return the next packets, if any */
748 if (s->handler && s->handler->parse_packet) {
749 /* timestamp should be overwritten by parse_packet, if not,
750 * the packet is left with pts == AV_NOPTS_VALUE */
751 timestamp = RTP_NOTS_VALUE;
752 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
753 s->st, pkt, ×tamp, NULL, 0, 0,
755 finalize_packet(s, pkt, timestamp);
763 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
765 if (RTP_PT_IS_RTCP(buf[1])) {
766 return rtcp_parse_packet(s, buf, len);
770 int64_t received = av_gettime();
771 uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
773 timestamp = AV_RB32(buf + 4);
774 // Calculate the jitter immediately, before queueing the packet
775 // into the reordering queue.
776 rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
779 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
780 /* First packet, or no reordering */
781 return rtp_parse_packet_internal(s, pkt, buf, len);
783 uint16_t seq = AV_RB16(buf + 2);
784 int16_t diff = seq - s->seq;
786 /* Packet older than the previously emitted one, drop */
787 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
788 "RTP: dropping old packet received too late\n");
790 } else if (diff <= 1) {
792 rv = rtp_parse_packet_internal(s, pkt, buf, len);
795 /* Still missing some packet, enqueue this one. */
796 enqueue_packet(s, buf, len);
798 /* Return the first enqueued packet if the queue is full,
799 * even if we're missing something */
800 if (s->queue_len >= s->queue_size)
801 return rtp_parse_queued_packet(s, pkt);
808 * Parse an RTP or RTCP packet directly sent as a buffer.
809 * @param s RTP parse context.
810 * @param pkt returned packet
811 * @param bufptr pointer to the input buffer or NULL to read the next packets
812 * @param len buffer len
813 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
814 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
816 int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
817 uint8_t **bufptr, int len)
820 if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
822 rv = rtp_parse_one_packet(s, pkt, bufptr, len);
824 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
825 rv = rtp_parse_queued_packet(s, pkt);
826 return rv ? rv : has_next_packet(s);
829 void ff_rtp_parse_close(RTPDemuxContext *s)
831 ff_rtp_reset_packet_queue(s);
832 ff_srtp_free(&s->srtp);
836 int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
837 int (*parse_fmtp)(AVStream *stream,
838 PayloadContext *data,
839 char *attr, char *value))
844 int value_size = strlen(p) + 1;
846 if (!(value = av_malloc(value_size))) {
847 av_log(NULL, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
848 return AVERROR(ENOMEM);
851 // remove protocol identifier
852 while (*p && *p == ' ')
854 while (*p && *p != ' ')
855 p++; // eat protocol identifier
856 while (*p && *p == ' ')
857 p++; // strip trailing spaces
859 while (ff_rtsp_next_attr_and_value(&p,
861 value, value_size)) {
862 res = parse_fmtp(stream, data, attr, value);
863 if (res < 0 && res != AVERROR_PATCHWELCOME) {
872 int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
877 pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
878 pkt->stream_index = stream_idx;
880 if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) {
881 av_freep(&pkt->data);