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_h261_dynamic_handler);
76 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
77 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
78 ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
79 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
80 ff_register_dynamic_payload_handler(&ff_h265_dynamic_handler);
81 ff_register_dynamic_payload_handler(&ff_hevc_dynamic_handler);
82 ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
83 ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler);
84 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
85 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
86 ff_register_dynamic_payload_handler(&ff_mpeg_audio_dynamic_handler);
87 ff_register_dynamic_payload_handler(&ff_mpeg_video_dynamic_handler);
88 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
89 ff_register_dynamic_payload_handler(&ff_mpegts_dynamic_handler);
90 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
91 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
92 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
93 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
94 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
95 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
96 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
97 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
98 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
99 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
100 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
101 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
102 ff_register_dynamic_payload_handler(&gsm_dynamic_handler);
103 ff_register_dynamic_payload_handler(&opus_dynamic_handler);
104 ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
105 ff_register_dynamic_payload_handler(&speex_dynamic_handler);
108 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
109 enum AVMediaType codec_type)
111 RTPDynamicProtocolHandler *handler;
112 for (handler = rtp_first_dynamic_payload_handler;
113 handler; handler = handler->next)
114 if (!av_strcasecmp(name, handler->enc_name) &&
115 codec_type == handler->codec_type)
120 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
121 enum AVMediaType codec_type)
123 RTPDynamicProtocolHandler *handler;
124 for (handler = rtp_first_dynamic_payload_handler;
125 handler; handler = handler->next)
126 if (handler->static_payload_id && handler->static_payload_id == id &&
127 codec_type == handler->codec_type)
132 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
137 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
141 if (payload_len < 20) {
142 av_log(NULL, AV_LOG_ERROR,
143 "Invalid length for RTCP SR packet\n");
144 return AVERROR_INVALIDDATA;
147 s->last_rtcp_reception_time = av_gettime();
148 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
149 s->last_rtcp_timestamp = AV_RB32(buf + 16);
150 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
151 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
152 if (!s->base_timestamp)
153 s->base_timestamp = s->last_rtcp_timestamp;
154 s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
168 #define RTP_SEQ_MOD (1 << 16)
170 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
172 memset(s, 0, sizeof(RTPStatistics));
173 s->max_seq = base_sequence;
178 * Called whenever there is a large jump in sequence numbers,
179 * or when they get out of probation...
181 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
185 s->base_seq = seq - 1;
186 s->bad_seq = RTP_SEQ_MOD + 1;
188 s->expected_prior = 0;
189 s->received_prior = 0;
194 /* 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
203 * seq. numbers have been received */
205 if (seq == s->max_seq + 1) {
208 if (s->probation == 0) {
209 rtp_init_sequence(s, seq);
214 s->probation = MIN_SEQUENTIAL - 1;
217 } else if (udelta < MAX_DROPOUT) {
218 // in order, with permissible gap
219 if (seq < s->max_seq) {
220 // sequence number wrapped; count another 64k cycles
221 s->cycles += RTP_SEQ_MOD;
224 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
225 // sequence made a large jump...
226 if (seq == s->bad_seq) {
227 /* two sequential packets -- assume that the other side
228 * restarted without telling us; just resync. */
229 rtp_init_sequence(s, seq);
231 s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
235 // duplicate or reordered packet...
241 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp,
242 uint32_t arrival_timestamp)
244 // Most of this is pretty straight from RFC 3550 appendix A.8
245 uint32_t transit = arrival_timestamp - sent_timestamp;
246 uint32_t prev_transit = s->transit;
247 int32_t d = transit - prev_transit;
248 // Doing the FFABS() call directly on the "transit - prev_transit"
249 // expression doesn't work, since it's an unsigned expression. Doing the
250 // transit calculation in unsigned is desired though, since it most
251 // probably will need to wrap around.
253 s->transit = transit;
256 s->jitter += d - (int32_t) ((s->jitter + 8) >> 4);
259 int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
260 AVIOContext *avio, int count)
266 RTPStatistics *stats = &s->statistics;
268 uint32_t extended_max;
269 uint32_t expected_interval;
270 uint32_t received_interval;
271 int32_t lost_interval;
275 if ((!fd && !avio) || (count < 1))
278 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
279 /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
280 s->octet_count += count;
281 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
283 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
286 s->last_octet_count = s->octet_count;
290 else if (avio_open_dyn_buf(&pb) < 0)
294 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
295 avio_w8(pb, RTCP_RR);
296 avio_wb16(pb, 7); /* length in words - 1 */
297 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
298 avio_wb32(pb, s->ssrc + 1);
299 avio_wb32(pb, s->ssrc); // server SSRC
300 // some placeholders we should really fill...
302 extended_max = stats->cycles + stats->max_seq;
303 expected = extended_max - stats->base_seq;
304 lost = expected - stats->received;
305 lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
306 expected_interval = expected - stats->expected_prior;
307 stats->expected_prior = expected;
308 received_interval = stats->received - stats->received_prior;
309 stats->received_prior = stats->received;
310 lost_interval = expected_interval - received_interval;
311 if (expected_interval == 0 || lost_interval <= 0)
314 fraction = (lost_interval << 8) / expected_interval;
316 fraction = (fraction << 24) | lost;
318 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
319 avio_wb32(pb, extended_max); /* max sequence received */
320 avio_wb32(pb, stats->jitter >> 4); /* jitter */
322 if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) {
323 avio_wb32(pb, 0); /* last SR timestamp */
324 avio_wb32(pb, 0); /* delay since last SR */
326 uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special?
327 uint32_t delay_since_last = av_rescale(av_gettime() - s->last_rtcp_reception_time,
328 65536, AV_TIME_BASE);
330 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
331 avio_wb32(pb, delay_since_last); /* delay since last SR */
335 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
336 avio_w8(pb, RTCP_SDES);
337 len = strlen(s->hostname);
338 avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
339 avio_wb32(pb, s->ssrc + 1);
342 avio_write(pb, s->hostname, len);
343 avio_w8(pb, 0); /* END */
345 for (len = (7 + len) % 4; len % 4; len++)
351 len = avio_close_dyn_buf(pb, &buf);
352 if ((len > 0) && buf) {
353 int av_unused result;
354 av_dlog(s->ic, "sending %d bytes of RR\n", len);
355 result = ffurl_write(fd, buf, len);
356 av_dlog(s->ic, "result from ffurl_write: %d\n", result);
362 void ff_rtp_send_punch_packets(URLContext *rtp_handle)
368 /* Send a small RTP packet */
369 if (avio_open_dyn_buf(&pb) < 0)
372 avio_w8(pb, (RTP_VERSION << 6));
373 avio_w8(pb, 0); /* Payload type */
374 avio_wb16(pb, 0); /* Seq */
375 avio_wb32(pb, 0); /* Timestamp */
376 avio_wb32(pb, 0); /* SSRC */
379 len = avio_close_dyn_buf(pb, &buf);
380 if ((len > 0) && buf)
381 ffurl_write(rtp_handle, buf, len);
384 /* Send a minimal RTCP RR */
385 if (avio_open_dyn_buf(&pb) < 0)
388 avio_w8(pb, (RTP_VERSION << 6));
389 avio_w8(pb, RTCP_RR); /* receiver report */
390 avio_wb16(pb, 1); /* length in words - 1 */
391 avio_wb32(pb, 0); /* our own SSRC */
394 len = avio_close_dyn_buf(pb, &buf);
395 if ((len > 0) && buf)
396 ffurl_write(rtp_handle, buf, len);
400 static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
401 uint16_t *missing_mask)
404 uint16_t next_seq = s->seq + 1;
405 RTPPacket *pkt = s->queue;
407 if (!pkt || pkt->seq == next_seq)
411 for (i = 1; i <= 16; i++) {
412 uint16_t missing_seq = next_seq + i;
414 int16_t diff = pkt->seq - missing_seq;
421 if (pkt->seq == missing_seq)
423 *missing_mask |= 1 << (i - 1);
426 *first_missing = next_seq;
430 int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
433 int len, need_keyframe, missing_packets;
437 uint16_t first_missing = 0, missing_mask = 0;
442 need_keyframe = s->handler && s->handler->need_keyframe &&
443 s->handler->need_keyframe(s->dynamic_protocol_context);
444 missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
446 if (!need_keyframe && !missing_packets)
449 /* Send new feedback if enough time has elapsed since the last
450 * feedback packet. */
453 if (s->last_feedback_time &&
454 (now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL)
456 s->last_feedback_time = now;
460 else if (avio_open_dyn_buf(&pb) < 0)
464 avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
465 avio_w8(pb, RTCP_PSFB);
466 avio_wb16(pb, 2); /* length in words - 1 */
467 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
468 avio_wb32(pb, s->ssrc + 1);
469 avio_wb32(pb, s->ssrc); // server SSRC
472 if (missing_packets) {
473 avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
474 avio_w8(pb, RTCP_RTPFB);
475 avio_wb16(pb, 3); /* length in words - 1 */
476 avio_wb32(pb, s->ssrc + 1);
477 avio_wb32(pb, s->ssrc); // server SSRC
479 avio_wb16(pb, first_missing);
480 avio_wb16(pb, missing_mask);
486 len = avio_close_dyn_buf(pb, &buf);
487 if (len > 0 && buf) {
488 ffurl_write(fd, buf, len);
495 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
498 RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
499 int payload_type, int queue_size)
503 s = av_mallocz(sizeof(RTPDemuxContext));
506 s->payload_type = payload_type;
507 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
508 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
511 s->queue_size = queue_size;
512 rtp_init_statistics(&s->statistics, 0);
514 switch (st->codec->codec_id) {
515 case AV_CODEC_ID_ADPCM_G722:
516 /* According to RFC 3551, the stream clock rate is 8000
517 * even if the sample rate is 16000. */
518 if (st->codec->sample_rate == 8000)
519 st->codec->sample_rate = 16000;
525 // needed to send back RTCP RR in RTSP sessions
526 gethostname(s->hostname, sizeof(s->hostname));
530 void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
531 RTPDynamicProtocolHandler *handler)
533 s->dynamic_protocol_context = ctx;
534 s->handler = handler;
537 void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
540 if (!ff_srtp_set_crypto(&s->srtp, suite, params))
545 * This was the second switch in rtp_parse packet.
546 * Normalizes time, if required, sets stream_index, etc.
548 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
550 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
551 return; /* Timestamp already set by depacketizer */
552 if (timestamp == RTP_NOTS_VALUE)
555 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
559 /* compute pts from timestamp with received ntp_time */
560 delta_timestamp = timestamp - s->last_rtcp_timestamp;
561 /* convert to the PTS timebase */
562 addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time,
563 s->st->time_base.den,
564 (uint64_t) s->st->time_base.num << 32);
565 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
570 if (!s->base_timestamp)
571 s->base_timestamp = timestamp;
572 /* assume that the difference is INT32_MIN < x < INT32_MAX,
573 * but allow the first timestamp to exceed INT32_MAX */
575 s->unwrapped_timestamp += timestamp;
577 s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
578 s->timestamp = timestamp;
579 pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
583 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
584 const uint8_t *buf, int len)
587 int payload_type, seq, flags = 0;
593 csrc = buf[0] & 0x0f;
595 payload_type = buf[1] & 0x7f;
597 flags |= RTP_FLAG_MARKER;
598 seq = AV_RB16(buf + 2);
599 timestamp = AV_RB32(buf + 4);
600 ssrc = AV_RB32(buf + 8);
601 /* store the ssrc in the RTPDemuxContext */
604 /* NOTE: we can handle only one payload type */
605 if (s->payload_type != payload_type)
609 // only do something with this if all the rtp checks pass...
610 if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
611 av_log(st ? st->codec : NULL, AV_LOG_ERROR,
612 "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
613 payload_type, seq, ((s->seq + 1) & 0xffff));
618 int padding = buf[len - 1];
619 if (len >= 12 + padding)
630 return AVERROR_INVALIDDATA;
632 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
636 /* calculate the header extension length (stored as number
637 * of 32-bit words) */
638 ext = (AV_RB16(buf + 2) + 1) << 2;
642 // skip past RTP header extension
647 if (s->handler && s->handler->parse_packet) {
648 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
649 s->st, pkt, ×tamp, buf, len, seq,
652 if ((rv = av_new_packet(pkt, len)) < 0)
654 memcpy(pkt->data, buf, len);
655 pkt->stream_index = st->index;
657 return AVERROR(EINVAL);
660 // now perform timestamp things....
661 finalize_packet(s, pkt, timestamp);
666 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
669 RTPPacket *next = s->queue->next;
670 av_free(s->queue->buf);
679 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
681 uint16_t seq = AV_RB16(buf + 2);
682 RTPPacket **cur = &s->queue, *packet;
684 /* Find the correct place in the queue to insert the packet */
686 int16_t diff = seq - (*cur)->seq;
692 packet = av_mallocz(sizeof(*packet));
695 packet->recvtime = av_gettime();
704 static int has_next_packet(RTPDemuxContext *s)
706 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
709 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
711 return s->queue ? s->queue->recvtime : 0;
714 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
719 if (s->queue_len <= 0)
722 if (!has_next_packet(s))
723 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
724 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
726 /* Parse the first packet in the queue, and dequeue it */
727 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
728 next = s->queue->next;
729 av_free(s->queue->buf);
736 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
737 uint8_t **bufptr, int len)
739 uint8_t *buf = bufptr ? *bufptr : NULL;
745 /* If parsing of the previous packet actually returned 0 or an error,
746 * there's nothing more to be parsed from that packet, but we may have
747 * indicated that we can return the next enqueued packet. */
748 if (s->prev_ret <= 0)
749 return rtp_parse_queued_packet(s, pkt);
750 /* return the next packets, if any */
751 if (s->handler && s->handler->parse_packet) {
752 /* timestamp should be overwritten by parse_packet, if not,
753 * the packet is left with pts == AV_NOPTS_VALUE */
754 timestamp = RTP_NOTS_VALUE;
755 rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
756 s->st, pkt, ×tamp, NULL, 0, 0,
758 finalize_packet(s, pkt, timestamp);
766 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
768 if (RTP_PT_IS_RTCP(buf[1])) {
769 return rtcp_parse_packet(s, buf, len);
773 int64_t received = av_gettime();
774 uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
776 timestamp = AV_RB32(buf + 4);
777 // Calculate the jitter immediately, before queueing the packet
778 // into the reordering queue.
779 rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
782 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
783 /* First packet, or no reordering */
784 return rtp_parse_packet_internal(s, pkt, buf, len);
786 uint16_t seq = AV_RB16(buf + 2);
787 int16_t diff = seq - s->seq;
789 /* Packet older than the previously emitted one, drop */
790 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
791 "RTP: dropping old packet received too late\n");
793 } else if (diff <= 1) {
795 rv = rtp_parse_packet_internal(s, pkt, buf, len);
798 /* Still missing some packet, enqueue this one. */
799 enqueue_packet(s, buf, len);
801 /* Return the first enqueued packet if the queue is full,
802 * even if we're missing something */
803 if (s->queue_len >= s->queue_size)
804 return rtp_parse_queued_packet(s, pkt);
811 * Parse an RTP or RTCP packet directly sent as a buffer.
812 * @param s RTP parse context.
813 * @param pkt returned packet
814 * @param bufptr pointer to the input buffer or NULL to read the next packets
815 * @param len buffer len
816 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
817 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
819 int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
820 uint8_t **bufptr, int len)
823 if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
825 rv = rtp_parse_one_packet(s, pkt, bufptr, len);
827 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
828 rv = rtp_parse_queued_packet(s, pkt);
829 return rv ? rv : has_next_packet(s);
832 void ff_rtp_parse_close(RTPDemuxContext *s)
834 ff_rtp_reset_packet_queue(s);
835 ff_srtp_free(&s->srtp);
839 int ff_parse_fmtp(AVFormatContext *s,
840 AVStream *stream, PayloadContext *data, const char *p,
841 int (*parse_fmtp)(AVFormatContext *s,
843 PayloadContext *data,
844 char *attr, char *value))
849 int value_size = strlen(p) + 1;
851 if (!(value = av_malloc(value_size))) {
852 av_log(NULL, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
853 return AVERROR(ENOMEM);
856 // remove protocol identifier
857 while (*p && *p == ' ')
859 while (*p && *p != ' ')
860 p++; // eat protocol identifier
861 while (*p && *p == ' ')
862 p++; // strip trailing spaces
864 while (ff_rtsp_next_attr_and_value(&p,
866 value, value_size)) {
867 res = parse_fmtp(s, stream, data, attr, value);
868 if (res < 0 && res != AVERROR_PATCHWELCOME) {
877 int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
882 pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
883 pkt->stream_index = stream_idx;
885 if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) {
886 av_freep(&pkt->data);