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"
34 #include "rtpdec_formats.h"
38 /* TODO: - add RTCP statistics reporting (should be optional).
40 - add support for h263/mpeg4 packetized output : IDEA: send a
41 buffer to 'rtp_write_packet' contains all the packets for ONE
42 frame. Each packet should have a four byte header containing
43 the length in big endian format (same trick as
44 'ffio_open_dyn_packet_buf')
47 static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
48 .enc_name = "X-MP3-draft-00",
49 .codec_type = AVMEDIA_TYPE_AUDIO,
50 .codec_id = AV_CODEC_ID_MP3ADU,
53 /* statistics functions */
54 static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
56 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
58 handler->next= RTPFirstDynamicPayloadHandler;
59 RTPFirstDynamicPayloadHandler= handler;
62 void av_register_rtp_dynamic_payload_handlers(void)
64 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
65 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
66 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
67 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
68 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
69 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
70 ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
71 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
72 ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
73 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
74 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
75 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
76 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
77 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
78 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
79 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
80 ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);
82 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
83 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
85 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
86 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
87 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
88 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
90 ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
91 ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
92 ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
93 ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
96 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
97 enum AVMediaType codec_type)
99 RTPDynamicProtocolHandler *handler;
100 for (handler = RTPFirstDynamicPayloadHandler;
101 handler; handler = handler->next)
102 if (!av_strcasecmp(name, handler->enc_name) &&
103 codec_type == handler->codec_type)
108 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
109 enum AVMediaType codec_type)
111 RTPDynamicProtocolHandler *handler;
112 for (handler = RTPFirstDynamicPayloadHandler;
113 handler; handler = handler->next)
114 if (handler->static_payload_id && handler->static_payload_id == id &&
115 codec_type == handler->codec_type)
120 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
124 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
128 if (payload_len < 20) {
129 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
130 return AVERROR_INVALIDDATA;
133 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
134 s->last_rtcp_timestamp = AV_RB32(buf + 16);
135 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
136 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
137 if (!s->base_timestamp)
138 s->base_timestamp = s->last_rtcp_timestamp;
139 s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
153 #define RTP_SEQ_MOD (1<<16)
156 * called on parse open packet
158 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
160 memset(s, 0, sizeof(RTPStatistics));
161 s->max_seq= base_sequence;
166 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
168 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
173 s->bad_seq= RTP_SEQ_MOD + 1;
175 s->expected_prior= 0;
176 s->received_prior= 0;
182 * returns 1 if we should handle this packet.
184 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
186 uint16_t udelta= seq - s->max_seq;
187 const int MAX_DROPOUT= 3000;
188 const int MAX_MISORDER = 100;
189 const int MIN_SEQUENTIAL = 2;
191 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
194 if(seq==s->max_seq + 1) {
197 if(s->probation==0) {
198 rtp_init_sequence(s, seq);
203 s->probation= MIN_SEQUENTIAL - 1;
206 } else if (udelta < MAX_DROPOUT) {
207 // in order, with permissible gap
208 if(seq < s->max_seq) {
209 //sequence number wrapped; count antother 64k cycles
210 s->cycles += RTP_SEQ_MOD;
213 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
214 // sequence made a large jump...
215 if(seq==s->bad_seq) {
216 // two sequential packets-- assume that the other side 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) fraction= 0;
280 else fraction = (lost_interval<<8)/expected_interval;
282 fraction= (fraction<<24) | lost;
284 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
285 avio_wb32(pb, extended_max); /* max sequence received */
286 avio_wb32(pb, stats->jitter>>4); /* jitter */
288 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
290 avio_wb32(pb, 0); /* last SR timestamp */
291 avio_wb32(pb, 0); /* delay since last SR */
293 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
294 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
296 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
297 avio_wb32(pb, delay_since_last); /* delay since last SR */
301 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
302 avio_w8(pb, RTCP_SDES);
303 len = strlen(s->hostname);
304 avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
305 avio_wb32(pb, s->ssrc + 1);
308 avio_write(pb, s->hostname, len);
310 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);
366 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
367 * MPEG2TS streams to indicate that they should be demuxed inside the
368 * rtp demux (otherwise AV_CODEC_ID_MPEG2TS packets are returned)
370 RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
374 s = av_mallocz(sizeof(RTPDemuxContext));
377 s->payload_type = payload_type;
378 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
379 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
382 s->queue_size = queue_size;
383 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
384 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
385 s->ts = ff_mpegts_parse_open(s->ic);
391 switch(st->codec->codec_id) {
392 case AV_CODEC_ID_MPEG1VIDEO:
393 case AV_CODEC_ID_MPEG2VIDEO:
394 case AV_CODEC_ID_MP2:
395 case AV_CODEC_ID_MP3:
396 case AV_CODEC_ID_MPEG4:
397 case AV_CODEC_ID_H263:
398 case AV_CODEC_ID_H264:
399 st->need_parsing = AVSTREAM_PARSE_FULL;
401 case AV_CODEC_ID_VORBIS:
402 st->need_parsing = AVSTREAM_PARSE_HEADERS;
404 case AV_CODEC_ID_ADPCM_G722:
405 /* According to RFC 3551, the stream clock rate is 8000
406 * even if the sample rate is 16000. */
407 if (st->codec->sample_rate == 8000)
408 st->codec->sample_rate = 16000;
414 // needed to send back RTCP RR in RTSP sessions
416 gethostname(s->hostname, sizeof(s->hostname));
421 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. Normalizes time, if required, sets stream_index, etc.
431 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
433 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
434 return; /* Timestamp already set by depacketizer */
435 if (timestamp == RTP_NOTS_VALUE)
438 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
442 /* compute pts from timestamp with received ntp_time */
443 delta_timestamp = timestamp - s->last_rtcp_timestamp;
444 /* convert to the PTS timebase */
445 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);
446 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
451 if (!s->base_timestamp)
452 s->base_timestamp = timestamp;
453 /* assume that the difference is INT32_MIN < x < INT32_MAX, but allow the first timestamp to exceed INT32_MAX */
455 s->unwrapped_timestamp += timestamp;
457 s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
458 s->timestamp = timestamp;
459 pkt->pts = s->unwrapped_timestamp + s->range_start_offset - s->base_timestamp;
462 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
463 const uint8_t *buf, int len)
465 unsigned int ssrc, h;
466 int payload_type, seq, ret, flags = 0;
473 payload_type = buf[1] & 0x7f;
475 flags |= RTP_FLAG_MARKER;
476 seq = AV_RB16(buf + 2);
477 timestamp = AV_RB32(buf + 4);
478 ssrc = AV_RB32(buf + 8);
479 /* store the ssrc in the RTPDemuxContext */
482 /* NOTE: we can handle only one payload type */
483 if (s->payload_type != payload_type)
487 // only do something with this if all the rtp checks pass...
488 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
490 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
491 payload_type, seq, ((s->seq + 1) & 0xffff));
496 int padding = buf[len - 1];
497 if (len >= 12 + padding)
505 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
509 /* calculate the header extension length (stored as number
510 * of 32-bit words) */
511 ext = (AV_RB16(buf + 2) + 1) << 2;
515 // skip past RTP header extension
521 /* specific MPEG2TS demux support */
522 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
523 /* The only error that can be returned from ff_mpegts_parse_packet
524 * is "no more data to return from the provided buffer", so return
525 * AVERROR(EAGAIN) for all errors */
527 return AVERROR(EAGAIN);
529 s->read_buf_size = len - ret;
530 memcpy(s->buf, buf + ret, s->read_buf_size);
531 s->read_buf_index = 0;
535 } else if (s->parse_packet) {
536 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
537 s->st, pkt, ×tamp, buf, len, flags);
539 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
540 switch(st->codec->codec_id) {
541 case AV_CODEC_ID_MP2:
542 case AV_CODEC_ID_MP3:
543 /* better than nothing: skip mpeg audio RTP header */
549 av_new_packet(pkt, len);
550 memcpy(pkt->data, buf, len);
552 case AV_CODEC_ID_MPEG1VIDEO:
553 case AV_CODEC_ID_MPEG2VIDEO:
554 /* better than nothing: skip mpeg video RTP header */
567 av_new_packet(pkt, len);
568 memcpy(pkt->data, buf, len);
571 av_new_packet(pkt, len);
572 memcpy(pkt->data, buf, len);
576 pkt->stream_index = st->index;
579 // now perform timestamp things....
580 finalize_packet(s, pkt, timestamp);
585 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
588 RTPPacket *next = s->queue->next;
589 av_free(s->queue->buf);
598 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
600 uint16_t seq = AV_RB16(buf + 2);
601 RTPPacket *cur = s->queue, *prev = NULL, *packet;
603 /* Find the correct place in the queue to insert the packet */
605 int16_t diff = seq - cur->seq;
612 packet = av_mallocz(sizeof(*packet));
615 packet->recvtime = av_gettime();
627 static int has_next_packet(RTPDemuxContext *s)
629 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
632 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
634 return s->queue ? s->queue->recvtime : 0;
637 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
642 if (s->queue_len <= 0)
645 if (!has_next_packet(s))
646 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
647 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
649 /* Parse the first packet in the queue, and dequeue it */
650 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
651 next = s->queue->next;
652 av_free(s->queue->buf);
659 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
660 uint8_t **bufptr, int len)
662 uint8_t* buf = bufptr ? *bufptr : NULL;
668 /* If parsing of the previous packet actually returned 0 or an error,
669 * there's nothing more to be parsed from that packet, but we may have
670 * indicated that we can return the next enqueued packet. */
671 if (s->prev_ret <= 0)
672 return rtp_parse_queued_packet(s, pkt);
673 /* return the next packets, if any */
674 if(s->st && s->parse_packet) {
675 /* timestamp should be overwritten by parse_packet, if not,
676 * the packet is left with pts == AV_NOPTS_VALUE */
677 timestamp = RTP_NOTS_VALUE;
678 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
679 s->st, pkt, ×tamp, NULL, 0, flags);
680 finalize_packet(s, pkt, timestamp);
683 // TODO: Move to a dynamic packet handler (like above)
684 if (s->read_buf_index >= s->read_buf_size)
685 return AVERROR(EAGAIN);
686 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
687 s->read_buf_size - s->read_buf_index);
689 return AVERROR(EAGAIN);
690 s->read_buf_index += ret;
691 if (s->read_buf_index < s->read_buf_size)
701 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
703 if (RTP_PT_IS_RTCP(buf[1])) {
704 return rtcp_parse_packet(s, buf, len);
707 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
708 /* First packet, or no reordering */
709 return rtp_parse_packet_internal(s, pkt, buf, len);
711 uint16_t seq = AV_RB16(buf + 2);
712 int16_t diff = seq - s->seq;
714 /* Packet older than the previously emitted one, drop */
715 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
716 "RTP: dropping old packet received too late\n");
718 } else if (diff <= 1) {
720 rv = rtp_parse_packet_internal(s, pkt, buf, len);
723 /* Still missing some packet, enqueue this one. */
724 enqueue_packet(s, buf, len);
726 /* Return the first enqueued packet if the queue is full,
727 * even if we're missing something */
728 if (s->queue_len >= s->queue_size)
729 return rtp_parse_queued_packet(s, pkt);
736 * Parse an RTP or RTCP packet directly sent as a buffer.
737 * @param s RTP parse context.
738 * @param pkt returned packet
739 * @param bufptr pointer to the input buffer or NULL to read the next packets
740 * @param len buffer len
741 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
742 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
744 int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
745 uint8_t **bufptr, int len)
747 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
749 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
750 rv = rtp_parse_queued_packet(s, pkt);
751 return rv ? rv : has_next_packet(s);
754 void ff_rtp_parse_close(RTPDemuxContext *s)
756 ff_rtp_reset_packet_queue(s);
757 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
758 ff_mpegts_parse_close(s->ts);
763 int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
764 int (*parse_fmtp)(AVStream *stream,
765 PayloadContext *data,
766 char *attr, char *value))
771 int value_size = strlen(p) + 1;
773 if (!(value = av_malloc(value_size))) {
774 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
775 return AVERROR(ENOMEM);
778 // remove protocol identifier
779 while (*p && *p == ' ') p++; // strip spaces
780 while (*p && *p != ' ') p++; // eat protocol identifier
781 while (*p && *p == ' ') p++; // strip trailing spaces
783 while (ff_rtsp_next_attr_and_value(&p,
785 value, value_size)) {
787 res = parse_fmtp(stream, data, attr, value);
788 if (res < 0 && res != AVERROR_PATCHWELCOME) {