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 "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 ff_realmedia_mp3_dynamic_handler = {
47 .enc_name = "X-MP3-draft-00",
48 .codec_type = AVMEDIA_TYPE_AUDIO,
49 .codec_id = CODEC_ID_MP3ADU,
52 /* statistics functions */
53 static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
55 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
57 handler->next= RTPFirstDynamicPayloadHandler;
58 RTPFirstDynamicPayloadHandler= handler;
61 void av_register_rtp_dynamic_payload_handlers(void)
63 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
64 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
65 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
66 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
67 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
68 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
69 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
70 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
71 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
72 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
73 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
74 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
75 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
76 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
77 ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);
79 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
80 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
82 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
83 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
84 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
85 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
87 ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
88 ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
89 ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
90 ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
93 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
94 enum AVMediaType codec_type)
96 RTPDynamicProtocolHandler *handler;
97 for (handler = RTPFirstDynamicPayloadHandler;
98 handler; handler = handler->next)
99 if (!av_strcasecmp(name, handler->enc_name) &&
100 codec_type == handler->codec_type)
105 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
106 enum AVMediaType codec_type)
108 RTPDynamicProtocolHandler *handler;
109 for (handler = RTPFirstDynamicPayloadHandler;
110 handler; handler = handler->next)
111 if (handler->static_payload_id && handler->static_payload_id == id &&
112 codec_type == handler->codec_type)
117 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
121 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
125 if (payload_len < 20) {
126 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
127 return AVERROR_INVALIDDATA;
130 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
131 s->last_rtcp_timestamp = AV_RB32(buf + 16);
132 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
133 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
134 if (!s->base_timestamp)
135 s->base_timestamp = s->last_rtcp_timestamp;
136 s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
150 #define RTP_SEQ_MOD (1<<16)
153 * called on parse open packet
155 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
157 memset(s, 0, sizeof(RTPStatistics));
158 s->max_seq= base_sequence;
163 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
165 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
170 s->bad_seq= RTP_SEQ_MOD + 1;
172 s->expected_prior= 0;
173 s->received_prior= 0;
179 * returns 1 if we should handle this packet.
181 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
183 uint16_t udelta= seq - s->max_seq;
184 const int MAX_DROPOUT= 3000;
185 const int MAX_MISORDER = 100;
186 const int MIN_SEQUENTIAL = 2;
188 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
191 if(seq==s->max_seq + 1) {
194 if(s->probation==0) {
195 rtp_init_sequence(s, seq);
200 s->probation= MIN_SEQUENTIAL - 1;
203 } else if (udelta < MAX_DROPOUT) {
204 // in order, with permissible gap
205 if(seq < s->max_seq) {
206 //sequence number wrapped; count antother 64k cycles
207 s->cycles += RTP_SEQ_MOD;
210 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
211 // sequence made a large jump...
212 if(seq==s->bad_seq) {
213 // two sequential packets-- assume that the other side restarted without telling us; just resync.
214 rtp_init_sequence(s, seq);
216 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
220 // duplicate or reordered packet...
226 int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
232 RTPStatistics *stats= &s->statistics;
234 uint32_t extended_max;
235 uint32_t expected_interval;
236 uint32_t received_interval;
237 uint32_t lost_interval;
240 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
242 if (!s->rtp_ctx || (count < 1))
245 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
246 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
247 s->octet_count += count;
248 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
250 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
253 s->last_octet_count = s->octet_count;
255 if (avio_open_dyn_buf(&pb) < 0)
259 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
260 avio_w8(pb, RTCP_RR);
261 avio_wb16(pb, 7); /* length in words - 1 */
262 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
263 avio_wb32(pb, s->ssrc + 1);
264 avio_wb32(pb, s->ssrc); // server SSRC
265 // some placeholders we should really fill...
267 extended_max= stats->cycles + stats->max_seq;
268 expected= extended_max - stats->base_seq + 1;
269 lost= expected - stats->received;
270 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
271 expected_interval= expected - stats->expected_prior;
272 stats->expected_prior= expected;
273 received_interval= stats->received - stats->received_prior;
274 stats->received_prior= stats->received;
275 lost_interval= expected_interval - received_interval;
276 if (expected_interval==0 || lost_interval<=0) fraction= 0;
277 else fraction = (lost_interval<<8)/expected_interval;
279 fraction= (fraction<<24) | lost;
281 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
282 avio_wb32(pb, extended_max); /* max sequence received */
283 avio_wb32(pb, stats->jitter>>4); /* jitter */
285 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
287 avio_wb32(pb, 0); /* last SR timestamp */
288 avio_wb32(pb, 0); /* delay since last SR */
290 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
291 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
293 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
294 avio_wb32(pb, delay_since_last); /* delay since last SR */
298 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
299 avio_w8(pb, RTCP_SDES);
300 len = strlen(s->hostname);
301 avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
302 avio_wb32(pb, s->ssrc);
305 avio_write(pb, s->hostname, len);
307 for (len = (6 + len) % 4; len % 4; len++) {
312 len = avio_close_dyn_buf(pb, &buf);
313 if ((len > 0) && buf) {
314 int av_unused result;
315 av_dlog(s->ic, "sending %d bytes of RR\n", len);
316 result= ffurl_write(s->rtp_ctx, buf, len);
317 av_dlog(s->ic, "result from ffurl_write: %d\n", result);
323 void ff_rtp_send_punch_packets(URLContext* rtp_handle)
329 /* Send a small RTP packet */
330 if (avio_open_dyn_buf(&pb) < 0)
333 avio_w8(pb, (RTP_VERSION << 6));
334 avio_w8(pb, 0); /* Payload type */
335 avio_wb16(pb, 0); /* Seq */
336 avio_wb32(pb, 0); /* Timestamp */
337 avio_wb32(pb, 0); /* SSRC */
340 len = avio_close_dyn_buf(pb, &buf);
341 if ((len > 0) && buf)
342 ffurl_write(rtp_handle, buf, len);
345 /* Send a minimal RTCP RR */
346 if (avio_open_dyn_buf(&pb) < 0)
349 avio_w8(pb, (RTP_VERSION << 6));
350 avio_w8(pb, RTCP_RR); /* receiver report */
351 avio_wb16(pb, 1); /* length in words - 1 */
352 avio_wb32(pb, 0); /* our own SSRC */
355 len = avio_close_dyn_buf(pb, &buf);
356 if ((len > 0) && buf)
357 ffurl_write(rtp_handle, buf, len);
363 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
364 * MPEG2TS streams to indicate that they should be demuxed inside the
365 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
367 RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
371 s = av_mallocz(sizeof(RTPDemuxContext));
374 s->payload_type = payload_type;
375 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
376 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
379 s->queue_size = queue_size;
380 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
381 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
382 s->ts = ff_mpegts_parse_open(s->ic);
388 switch(st->codec->codec_id) {
389 case CODEC_ID_MPEG1VIDEO:
390 case CODEC_ID_MPEG2VIDEO:
396 st->need_parsing = AVSTREAM_PARSE_FULL;
398 case CODEC_ID_ADPCM_G722:
399 /* According to RFC 3551, the stream clock rate is 8000
400 * even if the sample rate is 16000. */
401 if (st->codec->sample_rate == 8000)
402 st->codec->sample_rate = 16000;
408 // needed to send back RTCP RR in RTSP sessions
410 gethostname(s->hostname, sizeof(s->hostname));
415 ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
416 RTPDynamicProtocolHandler *handler)
418 s->dynamic_protocol_context = ctx;
419 s->parse_packet = handler->parse_packet;
423 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
425 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
427 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
428 return; /* Timestamp already set by depacketizer */
429 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
433 /* compute pts from timestamp with received ntp_time */
434 delta_timestamp = timestamp - s->last_rtcp_timestamp;
435 /* convert to the PTS timebase */
436 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);
437 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
441 if (timestamp == RTP_NOTS_VALUE)
443 if (!s->base_timestamp)
444 s->base_timestamp = timestamp;
445 pkt->pts = s->range_start_offset + timestamp - s->base_timestamp;
448 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
449 const uint8_t *buf, int len)
451 unsigned int ssrc, h;
452 int payload_type, seq, ret, flags = 0;
459 payload_type = buf[1] & 0x7f;
461 flags |= RTP_FLAG_MARKER;
462 seq = AV_RB16(buf + 2);
463 timestamp = AV_RB32(buf + 4);
464 ssrc = AV_RB32(buf + 8);
465 /* store the ssrc in the RTPDemuxContext */
468 /* NOTE: we can handle only one payload type */
469 if (s->payload_type != payload_type)
473 // only do something with this if all the rtp checks pass...
474 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
476 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
477 payload_type, seq, ((s->seq + 1) & 0xffff));
482 int padding = buf[len - 1];
483 if (len >= 12 + padding)
491 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
495 /* calculate the header extension length (stored as number
496 * of 32-bit words) */
497 ext = (AV_RB16(buf + 2) + 1) << 2;
501 // skip past RTP header extension
507 /* specific MPEG2TS demux support */
508 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
509 /* The only error that can be returned from ff_mpegts_parse_packet
510 * is "no more data to return from the provided buffer", so return
511 * AVERROR(EAGAIN) for all errors */
513 return AVERROR(EAGAIN);
515 s->read_buf_size = len - ret;
516 memcpy(s->buf, buf + ret, s->read_buf_size);
517 s->read_buf_index = 0;
521 } else if (s->parse_packet) {
522 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
523 s->st, pkt, ×tamp, buf, len, flags);
525 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
526 switch(st->codec->codec_id) {
529 /* better than nothing: skip mpeg audio RTP header */
535 av_new_packet(pkt, len);
536 memcpy(pkt->data, buf, len);
538 case CODEC_ID_MPEG1VIDEO:
539 case CODEC_ID_MPEG2VIDEO:
540 /* better than nothing: skip mpeg video RTP header */
553 av_new_packet(pkt, len);
554 memcpy(pkt->data, buf, len);
557 av_new_packet(pkt, len);
558 memcpy(pkt->data, buf, len);
562 pkt->stream_index = st->index;
565 // now perform timestamp things....
566 finalize_packet(s, pkt, timestamp);
571 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
574 RTPPacket *next = s->queue->next;
575 av_free(s->queue->buf);
584 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
586 uint16_t seq = AV_RB16(buf + 2);
587 RTPPacket *cur = s->queue, *prev = NULL, *packet;
589 /* Find the correct place in the queue to insert the packet */
591 int16_t diff = seq - cur->seq;
598 packet = av_mallocz(sizeof(*packet));
601 packet->recvtime = av_gettime();
613 static int has_next_packet(RTPDemuxContext *s)
615 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
618 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
620 return s->queue ? s->queue->recvtime : 0;
623 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
628 if (s->queue_len <= 0)
631 if (!has_next_packet(s))
632 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
633 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
635 /* Parse the first packet in the queue, and dequeue it */
636 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
637 next = s->queue->next;
638 av_free(s->queue->buf);
645 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
646 uint8_t **bufptr, int len)
648 uint8_t* buf = bufptr ? *bufptr : NULL;
654 /* If parsing of the previous packet actually returned 0 or an error,
655 * there's nothing more to be parsed from that packet, but we may have
656 * indicated that we can return the next enqueued packet. */
657 if (s->prev_ret <= 0)
658 return rtp_parse_queued_packet(s, pkt);
659 /* return the next packets, if any */
660 if(s->st && s->parse_packet) {
661 /* timestamp should be overwritten by parse_packet, if not,
662 * the packet is left with pts == AV_NOPTS_VALUE */
663 timestamp = RTP_NOTS_VALUE;
664 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
665 s->st, pkt, ×tamp, NULL, 0, flags);
666 finalize_packet(s, pkt, timestamp);
669 // TODO: Move to a dynamic packet handler (like above)
670 if (s->read_buf_index >= s->read_buf_size)
671 return AVERROR(EAGAIN);
672 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
673 s->read_buf_size - s->read_buf_index);
675 return AVERROR(EAGAIN);
676 s->read_buf_index += ret;
677 if (s->read_buf_index < s->read_buf_size)
687 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
689 if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
690 return rtcp_parse_packet(s, buf, len);
693 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
694 /* First packet, or no reordering */
695 return rtp_parse_packet_internal(s, pkt, buf, len);
697 uint16_t seq = AV_RB16(buf + 2);
698 int16_t diff = seq - s->seq;
700 /* Packet older than the previously emitted one, drop */
701 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
702 "RTP: dropping old packet received too late\n");
704 } else if (diff <= 1) {
706 rv = rtp_parse_packet_internal(s, pkt, buf, len);
709 /* Still missing some packet, enqueue this one. */
710 enqueue_packet(s, buf, len);
712 /* Return the first enqueued packet if the queue is full,
713 * even if we're missing something */
714 if (s->queue_len >= s->queue_size)
715 return rtp_parse_queued_packet(s, pkt);
722 * Parse an RTP or RTCP packet directly sent as a buffer.
723 * @param s RTP parse context.
724 * @param pkt returned packet
725 * @param bufptr pointer to the input buffer or NULL to read the next packets
726 * @param len buffer len
727 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
728 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
730 int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
731 uint8_t **bufptr, int len)
733 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
735 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
736 rv = rtp_parse_queued_packet(s, pkt);
737 return rv ? rv : has_next_packet(s);
740 void ff_rtp_parse_close(RTPDemuxContext *s)
742 ff_rtp_reset_packet_queue(s);
743 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
744 ff_mpegts_parse_close(s->ts);
749 int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
750 int (*parse_fmtp)(AVStream *stream,
751 PayloadContext *data,
752 char *attr, char *value))
757 int value_size = strlen(p) + 1;
759 if (!(value = av_malloc(value_size))) {
760 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
761 return AVERROR(ENOMEM);
764 // remove protocol identifier
765 while (*p && *p == ' ') p++; // strip spaces
766 while (*p && *p != ' ') p++; // eat protocol identifier
767 while (*p && *p == ' ') p++; // strip trailing spaces
769 while (ff_rtsp_next_attr_and_value(&p,
771 value, value_size)) {
773 res = parse_fmtp(stream, data, attr, value);
774 if (res < 0 && res != AVERROR_PATCHWELCOME) {