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 "libavcodec/get_bits.h"
32 #include "rtpdec_formats.h"
36 /* TODO: - add RTCP statistics reporting (should be optional).
38 - add support for h263/mpeg4 packetized output : IDEA: send a
39 buffer to 'rtp_write_packet' contains all the packets for ONE
40 frame. Each packet should have a four byte header containing
41 the length in big endian format (same trick as
42 'ffio_open_dyn_packet_buf')
45 static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
46 .enc_name = "X-MP3-draft-00",
47 .codec_type = AVMEDIA_TYPE_AUDIO,
48 .codec_id = CODEC_ID_MP3ADU,
51 /* statistics functions */
52 static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
54 void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
56 handler->next= RTPFirstDynamicPayloadHandler;
57 RTPFirstDynamicPayloadHandler= 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_h264_dynamic_handler);
69 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
70 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
71 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
72 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
73 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
74 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
75 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
76 ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);
78 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
79 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
81 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
82 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
83 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
84 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
87 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
88 enum AVMediaType codec_type)
90 RTPDynamicProtocolHandler *handler;
91 for (handler = RTPFirstDynamicPayloadHandler;
92 handler; handler = handler->next)
93 if (!strcasecmp(name, handler->enc_name) &&
94 codec_type == handler->codec_type)
99 RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
100 enum AVMediaType codec_type)
102 RTPDynamicProtocolHandler *handler;
103 for (handler = RTPFirstDynamicPayloadHandler;
104 handler; handler = handler->next)
105 if (handler->static_payload_id && handler->static_payload_id == id &&
106 codec_type == handler->codec_type)
111 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
118 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
119 return AVERROR_INVALIDDATA;
121 payload_len = (AV_RB16(buf + 2) + 1) * 4;
123 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
124 s->last_rtcp_timestamp = AV_RB32(buf + 16);
125 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
126 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
127 if (!s->base_timestamp)
128 s->base_timestamp = s->last_rtcp_timestamp;
129 s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
144 #define RTP_SEQ_MOD (1<<16)
147 * called on parse open packet
149 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
151 memset(s, 0, sizeof(RTPStatistics));
152 s->max_seq= base_sequence;
157 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
159 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
164 s->bad_seq= RTP_SEQ_MOD + 1;
166 s->expected_prior= 0;
167 s->received_prior= 0;
173 * returns 1 if we should handle this packet.
175 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
177 uint16_t udelta= seq - s->max_seq;
178 const int MAX_DROPOUT= 3000;
179 const int MAX_MISORDER = 100;
180 const int MIN_SEQUENTIAL = 2;
182 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
185 if(seq==s->max_seq + 1) {
188 if(s->probation==0) {
189 rtp_init_sequence(s, seq);
194 s->probation= MIN_SEQUENTIAL - 1;
197 } else if (udelta < MAX_DROPOUT) {
198 // in order, with permissible gap
199 if(seq < s->max_seq) {
200 //sequence number wrapped; count antother 64k cycles
201 s->cycles += RTP_SEQ_MOD;
204 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
205 // sequence made a large jump...
206 if(seq==s->bad_seq) {
207 // two sequential packets-- assume that the other side restarted without telling us; just resync.
208 rtp_init_sequence(s, seq);
210 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
214 // duplicate or reordered packet...
222 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
223 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
224 * never change. I left this in in case someone else can see a way. (rdm)
226 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
228 uint32_t transit= arrival_timestamp - sent_timestamp;
231 d= FFABS(transit - s->transit);
232 s->jitter += d - ((s->jitter + 8)>>4);
236 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
242 RTPStatistics *stats= &s->statistics;
244 uint32_t extended_max;
245 uint32_t expected_interval;
246 uint32_t received_interval;
247 uint32_t lost_interval;
250 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
252 if (!s->rtp_ctx || (count < 1))
255 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
256 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
257 s->octet_count += count;
258 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
260 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
263 s->last_octet_count = s->octet_count;
265 if (avio_open_dyn_buf(&pb) < 0)
269 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
270 avio_w8(pb, RTCP_RR);
271 avio_wb16(pb, 7); /* length in words - 1 */
272 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
273 avio_wb32(pb, s->ssrc + 1);
274 avio_wb32(pb, s->ssrc); // server SSRC
275 // some placeholders we should really fill...
277 extended_max= stats->cycles + stats->max_seq;
278 expected= extended_max - stats->base_seq + 1;
279 lost= expected - stats->received;
280 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
281 expected_interval= expected - stats->expected_prior;
282 stats->expected_prior= expected;
283 received_interval= stats->received - stats->received_prior;
284 stats->received_prior= stats->received;
285 lost_interval= expected_interval - received_interval;
286 if (expected_interval==0 || lost_interval<=0) fraction= 0;
287 else fraction = (lost_interval<<8)/expected_interval;
289 fraction= (fraction<<24) | lost;
291 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
292 avio_wb32(pb, extended_max); /* max sequence received */
293 avio_wb32(pb, stats->jitter>>4); /* jitter */
295 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
297 avio_wb32(pb, 0); /* last SR timestamp */
298 avio_wb32(pb, 0); /* delay since last SR */
300 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
301 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
303 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
304 avio_wb32(pb, delay_since_last); /* delay since last SR */
308 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
309 avio_w8(pb, RTCP_SDES);
310 len = strlen(s->hostname);
311 avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
312 avio_wb32(pb, s->ssrc);
315 avio_write(pb, s->hostname, len);
317 for (len = (6 + len) % 4; len % 4; len++) {
322 len = avio_close_dyn_buf(pb, &buf);
323 if ((len > 0) && buf) {
324 int av_unused result;
325 av_dlog(s->ic, "sending %d bytes of RR\n", len);
326 result= ffurl_write(s->rtp_ctx, buf, len);
327 av_dlog(s->ic, "result from ffurl_write: %d\n", result);
333 void rtp_send_punch_packets(URLContext* rtp_handle)
339 /* Send a small RTP packet */
340 if (avio_open_dyn_buf(&pb) < 0)
343 avio_w8(pb, (RTP_VERSION << 6));
344 avio_w8(pb, 0); /* Payload type */
345 avio_wb16(pb, 0); /* Seq */
346 avio_wb32(pb, 0); /* Timestamp */
347 avio_wb32(pb, 0); /* SSRC */
350 len = avio_close_dyn_buf(pb, &buf);
351 if ((len > 0) && buf)
352 ffurl_write(rtp_handle, buf, len);
355 /* Send a minimal RTCP RR */
356 if (avio_open_dyn_buf(&pb) < 0)
359 avio_w8(pb, (RTP_VERSION << 6));
360 avio_w8(pb, RTCP_RR); /* receiver report */
361 avio_wb16(pb, 1); /* length in words - 1 */
362 avio_wb32(pb, 0); /* our own SSRC */
365 len = avio_close_dyn_buf(pb, &buf);
366 if ((len > 0) && buf)
367 ffurl_write(rtp_handle, buf, len);
373 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
374 * MPEG2TS streams to indicate that they should be demuxed inside the
375 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
377 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
381 s = av_mallocz(sizeof(RTPDemuxContext));
384 s->payload_type = payload_type;
385 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
386 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
389 s->queue_size = queue_size;
390 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
391 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
392 s->ts = ff_mpegts_parse_open(s->ic);
398 switch(st->codec->codec_id) {
399 case CODEC_ID_MPEG1VIDEO:
400 case CODEC_ID_MPEG2VIDEO:
406 st->need_parsing = AVSTREAM_PARSE_FULL;
408 case CODEC_ID_ADPCM_G722:
409 /* According to RFC 3551, the stream clock rate is 8000
410 * even if the sample rate is 16000. */
411 if (st->codec->sample_rate == 8000)
412 st->codec->sample_rate = 16000;
418 // needed to send back RTCP RR in RTSP sessions
420 gethostname(s->hostname, sizeof(s->hostname));
425 rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
426 RTPDynamicProtocolHandler *handler)
428 s->dynamic_protocol_context = ctx;
429 s->parse_packet = handler->parse_packet;
433 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
435 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
437 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
438 return; /* Timestamp already set by depacketizer */
439 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
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, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);
447 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
451 if (timestamp == RTP_NOTS_VALUE)
453 if (!s->base_timestamp)
454 s->base_timestamp = timestamp;
455 pkt->pts = s->range_start_offset + timestamp - s->base_timestamp;
458 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
459 const uint8_t *buf, int len)
461 unsigned int ssrc, h;
462 int payload_type, seq, ret, flags = 0;
469 payload_type = buf[1] & 0x7f;
471 flags |= RTP_FLAG_MARKER;
472 seq = AV_RB16(buf + 2);
473 timestamp = AV_RB32(buf + 4);
474 ssrc = AV_RB32(buf + 8);
475 /* store the ssrc in the RTPDemuxContext */
478 /* NOTE: we can handle only one payload type */
479 if (s->payload_type != payload_type)
483 // only do something with this if all the rtp checks pass...
484 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
486 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
487 payload_type, seq, ((s->seq + 1) & 0xffff));
492 int padding = buf[len - 1];
493 if (len >= 12 + padding)
501 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
505 /* calculate the header extension length (stored as number
506 * of 32-bit words) */
507 ext = (AV_RB16(buf + 2) + 1) << 2;
511 // skip past RTP header extension
517 /* specific MPEG2TS demux support */
518 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
519 /* The only error that can be returned from ff_mpegts_parse_packet
520 * is "no more data to return from the provided buffer", so return
521 * AVERROR(EAGAIN) for all errors */
523 return AVERROR(EAGAIN);
525 s->read_buf_size = len - ret;
526 memcpy(s->buf, buf + ret, s->read_buf_size);
527 s->read_buf_index = 0;
531 } else if (s->parse_packet) {
532 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
533 s->st, pkt, ×tamp, buf, len, flags);
535 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
536 switch(st->codec->codec_id) {
539 /* better than nothing: skip mpeg audio RTP header */
545 av_new_packet(pkt, len);
546 memcpy(pkt->data, buf, len);
548 case CODEC_ID_MPEG1VIDEO:
549 case CODEC_ID_MPEG2VIDEO:
550 /* better than nothing: skip mpeg video RTP header */
563 av_new_packet(pkt, len);
564 memcpy(pkt->data, buf, len);
567 av_new_packet(pkt, len);
568 memcpy(pkt->data, buf, len);
572 pkt->stream_index = st->index;
575 // now perform timestamp things....
576 finalize_packet(s, pkt, timestamp);
581 void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
584 RTPPacket *next = s->queue->next;
585 av_free(s->queue->buf);
594 static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
596 uint16_t seq = AV_RB16(buf + 2);
597 RTPPacket *cur = s->queue, *prev = NULL, *packet;
599 /* Find the correct place in the queue to insert the packet */
601 int16_t diff = seq - cur->seq;
608 packet = av_mallocz(sizeof(*packet));
611 packet->recvtime = av_gettime();
623 static int has_next_packet(RTPDemuxContext *s)
625 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
628 int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
630 return s->queue ? s->queue->recvtime : 0;
633 static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
638 if (s->queue_len <= 0)
641 if (!has_next_packet(s))
642 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
643 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
645 /* Parse the first packet in the queue, and dequeue it */
646 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
647 next = s->queue->next;
648 av_free(s->queue->buf);
655 static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
656 uint8_t **bufptr, int len)
658 uint8_t* buf = bufptr ? *bufptr : NULL;
664 /* If parsing of the previous packet actually returned 0 or an error,
665 * there's nothing more to be parsed from that packet, but we may have
666 * indicated that we can return the next enqueued packet. */
667 if (s->prev_ret <= 0)
668 return rtp_parse_queued_packet(s, pkt);
669 /* return the next packets, if any */
670 if(s->st && s->parse_packet) {
671 /* timestamp should be overwritten by parse_packet, if not,
672 * the packet is left with pts == AV_NOPTS_VALUE */
673 timestamp = RTP_NOTS_VALUE;
674 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
675 s->st, pkt, ×tamp, NULL, 0, flags);
676 finalize_packet(s, pkt, timestamp);
679 // TODO: Move to a dynamic packet handler (like above)
680 if (s->read_buf_index >= s->read_buf_size)
681 return AVERROR(EAGAIN);
682 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
683 s->read_buf_size - s->read_buf_index);
685 return AVERROR(EAGAIN);
686 s->read_buf_index += ret;
687 if (s->read_buf_index < s->read_buf_size)
697 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
699 if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
700 return rtcp_parse_packet(s, buf, len);
703 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
704 /* First packet, or no reordering */
705 return rtp_parse_packet_internal(s, pkt, buf, len);
707 uint16_t seq = AV_RB16(buf + 2);
708 int16_t diff = seq - s->seq;
710 /* Packet older than the previously emitted one, drop */
711 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
712 "RTP: dropping old packet received too late\n");
714 } else if (diff <= 1) {
716 rv = rtp_parse_packet_internal(s, pkt, buf, len);
719 /* Still missing some packet, enqueue this one. */
720 enqueue_packet(s, buf, len);
722 /* Return the first enqueued packet if the queue is full,
723 * even if we're missing something */
724 if (s->queue_len >= s->queue_size)
725 return rtp_parse_queued_packet(s, pkt);
732 * Parse an RTP or RTCP packet directly sent as a buffer.
733 * @param s RTP parse context.
734 * @param pkt returned packet
735 * @param bufptr pointer to the input buffer or NULL to read the next packets
736 * @param len buffer len
737 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
738 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
740 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
741 uint8_t **bufptr, int len)
743 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
745 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
746 rv = rtp_parse_queued_packet(s, pkt);
747 return rv ? rv : has_next_packet(s);
750 void rtp_parse_close(RTPDemuxContext *s)
752 ff_rtp_reset_packet_queue(s);
753 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
754 ff_mpegts_parse_close(s->ts);
759 int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
760 int (*parse_fmtp)(AVStream *stream,
761 PayloadContext *data,
762 char *attr, char *value))
767 int value_size = strlen(p) + 1;
769 if (!(value = av_malloc(value_size))) {
770 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
771 return AVERROR(ENOMEM);
774 // remove protocol identifier
775 while (*p && *p == ' ') p++; // strip spaces
776 while (*p && *p != ' ') p++; // eat protocol identifier
777 while (*p && *p == ' ') p++; // strip trailing spaces
779 while (ff_rtsp_next_attr_and_value(&p,
781 value, value_size)) {
783 res = parse_fmtp(stream, data, attr, value);
784 if (res < 0 && res != AVERROR_PATCHWELCOME) {