2 * RTP input/output format
3 * Copyright (c) 2002 Fabrice Bellard.
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Lesser General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Lesser General Public License for more details.
15 * You should have received a copy of the GNU Lesser General Public
16 * License along with this library; if not, write to the Free Software
17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
22 #include <sys/types.h>
23 #include <sys/socket.h>
24 #include <netinet/in.h>
26 # include <arpa/inet.h>
28 # include "barpainet.h"
35 /* TODO: - add RTCP statistics reporting (should be optional).
37 - add support for h263/mpeg4 packetized output : IDEA: send a
38 buffer to 'rtp_write_packet' contains all the packets for ONE
39 frame. Each packet should have a four byte header containing
40 the length in big endian format (same trick as
41 'url_open_dyn_packet_buf')
46 #define RTP_MAX_SDES 256 /* maximum text length for SDES */
48 /* RTCP paquets use 0.5 % of the bandwidth */
49 #define RTCP_TX_RATIO_NUM 5
50 #define RTCP_TX_RATIO_DEN 1000
80 RTP_PT_S16BE_STEREO = 10,
81 RTP_PT_S16BE_MONO = 11,
82 RTP_PT_MPEGAUDIO = 14,
85 RTP_PT_MPEGVIDEO = 32,
87 RTP_PT_H263 = 34, /* old H263 encapsulation */
91 typedef struct RTPContext {
96 uint32_t base_timestamp;
97 uint32_t cur_timestamp;
99 /* rtcp sender statistics receive */
100 int64_t last_rtcp_ntp_time;
101 int64_t first_rtcp_ntp_time;
102 uint32_t last_rtcp_timestamp;
103 /* rtcp sender statistics */
104 unsigned int packet_count;
105 unsigned int octet_count;
106 unsigned int last_octet_count;
108 /* buffer for output */
109 uint8_t buf[RTP_MAX_PACKET_LENGTH];
113 int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
115 switch(payload_type) {
117 codec->codec_id = CODEC_ID_PCM_MULAW;
119 codec->sample_rate = 8000;
122 codec->codec_id = CODEC_ID_PCM_ALAW;
124 codec->sample_rate = 8000;
126 case RTP_PT_S16BE_STEREO:
127 codec->codec_id = CODEC_ID_PCM_S16BE;
129 codec->sample_rate = 44100;
131 case RTP_PT_S16BE_MONO:
132 codec->codec_id = CODEC_ID_PCM_S16BE;
134 codec->sample_rate = 44100;
136 case RTP_PT_MPEGAUDIO:
137 codec->codec_id = CODEC_ID_MP2;
140 codec->codec_id = CODEC_ID_MJPEG;
142 case RTP_PT_MPEGVIDEO:
143 codec->codec_id = CODEC_ID_MPEG1VIDEO;
151 /* return < 0 if unknown payload type */
152 int rtp_get_payload_type(AVCodecContext *codec)
156 /* compute the payload type */
158 switch(codec->codec_id) {
159 case CODEC_ID_PCM_MULAW:
160 payload_type = RTP_PT_ULAW;
162 case CODEC_ID_PCM_ALAW:
163 payload_type = RTP_PT_ALAW;
165 case CODEC_ID_PCM_S16BE:
166 if (codec->channels == 1) {
167 payload_type = RTP_PT_S16BE_MONO;
168 } else if (codec->channels == 2) {
169 payload_type = RTP_PT_S16BE_STEREO;
173 case CODEC_ID_MP3LAME:
174 payload_type = RTP_PT_MPEGAUDIO;
177 payload_type = RTP_PT_JPEG;
179 case CODEC_ID_MPEG1VIDEO:
180 payload_type = RTP_PT_MPEGVIDEO;
188 static inline uint32_t decode_be32(const uint8_t *p)
190 return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
193 static inline uint64_t decode_be64(const uint8_t *p)
195 return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4);
198 static int rtcp_parse_packet(AVFormatContext *s1, const unsigned char *buf, int len)
200 RTPContext *s = s1->priv_data;
204 s->last_rtcp_ntp_time = decode_be64(buf + 8);
205 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
206 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
207 s->last_rtcp_timestamp = decode_be32(buf + 16);
212 * Parse an RTP packet directly sent as raw data. Can only be used if
213 * 'raw' is given as input file
214 * @param s1 media file context
215 * @param pkt returned packet
216 * @param buf input buffer
217 * @param len buffer len
218 * @return zero if no error.
220 int rtp_parse_packet(AVFormatContext *s1, AVPacket *pkt,
221 const unsigned char *buf, int len)
223 RTPContext *s = s1->priv_data;
224 unsigned int ssrc, h;
225 int payload_type, seq, delta_timestamp;
232 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
234 if (buf[1] >= 200 && buf[1] <= 204) {
235 rtcp_parse_packet(s1, buf, len);
238 payload_type = buf[1] & 0x7f;
239 seq = (buf[2] << 8) | buf[3];
240 timestamp = decode_be32(buf + 4);
241 ssrc = decode_be32(buf + 8);
243 if (s->payload_type < 0) {
244 s->payload_type = payload_type;
246 if (payload_type == RTP_PT_MPEG2TS) {
247 /* XXX: special case : not a single codec but a whole stream */
250 st = av_new_stream(s1, 0);
253 rtp_get_codec_info(&st->codec, payload_type);
257 /* NOTE: we can handle only one payload type */
258 if (s->payload_type != payload_type)
260 #if defined(DEBUG) || 1
261 if (seq != ((s->seq + 1) & 0xffff)) {
262 printf("RTP: PT=%02x: bad cseq %04x expected=%04x\n",
263 payload_type, seq, ((s->seq + 1) & 0xffff));
270 switch(st->codec.codec_id) {
272 /* better than nothing: skip mpeg audio RTP header */
275 h = decode_be32(buf);
278 av_new_packet(pkt, len);
279 memcpy(pkt->data, buf, len);
281 case CODEC_ID_MPEG1VIDEO:
282 /* better than nothing: skip mpeg audio RTP header */
285 h = decode_be32(buf);
295 av_new_packet(pkt, len);
296 memcpy(pkt->data, buf, len);
299 av_new_packet(pkt, len);
300 memcpy(pkt->data, buf, len);
304 switch(st->codec.codec_id) {
306 case CODEC_ID_MPEG1VIDEO:
307 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
309 /* XXX: is it really necessary to unify the timestamp base ? */
310 /* compute pts from timestamp with received ntp_time */
311 delta_timestamp = timestamp - s->last_rtcp_timestamp;
312 /* convert to 90 kHz without overflow */
313 addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
314 addend = (addend * 5625) >> 14;
315 pkt->pts = addend + delta_timestamp;
319 /* no timestamp info yet */
325 static int rtp_read_header(AVFormatContext *s1,
326 AVFormatParameters *ap)
328 RTPContext *s = s1->priv_data;
329 s->payload_type = -1;
330 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
331 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
335 static int rtp_read_packet(AVFormatContext *s1, AVPacket *pkt)
337 char buf[RTP_MAX_PACKET_LENGTH];
340 /* XXX: needs a better API for packet handling ? */
342 ret = url_read(url_fileno(&s1->pb), buf, sizeof(buf));
345 if (rtp_parse_packet(s1, pkt, buf, ret) == 0)
351 static int rtp_read_close(AVFormatContext *s1)
353 // RTPContext *s = s1->priv_data;
357 static int rtp_probe(AVProbeData *p)
359 if (strstart(p->filename, "rtp://", NULL))
360 return AVPROBE_SCORE_MAX;
366 static int rtp_write_header(AVFormatContext *s1)
368 RTPContext *s = s1->priv_data;
369 int payload_type, max_packet_size;
372 if (s1->nb_streams != 1)
376 payload_type = rtp_get_payload_type(&st->codec);
377 if (payload_type < 0)
378 payload_type = RTP_PT_PRIVATE; /* private payload type */
379 s->payload_type = payload_type;
381 s->base_timestamp = random();
382 s->timestamp = s->base_timestamp;
386 max_packet_size = url_fget_max_packet_size(&s1->pb);
387 if (max_packet_size <= 12)
389 s->max_payload_size = max_packet_size - 12;
391 switch(st->codec.codec_id) {
393 case CODEC_ID_MP3LAME:
394 s->buf_ptr = s->buf + 4;
395 s->cur_timestamp = 0;
397 case CODEC_ID_MPEG1VIDEO:
398 s->cur_timestamp = 0;
408 /* send an rtcp sender report packet */
409 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
411 RTPContext *s = s1->priv_data;
413 printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
415 put_byte(&s1->pb, (RTP_VERSION << 6));
416 put_byte(&s1->pb, 200);
417 put_be16(&s1->pb, 6); /* length in words - 1 */
418 put_be32(&s1->pb, s->ssrc);
419 put_be64(&s1->pb, ntp_time);
420 put_be32(&s1->pb, s->timestamp);
421 put_be32(&s1->pb, s->packet_count);
422 put_be32(&s1->pb, s->octet_count);
423 put_flush_packet(&s1->pb);
426 /* send an rtp packet. sequence number is incremented, but the caller
427 must update the timestamp itself */
428 static void rtp_send_data(AVFormatContext *s1, uint8_t *buf1, int len)
430 RTPContext *s = s1->priv_data;
433 printf("rtp_send_data size=%d\n", len);
436 /* build the RTP header */
437 put_byte(&s1->pb, (RTP_VERSION << 6));
438 put_byte(&s1->pb, s->payload_type & 0x7f);
439 put_be16(&s1->pb, s->seq);
440 put_be32(&s1->pb, s->timestamp);
441 put_be32(&s1->pb, s->ssrc);
443 put_buffer(&s1->pb, buf1, len);
444 put_flush_packet(&s1->pb);
447 s->octet_count += len;
451 /* send an integer number of samples and compute time stamp and fill
452 the rtp send buffer before sending. */
453 static void rtp_send_samples(AVFormatContext *s1,
454 uint8_t *buf1, int size, int sample_size)
456 RTPContext *s = s1->priv_data;
457 int len, max_packet_size, n;
459 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
460 /* not needed, but who nows */
461 if ((size % sample_size) != 0)
464 len = (max_packet_size - (s->buf_ptr - s->buf));
469 memcpy(s->buf_ptr, buf1, len);
473 n = (s->buf_ptr - s->buf);
474 /* if buffer full, then send it */
475 if (n >= max_packet_size) {
476 rtp_send_data(s1, s->buf, n);
478 /* update timestamp */
479 s->timestamp += n / sample_size;
484 /* NOTE: we suppose that exactly one frame is given as argument here */
486 static void rtp_send_mpegaudio(AVFormatContext *s1,
487 uint8_t *buf1, int size)
489 RTPContext *s = s1->priv_data;
490 AVStream *st = s1->streams[0];
491 int len, count, max_packet_size;
493 max_packet_size = s->max_payload_size;
495 /* test if we must flush because not enough space */
496 len = (s->buf_ptr - s->buf);
497 if ((len + size) > max_packet_size) {
499 rtp_send_data(s1, s->buf, s->buf_ptr - s->buf);
500 s->buf_ptr = s->buf + 4;
501 /* 90 KHz time stamp */
502 s->timestamp = s->base_timestamp +
503 (s->cur_timestamp * 90000LL) / st->codec.sample_rate;
508 if (size > max_packet_size) {
509 /* big packet: fragment */
512 len = max_packet_size - 4;
515 /* build fragmented packet */
518 s->buf[2] = count >> 8;
520 memcpy(s->buf + 4, buf1, len);
521 rtp_send_data(s1, s->buf, len + 4);
527 if (s->buf_ptr == s->buf + 4) {
528 /* no fragmentation possible */
534 memcpy(s->buf_ptr, buf1, size);
537 s->cur_timestamp += st->codec.frame_size;
540 /* NOTE: a single frame must be passed with sequence header if
541 needed. XXX: use slices. */
542 static void rtp_send_mpegvideo(AVFormatContext *s1,
543 uint8_t *buf1, int size)
545 RTPContext *s = s1->priv_data;
546 AVStream *st = s1->streams[0];
547 int len, h, max_packet_size;
550 max_packet_size = s->max_payload_size;
553 /* XXX: more correct headers */
555 if (st->codec.sub_id == 2)
556 h |= 1 << 26; /* mpeg 2 indicator */
563 if (st->codec.sub_id == 2) {
571 len = max_packet_size - (q - s->buf);
575 memcpy(q, buf1, len);
578 /* 90 KHz time stamp */
579 s->timestamp = s->base_timestamp +
580 av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate);
581 rtp_send_data(s1, s->buf, q - s->buf);
589 static void rtp_send_raw(AVFormatContext *s1,
590 uint8_t *buf1, int size)
592 RTPContext *s = s1->priv_data;
593 AVStream *st = s1->streams[0];
594 int len, max_packet_size;
596 max_packet_size = s->max_payload_size;
599 len = max_packet_size;
603 /* 90 KHz time stamp */
604 s->timestamp = s->base_timestamp +
605 av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate);
606 rtp_send_data(s1, buf1, len);
614 /* write an RTP packet. 'buf1' must contain a single specific frame. */
615 static int rtp_write_packet(AVFormatContext *s1, int stream_index,
616 uint8_t *buf1, int size, int force_pts)
618 RTPContext *s = s1->priv_data;
619 AVStream *st = s1->streams[0];
624 printf("%d: write len=%d\n", stream_index, size);
627 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
628 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
630 if (s->first_packet || rtcp_bytes >= 28) {
631 /* compute NTP time */
632 /* XXX: 90 kHz timestamp hardcoded */
633 ntp_time = ((int64_t)force_pts << 28) / 5625;
634 rtcp_send_sr(s1, ntp_time);
635 s->last_octet_count = s->octet_count;
639 switch(st->codec.codec_id) {
640 case CODEC_ID_PCM_MULAW:
641 case CODEC_ID_PCM_ALAW:
642 case CODEC_ID_PCM_U8:
643 case CODEC_ID_PCM_S8:
644 rtp_send_samples(s1, buf1, size, 1 * st->codec.channels);
646 case CODEC_ID_PCM_U16BE:
647 case CODEC_ID_PCM_U16LE:
648 case CODEC_ID_PCM_S16BE:
649 case CODEC_ID_PCM_S16LE:
650 rtp_send_samples(s1, buf1, size, 2 * st->codec.channels);
653 case CODEC_ID_MP3LAME:
654 rtp_send_mpegaudio(s1, buf1, size);
656 case CODEC_ID_MPEG1VIDEO:
657 rtp_send_mpegvideo(s1, buf1, size);
660 /* better than nothing : send the codec raw data */
661 rtp_send_raw(s1, buf1, size);
667 static int rtp_write_trailer(AVFormatContext *s1)
669 // RTPContext *s = s1->priv_data;
673 AVInputFormat rtp_demux = {
681 .flags = AVFMT_NOHEADER,
684 AVOutputFormat rtp_mux = {
699 av_register_output_format(&rtp_mux);
700 av_register_input_format(&rtp_demux);