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>
25 #include <arpa/inet.h>
31 /* TODO: - add RTCP statistics reporting (should be optional).
33 - add support for h263/mpeg4 packetized output : IDEA: send a
34 buffer to 'rtp_write_packet' contains all the packets for ONE
35 frame. Each packet should have a four byte header containing
36 the length in big endian format (same trick as
37 'url_open_dyn_packet_buf')
42 #define RTP_MAX_SDES 256 /* maximum text length for SDES */
44 /* RTCP paquets use 0.5 % of the bandwidth */
45 #define RTCP_TX_RATIO_NUM 5
46 #define RTCP_TX_RATIO_DEN 1000
76 RTP_PT_S16BE_STEREO = 10,
77 RTP_PT_S16BE_MONO = 11,
78 RTP_PT_MPEGAUDIO = 14,
81 RTP_PT_MPEGVIDEO = 32,
83 RTP_PT_H263 = 34, /* old H263 encapsulation */
86 typedef struct RTPContext {
91 UINT32 base_timestamp;
94 /* rtcp sender statistics receive */
95 INT64 last_rtcp_ntp_time;
96 UINT32 last_rtcp_timestamp;
97 /* rtcp sender statistics */
98 unsigned int packet_count;
99 unsigned int octet_count;
100 unsigned int last_octet_count;
102 /* buffer for output */
103 UINT8 buf[RTP_MAX_PACKET_LENGTH];
107 int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
109 switch(payload_type) {
111 codec->codec_id = CODEC_ID_PCM_MULAW;
113 codec->sample_rate = 8000;
116 codec->codec_id = CODEC_ID_PCM_ALAW;
118 codec->sample_rate = 8000;
120 case RTP_PT_S16BE_STEREO:
121 codec->codec_id = CODEC_ID_PCM_S16BE;
123 codec->sample_rate = 44100;
125 case RTP_PT_S16BE_MONO:
126 codec->codec_id = CODEC_ID_PCM_S16BE;
128 codec->sample_rate = 44100;
130 case RTP_PT_MPEGAUDIO:
131 codec->codec_id = CODEC_ID_MP2;
134 codec->codec_id = CODEC_ID_MJPEG;
136 case RTP_PT_MPEGVIDEO:
137 codec->codec_id = CODEC_ID_MPEG1VIDEO;
145 /* return < 0 if unknown payload type */
146 int rtp_get_payload_type(AVCodecContext *codec)
150 /* compute the payload type */
152 switch(codec->codec_id) {
153 case CODEC_ID_PCM_MULAW:
154 payload_type = RTP_PT_ULAW;
156 case CODEC_ID_PCM_ALAW:
157 payload_type = RTP_PT_ALAW;
159 case CODEC_ID_PCM_S16BE:
160 if (codec->channels == 1) {
161 payload_type = RTP_PT_S16BE_MONO;
162 } else if (codec->channels == 2) {
163 payload_type = RTP_PT_S16BE_STEREO;
167 case CODEC_ID_MP3LAME:
168 payload_type = RTP_PT_MPEGAUDIO;
171 payload_type = RTP_PT_JPEG;
173 case CODEC_ID_MPEG1VIDEO:
174 payload_type = RTP_PT_MPEGVIDEO;
182 static inline UINT32 decode_be32(const UINT8 *p)
184 return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
187 static inline UINT32 decode_be64(const UINT8 *p)
189 return ((UINT64)decode_be32(p) << 32) | decode_be32(p + 4);
192 static int rtcp_parse_packet(AVFormatContext *s1, const unsigned char *buf, int len)
194 RTPContext *s = s1->priv_data;
198 s->last_rtcp_ntp_time = decode_be64(buf + 8);
199 s->last_rtcp_timestamp = decode_be32(buf + 16);
204 * Parse an RTP packet directly sent as raw data. Can only be used if
205 * 'raw' is given as input file
206 * @param s1 media file context
207 * @param pkt returned packet
208 * @param buf input buffer
209 * @param len buffer len
210 * @return zero if no error.
212 int rtp_parse_packet(AVFormatContext *s1, AVPacket *pkt,
213 const unsigned char *buf, int len)
215 RTPContext *s = s1->priv_data;
216 unsigned int ssrc, h;
217 int payload_type, seq, delta_timestamp;
224 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
226 if (buf[1] >= 200 && buf[1] <= 204) {
227 rtcp_parse_packet(s1, buf, len);
230 payload_type = buf[1] & 0x7f;
231 seq = (buf[2] << 8) | buf[3];
232 timestamp = decode_be32(buf + 4);
233 ssrc = decode_be32(buf + 8);
235 if (s->payload_type < 0) {
236 s->payload_type = payload_type;
238 if (payload_type == RTP_PT_MPEG2TS) {
239 /* XXX: special case : not a single codec but a whole stream */
242 st = av_new_stream(s1, 0);
245 rtp_get_codec_info(&st->codec, payload_type);
249 /* NOTE: we can handle only one payload type */
250 if (s->payload_type != payload_type)
252 #if defined(DEBUG) || 1
253 if (seq != ((s->seq + 1) & 0xffff)) {
254 printf("RTP: PT=%02x: bad cseq %04x expected=%04x\n",
255 payload_type, seq, ((s->seq + 1) & 0xffff));
262 switch(st->codec.codec_id) {
264 /* better than nothing: skip mpeg audio RTP header */
267 h = decode_be32(buf);
270 av_new_packet(pkt, len);
271 memcpy(pkt->data, buf, len);
273 case CODEC_ID_MPEG1VIDEO:
274 /* better than nothing: skip mpeg audio RTP header */
277 h = decode_be32(buf);
287 av_new_packet(pkt, len);
288 memcpy(pkt->data, buf, len);
291 av_new_packet(pkt, len);
292 memcpy(pkt->data, buf, len);
296 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
297 /* compute pts from timestamp with received ntp_time */
298 delta_timestamp = timestamp - s->last_rtcp_timestamp;
299 /* XXX: do conversion, but not needed for mpeg at 90 KhZ */
300 pkt->pts = s->last_rtcp_ntp_time + delta_timestamp;
305 static int rtp_read_header(AVFormatContext *s1,
306 AVFormatParameters *ap)
308 RTPContext *s = s1->priv_data;
309 s->payload_type = -1;
310 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
314 static int rtp_read_packet(AVFormatContext *s1, AVPacket *pkt)
316 char buf[RTP_MAX_PACKET_LENGTH];
319 /* XXX: needs a better API for packet handling ? */
321 ret = url_read(url_fileno(&s1->pb), buf, sizeof(buf));
324 if (rtp_parse_packet(s1, pkt, buf, ret) == 0)
330 static int rtp_read_close(AVFormatContext *s1)
332 // RTPContext *s = s1->priv_data;
336 static int rtp_probe(AVProbeData *p)
338 if (strstart(p->filename, "rtp://", NULL))
339 return AVPROBE_SCORE_MAX;
345 static int rtp_write_header(AVFormatContext *s1)
347 RTPContext *s = s1->priv_data;
348 int payload_type, max_packet_size;
351 if (s1->nb_streams != 1)
355 payload_type = rtp_get_payload_type(&st->codec);
356 if (payload_type < 0)
358 s->payload_type = payload_type;
360 s->base_timestamp = random();
361 s->timestamp = s->base_timestamp;
365 max_packet_size = url_fget_max_packet_size(&s1->pb);
366 if (max_packet_size <= 12)
368 s->max_payload_size = max_packet_size - 12;
370 switch(st->codec.codec_id) {
372 case CODEC_ID_MP3LAME:
373 s->buf_ptr = s->buf + 4;
374 s->cur_timestamp = 0;
376 case CODEC_ID_MPEG1VIDEO:
377 s->cur_timestamp = 0;
387 /* send an rtcp sender report packet */
388 static void rtcp_send_sr(AVFormatContext *s1, INT64 ntp_time)
390 RTPContext *s = s1->priv_data;
392 printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
394 put_byte(&s1->pb, (RTP_VERSION << 6));
395 put_byte(&s1->pb, 200);
396 put_be16(&s1->pb, 6); /* length in words - 1 */
397 put_be32(&s1->pb, s->ssrc);
398 put_be64(&s1->pb, ntp_time);
399 put_be32(&s1->pb, s->timestamp);
400 put_be32(&s1->pb, s->packet_count);
401 put_be32(&s1->pb, s->octet_count);
402 put_flush_packet(&s1->pb);
405 /* send an rtp packet. sequence number is incremented, but the caller
406 must update the timestamp itself */
407 static void rtp_send_data(AVFormatContext *s1, UINT8 *buf1, int len)
409 RTPContext *s = s1->priv_data;
412 printf("rtp_send_data size=%d\n", len);
415 /* build the RTP header */
416 put_byte(&s1->pb, (RTP_VERSION << 6));
417 put_byte(&s1->pb, s->payload_type & 0x7f);
418 put_be16(&s1->pb, s->seq);
419 put_be32(&s1->pb, s->timestamp);
420 put_be32(&s1->pb, s->ssrc);
422 put_buffer(&s1->pb, buf1, len);
423 put_flush_packet(&s1->pb);
426 s->octet_count += len;
430 /* send an integer number of samples and compute time stamp and fill
431 the rtp send buffer before sending. */
432 static void rtp_send_samples(AVFormatContext *s1,
433 UINT8 *buf1, int size, int sample_size)
435 RTPContext *s = s1->priv_data;
436 int len, max_packet_size, n;
438 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
439 /* not needed, but who nows */
440 if ((size % sample_size) != 0)
443 len = (max_packet_size - (s->buf_ptr - s->buf));
448 memcpy(s->buf_ptr, buf1, len);
452 n = (s->buf_ptr - s->buf);
453 /* if buffer full, then send it */
454 if (n >= max_packet_size) {
455 rtp_send_data(s1, s->buf, n);
457 /* update timestamp */
458 s->timestamp += n / sample_size;
463 /* NOTE: we suppose that exactly one frame is given as argument here */
465 static void rtp_send_mpegaudio(AVFormatContext *s1,
466 UINT8 *buf1, int size)
468 RTPContext *s = s1->priv_data;
469 AVStream *st = s1->streams[0];
470 int len, count, max_packet_size;
472 max_packet_size = s->max_payload_size;
474 /* test if we must flush because not enough space */
475 len = (s->buf_ptr - s->buf);
476 if ((len + size) > max_packet_size) {
478 rtp_send_data(s1, s->buf, s->buf_ptr - s->buf);
479 s->buf_ptr = s->buf + 4;
480 /* 90 KHz time stamp */
481 s->timestamp = s->base_timestamp +
482 (s->cur_timestamp * 90000LL) / st->codec.sample_rate;
487 if (size > max_packet_size) {
488 /* big packet: fragment */
491 len = max_packet_size - 4;
494 /* build fragmented packet */
497 s->buf[2] = count >> 8;
499 memcpy(s->buf + 4, buf1, len);
500 rtp_send_data(s1, s->buf, len + 4);
506 if (s->buf_ptr == s->buf + 4) {
507 /* no fragmentation possible */
513 memcpy(s->buf_ptr, buf1, size);
516 s->cur_timestamp += st->codec.frame_size;
519 /* NOTE: a single frame must be passed with sequence header if
520 needed. XXX: use slices. */
521 static void rtp_send_mpegvideo(AVFormatContext *s1,
522 UINT8 *buf1, int size)
524 RTPContext *s = s1->priv_data;
525 AVStream *st = s1->streams[0];
526 int len, h, max_packet_size;
529 max_packet_size = s->max_payload_size;
532 /* XXX: more correct headers */
534 if (st->codec.sub_id == 2)
535 h |= 1 << 26; /* mpeg 2 indicator */
542 if (st->codec.sub_id == 2) {
550 len = max_packet_size - (q - s->buf);
554 memcpy(q, buf1, len);
557 /* 90 KHz time stamp */
559 s->timestamp = s->base_timestamp +
560 (s->cur_timestamp * 90000LL * FRAME_RATE_BASE) / st->codec.frame_rate;
561 rtp_send_data(s1, s->buf, q - s->buf);
569 /* write an RTP packet. 'buf1' must contain a single specific frame. */
570 static int rtp_write_packet(AVFormatContext *s1, int stream_index,
571 UINT8 *buf1, int size, int force_pts)
573 RTPContext *s = s1->priv_data;
574 AVStream *st = s1->streams[0];
579 printf("%d: write len=%d\n", stream_index, size);
582 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
583 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
585 if (s->first_packet || rtcp_bytes >= 28) {
586 /* compute NTP time */
587 ntp_time = force_pts; // ((INT64)force_pts << 28) / 5625
588 rtcp_send_sr(s1, ntp_time);
589 s->last_octet_count = s->octet_count;
593 switch(st->codec.codec_id) {
594 case CODEC_ID_PCM_MULAW:
595 case CODEC_ID_PCM_ALAW:
596 case CODEC_ID_PCM_U8:
597 case CODEC_ID_PCM_S8:
598 rtp_send_samples(s1, buf1, size, 1 * st->codec.channels);
600 case CODEC_ID_PCM_U16BE:
601 case CODEC_ID_PCM_U16LE:
602 case CODEC_ID_PCM_S16BE:
603 case CODEC_ID_PCM_S16LE:
604 rtp_send_samples(s1, buf1, size, 2 * st->codec.channels);
607 case CODEC_ID_MP3LAME:
608 rtp_send_mpegaudio(s1, buf1, size);
610 case CODEC_ID_MPEG1VIDEO:
611 rtp_send_mpegvideo(s1, buf1, size);
619 static int rtp_write_trailer(AVFormatContext *s1)
621 // RTPContext *s = s1->priv_data;
625 AVInputFormat rtp_demux = {
633 .flags = AVFMT_NOHEADER,
636 AVOutputFormat rtp_mux = {
651 av_register_output_format(&rtp_mux);
652 av_register_input_format(&rtp_demux);