3 * Copyright (c) 2002 Fabrice Bellard
5 * This file is part of Libav.
7 * Libav 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 * Libav 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 Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
25 #include "libavutil/mathematics.h"
26 #include "libavutil/random_seed.h"
27 #include "libavutil/opt.h"
31 static const AVOption options[] = {
32 FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
33 { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
34 { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
35 { "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM },
36 { "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM },
40 static const AVClass rtp_muxer_class = {
41 .class_name = "RTP muxer",
42 .item_name = av_default_item_name,
44 .version = LIBAVUTIL_VERSION_INT,
47 #define RTCP_SR_SIZE 28
49 static int is_supported(enum AVCodecID id)
52 case AV_CODEC_ID_H263:
53 case AV_CODEC_ID_H263P:
54 case AV_CODEC_ID_H264:
55 case AV_CODEC_ID_MPEG1VIDEO:
56 case AV_CODEC_ID_MPEG2VIDEO:
57 case AV_CODEC_ID_MPEG4:
61 case AV_CODEC_ID_PCM_ALAW:
62 case AV_CODEC_ID_PCM_MULAW:
63 case AV_CODEC_ID_PCM_S8:
64 case AV_CODEC_ID_PCM_S16BE:
65 case AV_CODEC_ID_PCM_S16LE:
66 case AV_CODEC_ID_PCM_U16BE:
67 case AV_CODEC_ID_PCM_U16LE:
68 case AV_CODEC_ID_PCM_U8:
69 case AV_CODEC_ID_MPEG2TS:
70 case AV_CODEC_ID_AMR_NB:
71 case AV_CODEC_ID_AMR_WB:
72 case AV_CODEC_ID_VORBIS:
73 case AV_CODEC_ID_THEORA:
75 case AV_CODEC_ID_ADPCM_G722:
76 case AV_CODEC_ID_ADPCM_G726:
77 case AV_CODEC_ID_ILBC:
78 case AV_CODEC_ID_MJPEG:
79 case AV_CODEC_ID_SPEEX:
80 case AV_CODEC_ID_OPUS:
87 static int rtp_write_header(AVFormatContext *s1)
89 RTPMuxContext *s = s1->priv_data;
93 if (s1->nb_streams != 1) {
94 av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
95 return AVERROR(EINVAL);
98 if (!is_supported(st->codec->codec_id)) {
99 av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
104 if (s->payload_type < 0) {
105 /* Re-validate non-dynamic payload types */
106 if (st->id < RTP_PT_PRIVATE)
107 st->id = ff_rtp_get_payload_type(s1, st->codec, -1);
109 s->payload_type = st->id;
111 /* private option takes priority */
112 st->id = s->payload_type;
115 s->base_timestamp = av_get_random_seed();
116 s->timestamp = s->base_timestamp;
117 s->cur_timestamp = 0;
119 s->ssrc = av_get_random_seed();
121 s->first_rtcp_ntp_time = ff_ntp_time();
122 if (s1->start_time_realtime)
123 /* Round the NTP time to whole milliseconds. */
124 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
126 // Pick a random sequence start number, but in the lower end of the
127 // available range, so that any wraparound doesn't happen immediately.
128 // (Immediate wraparound would be an issue for SRTP.)
130 s->seq = av_get_random_seed() & 0x0fff;
132 s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
134 if (s1->packet_size) {
135 if (s1->pb->max_packet_size)
136 s1->packet_size = FFMIN(s1->packet_size,
137 s1->pb->max_packet_size);
139 s1->packet_size = s1->pb->max_packet_size;
140 if (s1->packet_size <= 12) {
141 av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
144 s->buf = av_malloc(s1->packet_size);
145 if (s->buf == NULL) {
146 return AVERROR(ENOMEM);
148 s->max_payload_size = s1->packet_size - 12;
150 s->max_frames_per_packet = 0;
151 if (s1->max_delay > 0) {
152 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
153 int frame_size = av_get_audio_frame_duration(st->codec, 0);
155 frame_size = st->codec->frame_size;
156 if (frame_size == 0) {
157 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
159 s->max_frames_per_packet =
160 av_rescale_q_rnd(s1->max_delay,
162 (AVRational){ frame_size, st->codec->sample_rate },
166 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
167 /* FIXME: We should round down here... */
168 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
172 avpriv_set_pts_info(st, 32, 1, 90000);
173 switch(st->codec->codec_id) {
174 case AV_CODEC_ID_MP2:
175 case AV_CODEC_ID_MP3:
176 s->buf_ptr = s->buf + 4;
178 case AV_CODEC_ID_MPEG1VIDEO:
179 case AV_CODEC_ID_MPEG2VIDEO:
181 case AV_CODEC_ID_MPEG2TS:
182 n = s->max_payload_size / TS_PACKET_SIZE;
185 s->max_payload_size = n * TS_PACKET_SIZE;
188 case AV_CODEC_ID_H264:
189 /* check for H.264 MP4 syntax */
190 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
191 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
194 case AV_CODEC_ID_VORBIS:
195 case AV_CODEC_ID_THEORA:
196 if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
197 s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
198 s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
201 case AV_CODEC_ID_ADPCM_G722:
202 /* Due to a historical error, the clock rate for G722 in RTP is
203 * 8000, even if the sample rate is 16000. See RFC 3551. */
204 avpriv_set_pts_info(st, 32, 1, 8000);
206 case AV_CODEC_ID_OPUS:
207 if (st->codec->channels > 2) {
208 av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
211 /* The opus RTP RFC says that all opus streams should use 48000 Hz
212 * as clock rate, since all opus sample rates can be expressed in
213 * this clock rate, and sample rate changes on the fly are supported. */
214 avpriv_set_pts_info(st, 32, 1, 48000);
216 case AV_CODEC_ID_ILBC:
217 if (st->codec->block_align != 38 && st->codec->block_align != 50) {
218 av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
221 if (!s->max_frames_per_packet)
222 s->max_frames_per_packet = 1;
223 s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
224 s->max_payload_size / st->codec->block_align);
226 case AV_CODEC_ID_AMR_NB:
227 case AV_CODEC_ID_AMR_WB:
228 if (!s->max_frames_per_packet)
229 s->max_frames_per_packet = 12;
230 if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
234 /* max_header_toc_size + the largest AMR payload must fit */
235 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
236 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
239 if (st->codec->channels != 1) {
240 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
243 case AV_CODEC_ID_AAC:
247 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
248 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
258 return AVERROR(EINVAL);
261 /* send an rtcp sender report packet */
262 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
264 RTPMuxContext *s = s1->priv_data;
267 av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
269 s->last_rtcp_ntp_time = ntp_time;
270 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
271 s1->streams[0]->time_base) + s->base_timestamp;
272 avio_w8(s1->pb, RTP_VERSION << 6);
273 avio_w8(s1->pb, RTCP_SR);
274 avio_wb16(s1->pb, 6); /* length in words - 1 */
275 avio_wb32(s1->pb, s->ssrc);
276 avio_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time));
277 avio_wb32(s1->pb, rtp_ts);
278 avio_wb32(s1->pb, s->packet_count);
279 avio_wb32(s1->pb, s->octet_count);
282 int len = FFMIN(strlen(s->cname), 255);
283 avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
284 avio_w8(s1->pb, RTCP_SDES);
285 avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
287 avio_wb32(s1->pb, s->ssrc);
288 avio_w8(s1->pb, 0x01); /* CNAME */
289 avio_w8(s1->pb, len);
290 avio_write(s1->pb, s->cname, len);
291 avio_w8(s1->pb, 0); /* END */
292 for (len = (7 + len) % 4; len % 4; len++)
297 avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
298 avio_w8(s1->pb, RTCP_BYE);
299 avio_wb16(s1->pb, 1); /* length in words - 1 */
300 avio_wb32(s1->pb, s->ssrc);
306 /* send an rtp packet. sequence number is incremented, but the caller
307 must update the timestamp itself */
308 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
310 RTPMuxContext *s = s1->priv_data;
312 av_dlog(s1, "rtp_send_data size=%d\n", len);
314 /* build the RTP header */
315 avio_w8(s1->pb, RTP_VERSION << 6);
316 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
317 avio_wb16(s1->pb, s->seq);
318 avio_wb32(s1->pb, s->timestamp);
319 avio_wb32(s1->pb, s->ssrc);
321 avio_write(s1->pb, buf1, len);
324 s->seq = (s->seq + 1) & 0xffff;
325 s->octet_count += len;
329 /* send an integer number of samples and compute time stamp and fill
330 the rtp send buffer before sending. */
331 static int rtp_send_samples(AVFormatContext *s1,
332 const uint8_t *buf1, int size, int sample_size_bits)
334 RTPMuxContext *s = s1->priv_data;
335 int len, max_packet_size, n;
336 /* Calculate the number of bytes to get samples aligned on a byte border */
337 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
339 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
340 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
341 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
342 return AVERROR(EINVAL);
346 len = FFMIN(max_packet_size, size);
349 memcpy(s->buf_ptr, buf1, len);
353 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
354 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
355 n += (s->buf_ptr - s->buf);
360 static void rtp_send_mpegaudio(AVFormatContext *s1,
361 const uint8_t *buf1, int size)
363 RTPMuxContext *s = s1->priv_data;
364 int len, count, max_packet_size;
366 max_packet_size = s->max_payload_size;
368 /* test if we must flush because not enough space */
369 len = (s->buf_ptr - s->buf);
370 if ((len + size) > max_packet_size) {
372 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
373 s->buf_ptr = s->buf + 4;
376 if (s->buf_ptr == s->buf + 4) {
377 s->timestamp = s->cur_timestamp;
381 if (size > max_packet_size) {
382 /* big packet: fragment */
385 len = max_packet_size - 4;
388 /* build fragmented packet */
391 s->buf[2] = count >> 8;
393 memcpy(s->buf + 4, buf1, len);
394 ff_rtp_send_data(s1, s->buf, len + 4, 0);
400 if (s->buf_ptr == s->buf + 4) {
401 /* no fragmentation possible */
407 memcpy(s->buf_ptr, buf1, size);
412 static void rtp_send_raw(AVFormatContext *s1,
413 const uint8_t *buf1, int size)
415 RTPMuxContext *s = s1->priv_data;
416 int len, max_packet_size;
418 max_packet_size = s->max_payload_size;
421 len = max_packet_size;
425 s->timestamp = s->cur_timestamp;
426 ff_rtp_send_data(s1, buf1, len, (len == size));
433 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
434 static void rtp_send_mpegts_raw(AVFormatContext *s1,
435 const uint8_t *buf1, int size)
437 RTPMuxContext *s = s1->priv_data;
440 while (size >= TS_PACKET_SIZE) {
441 len = s->max_payload_size - (s->buf_ptr - s->buf);
444 memcpy(s->buf_ptr, buf1, len);
449 out_len = s->buf_ptr - s->buf;
450 if (out_len >= s->max_payload_size) {
451 ff_rtp_send_data(s1, s->buf, out_len, 0);
457 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
459 RTPMuxContext *s = s1->priv_data;
460 AVStream *st = s1->streams[0];
461 int frame_duration = av_get_audio_frame_duration(st->codec, 0);
462 int frame_size = st->codec->block_align;
463 int frames = size / frame_size;
466 int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
468 if (!s->num_frames) {
470 s->timestamp = s->cur_timestamp;
472 memcpy(s->buf_ptr, buf, n * frame_size);
475 s->buf_ptr += n * frame_size;
476 buf += n * frame_size;
477 s->cur_timestamp += n * frame_duration;
479 if (s->num_frames == s->max_frames_per_packet) {
480 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
487 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
489 RTPMuxContext *s = s1->priv_data;
490 AVStream *st = s1->streams[0];
494 av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
496 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
498 if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
499 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
500 !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
501 rtcp_send_sr(s1, ff_ntp_time(), 0);
502 s->last_octet_count = s->octet_count;
505 s->cur_timestamp = s->base_timestamp + pkt->pts;
507 switch(st->codec->codec_id) {
508 case AV_CODEC_ID_PCM_MULAW:
509 case AV_CODEC_ID_PCM_ALAW:
510 case AV_CODEC_ID_PCM_U8:
511 case AV_CODEC_ID_PCM_S8:
512 return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
513 case AV_CODEC_ID_PCM_U16BE:
514 case AV_CODEC_ID_PCM_U16LE:
515 case AV_CODEC_ID_PCM_S16BE:
516 case AV_CODEC_ID_PCM_S16LE:
517 return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
518 case AV_CODEC_ID_ADPCM_G722:
519 /* The actual sample size is half a byte per sample, but since the
520 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
521 * the correct parameter for send_samples_bits is 8 bits per stream
523 return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
524 case AV_CODEC_ID_ADPCM_G726:
525 return rtp_send_samples(s1, pkt->data, size,
526 st->codec->bits_per_coded_sample * st->codec->channels);
527 case AV_CODEC_ID_MP2:
528 case AV_CODEC_ID_MP3:
529 rtp_send_mpegaudio(s1, pkt->data, size);
531 case AV_CODEC_ID_MPEG1VIDEO:
532 case AV_CODEC_ID_MPEG2VIDEO:
533 ff_rtp_send_mpegvideo(s1, pkt->data, size);
535 case AV_CODEC_ID_AAC:
536 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
537 ff_rtp_send_latm(s1, pkt->data, size);
539 ff_rtp_send_aac(s1, pkt->data, size);
541 case AV_CODEC_ID_AMR_NB:
542 case AV_CODEC_ID_AMR_WB:
543 ff_rtp_send_amr(s1, pkt->data, size);
545 case AV_CODEC_ID_MPEG2TS:
546 rtp_send_mpegts_raw(s1, pkt->data, size);
548 case AV_CODEC_ID_H264:
549 ff_rtp_send_h264(s1, pkt->data, size);
551 case AV_CODEC_ID_H263:
552 if (s->flags & FF_RTP_FLAG_RFC2190) {
553 int mb_info_size = 0;
554 const uint8_t *mb_info =
555 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
557 ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
561 case AV_CODEC_ID_H263P:
562 ff_rtp_send_h263(s1, pkt->data, size);
564 case AV_CODEC_ID_VORBIS:
565 case AV_CODEC_ID_THEORA:
566 ff_rtp_send_xiph(s1, pkt->data, size);
568 case AV_CODEC_ID_VP8:
569 ff_rtp_send_vp8(s1, pkt->data, size);
571 case AV_CODEC_ID_ILBC:
572 rtp_send_ilbc(s1, pkt->data, size);
574 case AV_CODEC_ID_MJPEG:
575 ff_rtp_send_jpeg(s1, pkt->data, size);
577 case AV_CODEC_ID_OPUS:
578 if (size > s->max_payload_size) {
579 av_log(s1, AV_LOG_ERROR,
580 "Packet size %d too large for max RTP payload size %d\n",
581 size, s->max_payload_size);
582 return AVERROR(EINVAL);
584 /* Intentional fallthrough */
586 /* better than nothing : send the codec raw data */
587 rtp_send_raw(s1, pkt->data, size);
593 static int rtp_write_trailer(AVFormatContext *s1)
595 RTPMuxContext *s = s1->priv_data;
597 /* If the caller closes and recreates ->pb, this might actually
598 * be NULL here even if it was successfully allocated at the start. */
599 if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
600 rtcp_send_sr(s1, ff_ntp_time(), 1);
606 AVOutputFormat ff_rtp_muxer = {
608 .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
609 .priv_data_size = sizeof(RTPMuxContext),
610 .audio_codec = AV_CODEC_ID_PCM_MULAW,
611 .video_codec = AV_CODEC_ID_MPEG4,
612 .write_header = rtp_write_header,
613 .write_packet = rtp_write_packet,
614 .write_trailer = rtp_write_trailer,
615 .priv_class = &rtp_muxer_class,