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)
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_wb32(s1->pb, ntp_time / 1000000);
277 avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
278 avio_wb32(s1->pb, rtp_ts);
279 avio_wb32(s1->pb, s->packet_count);
280 avio_wb32(s1->pb, s->octet_count);
283 int len = FFMIN(strlen(s->cname), 255);
284 avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
285 avio_w8(s1->pb, RTCP_SDES);
286 avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
288 avio_wb32(s1->pb, s->ssrc);
289 avio_w8(s1->pb, 0x01); /* CNAME */
290 avio_w8(s1->pb, len);
291 avio_write(s1->pb, s->cname, len);
292 avio_w8(s1->pb, 0); /* END */
293 for (len = (7 + len) % 4; len % 4; len++)
300 /* send an rtp packet. sequence number is incremented, but the caller
301 must update the timestamp itself */
302 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
304 RTPMuxContext *s = s1->priv_data;
306 av_dlog(s1, "rtp_send_data size=%d\n", len);
308 /* build the RTP header */
309 avio_w8(s1->pb, (RTP_VERSION << 6));
310 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
311 avio_wb16(s1->pb, s->seq);
312 avio_wb32(s1->pb, s->timestamp);
313 avio_wb32(s1->pb, s->ssrc);
315 avio_write(s1->pb, buf1, len);
318 s->seq = (s->seq + 1) & 0xffff;
319 s->octet_count += len;
323 /* send an integer number of samples and compute time stamp and fill
324 the rtp send buffer before sending. */
325 static int rtp_send_samples(AVFormatContext *s1,
326 const uint8_t *buf1, int size, int sample_size_bits)
328 RTPMuxContext *s = s1->priv_data;
329 int len, max_packet_size, n;
330 /* Calculate the number of bytes to get samples aligned on a byte border */
331 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
333 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
334 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
335 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
336 return AVERROR(EINVAL);
340 len = FFMIN(max_packet_size, size);
343 memcpy(s->buf_ptr, buf1, len);
347 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
348 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
349 n += (s->buf_ptr - s->buf);
354 static void rtp_send_mpegaudio(AVFormatContext *s1,
355 const uint8_t *buf1, int size)
357 RTPMuxContext *s = s1->priv_data;
358 int len, count, max_packet_size;
360 max_packet_size = s->max_payload_size;
362 /* test if we must flush because not enough space */
363 len = (s->buf_ptr - s->buf);
364 if ((len + size) > max_packet_size) {
366 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
367 s->buf_ptr = s->buf + 4;
370 if (s->buf_ptr == s->buf + 4) {
371 s->timestamp = s->cur_timestamp;
375 if (size > max_packet_size) {
376 /* big packet: fragment */
379 len = max_packet_size - 4;
382 /* build fragmented packet */
385 s->buf[2] = count >> 8;
387 memcpy(s->buf + 4, buf1, len);
388 ff_rtp_send_data(s1, s->buf, len + 4, 0);
394 if (s->buf_ptr == s->buf + 4) {
395 /* no fragmentation possible */
401 memcpy(s->buf_ptr, buf1, size);
406 static void rtp_send_raw(AVFormatContext *s1,
407 const uint8_t *buf1, int size)
409 RTPMuxContext *s = s1->priv_data;
410 int len, max_packet_size;
412 max_packet_size = s->max_payload_size;
415 len = max_packet_size;
419 s->timestamp = s->cur_timestamp;
420 ff_rtp_send_data(s1, buf1, len, (len == size));
427 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
428 static void rtp_send_mpegts_raw(AVFormatContext *s1,
429 const uint8_t *buf1, int size)
431 RTPMuxContext *s = s1->priv_data;
434 while (size >= TS_PACKET_SIZE) {
435 len = s->max_payload_size - (s->buf_ptr - s->buf);
438 memcpy(s->buf_ptr, buf1, len);
443 out_len = s->buf_ptr - s->buf;
444 if (out_len >= s->max_payload_size) {
445 ff_rtp_send_data(s1, s->buf, out_len, 0);
451 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
453 RTPMuxContext *s = s1->priv_data;
454 AVStream *st = s1->streams[0];
455 int frame_duration = av_get_audio_frame_duration(st->codec, 0);
456 int frame_size = st->codec->block_align;
457 int frames = size / frame_size;
460 int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
462 if (!s->num_frames) {
464 s->timestamp = s->cur_timestamp;
466 memcpy(s->buf_ptr, buf, n * frame_size);
469 s->buf_ptr += n * frame_size;
470 buf += n * frame_size;
471 s->cur_timestamp += n * frame_duration;
473 if (s->num_frames == s->max_frames_per_packet) {
474 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
481 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
483 RTPMuxContext *s = s1->priv_data;
484 AVStream *st = s1->streams[0];
488 av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
490 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
492 if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
493 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
494 !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
495 rtcp_send_sr(s1, ff_ntp_time());
496 s->last_octet_count = s->octet_count;
499 s->cur_timestamp = s->base_timestamp + pkt->pts;
501 switch(st->codec->codec_id) {
502 case AV_CODEC_ID_PCM_MULAW:
503 case AV_CODEC_ID_PCM_ALAW:
504 case AV_CODEC_ID_PCM_U8:
505 case AV_CODEC_ID_PCM_S8:
506 return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
507 case AV_CODEC_ID_PCM_U16BE:
508 case AV_CODEC_ID_PCM_U16LE:
509 case AV_CODEC_ID_PCM_S16BE:
510 case AV_CODEC_ID_PCM_S16LE:
511 return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
512 case AV_CODEC_ID_ADPCM_G722:
513 /* The actual sample size is half a byte per sample, but since the
514 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
515 * the correct parameter for send_samples_bits is 8 bits per stream
517 return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
518 case AV_CODEC_ID_ADPCM_G726:
519 return rtp_send_samples(s1, pkt->data, size,
520 st->codec->bits_per_coded_sample * st->codec->channels);
521 case AV_CODEC_ID_MP2:
522 case AV_CODEC_ID_MP3:
523 rtp_send_mpegaudio(s1, pkt->data, size);
525 case AV_CODEC_ID_MPEG1VIDEO:
526 case AV_CODEC_ID_MPEG2VIDEO:
527 ff_rtp_send_mpegvideo(s1, pkt->data, size);
529 case AV_CODEC_ID_AAC:
530 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
531 ff_rtp_send_latm(s1, pkt->data, size);
533 ff_rtp_send_aac(s1, pkt->data, size);
535 case AV_CODEC_ID_AMR_NB:
536 case AV_CODEC_ID_AMR_WB:
537 ff_rtp_send_amr(s1, pkt->data, size);
539 case AV_CODEC_ID_MPEG2TS:
540 rtp_send_mpegts_raw(s1, pkt->data, size);
542 case AV_CODEC_ID_H264:
543 ff_rtp_send_h264(s1, pkt->data, size);
545 case AV_CODEC_ID_H263:
546 if (s->flags & FF_RTP_FLAG_RFC2190) {
547 int mb_info_size = 0;
548 const uint8_t *mb_info =
549 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
551 ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
555 case AV_CODEC_ID_H263P:
556 ff_rtp_send_h263(s1, pkt->data, size);
558 case AV_CODEC_ID_VORBIS:
559 case AV_CODEC_ID_THEORA:
560 ff_rtp_send_xiph(s1, pkt->data, size);
562 case AV_CODEC_ID_VP8:
563 ff_rtp_send_vp8(s1, pkt->data, size);
565 case AV_CODEC_ID_ILBC:
566 rtp_send_ilbc(s1, pkt->data, size);
568 case AV_CODEC_ID_MJPEG:
569 ff_rtp_send_jpeg(s1, pkt->data, size);
571 case AV_CODEC_ID_OPUS:
572 if (size > s->max_payload_size) {
573 av_log(s1, AV_LOG_ERROR,
574 "Packet size %d too large for max RTP payload size %d\n",
575 size, s->max_payload_size);
576 return AVERROR(EINVAL);
578 /* Intentional fallthrough */
580 /* better than nothing : send the codec raw data */
581 rtp_send_raw(s1, pkt->data, size);
587 static int rtp_write_trailer(AVFormatContext *s1)
589 RTPMuxContext *s = s1->priv_data;
596 AVOutputFormat ff_rtp_muxer = {
598 .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
599 .priv_data_size = sizeof(RTPMuxContext),
600 .audio_codec = AV_CODEC_ID_PCM_MULAW,
601 .video_codec = AV_CODEC_ID_MPEG4,
602 .write_header = rtp_write_header,
603 .write_packet = rtp_write_packet,
604 .write_trailer = rtp_write_trailer,
605 .priv_class = &rtp_muxer_class,