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
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_DIRAC:
53 case AV_CODEC_ID_H261:
54 case AV_CODEC_ID_H263:
55 case AV_CODEC_ID_H263P:
56 case AV_CODEC_ID_H264:
57 case AV_CODEC_ID_HEVC:
58 case AV_CODEC_ID_MPEG1VIDEO:
59 case AV_CODEC_ID_MPEG2VIDEO:
60 case AV_CODEC_ID_MPEG4:
64 case AV_CODEC_ID_PCM_ALAW:
65 case AV_CODEC_ID_PCM_MULAW:
66 case AV_CODEC_ID_PCM_S8:
67 case AV_CODEC_ID_PCM_S16BE:
68 case AV_CODEC_ID_PCM_S16LE:
69 case AV_CODEC_ID_PCM_U16BE:
70 case AV_CODEC_ID_PCM_U16LE:
71 case AV_CODEC_ID_PCM_U8:
72 case AV_CODEC_ID_MPEG2TS:
73 case AV_CODEC_ID_AMR_NB:
74 case AV_CODEC_ID_AMR_WB:
75 case AV_CODEC_ID_VORBIS:
76 case AV_CODEC_ID_THEORA:
78 case AV_CODEC_ID_ADPCM_G722:
79 case AV_CODEC_ID_ADPCM_G726:
80 case AV_CODEC_ID_ILBC:
81 case AV_CODEC_ID_MJPEG:
82 case AV_CODEC_ID_SPEEX:
83 case AV_CODEC_ID_OPUS:
90 static int rtp_write_header(AVFormatContext *s1)
92 RTPMuxContext *s = s1->priv_data;
93 int n, ret = AVERROR(EINVAL);
96 if (s1->nb_streams != 1) {
97 av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
98 return AVERROR(EINVAL);
101 if (!is_supported(st->codecpar->codec_id)) {
102 av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codecpar->codec_id));
107 if (s->payload_type < 0) {
108 /* Re-validate non-dynamic payload types */
109 if (st->id < RTP_PT_PRIVATE)
110 st->id = ff_rtp_get_payload_type(s1, st->codecpar, -1);
112 s->payload_type = st->id;
114 /* private option takes priority */
115 st->id = s->payload_type;
118 s->base_timestamp = av_get_random_seed();
119 s->timestamp = s->base_timestamp;
120 s->cur_timestamp = 0;
122 s->ssrc = av_get_random_seed();
124 s->first_rtcp_ntp_time = ff_ntp_time();
125 if (s1->start_time_realtime != 0 && s1->start_time_realtime != AV_NOPTS_VALUE)
126 /* Round the NTP time to whole milliseconds. */
127 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
129 // Pick a random sequence start number, but in the lower end of the
130 // available range, so that any wraparound doesn't happen immediately.
131 // (Immediate wraparound would be an issue for SRTP.)
133 if (s1->flags & AVFMT_FLAG_BITEXACT) {
136 s->seq = av_get_random_seed() & 0x0fff;
138 s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
140 if (s1->packet_size) {
141 if (s1->pb->max_packet_size)
142 s1->packet_size = FFMIN(s1->packet_size,
143 s1->pb->max_packet_size);
145 s1->packet_size = s1->pb->max_packet_size;
146 if (s1->packet_size <= 12) {
147 av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
150 s->buf = av_malloc(s1->packet_size);
152 return AVERROR(ENOMEM);
154 s->max_payload_size = s1->packet_size - 12;
156 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
157 avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
159 avpriv_set_pts_info(st, 32, 1, 90000);
162 switch(st->codecpar->codec_id) {
163 case AV_CODEC_ID_MP2:
164 case AV_CODEC_ID_MP3:
165 s->buf_ptr = s->buf + 4;
166 avpriv_set_pts_info(st, 32, 1, 90000);
168 case AV_CODEC_ID_MPEG1VIDEO:
169 case AV_CODEC_ID_MPEG2VIDEO:
171 case AV_CODEC_ID_MPEG2TS:
172 n = s->max_payload_size / TS_PACKET_SIZE;
175 s->max_payload_size = n * TS_PACKET_SIZE;
177 case AV_CODEC_ID_DIRAC:
178 if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
179 av_log(s, AV_LOG_ERROR,
180 "Packetizing VC-2 is experimental and does not use all values "
181 "of the specification "
182 "(even though most receivers may handle it just fine). "
183 "Please set -strict experimental in order to enable it.\n");
184 ret = AVERROR_EXPERIMENTAL;
188 case AV_CODEC_ID_H261:
189 if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
190 av_log(s, AV_LOG_ERROR,
191 "Packetizing H261 is experimental and produces incorrect "
192 "packetization for cases where GOBs don't fit into packets "
193 "(even though most receivers may handle it just fine). "
194 "Please set -f_strict experimental in order to enable it.\n");
195 ret = AVERROR_EXPERIMENTAL;
199 case AV_CODEC_ID_H264:
200 /* check for H.264 MP4 syntax */
201 if (st->codecpar->extradata_size > 4 && st->codecpar->extradata[0] == 1) {
202 s->nal_length_size = (st->codecpar->extradata[4] & 0x03) + 1;
205 case AV_CODEC_ID_HEVC:
206 /* Only check for the standardized hvcC version of extradata, keeping
207 * things simple and similar to the avcC/H264 case above, instead
208 * of trying to handle the pre-standardization versions (as in
209 * libavcodec/hevc.c). */
210 if (st->codecpar->extradata_size > 21 && st->codecpar->extradata[0] == 1) {
211 s->nal_length_size = (st->codecpar->extradata[21] & 0x03) + 1;
214 case AV_CODEC_ID_VORBIS:
215 case AV_CODEC_ID_THEORA:
216 s->max_frames_per_packet = 15;
218 case AV_CODEC_ID_ADPCM_G722:
219 /* Due to a historical error, the clock rate for G722 in RTP is
220 * 8000, even if the sample rate is 16000. See RFC 3551. */
221 avpriv_set_pts_info(st, 32, 1, 8000);
223 case AV_CODEC_ID_OPUS:
224 if (st->codecpar->channels > 2) {
225 av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
228 /* The opus RTP RFC says that all opus streams should use 48000 Hz
229 * as clock rate, since all opus sample rates can be expressed in
230 * this clock rate, and sample rate changes on the fly are supported. */
231 avpriv_set_pts_info(st, 32, 1, 48000);
233 case AV_CODEC_ID_ILBC:
234 if (st->codecpar->block_align != 38 && st->codecpar->block_align != 50) {
235 av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
238 s->max_frames_per_packet = s->max_payload_size / st->codecpar->block_align;
240 case AV_CODEC_ID_AMR_NB:
241 case AV_CODEC_ID_AMR_WB:
242 s->max_frames_per_packet = 50;
243 if (st->codecpar->codec_id == AV_CODEC_ID_AMR_NB)
247 /* max_header_toc_size + the largest AMR payload must fit */
248 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
249 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
252 if (st->codecpar->channels != 1) {
253 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
257 case AV_CODEC_ID_AAC:
258 s->max_frames_per_packet = 50;
271 /* send an rtcp sender report packet */
272 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
274 RTPMuxContext *s = s1->priv_data;
277 av_log(s1, AV_LOG_TRACE, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
279 s->last_rtcp_ntp_time = ntp_time;
280 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
281 s1->streams[0]->time_base) + s->base_timestamp;
282 avio_w8(s1->pb, RTP_VERSION << 6);
283 avio_w8(s1->pb, RTCP_SR);
284 avio_wb16(s1->pb, 6); /* length in words - 1 */
285 avio_wb32(s1->pb, s->ssrc);
286 avio_wb32(s1->pb, ntp_time / 1000000);
287 avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
288 avio_wb32(s1->pb, rtp_ts);
289 avio_wb32(s1->pb, s->packet_count);
290 avio_wb32(s1->pb, s->octet_count);
293 int len = FFMIN(strlen(s->cname), 255);
294 avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
295 avio_w8(s1->pb, RTCP_SDES);
296 avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
298 avio_wb32(s1->pb, s->ssrc);
299 avio_w8(s1->pb, 0x01); /* CNAME */
300 avio_w8(s1->pb, len);
301 avio_write(s1->pb, s->cname, len);
302 avio_w8(s1->pb, 0); /* END */
303 for (len = (7 + len) % 4; len % 4; len++)
308 avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
309 avio_w8(s1->pb, RTCP_BYE);
310 avio_wb16(s1->pb, 1); /* length in words - 1 */
311 avio_wb32(s1->pb, s->ssrc);
317 /* send an rtp packet. sequence number is incremented, but the caller
318 must update the timestamp itself */
319 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
321 RTPMuxContext *s = s1->priv_data;
323 av_log(s1, AV_LOG_TRACE, "rtp_send_data size=%d\n", len);
325 /* build the RTP header */
326 avio_w8(s1->pb, RTP_VERSION << 6);
327 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
328 avio_wb16(s1->pb, s->seq);
329 avio_wb32(s1->pb, s->timestamp);
330 avio_wb32(s1->pb, s->ssrc);
332 avio_write(s1->pb, buf1, len);
335 s->seq = (s->seq + 1) & 0xffff;
336 s->octet_count += len;
340 /* send an integer number of samples and compute time stamp and fill
341 the rtp send buffer before sending. */
342 static int rtp_send_samples(AVFormatContext *s1,
343 const uint8_t *buf1, int size, int sample_size_bits)
345 RTPMuxContext *s = s1->priv_data;
346 int len, max_packet_size, n;
347 /* Calculate the number of bytes to get samples aligned on a byte border */
348 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
350 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
351 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
352 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
353 return AVERROR(EINVAL);
357 len = FFMIN(max_packet_size, size);
360 memcpy(s->buf_ptr, buf1, len);
364 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
365 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
366 n += (s->buf_ptr - s->buf);
371 static void rtp_send_mpegaudio(AVFormatContext *s1,
372 const uint8_t *buf1, int size)
374 RTPMuxContext *s = s1->priv_data;
375 int len, count, max_packet_size;
377 max_packet_size = s->max_payload_size;
379 /* test if we must flush because not enough space */
380 len = (s->buf_ptr - s->buf);
381 if ((len + size) > max_packet_size) {
383 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
384 s->buf_ptr = s->buf + 4;
387 if (s->buf_ptr == s->buf + 4) {
388 s->timestamp = s->cur_timestamp;
392 if (size > max_packet_size) {
393 /* big packet: fragment */
396 len = max_packet_size - 4;
399 /* build fragmented packet */
402 s->buf[2] = count >> 8;
404 memcpy(s->buf + 4, buf1, len);
405 ff_rtp_send_data(s1, s->buf, len + 4, 0);
411 if (s->buf_ptr == s->buf + 4) {
412 /* no fragmentation possible */
418 memcpy(s->buf_ptr, buf1, size);
423 static void rtp_send_raw(AVFormatContext *s1,
424 const uint8_t *buf1, int size)
426 RTPMuxContext *s = s1->priv_data;
427 int len, max_packet_size;
429 max_packet_size = s->max_payload_size;
432 len = max_packet_size;
436 s->timestamp = s->cur_timestamp;
437 ff_rtp_send_data(s1, buf1, len, (len == size));
444 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
445 static void rtp_send_mpegts_raw(AVFormatContext *s1,
446 const uint8_t *buf1, int size)
448 RTPMuxContext *s = s1->priv_data;
451 s->timestamp = s->cur_timestamp;
452 while (size >= TS_PACKET_SIZE) {
453 len = s->max_payload_size - (s->buf_ptr - s->buf);
456 memcpy(s->buf_ptr, buf1, len);
461 out_len = s->buf_ptr - s->buf;
462 if (out_len >= s->max_payload_size) {
463 ff_rtp_send_data(s1, s->buf, out_len, 0);
469 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
471 RTPMuxContext *s = s1->priv_data;
472 AVStream *st = s1->streams[0];
473 int frame_duration = av_get_audio_frame_duration2(st->codecpar, 0);
474 int frame_size = st->codecpar->block_align;
475 int frames = size / frame_size;
478 if (s->num_frames > 0 &&
479 av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
480 s1->max_delay, AV_TIME_BASE_Q) >= 0) {
481 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
485 if (!s->num_frames) {
487 s->timestamp = s->cur_timestamp;
489 memcpy(s->buf_ptr, buf, frame_size);
492 s->buf_ptr += frame_size;
494 s->cur_timestamp += frame_duration;
496 if (s->num_frames == s->max_frames_per_packet) {
497 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
504 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
506 RTPMuxContext *s = s1->priv_data;
507 AVStream *st = s1->streams[0];
511 av_log(s1, AV_LOG_TRACE, "%d: write len=%d\n", pkt->stream_index, size);
513 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
515 if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
516 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
517 !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
518 rtcp_send_sr(s1, ff_ntp_time(), 0);
519 s->last_octet_count = s->octet_count;
522 s->cur_timestamp = s->base_timestamp + pkt->pts;
524 switch(st->codecpar->codec_id) {
525 case AV_CODEC_ID_PCM_MULAW:
526 case AV_CODEC_ID_PCM_ALAW:
527 case AV_CODEC_ID_PCM_U8:
528 case AV_CODEC_ID_PCM_S8:
529 return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
530 case AV_CODEC_ID_PCM_U16BE:
531 case AV_CODEC_ID_PCM_U16LE:
532 case AV_CODEC_ID_PCM_S16BE:
533 case AV_CODEC_ID_PCM_S16LE:
534 return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->channels);
535 case AV_CODEC_ID_ADPCM_G722:
536 /* The actual sample size is half a byte per sample, but since the
537 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
538 * the correct parameter for send_samples_bits is 8 bits per stream
540 return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
541 case AV_CODEC_ID_ADPCM_G726:
542 return rtp_send_samples(s1, pkt->data, size,
543 st->codecpar->bits_per_coded_sample * st->codecpar->channels);
544 case AV_CODEC_ID_MP2:
545 case AV_CODEC_ID_MP3:
546 rtp_send_mpegaudio(s1, pkt->data, size);
548 case AV_CODEC_ID_MPEG1VIDEO:
549 case AV_CODEC_ID_MPEG2VIDEO:
550 ff_rtp_send_mpegvideo(s1, pkt->data, size);
552 case AV_CODEC_ID_AAC:
553 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
554 ff_rtp_send_latm(s1, pkt->data, size);
556 ff_rtp_send_aac(s1, pkt->data, size);
558 case AV_CODEC_ID_AMR_NB:
559 case AV_CODEC_ID_AMR_WB:
560 ff_rtp_send_amr(s1, pkt->data, size);
562 case AV_CODEC_ID_MPEG2TS:
563 rtp_send_mpegts_raw(s1, pkt->data, size);
565 case AV_CODEC_ID_DIRAC:
566 ff_rtp_send_vc2hq(s1, pkt->data, size, st->codecpar->field_order != AV_FIELD_PROGRESSIVE ? 1 : 0);
568 case AV_CODEC_ID_H264:
569 ff_rtp_send_h264_hevc(s1, pkt->data, size);
571 case AV_CODEC_ID_H261:
572 ff_rtp_send_h261(s1, pkt->data, size);
574 case AV_CODEC_ID_H263:
575 if (s->flags & FF_RTP_FLAG_RFC2190) {
576 int mb_info_size = 0;
577 const uint8_t *mb_info =
578 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
580 ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
584 case AV_CODEC_ID_H263P:
585 ff_rtp_send_h263(s1, pkt->data, size);
587 case AV_CODEC_ID_HEVC:
588 ff_rtp_send_h264_hevc(s1, pkt->data, size);
590 case AV_CODEC_ID_VORBIS:
591 case AV_CODEC_ID_THEORA:
592 ff_rtp_send_xiph(s1, pkt->data, size);
594 case AV_CODEC_ID_VP8:
595 ff_rtp_send_vp8(s1, pkt->data, size);
597 case AV_CODEC_ID_ILBC:
598 rtp_send_ilbc(s1, pkt->data, size);
600 case AV_CODEC_ID_MJPEG:
601 ff_rtp_send_jpeg(s1, pkt->data, size);
603 case AV_CODEC_ID_OPUS:
604 if (size > s->max_payload_size) {
605 av_log(s1, AV_LOG_ERROR,
606 "Packet size %d too large for max RTP payload size %d\n",
607 size, s->max_payload_size);
608 return AVERROR(EINVAL);
610 /* Intentional fallthrough */
612 /* better than nothing : send the codec raw data */
613 rtp_send_raw(s1, pkt->data, size);
619 static int rtp_write_trailer(AVFormatContext *s1)
621 RTPMuxContext *s = s1->priv_data;
623 /* If the caller closes and recreates ->pb, this might actually
624 * be NULL here even if it was successfully allocated at the start. */
625 if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
626 rtcp_send_sr(s1, ff_ntp_time(), 1);
632 AVOutputFormat ff_rtp_muxer = {
634 .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
635 .priv_data_size = sizeof(RTPMuxContext),
636 .audio_codec = AV_CODEC_ID_PCM_MULAW,
637 .video_codec = AV_CODEC_ID_MPEG4,
638 .write_header = rtp_write_header,
639 .write_packet = rtp_write_packet,
640 .write_trailer = rtp_write_trailer,
641 .priv_class = &rtp_muxer_class,
642 .flags = AVFMT_TS_NONSTRICT,