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_S24BE:
70 case AV_CODEC_ID_PCM_U16BE:
71 case AV_CODEC_ID_PCM_U16LE:
72 case AV_CODEC_ID_PCM_U8:
73 case AV_CODEC_ID_MPEG2TS:
74 case AV_CODEC_ID_AMR_NB:
75 case AV_CODEC_ID_AMR_WB:
76 case AV_CODEC_ID_VORBIS:
77 case AV_CODEC_ID_THEORA:
80 case AV_CODEC_ID_ADPCM_G722:
81 case AV_CODEC_ID_ADPCM_G726:
82 case AV_CODEC_ID_ADPCM_G726LE:
83 case AV_CODEC_ID_ILBC:
84 case AV_CODEC_ID_MJPEG:
85 case AV_CODEC_ID_SPEEX:
86 case AV_CODEC_ID_OPUS:
93 static int rtp_write_header(AVFormatContext *s1)
95 RTPMuxContext *s = s1->priv_data;
96 int n, ret = AVERROR(EINVAL);
99 if (s1->nb_streams != 1) {
100 av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
101 return AVERROR(EINVAL);
104 if (!is_supported(st->codecpar->codec_id)) {
105 av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codecpar->codec_id));
110 if (s->payload_type < 0) {
111 /* Re-validate non-dynamic payload types */
112 if (st->id < RTP_PT_PRIVATE)
113 st->id = ff_rtp_get_payload_type(s1, st->codecpar, -1);
115 s->payload_type = st->id;
117 /* private option takes priority */
118 st->id = s->payload_type;
121 s->base_timestamp = av_get_random_seed();
122 s->timestamp = s->base_timestamp;
123 s->cur_timestamp = 0;
125 s->ssrc = av_get_random_seed();
127 s->first_rtcp_ntp_time = ff_ntp_time();
128 if (s1->start_time_realtime != 0 && s1->start_time_realtime != AV_NOPTS_VALUE)
129 /* Round the NTP time to whole milliseconds. */
130 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
132 // Pick a random sequence start number, but in the lower end of the
133 // available range, so that any wraparound doesn't happen immediately.
134 // (Immediate wraparound would be an issue for SRTP.)
136 if (s1->flags & AVFMT_FLAG_BITEXACT) {
139 s->seq = av_get_random_seed() & 0x0fff;
141 s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
143 if (s1->packet_size) {
144 if (s1->pb->max_packet_size)
145 s1->packet_size = FFMIN(s1->packet_size,
146 s1->pb->max_packet_size);
148 s1->packet_size = s1->pb->max_packet_size;
149 if (s1->packet_size <= 12) {
150 av_log(s1, AV_LOG_ERROR, "Max packet size %u too low\n", s1->packet_size);
153 s->buf = av_malloc(s1->packet_size);
155 return AVERROR(ENOMEM);
157 s->max_payload_size = s1->packet_size - 12;
159 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
160 avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
162 avpriv_set_pts_info(st, 32, 1, 90000);
165 switch(st->codecpar->codec_id) {
166 case AV_CODEC_ID_MP2:
167 case AV_CODEC_ID_MP3:
168 s->buf_ptr = s->buf + 4;
169 avpriv_set_pts_info(st, 32, 1, 90000);
171 case AV_CODEC_ID_MPEG1VIDEO:
172 case AV_CODEC_ID_MPEG2VIDEO:
174 case AV_CODEC_ID_MPEG2TS:
175 n = s->max_payload_size / TS_PACKET_SIZE;
178 s->max_payload_size = n * TS_PACKET_SIZE;
180 case AV_CODEC_ID_DIRAC:
181 if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
182 av_log(s, AV_LOG_ERROR,
183 "Packetizing VC-2 is experimental and does not use all values "
184 "of the specification "
185 "(even though most receivers may handle it just fine). "
186 "Please set -strict experimental in order to enable it.\n");
187 ret = AVERROR_EXPERIMENTAL;
191 case AV_CODEC_ID_H261:
192 if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
193 av_log(s, AV_LOG_ERROR,
194 "Packetizing H.261 is experimental and produces incorrect "
195 "packetization for cases where GOBs don't fit into packets "
196 "(even though most receivers may handle it just fine). "
197 "Please set -f_strict experimental in order to enable it.\n");
198 ret = AVERROR_EXPERIMENTAL;
202 case AV_CODEC_ID_H264:
203 /* check for H.264 MP4 syntax */
204 if (st->codecpar->extradata_size > 4 && st->codecpar->extradata[0] == 1) {
205 s->nal_length_size = (st->codecpar->extradata[4] & 0x03) + 1;
208 case AV_CODEC_ID_HEVC:
209 /* Only check for the standardized hvcC version of extradata, keeping
210 * things simple and similar to the avcC/H.264 case above, instead
211 * of trying to handle the pre-standardization versions (as in
212 * libavcodec/hevc.c). */
213 if (st->codecpar->extradata_size > 21 && st->codecpar->extradata[0] == 1) {
214 s->nal_length_size = (st->codecpar->extradata[21] & 0x03) + 1;
217 case AV_CODEC_ID_VP9:
218 if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
219 av_log(s, AV_LOG_ERROR,
220 "Packetizing VP9 is experimental and its specification is "
221 "still in draft state. "
222 "Please set -strict experimental in order to enable it.\n");
223 ret = AVERROR_EXPERIMENTAL;
227 case AV_CODEC_ID_VORBIS:
228 case AV_CODEC_ID_THEORA:
229 s->max_frames_per_packet = 15;
231 case AV_CODEC_ID_ADPCM_G722:
232 /* Due to a historical error, the clock rate for G722 in RTP is
233 * 8000, even if the sample rate is 16000. See RFC 3551. */
234 avpriv_set_pts_info(st, 32, 1, 8000);
236 case AV_CODEC_ID_OPUS:
237 if (st->codecpar->channels > 2) {
238 av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
241 /* The opus RTP RFC says that all opus streams should use 48000 Hz
242 * as clock rate, since all opus sample rates can be expressed in
243 * this clock rate, and sample rate changes on the fly are supported. */
244 avpriv_set_pts_info(st, 32, 1, 48000);
246 case AV_CODEC_ID_ILBC:
247 if (st->codecpar->block_align != 38 && st->codecpar->block_align != 50) {
248 av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
251 s->max_frames_per_packet = s->max_payload_size / st->codecpar->block_align;
253 case AV_CODEC_ID_AMR_NB:
254 case AV_CODEC_ID_AMR_WB:
255 s->max_frames_per_packet = 50;
256 if (st->codecpar->codec_id == AV_CODEC_ID_AMR_NB)
260 /* max_header_toc_size + the largest AMR payload must fit */
261 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
262 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
265 if (st->codecpar->channels != 1) {
266 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
270 case AV_CODEC_ID_AAC:
271 s->max_frames_per_packet = 50;
284 /* send an rtcp sender report packet */
285 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
287 RTPMuxContext *s = s1->priv_data;
290 av_log(s1, AV_LOG_TRACE, "RTCP: %02x %"PRIx64" %"PRIx32"\n", s->payload_type, ntp_time, s->timestamp);
292 s->last_rtcp_ntp_time = ntp_time;
293 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
294 s1->streams[0]->time_base) + s->base_timestamp;
295 avio_w8(s1->pb, RTP_VERSION << 6);
296 avio_w8(s1->pb, RTCP_SR);
297 avio_wb16(s1->pb, 6); /* length in words - 1 */
298 avio_wb32(s1->pb, s->ssrc);
299 avio_wb32(s1->pb, ntp_time / 1000000);
300 avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
301 avio_wb32(s1->pb, rtp_ts);
302 avio_wb32(s1->pb, s->packet_count);
303 avio_wb32(s1->pb, s->octet_count);
306 int len = FFMIN(strlen(s->cname), 255);
307 avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
308 avio_w8(s1->pb, RTCP_SDES);
309 avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
311 avio_wb32(s1->pb, s->ssrc);
312 avio_w8(s1->pb, 0x01); /* CNAME */
313 avio_w8(s1->pb, len);
314 avio_write(s1->pb, s->cname, len);
315 avio_w8(s1->pb, 0); /* END */
316 for (len = (7 + len) % 4; len % 4; len++)
321 avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
322 avio_w8(s1->pb, RTCP_BYE);
323 avio_wb16(s1->pb, 1); /* length in words - 1 */
324 avio_wb32(s1->pb, s->ssrc);
330 /* send an rtp packet. sequence number is incremented, but the caller
331 must update the timestamp itself */
332 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
334 RTPMuxContext *s = s1->priv_data;
336 av_log(s1, AV_LOG_TRACE, "rtp_send_data size=%d\n", len);
338 /* build the RTP header */
339 avio_w8(s1->pb, RTP_VERSION << 6);
340 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
341 avio_wb16(s1->pb, s->seq);
342 avio_wb32(s1->pb, s->timestamp);
343 avio_wb32(s1->pb, s->ssrc);
345 avio_write(s1->pb, buf1, len);
348 s->seq = (s->seq + 1) & 0xffff;
349 s->octet_count += len;
353 /* send an integer number of samples and compute time stamp and fill
354 the rtp send buffer before sending. */
355 static int rtp_send_samples(AVFormatContext *s1,
356 const uint8_t *buf1, int size, int sample_size_bits)
358 RTPMuxContext *s = s1->priv_data;
359 int len, max_packet_size, n;
360 /* Calculate the number of bytes to get samples aligned on a byte border */
361 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
363 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
364 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
365 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
366 return AVERROR(EINVAL);
370 len = FFMIN(max_packet_size, size);
373 memcpy(s->buf_ptr, buf1, len);
377 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
378 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
379 n += (s->buf_ptr - s->buf);
384 static void rtp_send_mpegaudio(AVFormatContext *s1,
385 const uint8_t *buf1, int size)
387 RTPMuxContext *s = s1->priv_data;
388 int len, count, max_packet_size;
390 max_packet_size = s->max_payload_size;
392 /* test if we must flush because not enough space */
393 len = (s->buf_ptr - s->buf);
394 if ((len + size) > max_packet_size) {
396 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
397 s->buf_ptr = s->buf + 4;
400 if (s->buf_ptr == s->buf + 4) {
401 s->timestamp = s->cur_timestamp;
405 if (size > max_packet_size) {
406 /* big packet: fragment */
409 len = max_packet_size - 4;
412 /* build fragmented packet */
415 s->buf[2] = count >> 8;
417 memcpy(s->buf + 4, buf1, len);
418 ff_rtp_send_data(s1, s->buf, len + 4, 0);
424 if (s->buf_ptr == s->buf + 4) {
425 /* no fragmentation possible */
431 memcpy(s->buf_ptr, buf1, size);
436 static void rtp_send_raw(AVFormatContext *s1,
437 const uint8_t *buf1, int size)
439 RTPMuxContext *s = s1->priv_data;
440 int len, max_packet_size;
442 max_packet_size = s->max_payload_size;
445 len = max_packet_size;
449 s->timestamp = s->cur_timestamp;
450 ff_rtp_send_data(s1, buf1, len, (len == size));
457 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
458 static void rtp_send_mpegts_raw(AVFormatContext *s1,
459 const uint8_t *buf1, int size)
461 RTPMuxContext *s = s1->priv_data;
464 s->timestamp = s->cur_timestamp;
465 while (size >= TS_PACKET_SIZE) {
466 len = s->max_payload_size - (s->buf_ptr - s->buf);
469 memcpy(s->buf_ptr, buf1, len);
474 out_len = s->buf_ptr - s->buf;
475 if (out_len >= s->max_payload_size) {
476 ff_rtp_send_data(s1, s->buf, out_len, 0);
482 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
484 RTPMuxContext *s = s1->priv_data;
485 AVStream *st = s1->streams[0];
486 int frame_duration = av_get_audio_frame_duration2(st->codecpar, 0);
487 int frame_size = st->codecpar->block_align;
488 int frames = size / frame_size;
491 if (s->num_frames > 0 &&
492 av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
493 s1->max_delay, AV_TIME_BASE_Q) >= 0) {
494 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
498 if (!s->num_frames) {
500 s->timestamp = s->cur_timestamp;
502 memcpy(s->buf_ptr, buf, frame_size);
505 s->buf_ptr += frame_size;
507 s->cur_timestamp += frame_duration;
509 if (s->num_frames == s->max_frames_per_packet) {
510 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
517 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
519 RTPMuxContext *s = s1->priv_data;
520 AVStream *st = s1->streams[0];
524 av_log(s1, AV_LOG_TRACE, "%d: write len=%d\n", pkt->stream_index, size);
526 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
528 if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
529 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
530 !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
531 rtcp_send_sr(s1, ff_ntp_time(), 0);
532 s->last_octet_count = s->octet_count;
535 s->cur_timestamp = s->base_timestamp + pkt->pts;
537 switch(st->codecpar->codec_id) {
538 case AV_CODEC_ID_PCM_MULAW:
539 case AV_CODEC_ID_PCM_ALAW:
540 case AV_CODEC_ID_PCM_U8:
541 case AV_CODEC_ID_PCM_S8:
542 return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
543 case AV_CODEC_ID_PCM_U16BE:
544 case AV_CODEC_ID_PCM_U16LE:
545 case AV_CODEC_ID_PCM_S16BE:
546 case AV_CODEC_ID_PCM_S16LE:
547 return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->channels);
548 case AV_CODEC_ID_PCM_S24BE:
549 return rtp_send_samples(s1, pkt->data, size, 24 * st->codecpar->channels);
550 case AV_CODEC_ID_ADPCM_G722:
551 /* The actual sample size is half a byte per sample, but since the
552 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
553 * the correct parameter for send_samples_bits is 8 bits per stream
555 return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
556 case AV_CODEC_ID_ADPCM_G726:
557 case AV_CODEC_ID_ADPCM_G726LE:
558 return rtp_send_samples(s1, pkt->data, size,
559 st->codecpar->bits_per_coded_sample * st->codecpar->channels);
560 case AV_CODEC_ID_MP2:
561 case AV_CODEC_ID_MP3:
562 rtp_send_mpegaudio(s1, pkt->data, size);
564 case AV_CODEC_ID_MPEG1VIDEO:
565 case AV_CODEC_ID_MPEG2VIDEO:
566 ff_rtp_send_mpegvideo(s1, pkt->data, size);
568 case AV_CODEC_ID_AAC:
569 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
570 ff_rtp_send_latm(s1, pkt->data, size);
572 ff_rtp_send_aac(s1, pkt->data, size);
574 case AV_CODEC_ID_AMR_NB:
575 case AV_CODEC_ID_AMR_WB:
576 ff_rtp_send_amr(s1, pkt->data, size);
578 case AV_CODEC_ID_MPEG2TS:
579 rtp_send_mpegts_raw(s1, pkt->data, size);
581 case AV_CODEC_ID_DIRAC:
582 ff_rtp_send_vc2hq(s1, pkt->data, size, st->codecpar->field_order != AV_FIELD_PROGRESSIVE ? 1 : 0);
584 case AV_CODEC_ID_H264:
585 ff_rtp_send_h264_hevc(s1, pkt->data, size);
587 case AV_CODEC_ID_H261:
588 ff_rtp_send_h261(s1, pkt->data, size);
590 case AV_CODEC_ID_H263:
591 if (s->flags & FF_RTP_FLAG_RFC2190) {
592 int mb_info_size = 0;
593 const uint8_t *mb_info =
594 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
596 ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
600 case AV_CODEC_ID_H263P:
601 ff_rtp_send_h263(s1, pkt->data, size);
603 case AV_CODEC_ID_HEVC:
604 ff_rtp_send_h264_hevc(s1, pkt->data, size);
606 case AV_CODEC_ID_VORBIS:
607 case AV_CODEC_ID_THEORA:
608 ff_rtp_send_xiph(s1, pkt->data, size);
610 case AV_CODEC_ID_VP8:
611 ff_rtp_send_vp8(s1, pkt->data, size);
613 case AV_CODEC_ID_VP9:
614 ff_rtp_send_vp9(s1, pkt->data, size);
616 case AV_CODEC_ID_ILBC:
617 rtp_send_ilbc(s1, pkt->data, size);
619 case AV_CODEC_ID_MJPEG:
620 ff_rtp_send_jpeg(s1, pkt->data, size);
622 case AV_CODEC_ID_OPUS:
623 if (size > s->max_payload_size) {
624 av_log(s1, AV_LOG_ERROR,
625 "Packet size %d too large for max RTP payload size %d\n",
626 size, s->max_payload_size);
627 return AVERROR(EINVAL);
629 /* Intentional fallthrough */
631 /* better than nothing : send the codec raw data */
632 rtp_send_raw(s1, pkt->data, size);
638 static int rtp_write_trailer(AVFormatContext *s1)
640 RTPMuxContext *s = s1->priv_data;
642 /* If the caller closes and recreates ->pb, this might actually
643 * be NULL here even if it was successfully allocated at the start. */
644 if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
645 rtcp_send_sr(s1, ff_ntp_time(), 1);
651 AVOutputFormat ff_rtp_muxer = {
653 .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
654 .priv_data_size = sizeof(RTPMuxContext),
655 .audio_codec = AV_CODEC_ID_PCM_MULAW,
656 .video_codec = AV_CODEC_ID_MPEG4,
657 .write_header = rtp_write_header,
658 .write_packet = rtp_write_packet,
659 .write_trailer = rtp_write_trailer,
660 .priv_class = &rtp_muxer_class,
661 .flags = AVFMT_TS_NONSTRICT,