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_H261:
53 case AV_CODEC_ID_H263:
54 case AV_CODEC_ID_H263P:
55 case AV_CODEC_ID_H264:
56 case AV_CODEC_ID_HEVC:
57 case AV_CODEC_ID_MPEG1VIDEO:
58 case AV_CODEC_ID_MPEG2VIDEO:
59 case AV_CODEC_ID_MPEG4:
63 case AV_CODEC_ID_PCM_ALAW:
64 case AV_CODEC_ID_PCM_MULAW:
65 case AV_CODEC_ID_PCM_S8:
66 case AV_CODEC_ID_PCM_S16BE:
67 case AV_CODEC_ID_PCM_S16LE:
68 case AV_CODEC_ID_PCM_U16BE:
69 case AV_CODEC_ID_PCM_U16LE:
70 case AV_CODEC_ID_PCM_U8:
71 case AV_CODEC_ID_MPEG2TS:
72 case AV_CODEC_ID_AMR_NB:
73 case AV_CODEC_ID_AMR_WB:
74 case AV_CODEC_ID_VORBIS:
75 case AV_CODEC_ID_THEORA:
77 case AV_CODEC_ID_ADPCM_G722:
78 case AV_CODEC_ID_ADPCM_G726:
79 case AV_CODEC_ID_ILBC:
80 case AV_CODEC_ID_MJPEG:
81 case AV_CODEC_ID_SPEEX:
82 case AV_CODEC_ID_OPUS:
89 static int rtp_write_header(AVFormatContext *s1)
91 RTPMuxContext *s = s1->priv_data;
92 int n, ret = AVERROR(EINVAL);
95 if (s1->nb_streams != 1) {
96 av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
97 return AVERROR(EINVAL);
100 if (!is_supported(st->codec->codec_id)) {
101 av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));
106 if (s->payload_type < 0) {
107 /* Re-validate non-dynamic payload types */
108 if (st->id < RTP_PT_PRIVATE)
109 st->id = ff_rtp_get_payload_type(s1, st->codec, -1);
111 s->payload_type = st->id;
113 /* private option takes priority */
114 st->id = s->payload_type;
117 s->base_timestamp = av_get_random_seed();
118 s->timestamp = s->base_timestamp;
119 s->cur_timestamp = 0;
121 s->ssrc = av_get_random_seed();
123 s->first_rtcp_ntp_time = ff_ntp_time();
124 if (s1->start_time_realtime != 0 && s1->start_time_realtime != AV_NOPTS_VALUE)
125 /* Round the NTP time to whole milliseconds. */
126 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
128 // Pick a random sequence start number, but in the lower end of the
129 // available range, so that any wraparound doesn't happen immediately.
130 // (Immediate wraparound would be an issue for SRTP.)
132 if (s1->flags & AVFMT_FLAG_BITEXACT) {
135 s->seq = av_get_random_seed() & 0x0fff;
137 s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
139 if (s1->packet_size) {
140 if (s1->pb->max_packet_size)
141 s1->packet_size = FFMIN(s1->packet_size,
142 s1->pb->max_packet_size);
144 s1->packet_size = s1->pb->max_packet_size;
145 if (s1->packet_size <= 12) {
146 av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
149 s->buf = av_malloc(s1->packet_size);
151 return AVERROR(ENOMEM);
153 s->max_payload_size = s1->packet_size - 12;
155 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
156 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
158 avpriv_set_pts_info(st, 32, 1, 90000);
161 switch(st->codec->codec_id) {
162 case AV_CODEC_ID_MP2:
163 case AV_CODEC_ID_MP3:
164 s->buf_ptr = s->buf + 4;
165 avpriv_set_pts_info(st, 32, 1, 90000);
167 case AV_CODEC_ID_MPEG1VIDEO:
168 case AV_CODEC_ID_MPEG2VIDEO:
170 case AV_CODEC_ID_MPEG2TS:
171 n = s->max_payload_size / TS_PACKET_SIZE;
174 s->max_payload_size = n * TS_PACKET_SIZE;
176 case AV_CODEC_ID_H261:
177 if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
178 av_log(s, AV_LOG_ERROR,
179 "Packetizing H261 is experimental and produces incorrect "
180 "packetization for cases where GOBs don't fit into packets "
181 "(even though most receivers may handle it just fine). "
182 "Please set -f_strict experimental in order to enable it.\n");
183 ret = AVERROR_EXPERIMENTAL;
187 case AV_CODEC_ID_H264:
188 /* check for H.264 MP4 syntax */
189 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
190 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
193 case AV_CODEC_ID_HEVC:
194 /* Only check for the standardized hvcC version of extradata, keeping
195 * things simple and similar to the avcC/H264 case above, instead
196 * of trying to handle the pre-standardization versions (as in
197 * libavcodec/hevc.c). */
198 if (st->codec->extradata_size > 21 && st->codec->extradata[0] == 1) {
199 s->nal_length_size = (st->codec->extradata[21] & 0x03) + 1;
202 case AV_CODEC_ID_VORBIS:
203 case AV_CODEC_ID_THEORA:
204 s->max_frames_per_packet = 15;
206 case AV_CODEC_ID_ADPCM_G722:
207 /* Due to a historical error, the clock rate for G722 in RTP is
208 * 8000, even if the sample rate is 16000. See RFC 3551. */
209 avpriv_set_pts_info(st, 32, 1, 8000);
211 case AV_CODEC_ID_OPUS:
212 if (st->codec->channels > 2) {
213 av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
216 /* The opus RTP RFC says that all opus streams should use 48000 Hz
217 * as clock rate, since all opus sample rates can be expressed in
218 * this clock rate, and sample rate changes on the fly are supported. */
219 avpriv_set_pts_info(st, 32, 1, 48000);
221 case AV_CODEC_ID_ILBC:
222 if (st->codec->block_align != 38 && st->codec->block_align != 50) {
223 av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
226 s->max_frames_per_packet = s->max_payload_size / st->codec->block_align;
228 case AV_CODEC_ID_AMR_NB:
229 case AV_CODEC_ID_AMR_WB:
230 s->max_frames_per_packet = 50;
231 if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
235 /* max_header_toc_size + the largest AMR payload must fit */
236 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
237 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
240 if (st->codec->channels != 1) {
241 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
245 case AV_CODEC_ID_AAC:
246 s->max_frames_per_packet = 50;
259 /* send an rtcp sender report packet */
260 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
262 RTPMuxContext *s = s1->priv_data;
265 av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
267 s->last_rtcp_ntp_time = ntp_time;
268 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
269 s1->streams[0]->time_base) + s->base_timestamp;
270 avio_w8(s1->pb, RTP_VERSION << 6);
271 avio_w8(s1->pb, RTCP_SR);
272 avio_wb16(s1->pb, 6); /* length in words - 1 */
273 avio_wb32(s1->pb, s->ssrc);
274 avio_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time));
275 avio_wb32(s1->pb, rtp_ts);
276 avio_wb32(s1->pb, s->packet_count);
277 avio_wb32(s1->pb, s->octet_count);
280 int len = FFMIN(strlen(s->cname), 255);
281 avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
282 avio_w8(s1->pb, RTCP_SDES);
283 avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
285 avio_wb32(s1->pb, s->ssrc);
286 avio_w8(s1->pb, 0x01); /* CNAME */
287 avio_w8(s1->pb, len);
288 avio_write(s1->pb, s->cname, len);
289 avio_w8(s1->pb, 0); /* END */
290 for (len = (7 + len) % 4; len % 4; len++)
295 avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
296 avio_w8(s1->pb, RTCP_BYE);
297 avio_wb16(s1->pb, 1); /* length in words - 1 */
298 avio_wb32(s1->pb, s->ssrc);
304 /* send an rtp packet. sequence number is incremented, but the caller
305 must update the timestamp itself */
306 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
308 RTPMuxContext *s = s1->priv_data;
310 av_dlog(s1, "rtp_send_data size=%d\n", len);
312 /* build the RTP header */
313 avio_w8(s1->pb, RTP_VERSION << 6);
314 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
315 avio_wb16(s1->pb, s->seq);
316 avio_wb32(s1->pb, s->timestamp);
317 avio_wb32(s1->pb, s->ssrc);
319 avio_write(s1->pb, buf1, len);
322 s->seq = (s->seq + 1) & 0xffff;
323 s->octet_count += len;
327 /* send an integer number of samples and compute time stamp and fill
328 the rtp send buffer before sending. */
329 static int rtp_send_samples(AVFormatContext *s1,
330 const uint8_t *buf1, int size, int sample_size_bits)
332 RTPMuxContext *s = s1->priv_data;
333 int len, max_packet_size, n;
334 /* Calculate the number of bytes to get samples aligned on a byte border */
335 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
337 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
338 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
339 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
340 return AVERROR(EINVAL);
344 len = FFMIN(max_packet_size, size);
347 memcpy(s->buf_ptr, buf1, len);
351 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
352 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
353 n += (s->buf_ptr - s->buf);
358 static void rtp_send_mpegaudio(AVFormatContext *s1,
359 const uint8_t *buf1, int size)
361 RTPMuxContext *s = s1->priv_data;
362 int len, count, max_packet_size;
364 max_packet_size = s->max_payload_size;
366 /* test if we must flush because not enough space */
367 len = (s->buf_ptr - s->buf);
368 if ((len + size) > max_packet_size) {
370 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
371 s->buf_ptr = s->buf + 4;
374 if (s->buf_ptr == s->buf + 4) {
375 s->timestamp = s->cur_timestamp;
379 if (size > max_packet_size) {
380 /* big packet: fragment */
383 len = max_packet_size - 4;
386 /* build fragmented packet */
389 s->buf[2] = count >> 8;
391 memcpy(s->buf + 4, buf1, len);
392 ff_rtp_send_data(s1, s->buf, len + 4, 0);
398 if (s->buf_ptr == s->buf + 4) {
399 /* no fragmentation possible */
405 memcpy(s->buf_ptr, buf1, size);
410 static void rtp_send_raw(AVFormatContext *s1,
411 const uint8_t *buf1, int size)
413 RTPMuxContext *s = s1->priv_data;
414 int len, max_packet_size;
416 max_packet_size = s->max_payload_size;
419 len = max_packet_size;
423 s->timestamp = s->cur_timestamp;
424 ff_rtp_send_data(s1, buf1, len, (len == size));
431 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
432 static void rtp_send_mpegts_raw(AVFormatContext *s1,
433 const uint8_t *buf1, int size)
435 RTPMuxContext *s = s1->priv_data;
438 s->timestamp = s->cur_timestamp;
439 while (size >= TS_PACKET_SIZE) {
440 len = s->max_payload_size - (s->buf_ptr - s->buf);
443 memcpy(s->buf_ptr, buf1, len);
448 out_len = s->buf_ptr - s->buf;
449 if (out_len >= s->max_payload_size) {
450 ff_rtp_send_data(s1, s->buf, out_len, 0);
456 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
458 RTPMuxContext *s = s1->priv_data;
459 AVStream *st = s1->streams[0];
460 int frame_duration = av_get_audio_frame_duration(st->codec, 0);
461 int frame_size = st->codec->block_align;
462 int frames = size / frame_size;
465 if (s->num_frames > 0 &&
466 av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
467 s1->max_delay, AV_TIME_BASE_Q) >= 0) {
468 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
472 if (!s->num_frames) {
474 s->timestamp = s->cur_timestamp;
476 memcpy(s->buf_ptr, buf, frame_size);
479 s->buf_ptr += frame_size;
481 s->cur_timestamp += frame_duration;
483 if (s->num_frames == s->max_frames_per_packet) {
484 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
491 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
493 RTPMuxContext *s = s1->priv_data;
494 AVStream *st = s1->streams[0];
498 av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
500 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
502 if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
503 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
504 !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
505 rtcp_send_sr(s1, ff_ntp_time(), 0);
506 s->last_octet_count = s->octet_count;
509 s->cur_timestamp = s->base_timestamp + pkt->pts;
511 switch(st->codec->codec_id) {
512 case AV_CODEC_ID_PCM_MULAW:
513 case AV_CODEC_ID_PCM_ALAW:
514 case AV_CODEC_ID_PCM_U8:
515 case AV_CODEC_ID_PCM_S8:
516 return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
517 case AV_CODEC_ID_PCM_U16BE:
518 case AV_CODEC_ID_PCM_U16LE:
519 case AV_CODEC_ID_PCM_S16BE:
520 case AV_CODEC_ID_PCM_S16LE:
521 return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
522 case AV_CODEC_ID_ADPCM_G722:
523 /* The actual sample size is half a byte per sample, but since the
524 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
525 * the correct parameter for send_samples_bits is 8 bits per stream
527 return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
528 case AV_CODEC_ID_ADPCM_G726:
529 return rtp_send_samples(s1, pkt->data, size,
530 st->codec->bits_per_coded_sample * st->codec->channels);
531 case AV_CODEC_ID_MP2:
532 case AV_CODEC_ID_MP3:
533 rtp_send_mpegaudio(s1, pkt->data, size);
535 case AV_CODEC_ID_MPEG1VIDEO:
536 case AV_CODEC_ID_MPEG2VIDEO:
537 ff_rtp_send_mpegvideo(s1, pkt->data, size);
539 case AV_CODEC_ID_AAC:
540 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
541 ff_rtp_send_latm(s1, pkt->data, size);
543 ff_rtp_send_aac(s1, pkt->data, size);
545 case AV_CODEC_ID_AMR_NB:
546 case AV_CODEC_ID_AMR_WB:
547 ff_rtp_send_amr(s1, pkt->data, size);
549 case AV_CODEC_ID_MPEG2TS:
550 rtp_send_mpegts_raw(s1, pkt->data, size);
552 case AV_CODEC_ID_H264:
553 ff_rtp_send_h264_hevc(s1, pkt->data, size);
555 case AV_CODEC_ID_H261:
556 ff_rtp_send_h261(s1, pkt->data, size);
558 case AV_CODEC_ID_H263:
559 if (s->flags & FF_RTP_FLAG_RFC2190) {
560 int mb_info_size = 0;
561 const uint8_t *mb_info =
562 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
565 av_log(s1, AV_LOG_ERROR, "failed to allocate side data\n");
566 return AVERROR(ENOMEM);
568 ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
572 case AV_CODEC_ID_H263P:
573 ff_rtp_send_h263(s1, pkt->data, size);
575 case AV_CODEC_ID_HEVC:
576 ff_rtp_send_h264_hevc(s1, pkt->data, size);
578 case AV_CODEC_ID_VORBIS:
579 case AV_CODEC_ID_THEORA:
580 ff_rtp_send_xiph(s1, pkt->data, size);
582 case AV_CODEC_ID_VP8:
583 ff_rtp_send_vp8(s1, pkt->data, size);
585 case AV_CODEC_ID_ILBC:
586 rtp_send_ilbc(s1, pkt->data, size);
588 case AV_CODEC_ID_MJPEG:
589 ff_rtp_send_jpeg(s1, pkt->data, size);
591 case AV_CODEC_ID_OPUS:
592 if (size > s->max_payload_size) {
593 av_log(s1, AV_LOG_ERROR,
594 "Packet size %d too large for max RTP payload size %d\n",
595 size, s->max_payload_size);
596 return AVERROR(EINVAL);
598 /* Intentional fallthrough */
600 /* better than nothing : send the codec raw data */
601 rtp_send_raw(s1, pkt->data, size);
607 static int rtp_write_trailer(AVFormatContext *s1)
609 RTPMuxContext *s = s1->priv_data;
611 /* If the caller closes and recreates ->pb, this might actually
612 * be NULL here even if it was successfully allocated at the start. */
613 if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
614 rtcp_send_sr(s1, ff_ntp_time(), 1);
620 AVOutputFormat ff_rtp_muxer = {
622 .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
623 .priv_data_size = sizeof(RTPMuxContext),
624 .audio_codec = AV_CODEC_ID_PCM_MULAW,
625 .video_codec = AV_CODEC_ID_MPEG4,
626 .write_header = rtp_write_header,
627 .write_packet = rtp_write_packet,
628 .write_trailer = rtp_write_trailer,
629 .priv_class = &rtp_muxer_class,
630 .flags = AVFMT_TS_NONSTRICT,