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_HEVC:
56 case AV_CODEC_ID_MPEG1VIDEO:
57 case AV_CODEC_ID_MPEG2VIDEO:
58 case AV_CODEC_ID_MPEG4:
62 case AV_CODEC_ID_PCM_ALAW:
63 case AV_CODEC_ID_PCM_MULAW:
64 case AV_CODEC_ID_PCM_S8:
65 case AV_CODEC_ID_PCM_S16BE:
66 case AV_CODEC_ID_PCM_S16LE:
67 case AV_CODEC_ID_PCM_U16BE:
68 case AV_CODEC_ID_PCM_U16LE:
69 case AV_CODEC_ID_PCM_U8:
70 case AV_CODEC_ID_MPEG2TS:
71 case AV_CODEC_ID_AMR_NB:
72 case AV_CODEC_ID_AMR_WB:
73 case AV_CODEC_ID_VORBIS:
74 case AV_CODEC_ID_THEORA:
76 case AV_CODEC_ID_ADPCM_G722:
77 case AV_CODEC_ID_ADPCM_G726:
78 case AV_CODEC_ID_ILBC:
79 case AV_CODEC_ID_MJPEG:
80 case AV_CODEC_ID_SPEEX:
81 case AV_CODEC_ID_OPUS:
88 static int rtp_write_header(AVFormatContext *s1)
90 RTPMuxContext *s = s1->priv_data;
94 if (s1->nb_streams != 1) {
95 av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
96 return AVERROR(EINVAL);
99 if (!is_supported(st->codec->codec_id)) {
100 av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
105 if (s->payload_type < 0) {
106 /* Re-validate non-dynamic payload types */
107 if (st->id < RTP_PT_PRIVATE)
108 st->id = ff_rtp_get_payload_type(s1, st->codec, -1);
110 s->payload_type = st->id;
112 /* private option takes priority */
113 st->id = s->payload_type;
116 s->base_timestamp = av_get_random_seed();
117 s->timestamp = s->base_timestamp;
118 s->cur_timestamp = 0;
120 s->ssrc = av_get_random_seed();
122 s->first_rtcp_ntp_time = ff_ntp_time();
123 if (s1->start_time_realtime)
124 /* Round the NTP time to whole milliseconds. */
125 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
127 // Pick a random sequence start number, but in the lower end of the
128 // available range, so that any wraparound doesn't happen immediately.
129 // (Immediate wraparound would be an issue for SRTP.)
131 s->seq = av_get_random_seed() & 0x0fff;
133 s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
135 if (s1->packet_size) {
136 if (s1->pb->max_packet_size)
137 s1->packet_size = FFMIN(s1->packet_size,
138 s1->pb->max_packet_size);
140 s1->packet_size = s1->pb->max_packet_size;
141 if (s1->packet_size <= 12) {
142 av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
145 s->buf = av_malloc(s1->packet_size);
147 return AVERROR(ENOMEM);
149 s->max_payload_size = s1->packet_size - 12;
151 s->max_frames_per_packet = 0;
152 if (s1->max_delay > 0) {
153 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
154 int frame_size = av_get_audio_frame_duration(st->codec, 0);
156 frame_size = st->codec->frame_size;
157 if (frame_size == 0) {
158 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
160 s->max_frames_per_packet =
161 av_rescale_q_rnd(s1->max_delay,
163 (AVRational){ frame_size, st->codec->sample_rate },
167 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
168 /* FIXME: We should round down here... */
169 if (st->avg_frame_rate.num > 0 && st->avg_frame_rate.den > 0) {
170 s->max_frames_per_packet = av_rescale_q(s1->max_delay,
171 (AVRational){1, 1000000},
172 av_inv_q(st->avg_frame_rate));
174 s->max_frames_per_packet = 1;
178 avpriv_set_pts_info(st, 32, 1, 90000);
179 switch(st->codec->codec_id) {
180 case AV_CODEC_ID_MP2:
181 case AV_CODEC_ID_MP3:
182 s->buf_ptr = s->buf + 4;
184 case AV_CODEC_ID_MPEG1VIDEO:
185 case AV_CODEC_ID_MPEG2VIDEO:
187 case AV_CODEC_ID_MPEG2TS:
188 n = s->max_payload_size / TS_PACKET_SIZE;
191 s->max_payload_size = n * TS_PACKET_SIZE;
194 case AV_CODEC_ID_H264:
195 /* check for H.264 MP4 syntax */
196 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
197 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
200 case AV_CODEC_ID_HEVC:
201 /* Only check for the standardized hvcC version of extradata, keeping
202 * things simple and similar to the avcC/H264 case above, instead
203 * of trying to handle the pre-standardization versions (as in
204 * libavcodec/hevc.c). */
205 if (st->codec->extradata_size > 21 && st->codec->extradata[0] == 1) {
206 s->nal_length_size = (st->codec->extradata[21] & 0x03) + 1;
209 case AV_CODEC_ID_VORBIS:
210 case AV_CODEC_ID_THEORA:
211 if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
212 s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
213 s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
216 case AV_CODEC_ID_ADPCM_G722:
217 /* Due to a historical error, the clock rate for G722 in RTP is
218 * 8000, even if the sample rate is 16000. See RFC 3551. */
219 avpriv_set_pts_info(st, 32, 1, 8000);
221 case AV_CODEC_ID_OPUS:
222 if (st->codec->channels > 2) {
223 av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
226 /* The opus RTP RFC says that all opus streams should use 48000 Hz
227 * as clock rate, since all opus sample rates can be expressed in
228 * this clock rate, and sample rate changes on the fly are supported. */
229 avpriv_set_pts_info(st, 32, 1, 48000);
231 case AV_CODEC_ID_ILBC:
232 if (st->codec->block_align != 38 && st->codec->block_align != 50) {
233 av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
236 if (!s->max_frames_per_packet)
237 s->max_frames_per_packet = 1;
238 s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
239 s->max_payload_size / st->codec->block_align);
241 case AV_CODEC_ID_AMR_NB:
242 case AV_CODEC_ID_AMR_WB:
243 if (!s->max_frames_per_packet)
244 s->max_frames_per_packet = 12;
245 if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
249 /* max_header_toc_size + the largest AMR payload must fit */
250 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
251 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
254 if (st->codec->channels != 1) {
255 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
258 case AV_CODEC_ID_AAC:
262 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
263 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
273 return AVERROR(EINVAL);
276 /* send an rtcp sender report packet */
277 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
279 RTPMuxContext *s = s1->priv_data;
282 av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
284 s->last_rtcp_ntp_time = ntp_time;
285 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
286 s1->streams[0]->time_base) + s->base_timestamp;
287 avio_w8(s1->pb, RTP_VERSION << 6);
288 avio_w8(s1->pb, RTCP_SR);
289 avio_wb16(s1->pb, 6); /* length in words - 1 */
290 avio_wb32(s1->pb, s->ssrc);
291 avio_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time));
292 avio_wb32(s1->pb, rtp_ts);
293 avio_wb32(s1->pb, s->packet_count);
294 avio_wb32(s1->pb, s->octet_count);
297 int len = FFMIN(strlen(s->cname), 255);
298 avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
299 avio_w8(s1->pb, RTCP_SDES);
300 avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
302 avio_wb32(s1->pb, s->ssrc);
303 avio_w8(s1->pb, 0x01); /* CNAME */
304 avio_w8(s1->pb, len);
305 avio_write(s1->pb, s->cname, len);
306 avio_w8(s1->pb, 0); /* END */
307 for (len = (7 + len) % 4; len % 4; len++)
312 avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
313 avio_w8(s1->pb, RTCP_BYE);
314 avio_wb16(s1->pb, 1); /* length in words - 1 */
315 avio_wb32(s1->pb, s->ssrc);
321 /* send an rtp packet. sequence number is incremented, but the caller
322 must update the timestamp itself */
323 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
325 RTPMuxContext *s = s1->priv_data;
327 av_dlog(s1, "rtp_send_data size=%d\n", len);
329 /* build the RTP header */
330 avio_w8(s1->pb, RTP_VERSION << 6);
331 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
332 avio_wb16(s1->pb, s->seq);
333 avio_wb32(s1->pb, s->timestamp);
334 avio_wb32(s1->pb, s->ssrc);
336 avio_write(s1->pb, buf1, len);
339 s->seq = (s->seq + 1) & 0xffff;
340 s->octet_count += len;
344 /* send an integer number of samples and compute time stamp and fill
345 the rtp send buffer before sending. */
346 static int rtp_send_samples(AVFormatContext *s1,
347 const uint8_t *buf1, int size, int sample_size_bits)
349 RTPMuxContext *s = s1->priv_data;
350 int len, max_packet_size, n;
351 /* Calculate the number of bytes to get samples aligned on a byte border */
352 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
354 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
355 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
356 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
357 return AVERROR(EINVAL);
361 len = FFMIN(max_packet_size, size);
364 memcpy(s->buf_ptr, buf1, len);
368 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
369 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
370 n += (s->buf_ptr - s->buf);
375 static void rtp_send_mpegaudio(AVFormatContext *s1,
376 const uint8_t *buf1, int size)
378 RTPMuxContext *s = s1->priv_data;
379 int len, count, max_packet_size;
381 max_packet_size = s->max_payload_size;
383 /* test if we must flush because not enough space */
384 len = (s->buf_ptr - s->buf);
385 if ((len + size) > max_packet_size) {
387 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
388 s->buf_ptr = s->buf + 4;
391 if (s->buf_ptr == s->buf + 4) {
392 s->timestamp = s->cur_timestamp;
396 if (size > max_packet_size) {
397 /* big packet: fragment */
400 len = max_packet_size - 4;
403 /* build fragmented packet */
406 s->buf[2] = count >> 8;
408 memcpy(s->buf + 4, buf1, len);
409 ff_rtp_send_data(s1, s->buf, len + 4, 0);
415 if (s->buf_ptr == s->buf + 4) {
416 /* no fragmentation possible */
422 memcpy(s->buf_ptr, buf1, size);
427 static void rtp_send_raw(AVFormatContext *s1,
428 const uint8_t *buf1, int size)
430 RTPMuxContext *s = s1->priv_data;
431 int len, max_packet_size;
433 max_packet_size = s->max_payload_size;
436 len = max_packet_size;
440 s->timestamp = s->cur_timestamp;
441 ff_rtp_send_data(s1, buf1, len, (len == size));
448 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
449 static void rtp_send_mpegts_raw(AVFormatContext *s1,
450 const uint8_t *buf1, int size)
452 RTPMuxContext *s = s1->priv_data;
455 while (size >= TS_PACKET_SIZE) {
456 len = s->max_payload_size - (s->buf_ptr - s->buf);
459 memcpy(s->buf_ptr, buf1, len);
464 out_len = s->buf_ptr - s->buf;
465 if (out_len >= s->max_payload_size) {
466 ff_rtp_send_data(s1, s->buf, out_len, 0);
472 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
474 RTPMuxContext *s = s1->priv_data;
475 AVStream *st = s1->streams[0];
476 int frame_duration = av_get_audio_frame_duration(st->codec, 0);
477 int frame_size = st->codec->block_align;
478 int frames = size / frame_size;
481 int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
483 if (!s->num_frames) {
485 s->timestamp = s->cur_timestamp;
487 memcpy(s->buf_ptr, buf, n * frame_size);
490 s->buf_ptr += n * frame_size;
491 buf += n * frame_size;
492 s->cur_timestamp += n * frame_duration;
494 if (s->num_frames == s->max_frames_per_packet) {
495 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
502 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
504 RTPMuxContext *s = s1->priv_data;
505 AVStream *st = s1->streams[0];
509 av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
511 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
513 if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
514 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
515 !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
516 rtcp_send_sr(s1, ff_ntp_time(), 0);
517 s->last_octet_count = s->octet_count;
520 s->cur_timestamp = s->base_timestamp + pkt->pts;
522 switch(st->codec->codec_id) {
523 case AV_CODEC_ID_PCM_MULAW:
524 case AV_CODEC_ID_PCM_ALAW:
525 case AV_CODEC_ID_PCM_U8:
526 case AV_CODEC_ID_PCM_S8:
527 return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
528 case AV_CODEC_ID_PCM_U16BE:
529 case AV_CODEC_ID_PCM_U16LE:
530 case AV_CODEC_ID_PCM_S16BE:
531 case AV_CODEC_ID_PCM_S16LE:
532 return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
533 case AV_CODEC_ID_ADPCM_G722:
534 /* The actual sample size is half a byte per sample, but since the
535 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
536 * the correct parameter for send_samples_bits is 8 bits per stream
538 return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
539 case AV_CODEC_ID_ADPCM_G726:
540 return rtp_send_samples(s1, pkt->data, size,
541 st->codec->bits_per_coded_sample * st->codec->channels);
542 case AV_CODEC_ID_MP2:
543 case AV_CODEC_ID_MP3:
544 rtp_send_mpegaudio(s1, pkt->data, size);
546 case AV_CODEC_ID_MPEG1VIDEO:
547 case AV_CODEC_ID_MPEG2VIDEO:
548 ff_rtp_send_mpegvideo(s1, pkt->data, size);
550 case AV_CODEC_ID_AAC:
551 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
552 ff_rtp_send_latm(s1, pkt->data, size);
554 ff_rtp_send_aac(s1, pkt->data, size);
556 case AV_CODEC_ID_AMR_NB:
557 case AV_CODEC_ID_AMR_WB:
558 ff_rtp_send_amr(s1, pkt->data, size);
560 case AV_CODEC_ID_MPEG2TS:
561 rtp_send_mpegts_raw(s1, pkt->data, size);
563 case AV_CODEC_ID_H264:
564 ff_rtp_send_h264(s1, pkt->data, size);
566 case AV_CODEC_ID_H263:
567 if (s->flags & FF_RTP_FLAG_RFC2190) {
568 int mb_info_size = 0;
569 const uint8_t *mb_info =
570 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
572 ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
576 case AV_CODEC_ID_H263P:
577 ff_rtp_send_h263(s1, pkt->data, size);
579 case AV_CODEC_ID_HEVC:
580 ff_rtp_send_hevc(s1, pkt->data, size);
582 case AV_CODEC_ID_VORBIS:
583 case AV_CODEC_ID_THEORA:
584 ff_rtp_send_xiph(s1, pkt->data, size);
586 case AV_CODEC_ID_VP8:
587 ff_rtp_send_vp8(s1, pkt->data, size);
589 case AV_CODEC_ID_ILBC:
590 rtp_send_ilbc(s1, pkt->data, size);
592 case AV_CODEC_ID_MJPEG:
593 ff_rtp_send_jpeg(s1, pkt->data, size);
595 case AV_CODEC_ID_OPUS:
596 if (size > s->max_payload_size) {
597 av_log(s1, AV_LOG_ERROR,
598 "Packet size %d too large for max RTP payload size %d\n",
599 size, s->max_payload_size);
600 return AVERROR(EINVAL);
602 /* Intentional fallthrough */
604 /* better than nothing : send the codec raw data */
605 rtp_send_raw(s1, pkt->data, size);
611 static int rtp_write_trailer(AVFormatContext *s1)
613 RTPMuxContext *s = s1->priv_data;
615 /* If the caller closes and recreates ->pb, this might actually
616 * be NULL here even if it was successfully allocated at the start. */
617 if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
618 rtcp_send_sr(s1, ff_ntp_time(), 1);
624 AVOutputFormat ff_rtp_muxer = {
626 .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
627 .priv_data_size = sizeof(RTPMuxContext),
628 .audio_codec = AV_CODEC_ID_PCM_MULAW,
629 .video_codec = AV_CODEC_ID_MPEG4,
630 .write_header = rtp_write_header,
631 .write_packet = rtp_write_packet,
632 .write_trailer = rtp_write_trailer,
633 .priv_class = &rtp_muxer_class,