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"
33 static const AVOption options[] = {
34 FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
35 { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
36 { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
37 { "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM },
41 static const AVClass rtp_muxer_class = {
42 .class_name = "RTP muxer",
43 .item_name = av_default_item_name,
45 .version = LIBAVUTIL_VERSION_INT,
48 #define RTCP_SR_SIZE 28
50 static int is_supported(enum AVCodecID id)
53 case AV_CODEC_ID_H263:
54 case AV_CODEC_ID_H263P:
55 case AV_CODEC_ID_H264:
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 %s\n", avcodec_get_name(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 +
128 if (s1->packet_size) {
129 if (s1->pb->max_packet_size)
130 s1->packet_size = FFMIN(s1->packet_size,
131 s1->pb->max_packet_size);
133 s1->packet_size = s1->pb->max_packet_size;
134 if (s1->packet_size <= 12) {
135 av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
138 s->buf = av_malloc(s1->packet_size);
139 if (s->buf == NULL) {
140 return AVERROR(ENOMEM);
142 s->max_payload_size = s1->packet_size - 12;
144 s->max_frames_per_packet = 0;
145 if (s1->max_delay > 0) {
146 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
147 int frame_size = av_get_audio_frame_duration(st->codec, 0);
149 frame_size = st->codec->frame_size;
150 if (frame_size == 0) {
151 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
153 s->max_frames_per_packet =
154 av_rescale_q_rnd(s1->max_delay,
156 (AVRational){ frame_size, st->codec->sample_rate },
160 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
161 /* FIXME: We should round down here... */
162 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
166 avpriv_set_pts_info(st, 32, 1, 90000);
167 switch(st->codec->codec_id) {
168 case AV_CODEC_ID_MP2:
169 case AV_CODEC_ID_MP3:
170 s->buf_ptr = s->buf + 4;
172 case AV_CODEC_ID_MPEG1VIDEO:
173 case AV_CODEC_ID_MPEG2VIDEO:
175 case AV_CODEC_ID_MPEG2TS:
176 n = s->max_payload_size / TS_PACKET_SIZE;
179 s->max_payload_size = n * TS_PACKET_SIZE;
182 case AV_CODEC_ID_H264:
183 /* check for H.264 MP4 syntax */
184 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
185 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
188 case AV_CODEC_ID_VORBIS:
189 case AV_CODEC_ID_THEORA:
190 if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
191 s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
192 s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
195 case AV_CODEC_ID_ADPCM_G722:
196 /* Due to a historical error, the clock rate for G722 in RTP is
197 * 8000, even if the sample rate is 16000. See RFC 3551. */
198 avpriv_set_pts_info(st, 32, 1, 8000);
200 case AV_CODEC_ID_OPUS:
201 if (st->codec->channels > 2) {
202 av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
205 /* The opus RTP RFC says that all opus streams should use 48000 Hz
206 * as clock rate, since all opus sample rates can be expressed in
207 * this clock rate, and sample rate changes on the fly are supported. */
208 avpriv_set_pts_info(st, 32, 1, 48000);
210 case AV_CODEC_ID_ILBC:
211 if (st->codec->block_align != 38 && st->codec->block_align != 50) {
212 av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
215 if (!s->max_frames_per_packet)
216 s->max_frames_per_packet = 1;
217 s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
218 s->max_payload_size / st->codec->block_align);
220 case AV_CODEC_ID_AMR_NB:
221 case AV_CODEC_ID_AMR_WB:
222 if (!s->max_frames_per_packet)
223 s->max_frames_per_packet = 12;
224 if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
228 /* max_header_toc_size + the largest AMR payload must fit */
229 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
230 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
233 if (st->codec->channels != 1) {
234 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
237 case AV_CODEC_ID_AAC:
241 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
242 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
252 return AVERROR(EINVAL);
255 /* send an rtcp sender report packet */
256 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
258 RTPMuxContext *s = s1->priv_data;
261 av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
263 s->last_rtcp_ntp_time = ntp_time;
264 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
265 s1->streams[0]->time_base) + s->base_timestamp;
266 avio_w8(s1->pb, (RTP_VERSION << 6));
267 avio_w8(s1->pb, RTCP_SR);
268 avio_wb16(s1->pb, 6); /* length in words - 1 */
269 avio_wb32(s1->pb, s->ssrc);
270 avio_wb32(s1->pb, ntp_time / 1000000);
271 avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
272 avio_wb32(s1->pb, rtp_ts);
273 avio_wb32(s1->pb, s->packet_count);
274 avio_wb32(s1->pb, s->octet_count);
277 int len = FFMIN(strlen(s->cname), 255);
278 avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
279 avio_w8(s1->pb, RTCP_SDES);
280 avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
282 avio_wb32(s1->pb, s->ssrc);
283 avio_w8(s1->pb, 0x01); /* CNAME */
284 avio_w8(s1->pb, len);
285 avio_write(s1->pb, s->cname, len);
286 avio_w8(s1->pb, 0); /* END */
287 for (len = (7 + len) % 4; len % 4; len++)
294 /* send an rtp packet. sequence number is incremented, but the caller
295 must update the timestamp itself */
296 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
298 RTPMuxContext *s = s1->priv_data;
300 av_dlog(s1, "rtp_send_data size=%d\n", len);
302 /* build the RTP header */
303 avio_w8(s1->pb, (RTP_VERSION << 6));
304 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
305 avio_wb16(s1->pb, s->seq);
306 avio_wb32(s1->pb, s->timestamp);
307 avio_wb32(s1->pb, s->ssrc);
309 avio_write(s1->pb, buf1, len);
313 s->octet_count += len;
317 /* send an integer number of samples and compute time stamp and fill
318 the rtp send buffer before sending. */
319 static int rtp_send_samples(AVFormatContext *s1,
320 const uint8_t *buf1, int size, int sample_size_bits)
322 RTPMuxContext *s = s1->priv_data;
323 int len, max_packet_size, n;
324 /* Calculate the number of bytes to get samples aligned on a byte border */
325 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
327 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
328 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
329 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
330 return AVERROR(EINVAL);
334 len = FFMIN(max_packet_size, size);
337 memcpy(s->buf_ptr, buf1, len);
341 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
342 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
343 n += (s->buf_ptr - s->buf);
348 static void rtp_send_mpegaudio(AVFormatContext *s1,
349 const uint8_t *buf1, int size)
351 RTPMuxContext *s = s1->priv_data;
352 int len, count, max_packet_size;
354 max_packet_size = s->max_payload_size;
356 /* test if we must flush because not enough space */
357 len = (s->buf_ptr - s->buf);
358 if ((len + size) > max_packet_size) {
360 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
361 s->buf_ptr = s->buf + 4;
364 if (s->buf_ptr == s->buf + 4) {
365 s->timestamp = s->cur_timestamp;
369 if (size > max_packet_size) {
370 /* big packet: fragment */
373 len = max_packet_size - 4;
376 /* build fragmented packet */
379 s->buf[2] = count >> 8;
381 memcpy(s->buf + 4, buf1, len);
382 ff_rtp_send_data(s1, s->buf, len + 4, 0);
388 if (s->buf_ptr == s->buf + 4) {
389 /* no fragmentation possible */
395 memcpy(s->buf_ptr, buf1, size);
400 static void rtp_send_raw(AVFormatContext *s1,
401 const uint8_t *buf1, int size)
403 RTPMuxContext *s = s1->priv_data;
404 int len, max_packet_size;
406 max_packet_size = s->max_payload_size;
409 len = max_packet_size;
413 s->timestamp = s->cur_timestamp;
414 ff_rtp_send_data(s1, buf1, len, (len == size));
421 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
422 static void rtp_send_mpegts_raw(AVFormatContext *s1,
423 const uint8_t *buf1, int size)
425 RTPMuxContext *s = s1->priv_data;
428 while (size >= TS_PACKET_SIZE) {
429 len = s->max_payload_size - (s->buf_ptr - s->buf);
432 memcpy(s->buf_ptr, buf1, len);
437 out_len = s->buf_ptr - s->buf;
438 if (out_len >= s->max_payload_size) {
439 ff_rtp_send_data(s1, s->buf, out_len, 0);
445 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
447 RTPMuxContext *s = s1->priv_data;
448 AVStream *st = s1->streams[0];
449 int frame_duration = av_get_audio_frame_duration(st->codec, 0);
450 int frame_size = st->codec->block_align;
451 int frames = size / frame_size;
454 int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
456 if (!s->num_frames) {
458 s->timestamp = s->cur_timestamp;
460 memcpy(s->buf_ptr, buf, n * frame_size);
463 s->buf_ptr += n * frame_size;
464 buf += n * frame_size;
465 s->cur_timestamp += n * frame_duration;
467 if (s->num_frames == s->max_frames_per_packet) {
468 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
475 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
477 RTPMuxContext *s = s1->priv_data;
478 AVStream *st = s1->streams[0];
482 av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
484 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
486 if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
487 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
488 !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
489 rtcp_send_sr(s1, ff_ntp_time());
490 s->last_octet_count = s->octet_count;
493 s->cur_timestamp = s->base_timestamp + pkt->pts;
495 switch(st->codec->codec_id) {
496 case AV_CODEC_ID_PCM_MULAW:
497 case AV_CODEC_ID_PCM_ALAW:
498 case AV_CODEC_ID_PCM_U8:
499 case AV_CODEC_ID_PCM_S8:
500 return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
501 case AV_CODEC_ID_PCM_U16BE:
502 case AV_CODEC_ID_PCM_U16LE:
503 case AV_CODEC_ID_PCM_S16BE:
504 case AV_CODEC_ID_PCM_S16LE:
505 return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
506 case AV_CODEC_ID_ADPCM_G722:
507 /* The actual sample size is half a byte per sample, but since the
508 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
509 * the correct parameter for send_samples_bits is 8 bits per stream
511 return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
512 case AV_CODEC_ID_ADPCM_G726:
513 return rtp_send_samples(s1, pkt->data, size,
514 st->codec->bits_per_coded_sample * st->codec->channels);
515 case AV_CODEC_ID_MP2:
516 case AV_CODEC_ID_MP3:
517 rtp_send_mpegaudio(s1, pkt->data, size);
519 case AV_CODEC_ID_MPEG1VIDEO:
520 case AV_CODEC_ID_MPEG2VIDEO:
521 ff_rtp_send_mpegvideo(s1, pkt->data, size);
523 case AV_CODEC_ID_AAC:
524 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
525 ff_rtp_send_latm(s1, pkt->data, size);
527 ff_rtp_send_aac(s1, pkt->data, size);
529 case AV_CODEC_ID_AMR_NB:
530 case AV_CODEC_ID_AMR_WB:
531 ff_rtp_send_amr(s1, pkt->data, size);
533 case AV_CODEC_ID_MPEG2TS:
534 rtp_send_mpegts_raw(s1, pkt->data, size);
536 case AV_CODEC_ID_H264:
537 ff_rtp_send_h264(s1, pkt->data, size);
539 case AV_CODEC_ID_H263:
540 if (s->flags & FF_RTP_FLAG_RFC2190) {
541 int mb_info_size = 0;
542 const uint8_t *mb_info =
543 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
545 ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
549 case AV_CODEC_ID_H263P:
550 ff_rtp_send_h263(s1, pkt->data, size);
552 case AV_CODEC_ID_VORBIS:
553 case AV_CODEC_ID_THEORA:
554 ff_rtp_send_xiph(s1, pkt->data, size);
556 case AV_CODEC_ID_VP8:
557 ff_rtp_send_vp8(s1, pkt->data, size);
559 case AV_CODEC_ID_ILBC:
560 rtp_send_ilbc(s1, pkt->data, size);
562 case AV_CODEC_ID_MJPEG:
563 ff_rtp_send_jpeg(s1, pkt->data, size);
565 case AV_CODEC_ID_OPUS:
566 if (size > s->max_payload_size) {
567 av_log(s1, AV_LOG_ERROR,
568 "Packet size %d too large for max RTP payload size %d\n",
569 size, s->max_payload_size);
570 return AVERROR(EINVAL);
572 /* Intentional fallthrough */
574 /* better than nothing : send the codec raw data */
575 rtp_send_raw(s1, pkt->data, size);
581 static int rtp_write_trailer(AVFormatContext *s1)
583 RTPMuxContext *s = s1->priv_data;
590 AVOutputFormat ff_rtp_muxer = {
592 .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
593 .priv_data_size = sizeof(RTPMuxContext),
594 .audio_codec = AV_CODEC_ID_PCM_MULAW,
595 .video_codec = AV_CODEC_ID_MPEG4,
596 .write_header = rtp_write_header,
597 .write_packet = rtp_write_packet,
598 .write_trailer = rtp_write_trailer,
599 .priv_class = &rtp_muxer_class,