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_MPEG1VIDEO:
56 case AV_CODEC_ID_MPEG2VIDEO:
57 case AV_CODEC_ID_MPEG4:
61 case AV_CODEC_ID_PCM_ALAW:
62 case AV_CODEC_ID_PCM_MULAW:
63 case AV_CODEC_ID_PCM_S8:
64 case AV_CODEC_ID_PCM_S16BE:
65 case AV_CODEC_ID_PCM_S16LE:
66 case AV_CODEC_ID_PCM_U16BE:
67 case AV_CODEC_ID_PCM_U16LE:
68 case AV_CODEC_ID_PCM_U8:
69 case AV_CODEC_ID_MPEG2TS:
70 case AV_CODEC_ID_AMR_NB:
71 case AV_CODEC_ID_AMR_WB:
72 case AV_CODEC_ID_VORBIS:
73 case AV_CODEC_ID_THEORA:
75 case AV_CODEC_ID_ADPCM_G722:
76 case AV_CODEC_ID_ADPCM_G726:
77 case AV_CODEC_ID_ILBC:
78 case AV_CODEC_ID_MJPEG:
79 case AV_CODEC_ID_SPEEX:
80 case AV_CODEC_ID_OPUS:
87 static int rtp_write_header(AVFormatContext *s1)
89 RTPMuxContext *s = s1->priv_data;
93 if (s1->nb_streams != 1) {
94 av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
95 return AVERROR(EINVAL);
98 if (!is_supported(st->codec->codec_id)) {
99 av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
104 if (s->payload_type < 0) {
105 /* Re-validate non-dynamic payload types */
106 if (st->id < RTP_PT_PRIVATE)
107 st->id = ff_rtp_get_payload_type(s1, st->codec, -1);
109 s->payload_type = st->id;
111 /* private option takes priority */
112 st->id = s->payload_type;
115 s->base_timestamp = av_get_random_seed();
116 s->timestamp = s->base_timestamp;
117 s->cur_timestamp = 0;
119 s->ssrc = av_get_random_seed();
121 s->first_rtcp_ntp_time = ff_ntp_time();
122 if (s1->start_time_realtime)
123 /* Round the NTP time to whole milliseconds. */
124 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
126 // Pick a random sequence start number, but in the lower end of the
127 // available range, so that any wraparound doesn't happen immediately.
128 // (Immediate wraparound would be an issue for SRTP.)
130 s->seq = av_get_random_seed() & 0x0fff;
132 s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
134 if (s1->packet_size) {
135 if (s1->pb->max_packet_size)
136 s1->packet_size = FFMIN(s1->packet_size,
137 s1->pb->max_packet_size);
139 s1->packet_size = s1->pb->max_packet_size;
140 if (s1->packet_size <= 12) {
141 av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
144 s->buf = av_malloc(s1->packet_size);
145 if (s->buf == NULL) {
146 return AVERROR(ENOMEM);
148 s->max_payload_size = s1->packet_size - 12;
150 s->max_frames_per_packet = 0;
151 if (s1->max_delay > 0) {
152 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
153 int frame_size = av_get_audio_frame_duration(st->codec, 0);
155 frame_size = st->codec->frame_size;
156 if (frame_size == 0) {
157 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
159 s->max_frames_per_packet =
160 av_rescale_q_rnd(s1->max_delay,
162 (AVRational){ frame_size, st->codec->sample_rate },
166 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
167 /* FIXME: We should round down here... */
168 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
172 avpriv_set_pts_info(st, 32, 1, 90000);
173 switch(st->codec->codec_id) {
174 case AV_CODEC_ID_MP2:
175 case AV_CODEC_ID_MP3:
176 s->buf_ptr = s->buf + 4;
178 case AV_CODEC_ID_MPEG1VIDEO:
179 case AV_CODEC_ID_MPEG2VIDEO:
181 case AV_CODEC_ID_MPEG2TS:
182 n = s->max_payload_size / TS_PACKET_SIZE;
185 s->max_payload_size = n * TS_PACKET_SIZE;
188 case AV_CODEC_ID_H264:
189 /* check for H.264 MP4 syntax */
190 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
191 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
194 case AV_CODEC_ID_VORBIS:
195 case AV_CODEC_ID_THEORA:
196 if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
197 s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
198 s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
201 case AV_CODEC_ID_ADPCM_G722:
202 /* Due to a historical error, the clock rate for G722 in RTP is
203 * 8000, even if the sample rate is 16000. See RFC 3551. */
204 avpriv_set_pts_info(st, 32, 1, 8000);
206 case AV_CODEC_ID_OPUS:
207 if (st->codec->channels > 2) {
208 av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
211 /* The opus RTP RFC says that all opus streams should use 48000 Hz
212 * as clock rate, since all opus sample rates can be expressed in
213 * this clock rate, and sample rate changes on the fly are supported. */
214 avpriv_set_pts_info(st, 32, 1, 48000);
216 case AV_CODEC_ID_ILBC:
217 if (st->codec->block_align != 38 && st->codec->block_align != 50) {
218 av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
221 if (!s->max_frames_per_packet)
222 s->max_frames_per_packet = 1;
223 s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
224 s->max_payload_size / st->codec->block_align);
226 case AV_CODEC_ID_AMR_NB:
227 case AV_CODEC_ID_AMR_WB:
228 if (!s->max_frames_per_packet)
229 s->max_frames_per_packet = 12;
230 if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
234 /* max_header_toc_size + the largest AMR payload must fit */
235 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
236 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
239 if (st->codec->channels != 1) {
240 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
243 case AV_CODEC_ID_AAC:
247 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
248 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
258 return AVERROR(EINVAL);
261 /* send an rtcp sender report packet */
262 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
264 RTPMuxContext *s = s1->priv_data;
267 av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
269 s->last_rtcp_ntp_time = ntp_time;
270 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
271 s1->streams[0]->time_base) + s->base_timestamp;
272 avio_w8(s1->pb, (RTP_VERSION << 6));
273 avio_w8(s1->pb, RTCP_SR);
274 avio_wb16(s1->pb, 6); /* length in words - 1 */
275 avio_wb32(s1->pb, s->ssrc);
276 avio_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time));
277 avio_wb32(s1->pb, rtp_ts);
278 avio_wb32(s1->pb, s->packet_count);
279 avio_wb32(s1->pb, s->octet_count);
282 int len = FFMIN(strlen(s->cname), 255);
283 avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
284 avio_w8(s1->pb, RTCP_SDES);
285 avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
287 avio_wb32(s1->pb, s->ssrc);
288 avio_w8(s1->pb, 0x01); /* CNAME */
289 avio_w8(s1->pb, len);
290 avio_write(s1->pb, s->cname, len);
291 avio_w8(s1->pb, 0); /* END */
292 for (len = (7 + len) % 4; len % 4; len++)
299 /* send an rtp packet. sequence number is incremented, but the caller
300 must update the timestamp itself */
301 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
303 RTPMuxContext *s = s1->priv_data;
305 av_dlog(s1, "rtp_send_data size=%d\n", len);
307 /* build the RTP header */
308 avio_w8(s1->pb, (RTP_VERSION << 6));
309 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
310 avio_wb16(s1->pb, s->seq);
311 avio_wb32(s1->pb, s->timestamp);
312 avio_wb32(s1->pb, s->ssrc);
314 avio_write(s1->pb, buf1, len);
317 s->seq = (s->seq + 1) & 0xffff;
318 s->octet_count += len;
322 /* send an integer number of samples and compute time stamp and fill
323 the rtp send buffer before sending. */
324 static int rtp_send_samples(AVFormatContext *s1,
325 const uint8_t *buf1, int size, int sample_size_bits)
327 RTPMuxContext *s = s1->priv_data;
328 int len, max_packet_size, n;
329 /* Calculate the number of bytes to get samples aligned on a byte border */
330 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
332 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
333 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
334 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
335 return AVERROR(EINVAL);
339 len = FFMIN(max_packet_size, size);
342 memcpy(s->buf_ptr, buf1, len);
346 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
347 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
348 n += (s->buf_ptr - s->buf);
353 static void rtp_send_mpegaudio(AVFormatContext *s1,
354 const uint8_t *buf1, int size)
356 RTPMuxContext *s = s1->priv_data;
357 int len, count, max_packet_size;
359 max_packet_size = s->max_payload_size;
361 /* test if we must flush because not enough space */
362 len = (s->buf_ptr - s->buf);
363 if ((len + size) > max_packet_size) {
365 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
366 s->buf_ptr = s->buf + 4;
369 if (s->buf_ptr == s->buf + 4) {
370 s->timestamp = s->cur_timestamp;
374 if (size > max_packet_size) {
375 /* big packet: fragment */
378 len = max_packet_size - 4;
381 /* build fragmented packet */
384 s->buf[2] = count >> 8;
386 memcpy(s->buf + 4, buf1, len);
387 ff_rtp_send_data(s1, s->buf, len + 4, 0);
393 if (s->buf_ptr == s->buf + 4) {
394 /* no fragmentation possible */
400 memcpy(s->buf_ptr, buf1, size);
405 static void rtp_send_raw(AVFormatContext *s1,
406 const uint8_t *buf1, int size)
408 RTPMuxContext *s = s1->priv_data;
409 int len, max_packet_size;
411 max_packet_size = s->max_payload_size;
414 len = max_packet_size;
418 s->timestamp = s->cur_timestamp;
419 ff_rtp_send_data(s1, buf1, len, (len == size));
426 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
427 static void rtp_send_mpegts_raw(AVFormatContext *s1,
428 const uint8_t *buf1, int size)
430 RTPMuxContext *s = s1->priv_data;
433 while (size >= TS_PACKET_SIZE) {
434 len = s->max_payload_size - (s->buf_ptr - s->buf);
437 memcpy(s->buf_ptr, buf1, len);
442 out_len = s->buf_ptr - s->buf;
443 if (out_len >= s->max_payload_size) {
444 ff_rtp_send_data(s1, s->buf, out_len, 0);
450 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
452 RTPMuxContext *s = s1->priv_data;
453 AVStream *st = s1->streams[0];
454 int frame_duration = av_get_audio_frame_duration(st->codec, 0);
455 int frame_size = st->codec->block_align;
456 int frames = size / frame_size;
459 int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
461 if (!s->num_frames) {
463 s->timestamp = s->cur_timestamp;
465 memcpy(s->buf_ptr, buf, n * frame_size);
468 s->buf_ptr += n * frame_size;
469 buf += n * frame_size;
470 s->cur_timestamp += n * frame_duration;
472 if (s->num_frames == s->max_frames_per_packet) {
473 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
480 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
482 RTPMuxContext *s = s1->priv_data;
483 AVStream *st = s1->streams[0];
487 av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
489 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
491 if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
492 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
493 !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
494 rtcp_send_sr(s1, ff_ntp_time());
495 s->last_octet_count = s->octet_count;
498 s->cur_timestamp = s->base_timestamp + pkt->pts;
500 switch(st->codec->codec_id) {
501 case AV_CODEC_ID_PCM_MULAW:
502 case AV_CODEC_ID_PCM_ALAW:
503 case AV_CODEC_ID_PCM_U8:
504 case AV_CODEC_ID_PCM_S8:
505 return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
506 case AV_CODEC_ID_PCM_U16BE:
507 case AV_CODEC_ID_PCM_U16LE:
508 case AV_CODEC_ID_PCM_S16BE:
509 case AV_CODEC_ID_PCM_S16LE:
510 return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
511 case AV_CODEC_ID_ADPCM_G722:
512 /* The actual sample size is half a byte per sample, but since the
513 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
514 * the correct parameter for send_samples_bits is 8 bits per stream
516 return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
517 case AV_CODEC_ID_ADPCM_G726:
518 return rtp_send_samples(s1, pkt->data, size,
519 st->codec->bits_per_coded_sample * st->codec->channels);
520 case AV_CODEC_ID_MP2:
521 case AV_CODEC_ID_MP3:
522 rtp_send_mpegaudio(s1, pkt->data, size);
524 case AV_CODEC_ID_MPEG1VIDEO:
525 case AV_CODEC_ID_MPEG2VIDEO:
526 ff_rtp_send_mpegvideo(s1, pkt->data, size);
528 case AV_CODEC_ID_AAC:
529 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
530 ff_rtp_send_latm(s1, pkt->data, size);
532 ff_rtp_send_aac(s1, pkt->data, size);
534 case AV_CODEC_ID_AMR_NB:
535 case AV_CODEC_ID_AMR_WB:
536 ff_rtp_send_amr(s1, pkt->data, size);
538 case AV_CODEC_ID_MPEG2TS:
539 rtp_send_mpegts_raw(s1, pkt->data, size);
541 case AV_CODEC_ID_H264:
542 ff_rtp_send_h264(s1, pkt->data, size);
544 case AV_CODEC_ID_H263:
545 if (s->flags & FF_RTP_FLAG_RFC2190) {
546 int mb_info_size = 0;
547 const uint8_t *mb_info =
548 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
550 ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
554 case AV_CODEC_ID_H263P:
555 ff_rtp_send_h263(s1, pkt->data, size);
557 case AV_CODEC_ID_VORBIS:
558 case AV_CODEC_ID_THEORA:
559 ff_rtp_send_xiph(s1, pkt->data, size);
561 case AV_CODEC_ID_VP8:
562 ff_rtp_send_vp8(s1, pkt->data, size);
564 case AV_CODEC_ID_ILBC:
565 rtp_send_ilbc(s1, pkt->data, size);
567 case AV_CODEC_ID_MJPEG:
568 ff_rtp_send_jpeg(s1, pkt->data, size);
570 case AV_CODEC_ID_OPUS:
571 if (size > s->max_payload_size) {
572 av_log(s1, AV_LOG_ERROR,
573 "Packet size %d too large for max RTP payload size %d\n",
574 size, s->max_payload_size);
575 return AVERROR(EINVAL);
577 /* Intentional fallthrough */
579 /* better than nothing : send the codec raw data */
580 rtp_send_raw(s1, pkt->data, size);
586 static int rtp_write_trailer(AVFormatContext *s1)
588 RTPMuxContext *s = s1->priv_data;
595 AVOutputFormat ff_rtp_muxer = {
597 .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
598 .priv_data_size = sizeof(RTPMuxContext),
599 .audio_codec = AV_CODEC_ID_PCM_MULAW,
600 .video_codec = AV_CODEC_ID_MPEG4,
601 .write_header = rtp_write_header,
602 .write_packet = rtp_write_packet,
603 .write_trailer = rtp_write_trailer,
604 .priv_class = &rtp_muxer_class,