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 },
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 %s\n", avcodec_get_name(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 +
127 if (s1->packet_size) {
128 if (s1->pb->max_packet_size)
129 s1->packet_size = FFMIN(s1->packet_size,
130 s1->pb->max_packet_size);
132 s1->packet_size = s1->pb->max_packet_size;
133 if (s1->packet_size <= 12) {
134 av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
137 s->buf = av_malloc(s1->packet_size);
138 if (s->buf == NULL) {
139 return AVERROR(ENOMEM);
141 s->max_payload_size = s1->packet_size - 12;
143 s->max_frames_per_packet = 0;
144 if (s1->max_delay > 0) {
145 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
146 int frame_size = av_get_audio_frame_duration(st->codec, 0);
148 frame_size = st->codec->frame_size;
149 if (frame_size == 0) {
150 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
152 s->max_frames_per_packet =
153 av_rescale_q_rnd(s1->max_delay,
155 (AVRational){ frame_size, st->codec->sample_rate },
159 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
160 /* FIXME: We should round down here... */
161 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
165 avpriv_set_pts_info(st, 32, 1, 90000);
166 switch(st->codec->codec_id) {
167 case AV_CODEC_ID_MP2:
168 case AV_CODEC_ID_MP3:
169 s->buf_ptr = s->buf + 4;
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;
181 case AV_CODEC_ID_H264:
182 /* check for H.264 MP4 syntax */
183 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
184 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
187 case AV_CODEC_ID_VORBIS:
188 case AV_CODEC_ID_THEORA:
189 if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
190 s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
191 s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
194 case AV_CODEC_ID_ADPCM_G722:
195 /* Due to a historical error, the clock rate for G722 in RTP is
196 * 8000, even if the sample rate is 16000. See RFC 3551. */
197 avpriv_set_pts_info(st, 32, 1, 8000);
199 case AV_CODEC_ID_OPUS:
200 if (st->codec->channels > 2) {
201 av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
204 /* The opus RTP RFC says that all opus streams should use 48000 Hz
205 * as clock rate, since all opus sample rates can be expressed in
206 * this clock rate, and sample rate changes on the fly are supported. */
207 avpriv_set_pts_info(st, 32, 1, 48000);
209 case AV_CODEC_ID_ILBC:
210 if (st->codec->block_align != 38 && st->codec->block_align != 50) {
211 av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
214 if (!s->max_frames_per_packet)
215 s->max_frames_per_packet = 1;
216 s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
217 s->max_payload_size / st->codec->block_align);
219 case AV_CODEC_ID_AMR_NB:
220 case AV_CODEC_ID_AMR_WB:
221 if (!s->max_frames_per_packet)
222 s->max_frames_per_packet = 12;
223 if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
227 /* max_header_toc_size + the largest AMR payload must fit */
228 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
229 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
232 if (st->codec->channels != 1) {
233 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
236 case AV_CODEC_ID_AAC:
240 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
241 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
251 return AVERROR(EINVAL);
254 /* send an rtcp sender report packet */
255 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
257 RTPMuxContext *s = s1->priv_data;
260 av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
262 s->last_rtcp_ntp_time = ntp_time;
263 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
264 s1->streams[0]->time_base) + s->base_timestamp;
265 avio_w8(s1->pb, (RTP_VERSION << 6));
266 avio_w8(s1->pb, RTCP_SR);
267 avio_wb16(s1->pb, 6); /* length in words - 1 */
268 avio_wb32(s1->pb, s->ssrc);
269 avio_wb32(s1->pb, ntp_time / 1000000);
270 avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
271 avio_wb32(s1->pb, rtp_ts);
272 avio_wb32(s1->pb, s->packet_count);
273 avio_wb32(s1->pb, s->octet_count);
277 /* send an rtp packet. sequence number is incremented, but the caller
278 must update the timestamp itself */
279 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
281 RTPMuxContext *s = s1->priv_data;
283 av_dlog(s1, "rtp_send_data size=%d\n", len);
285 /* build the RTP header */
286 avio_w8(s1->pb, (RTP_VERSION << 6));
287 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
288 avio_wb16(s1->pb, s->seq);
289 avio_wb32(s1->pb, s->timestamp);
290 avio_wb32(s1->pb, s->ssrc);
292 avio_write(s1->pb, buf1, len);
296 s->octet_count += len;
300 /* send an integer number of samples and compute time stamp and fill
301 the rtp send buffer before sending. */
302 static int rtp_send_samples(AVFormatContext *s1,
303 const uint8_t *buf1, int size, int sample_size_bits)
305 RTPMuxContext *s = s1->priv_data;
306 int len, max_packet_size, n;
307 /* Calculate the number of bytes to get samples aligned on a byte border */
308 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
310 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
311 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
312 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
313 return AVERROR(EINVAL);
317 len = FFMIN(max_packet_size, size);
320 memcpy(s->buf_ptr, buf1, len);
324 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
325 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
326 n += (s->buf_ptr - s->buf);
331 static void rtp_send_mpegaudio(AVFormatContext *s1,
332 const uint8_t *buf1, int size)
334 RTPMuxContext *s = s1->priv_data;
335 int len, count, max_packet_size;
337 max_packet_size = s->max_payload_size;
339 /* test if we must flush because not enough space */
340 len = (s->buf_ptr - s->buf);
341 if ((len + size) > max_packet_size) {
343 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
344 s->buf_ptr = s->buf + 4;
347 if (s->buf_ptr == s->buf + 4) {
348 s->timestamp = s->cur_timestamp;
352 if (size > max_packet_size) {
353 /* big packet: fragment */
356 len = max_packet_size - 4;
359 /* build fragmented packet */
362 s->buf[2] = count >> 8;
364 memcpy(s->buf + 4, buf1, len);
365 ff_rtp_send_data(s1, s->buf, len + 4, 0);
371 if (s->buf_ptr == s->buf + 4) {
372 /* no fragmentation possible */
378 memcpy(s->buf_ptr, buf1, size);
383 static void rtp_send_raw(AVFormatContext *s1,
384 const uint8_t *buf1, int size)
386 RTPMuxContext *s = s1->priv_data;
387 int len, max_packet_size;
389 max_packet_size = s->max_payload_size;
392 len = max_packet_size;
396 s->timestamp = s->cur_timestamp;
397 ff_rtp_send_data(s1, buf1, len, (len == size));
404 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
405 static void rtp_send_mpegts_raw(AVFormatContext *s1,
406 const uint8_t *buf1, int size)
408 RTPMuxContext *s = s1->priv_data;
411 while (size >= TS_PACKET_SIZE) {
412 len = s->max_payload_size - (s->buf_ptr - s->buf);
415 memcpy(s->buf_ptr, buf1, len);
420 out_len = s->buf_ptr - s->buf;
421 if (out_len >= s->max_payload_size) {
422 ff_rtp_send_data(s1, s->buf, out_len, 0);
428 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
430 RTPMuxContext *s = s1->priv_data;
431 AVStream *st = s1->streams[0];
432 int frame_duration = av_get_audio_frame_duration(st->codec, 0);
433 int frame_size = st->codec->block_align;
434 int frames = size / frame_size;
437 int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
439 if (!s->num_frames) {
441 s->timestamp = s->cur_timestamp;
443 memcpy(s->buf_ptr, buf, n * frame_size);
446 s->buf_ptr += n * frame_size;
447 buf += n * frame_size;
448 s->cur_timestamp += n * frame_duration;
450 if (s->num_frames == s->max_frames_per_packet) {
451 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
458 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
460 RTPMuxContext *s = s1->priv_data;
461 AVStream *st = s1->streams[0];
465 av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
467 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
469 if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
470 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
471 !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
472 rtcp_send_sr(s1, ff_ntp_time());
473 s->last_octet_count = s->octet_count;
476 s->cur_timestamp = s->base_timestamp + pkt->pts;
478 switch(st->codec->codec_id) {
479 case AV_CODEC_ID_PCM_MULAW:
480 case AV_CODEC_ID_PCM_ALAW:
481 case AV_CODEC_ID_PCM_U8:
482 case AV_CODEC_ID_PCM_S8:
483 return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
484 case AV_CODEC_ID_PCM_U16BE:
485 case AV_CODEC_ID_PCM_U16LE:
486 case AV_CODEC_ID_PCM_S16BE:
487 case AV_CODEC_ID_PCM_S16LE:
488 return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
489 case AV_CODEC_ID_ADPCM_G722:
490 /* The actual sample size is half a byte per sample, but since the
491 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
492 * the correct parameter for send_samples_bits is 8 bits per stream
494 return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
495 case AV_CODEC_ID_ADPCM_G726:
496 return rtp_send_samples(s1, pkt->data, size,
497 st->codec->bits_per_coded_sample * st->codec->channels);
498 case AV_CODEC_ID_MP2:
499 case AV_CODEC_ID_MP3:
500 rtp_send_mpegaudio(s1, pkt->data, size);
502 case AV_CODEC_ID_MPEG1VIDEO:
503 case AV_CODEC_ID_MPEG2VIDEO:
504 ff_rtp_send_mpegvideo(s1, pkt->data, size);
506 case AV_CODEC_ID_AAC:
507 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
508 ff_rtp_send_latm(s1, pkt->data, size);
510 ff_rtp_send_aac(s1, pkt->data, size);
512 case AV_CODEC_ID_AMR_NB:
513 case AV_CODEC_ID_AMR_WB:
514 ff_rtp_send_amr(s1, pkt->data, size);
516 case AV_CODEC_ID_MPEG2TS:
517 rtp_send_mpegts_raw(s1, pkt->data, size);
519 case AV_CODEC_ID_H264:
520 ff_rtp_send_h264(s1, pkt->data, size);
522 case AV_CODEC_ID_H263:
523 if (s->flags & FF_RTP_FLAG_RFC2190) {
524 int mb_info_size = 0;
525 const uint8_t *mb_info =
526 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
528 ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
532 case AV_CODEC_ID_H263P:
533 ff_rtp_send_h263(s1, pkt->data, size);
535 case AV_CODEC_ID_VORBIS:
536 case AV_CODEC_ID_THEORA:
537 ff_rtp_send_xiph(s1, pkt->data, size);
539 case AV_CODEC_ID_VP8:
540 ff_rtp_send_vp8(s1, pkt->data, size);
542 case AV_CODEC_ID_ILBC:
543 rtp_send_ilbc(s1, pkt->data, size);
545 case AV_CODEC_ID_MJPEG:
546 ff_rtp_send_jpeg(s1, pkt->data, size);
548 case AV_CODEC_ID_OPUS:
549 if (size > s->max_payload_size) {
550 av_log(s1, AV_LOG_ERROR,
551 "Packet size %d too large for max RTP payload size %d\n",
552 size, s->max_payload_size);
553 return AVERROR(EINVAL);
555 /* Intentional fallthrough */
557 /* better than nothing : send the codec raw data */
558 rtp_send_raw(s1, pkt->data, size);
564 static int rtp_write_trailer(AVFormatContext *s1)
566 RTPMuxContext *s = s1->priv_data;
573 AVOutputFormat ff_rtp_muxer = {
575 .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
576 .priv_data_size = sizeof(RTPMuxContext),
577 .audio_codec = AV_CODEC_ID_PCM_MULAW,
578 .video_codec = AV_CODEC_ID_MPEG4,
579 .write_header = rtp_write_header,
580 .write_packet = rtp_write_packet,
581 .write_trailer = rtp_write_trailer,
582 .priv_class = &rtp_muxer_class,