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 s->payload_type = ff_rtp_get_payload_type(s1, st->codec);
106 s->base_timestamp = av_get_random_seed();
107 s->timestamp = s->base_timestamp;
108 s->cur_timestamp = 0;
110 s->ssrc = av_get_random_seed();
112 s->first_rtcp_ntp_time = ff_ntp_time();
113 if (s1->start_time_realtime)
114 /* Round the NTP time to whole milliseconds. */
115 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
118 if (s1->packet_size) {
119 if (s1->pb->max_packet_size)
120 s1->packet_size = FFMIN(s1->packet_size,
121 s1->pb->max_packet_size);
123 s1->packet_size = s1->pb->max_packet_size;
124 if (s1->packet_size <= 12) {
125 av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
128 s->buf = av_malloc(s1->packet_size);
129 if (s->buf == NULL) {
130 return AVERROR(ENOMEM);
132 s->max_payload_size = s1->packet_size - 12;
134 s->max_frames_per_packet = 0;
135 if (s1->max_delay > 0) {
136 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
137 int frame_size = av_get_audio_frame_duration(st->codec, 0);
139 frame_size = st->codec->frame_size;
140 if (frame_size == 0) {
141 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
143 s->max_frames_per_packet =
144 av_rescale_q_rnd(s1->max_delay,
146 (AVRational){ frame_size, st->codec->sample_rate },
150 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
151 /* FIXME: We should round down here... */
152 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
156 avpriv_set_pts_info(st, 32, 1, 90000);
157 switch(st->codec->codec_id) {
158 case AV_CODEC_ID_MP2:
159 case AV_CODEC_ID_MP3:
160 s->buf_ptr = s->buf + 4;
162 case AV_CODEC_ID_MPEG1VIDEO:
163 case AV_CODEC_ID_MPEG2VIDEO:
165 case AV_CODEC_ID_MPEG2TS:
166 n = s->max_payload_size / TS_PACKET_SIZE;
169 s->max_payload_size = n * TS_PACKET_SIZE;
172 case AV_CODEC_ID_H264:
173 /* check for H.264 MP4 syntax */
174 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
175 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
178 case AV_CODEC_ID_VORBIS:
179 case AV_CODEC_ID_THEORA:
180 if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
181 s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
182 s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
185 case AV_CODEC_ID_ADPCM_G722:
186 /* Due to a historical error, the clock rate for G722 in RTP is
187 * 8000, even if the sample rate is 16000. See RFC 3551. */
188 avpriv_set_pts_info(st, 32, 1, 8000);
190 case AV_CODEC_ID_OPUS:
191 if (st->codec->channels > 2) {
192 av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
195 /* The opus RTP RFC says that all opus streams should use 48000 Hz
196 * as clock rate, since all opus sample rates can be expressed in
197 * this clock rate, and sample rate changes on the fly are supported. */
198 avpriv_set_pts_info(st, 32, 1, 48000);
200 case AV_CODEC_ID_ILBC:
201 if (st->codec->block_align != 38 && st->codec->block_align != 50) {
202 av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
205 if (!s->max_frames_per_packet)
206 s->max_frames_per_packet = 1;
207 s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
208 s->max_payload_size / st->codec->block_align);
210 case AV_CODEC_ID_AMR_NB:
211 case AV_CODEC_ID_AMR_WB:
212 if (!s->max_frames_per_packet)
213 s->max_frames_per_packet = 12;
214 if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
218 /* max_header_toc_size + the largest AMR payload must fit */
219 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
220 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
223 if (st->codec->channels != 1) {
224 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
227 case AV_CODEC_ID_AAC:
231 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
232 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
242 return AVERROR(EINVAL);
245 /* send an rtcp sender report packet */
246 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
248 RTPMuxContext *s = s1->priv_data;
251 av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
253 s->last_rtcp_ntp_time = ntp_time;
254 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
255 s1->streams[0]->time_base) + s->base_timestamp;
256 avio_w8(s1->pb, (RTP_VERSION << 6));
257 avio_w8(s1->pb, RTCP_SR);
258 avio_wb16(s1->pb, 6); /* length in words - 1 */
259 avio_wb32(s1->pb, s->ssrc);
260 avio_wb32(s1->pb, ntp_time / 1000000);
261 avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
262 avio_wb32(s1->pb, rtp_ts);
263 avio_wb32(s1->pb, s->packet_count);
264 avio_wb32(s1->pb, s->octet_count);
268 /* send an rtp packet. sequence number is incremented, but the caller
269 must update the timestamp itself */
270 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
272 RTPMuxContext *s = s1->priv_data;
274 av_dlog(s1, "rtp_send_data size=%d\n", len);
276 /* build the RTP header */
277 avio_w8(s1->pb, (RTP_VERSION << 6));
278 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
279 avio_wb16(s1->pb, s->seq);
280 avio_wb32(s1->pb, s->timestamp);
281 avio_wb32(s1->pb, s->ssrc);
283 avio_write(s1->pb, buf1, len);
287 s->octet_count += len;
291 /* send an integer number of samples and compute time stamp and fill
292 the rtp send buffer before sending. */
293 static int rtp_send_samples(AVFormatContext *s1,
294 const uint8_t *buf1, int size, int sample_size_bits)
296 RTPMuxContext *s = s1->priv_data;
297 int len, max_packet_size, n;
298 /* Calculate the number of bytes to get samples aligned on a byte border */
299 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
301 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
302 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
303 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
304 return AVERROR(EINVAL);
308 len = FFMIN(max_packet_size, size);
311 memcpy(s->buf_ptr, buf1, len);
315 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
316 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
317 n += (s->buf_ptr - s->buf);
322 static void rtp_send_mpegaudio(AVFormatContext *s1,
323 const uint8_t *buf1, int size)
325 RTPMuxContext *s = s1->priv_data;
326 int len, count, max_packet_size;
328 max_packet_size = s->max_payload_size;
330 /* test if we must flush because not enough space */
331 len = (s->buf_ptr - s->buf);
332 if ((len + size) > max_packet_size) {
334 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
335 s->buf_ptr = s->buf + 4;
338 if (s->buf_ptr == s->buf + 4) {
339 s->timestamp = s->cur_timestamp;
343 if (size > max_packet_size) {
344 /* big packet: fragment */
347 len = max_packet_size - 4;
350 /* build fragmented packet */
353 s->buf[2] = count >> 8;
355 memcpy(s->buf + 4, buf1, len);
356 ff_rtp_send_data(s1, s->buf, len + 4, 0);
362 if (s->buf_ptr == s->buf + 4) {
363 /* no fragmentation possible */
369 memcpy(s->buf_ptr, buf1, size);
374 static void rtp_send_raw(AVFormatContext *s1,
375 const uint8_t *buf1, int size)
377 RTPMuxContext *s = s1->priv_data;
378 int len, max_packet_size;
380 max_packet_size = s->max_payload_size;
383 len = max_packet_size;
387 s->timestamp = s->cur_timestamp;
388 ff_rtp_send_data(s1, buf1, len, (len == size));
395 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
396 static void rtp_send_mpegts_raw(AVFormatContext *s1,
397 const uint8_t *buf1, int size)
399 RTPMuxContext *s = s1->priv_data;
402 while (size >= TS_PACKET_SIZE) {
403 len = s->max_payload_size - (s->buf_ptr - s->buf);
406 memcpy(s->buf_ptr, buf1, len);
411 out_len = s->buf_ptr - s->buf;
412 if (out_len >= s->max_payload_size) {
413 ff_rtp_send_data(s1, s->buf, out_len, 0);
419 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
421 RTPMuxContext *s = s1->priv_data;
422 AVStream *st = s1->streams[0];
423 int frame_duration = av_get_audio_frame_duration(st->codec, 0);
424 int frame_size = st->codec->block_align;
425 int frames = size / frame_size;
428 int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
430 if (!s->num_frames) {
432 s->timestamp = s->cur_timestamp;
434 memcpy(s->buf_ptr, buf, n * frame_size);
437 s->buf_ptr += n * frame_size;
438 buf += n * frame_size;
439 s->cur_timestamp += n * frame_duration;
441 if (s->num_frames == s->max_frames_per_packet) {
442 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
449 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
451 RTPMuxContext *s = s1->priv_data;
452 AVStream *st = s1->streams[0];
456 av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
458 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
460 if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
461 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
462 !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
463 rtcp_send_sr(s1, ff_ntp_time());
464 s->last_octet_count = s->octet_count;
467 s->cur_timestamp = s->base_timestamp + pkt->pts;
469 switch(st->codec->codec_id) {
470 case AV_CODEC_ID_PCM_MULAW:
471 case AV_CODEC_ID_PCM_ALAW:
472 case AV_CODEC_ID_PCM_U8:
473 case AV_CODEC_ID_PCM_S8:
474 return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
475 case AV_CODEC_ID_PCM_U16BE:
476 case AV_CODEC_ID_PCM_U16LE:
477 case AV_CODEC_ID_PCM_S16BE:
478 case AV_CODEC_ID_PCM_S16LE:
479 return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
480 case AV_CODEC_ID_ADPCM_G722:
481 /* The actual sample size is half a byte per sample, but since the
482 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
483 * the correct parameter for send_samples_bits is 8 bits per stream
485 return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
486 case AV_CODEC_ID_ADPCM_G726:
487 return rtp_send_samples(s1, pkt->data, size,
488 st->codec->bits_per_coded_sample * st->codec->channels);
489 case AV_CODEC_ID_MP2:
490 case AV_CODEC_ID_MP3:
491 rtp_send_mpegaudio(s1, pkt->data, size);
493 case AV_CODEC_ID_MPEG1VIDEO:
494 case AV_CODEC_ID_MPEG2VIDEO:
495 ff_rtp_send_mpegvideo(s1, pkt->data, size);
497 case AV_CODEC_ID_AAC:
498 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
499 ff_rtp_send_latm(s1, pkt->data, size);
501 ff_rtp_send_aac(s1, pkt->data, size);
503 case AV_CODEC_ID_AMR_NB:
504 case AV_CODEC_ID_AMR_WB:
505 ff_rtp_send_amr(s1, pkt->data, size);
507 case AV_CODEC_ID_MPEG2TS:
508 rtp_send_mpegts_raw(s1, pkt->data, size);
510 case AV_CODEC_ID_H264:
511 ff_rtp_send_h264(s1, pkt->data, size);
513 case AV_CODEC_ID_H263:
514 if (s->flags & FF_RTP_FLAG_RFC2190) {
515 int mb_info_size = 0;
516 const uint8_t *mb_info =
517 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
519 ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
523 case AV_CODEC_ID_H263P:
524 ff_rtp_send_h263(s1, pkt->data, size);
526 case AV_CODEC_ID_VORBIS:
527 case AV_CODEC_ID_THEORA:
528 ff_rtp_send_xiph(s1, pkt->data, size);
530 case AV_CODEC_ID_VP8:
531 ff_rtp_send_vp8(s1, pkt->data, size);
533 case AV_CODEC_ID_ILBC:
534 rtp_send_ilbc(s1, pkt->data, size);
536 case AV_CODEC_ID_MJPEG:
537 ff_rtp_send_jpeg(s1, pkt->data, size);
539 case AV_CODEC_ID_OPUS:
540 if (size > s->max_payload_size) {
541 av_log(s1, AV_LOG_ERROR,
542 "Packet size %d too large for max RTP payload size %d\n",
543 size, s->max_payload_size);
544 return AVERROR(EINVAL);
546 /* Intentional fallthrough */
548 /* better than nothing : send the codec raw data */
549 rtp_send_raw(s1, pkt->data, size);
555 static int rtp_write_trailer(AVFormatContext *s1)
557 RTPMuxContext *s = s1->priv_data;
564 AVOutputFormat ff_rtp_muxer = {
566 .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
567 .priv_data_size = sizeof(RTPMuxContext),
568 .audio_codec = AV_CODEC_ID_PCM_MULAW,
569 .video_codec = AV_CODEC_ID_MPEG4,
570 .write_header = rtp_write_header,
571 .write_packet = rtp_write_packet,
572 .write_trailer = rtp_write_trailer,
573 .priv_class = &rtp_muxer_class,