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, {.dbl = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
36 { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { 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 CodecID id)
55 case CODEC_ID_MPEG1VIDEO:
56 case CODEC_ID_MPEG2VIDEO:
61 case CODEC_ID_PCM_ALAW:
62 case CODEC_ID_PCM_MULAW:
64 case CODEC_ID_PCM_S16BE:
65 case CODEC_ID_PCM_S16LE:
66 case CODEC_ID_PCM_U16BE:
67 case CODEC_ID_PCM_U16LE:
69 case CODEC_ID_MPEG2TS:
75 case CODEC_ID_ADPCM_G722:
76 case CODEC_ID_ADPCM_G726:
84 static int rtp_write_header(AVFormatContext *s1)
86 RTPMuxContext *s = s1->priv_data;
90 if (s1->nb_streams != 1) {
91 av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
92 return AVERROR(EINVAL);
95 if (!is_supported(st->codec->codec_id)) {
96 av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));
101 if (s->payload_type < 0)
102 s->payload_type = ff_rtp_get_payload_type(s1, st->codec);
103 s->base_timestamp = av_get_random_seed();
104 s->timestamp = s->base_timestamp;
105 s->cur_timestamp = 0;
107 s->ssrc = av_get_random_seed();
109 s->first_rtcp_ntp_time = ff_ntp_time();
110 if (s1->start_time_realtime)
111 /* Round the NTP time to whole milliseconds. */
112 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
115 if (s1->packet_size) {
116 if (s1->pb->max_packet_size)
117 s1->packet_size = FFMIN(s1->packet_size,
118 s1->pb->max_packet_size);
120 s1->packet_size = s1->pb->max_packet_size;
121 if (s1->packet_size <= 12) {
122 av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
125 s->buf = av_malloc(s1->packet_size);
126 if (s->buf == NULL) {
127 return AVERROR(ENOMEM);
129 s->max_payload_size = s1->packet_size - 12;
131 s->max_frames_per_packet = 0;
132 if (s1->max_delay > 0) {
133 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
134 int frame_size = av_get_audio_frame_duration(st->codec, 0);
136 frame_size = st->codec->frame_size;
137 if (frame_size == 0) {
138 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
140 s->max_frames_per_packet =
141 av_rescale_q_rnd(s1->max_delay,
143 (AVRational){ frame_size, st->codec->sample_rate },
147 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
148 /* FIXME: We should round down here... */
149 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
153 avpriv_set_pts_info(st, 32, 1, 90000);
154 switch(st->codec->codec_id) {
157 s->buf_ptr = s->buf + 4;
159 case CODEC_ID_MPEG1VIDEO:
160 case CODEC_ID_MPEG2VIDEO:
162 case CODEC_ID_MPEG2TS:
163 n = s->max_payload_size / TS_PACKET_SIZE;
166 s->max_payload_size = n * TS_PACKET_SIZE;
170 /* check for H.264 MP4 syntax */
171 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
172 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
175 case CODEC_ID_VORBIS:
176 case CODEC_ID_THEORA:
177 if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
178 s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
179 s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
183 av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
184 "incompatible with the latest spec drafts.\n");
186 case CODEC_ID_ADPCM_G722:
187 /* Due to a historical error, the clock rate for G722 in RTP is
188 * 8000, even if the sample rate is 16000. See RFC 3551. */
189 avpriv_set_pts_info(st, 32, 1, 8000);
192 if (st->codec->block_align != 38 && st->codec->block_align != 50) {
193 av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
196 if (!s->max_frames_per_packet)
197 s->max_frames_per_packet = 1;
198 s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
199 s->max_payload_size / st->codec->block_align);
201 case CODEC_ID_AMR_NB:
202 case CODEC_ID_AMR_WB:
203 if (!s->max_frames_per_packet)
204 s->max_frames_per_packet = 12;
205 if (st->codec->codec_id == CODEC_ID_AMR_NB)
209 /* max_header_toc_size + the largest AMR payload must fit */
210 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
211 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
214 if (st->codec->channels != 1) {
215 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
222 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
223 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
233 return AVERROR(EINVAL);
236 /* send an rtcp sender report packet */
237 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
239 RTPMuxContext *s = s1->priv_data;
242 av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
244 s->last_rtcp_ntp_time = ntp_time;
245 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
246 s1->streams[0]->time_base) + s->base_timestamp;
247 avio_w8(s1->pb, (RTP_VERSION << 6));
248 avio_w8(s1->pb, RTCP_SR);
249 avio_wb16(s1->pb, 6); /* length in words - 1 */
250 avio_wb32(s1->pb, s->ssrc);
251 avio_wb32(s1->pb, ntp_time / 1000000);
252 avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
253 avio_wb32(s1->pb, rtp_ts);
254 avio_wb32(s1->pb, s->packet_count);
255 avio_wb32(s1->pb, s->octet_count);
259 /* send an rtp packet. sequence number is incremented, but the caller
260 must update the timestamp itself */
261 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
263 RTPMuxContext *s = s1->priv_data;
265 av_dlog(s1, "rtp_send_data size=%d\n", len);
267 /* build the RTP header */
268 avio_w8(s1->pb, (RTP_VERSION << 6));
269 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
270 avio_wb16(s1->pb, s->seq);
271 avio_wb32(s1->pb, s->timestamp);
272 avio_wb32(s1->pb, s->ssrc);
274 avio_write(s1->pb, buf1, len);
278 s->octet_count += len;
282 /* send an integer number of samples and compute time stamp and fill
283 the rtp send buffer before sending. */
284 static void rtp_send_samples(AVFormatContext *s1,
285 const uint8_t *buf1, int size, int sample_size_bits)
287 RTPMuxContext *s = s1->priv_data;
288 int len, max_packet_size, n;
289 /* Calculate the number of bytes to get samples aligned on a byte border */
290 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
292 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
293 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
294 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
299 len = FFMIN(max_packet_size, size);
302 memcpy(s->buf_ptr, buf1, len);
306 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
307 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
308 n += (s->buf_ptr - s->buf);
312 static void rtp_send_mpegaudio(AVFormatContext *s1,
313 const uint8_t *buf1, int size)
315 RTPMuxContext *s = s1->priv_data;
316 int len, count, max_packet_size;
318 max_packet_size = s->max_payload_size;
320 /* test if we must flush because not enough space */
321 len = (s->buf_ptr - s->buf);
322 if ((len + size) > max_packet_size) {
324 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
325 s->buf_ptr = s->buf + 4;
328 if (s->buf_ptr == s->buf + 4) {
329 s->timestamp = s->cur_timestamp;
333 if (size > max_packet_size) {
334 /* big packet: fragment */
337 len = max_packet_size - 4;
340 /* build fragmented packet */
343 s->buf[2] = count >> 8;
345 memcpy(s->buf + 4, buf1, len);
346 ff_rtp_send_data(s1, s->buf, len + 4, 0);
352 if (s->buf_ptr == s->buf + 4) {
353 /* no fragmentation possible */
359 memcpy(s->buf_ptr, buf1, size);
364 static void rtp_send_raw(AVFormatContext *s1,
365 const uint8_t *buf1, int size)
367 RTPMuxContext *s = s1->priv_data;
368 int len, max_packet_size;
370 max_packet_size = s->max_payload_size;
373 len = max_packet_size;
377 s->timestamp = s->cur_timestamp;
378 ff_rtp_send_data(s1, buf1, len, (len == size));
385 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
386 static void rtp_send_mpegts_raw(AVFormatContext *s1,
387 const uint8_t *buf1, int size)
389 RTPMuxContext *s = s1->priv_data;
392 while (size >= TS_PACKET_SIZE) {
393 len = s->max_payload_size - (s->buf_ptr - s->buf);
396 memcpy(s->buf_ptr, buf1, len);
401 out_len = s->buf_ptr - s->buf;
402 if (out_len >= s->max_payload_size) {
403 ff_rtp_send_data(s1, s->buf, out_len, 0);
409 static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
411 RTPMuxContext *s = s1->priv_data;
412 AVStream *st = s1->streams[0];
413 int frame_duration = av_get_audio_frame_duration(st->codec, 0);
414 int frame_size = st->codec->block_align;
415 int frames = size / frame_size;
418 int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
420 if (!s->num_frames) {
422 s->timestamp = s->cur_timestamp;
424 memcpy(s->buf_ptr, buf, n * frame_size);
427 s->buf_ptr += n * frame_size;
428 buf += n * frame_size;
429 s->cur_timestamp += n * frame_duration;
431 if (s->num_frames == s->max_frames_per_packet) {
432 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
439 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
441 RTPMuxContext *s = s1->priv_data;
442 AVStream *st = s1->streams[0];
446 av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
448 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
450 if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
451 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
452 !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
453 rtcp_send_sr(s1, ff_ntp_time());
454 s->last_octet_count = s->octet_count;
457 s->cur_timestamp = s->base_timestamp + pkt->pts;
459 switch(st->codec->codec_id) {
460 case CODEC_ID_PCM_MULAW:
461 case CODEC_ID_PCM_ALAW:
462 case CODEC_ID_PCM_U8:
463 case CODEC_ID_PCM_S8:
464 rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
466 case CODEC_ID_PCM_U16BE:
467 case CODEC_ID_PCM_U16LE:
468 case CODEC_ID_PCM_S16BE:
469 case CODEC_ID_PCM_S16LE:
470 rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
472 case CODEC_ID_ADPCM_G722:
473 /* The actual sample size is half a byte per sample, but since the
474 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
475 * the correct parameter for send_samples_bits is 8 bits per stream
477 rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
479 case CODEC_ID_ADPCM_G726:
480 rtp_send_samples(s1, pkt->data, size,
481 st->codec->bits_per_coded_sample * st->codec->channels);
485 rtp_send_mpegaudio(s1, pkt->data, size);
487 case CODEC_ID_MPEG1VIDEO:
488 case CODEC_ID_MPEG2VIDEO:
489 ff_rtp_send_mpegvideo(s1, pkt->data, size);
492 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
493 ff_rtp_send_latm(s1, pkt->data, size);
495 ff_rtp_send_aac(s1, pkt->data, size);
497 case CODEC_ID_AMR_NB:
498 case CODEC_ID_AMR_WB:
499 ff_rtp_send_amr(s1, pkt->data, size);
501 case CODEC_ID_MPEG2TS:
502 rtp_send_mpegts_raw(s1, pkt->data, size);
505 ff_rtp_send_h264(s1, pkt->data, size);
508 if (s->flags & FF_RTP_FLAG_RFC2190) {
509 int mb_info_size = 0;
510 const uint8_t *mb_info =
511 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
513 ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
518 ff_rtp_send_h263(s1, pkt->data, size);
520 case CODEC_ID_VORBIS:
521 case CODEC_ID_THEORA:
522 ff_rtp_send_xiph(s1, pkt->data, size);
525 ff_rtp_send_vp8(s1, pkt->data, size);
528 rtp_send_ilbc(s1, pkt->data, size);
531 /* better than nothing : send the codec raw data */
532 rtp_send_raw(s1, pkt->data, size);
538 static int rtp_write_trailer(AVFormatContext *s1)
540 RTPMuxContext *s = s1->priv_data;
547 AVOutputFormat ff_rtp_muxer = {
549 .long_name = NULL_IF_CONFIG_SMALL("RTP output format"),
550 .priv_data_size = sizeof(RTPMuxContext),
551 .audio_codec = CODEC_ID_PCM_MULAW,
552 .video_codec = CODEC_ID_MPEG4,
553 .write_header = rtp_write_header,
554 .write_packet = rtp_write_packet,
555 .write_trailer = rtp_write_trailer,
556 .priv_class = &rtp_muxer_class,