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 },
39 static const AVClass rtp_muxer_class = {
40 .class_name = "RTP muxer",
41 .item_name = av_default_item_name,
43 .version = LIBAVUTIL_VERSION_INT,
46 #define RTCP_SR_SIZE 28
48 static int is_supported(enum CodecID id)
54 case CODEC_ID_MPEG1VIDEO:
55 case CODEC_ID_MPEG2VIDEO:
60 case CODEC_ID_PCM_ALAW:
61 case CODEC_ID_PCM_MULAW:
63 case CODEC_ID_PCM_S16BE:
64 case CODEC_ID_PCM_S16LE:
65 case CODEC_ID_PCM_U16BE:
66 case CODEC_ID_PCM_U16LE:
68 case CODEC_ID_MPEG2TS:
74 case CODEC_ID_ADPCM_G722:
75 case CODEC_ID_ADPCM_G726:
82 static int rtp_write_header(AVFormatContext *s1)
84 RTPMuxContext *s = s1->priv_data;
88 if (s1->nb_streams != 1) {
89 av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
90 return AVERROR(EINVAL);
93 if (!is_supported(st->codec->codec_id)) {
94 av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));
99 if (s->payload_type < 0)
100 s->payload_type = ff_rtp_get_payload_type(s1, st->codec);
101 s->base_timestamp = av_get_random_seed();
102 s->timestamp = s->base_timestamp;
103 s->cur_timestamp = 0;
104 s->ssrc = av_get_random_seed();
106 s->first_rtcp_ntp_time = ff_ntp_time();
107 if (s1->start_time_realtime)
108 /* Round the NTP time to whole milliseconds. */
109 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
112 if (s1->packet_size) {
113 if (s1->pb->max_packet_size)
114 s1->packet_size = FFMIN(s1->packet_size,
115 s1->pb->max_packet_size);
117 s1->packet_size = s1->pb->max_packet_size;
118 if (s1->packet_size <= 12) {
119 av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
122 s->buf = av_malloc(s1->packet_size);
123 if (s->buf == NULL) {
124 return AVERROR(ENOMEM);
126 s->max_payload_size = s1->packet_size - 12;
128 s->max_frames_per_packet = 0;
129 if (s1->max_delay > 0) {
130 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
131 int frame_size = av_get_audio_frame_duration(st->codec, 0);
133 frame_size = st->codec->frame_size;
134 if (frame_size == 0) {
135 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
137 s->max_frames_per_packet =
138 av_rescale_q_rnd(s1->max_delay,
140 (AVRational){ frame_size, st->codec->sample_rate },
144 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
145 /* FIXME: We should round down here... */
146 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
150 avpriv_set_pts_info(st, 32, 1, 90000);
151 switch(st->codec->codec_id) {
154 s->buf_ptr = s->buf + 4;
156 case CODEC_ID_MPEG1VIDEO:
157 case CODEC_ID_MPEG2VIDEO:
159 case CODEC_ID_MPEG2TS:
160 n = s->max_payload_size / TS_PACKET_SIZE;
163 s->max_payload_size = n * TS_PACKET_SIZE;
167 /* check for H.264 MP4 syntax */
168 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
169 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
172 case CODEC_ID_VORBIS:
173 case CODEC_ID_THEORA:
174 if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
175 s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
176 s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
180 av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
181 "incompatible with the latest spec drafts.\n");
183 case CODEC_ID_ADPCM_G722:
184 /* Due to a historical error, the clock rate for G722 in RTP is
185 * 8000, even if the sample rate is 16000. See RFC 3551. */
186 avpriv_set_pts_info(st, 32, 1, 8000);
188 case CODEC_ID_AMR_NB:
189 case CODEC_ID_AMR_WB:
190 if (!s->max_frames_per_packet)
191 s->max_frames_per_packet = 12;
192 if (st->codec->codec_id == CODEC_ID_AMR_NB)
196 /* max_header_toc_size + the largest AMR payload must fit */
197 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
198 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
201 if (st->codec->channels != 1) {
202 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
209 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
210 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
219 /* send an rtcp sender report packet */
220 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
222 RTPMuxContext *s = s1->priv_data;
225 av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
227 s->last_rtcp_ntp_time = ntp_time;
228 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
229 s1->streams[0]->time_base) + s->base_timestamp;
230 avio_w8(s1->pb, (RTP_VERSION << 6));
231 avio_w8(s1->pb, RTCP_SR);
232 avio_wb16(s1->pb, 6); /* length in words - 1 */
233 avio_wb32(s1->pb, s->ssrc);
234 avio_wb32(s1->pb, ntp_time / 1000000);
235 avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
236 avio_wb32(s1->pb, rtp_ts);
237 avio_wb32(s1->pb, s->packet_count);
238 avio_wb32(s1->pb, s->octet_count);
242 /* send an rtp packet. sequence number is incremented, but the caller
243 must update the timestamp itself */
244 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
246 RTPMuxContext *s = s1->priv_data;
248 av_dlog(s1, "rtp_send_data size=%d\n", len);
250 /* build the RTP header */
251 avio_w8(s1->pb, (RTP_VERSION << 6));
252 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
253 avio_wb16(s1->pb, s->seq);
254 avio_wb32(s1->pb, s->timestamp);
255 avio_wb32(s1->pb, s->ssrc);
257 avio_write(s1->pb, buf1, len);
261 s->octet_count += len;
265 /* send an integer number of samples and compute time stamp and fill
266 the rtp send buffer before sending. */
267 static void rtp_send_samples(AVFormatContext *s1,
268 const uint8_t *buf1, int size, int sample_size_bits)
270 RTPMuxContext *s = s1->priv_data;
271 int len, max_packet_size, n;
272 /* Calculate the number of bytes to get samples aligned on a byte border */
273 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
275 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
276 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
277 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
282 len = FFMIN(max_packet_size, size);
285 memcpy(s->buf_ptr, buf1, len);
289 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
290 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
291 n += (s->buf_ptr - s->buf);
295 static void rtp_send_mpegaudio(AVFormatContext *s1,
296 const uint8_t *buf1, int size)
298 RTPMuxContext *s = s1->priv_data;
299 int len, count, max_packet_size;
301 max_packet_size = s->max_payload_size;
303 /* test if we must flush because not enough space */
304 len = (s->buf_ptr - s->buf);
305 if ((len + size) > max_packet_size) {
307 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
308 s->buf_ptr = s->buf + 4;
311 if (s->buf_ptr == s->buf + 4) {
312 s->timestamp = s->cur_timestamp;
316 if (size > max_packet_size) {
317 /* big packet: fragment */
320 len = max_packet_size - 4;
323 /* build fragmented packet */
326 s->buf[2] = count >> 8;
328 memcpy(s->buf + 4, buf1, len);
329 ff_rtp_send_data(s1, s->buf, len + 4, 0);
335 if (s->buf_ptr == s->buf + 4) {
336 /* no fragmentation possible */
342 memcpy(s->buf_ptr, buf1, size);
347 static void rtp_send_raw(AVFormatContext *s1,
348 const uint8_t *buf1, int size)
350 RTPMuxContext *s = s1->priv_data;
351 int len, max_packet_size;
353 max_packet_size = s->max_payload_size;
356 len = max_packet_size;
360 s->timestamp = s->cur_timestamp;
361 ff_rtp_send_data(s1, buf1, len, (len == size));
368 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
369 static void rtp_send_mpegts_raw(AVFormatContext *s1,
370 const uint8_t *buf1, int size)
372 RTPMuxContext *s = s1->priv_data;
375 while (size >= TS_PACKET_SIZE) {
376 len = s->max_payload_size - (s->buf_ptr - s->buf);
379 memcpy(s->buf_ptr, buf1, len);
384 out_len = s->buf_ptr - s->buf;
385 if (out_len >= s->max_payload_size) {
386 ff_rtp_send_data(s1, s->buf, out_len, 0);
392 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
394 RTPMuxContext *s = s1->priv_data;
395 AVStream *st = s1->streams[0];
399 av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
401 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
403 if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
404 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
405 !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
406 rtcp_send_sr(s1, ff_ntp_time());
407 s->last_octet_count = s->octet_count;
410 s->cur_timestamp = s->base_timestamp + pkt->pts;
412 switch(st->codec->codec_id) {
413 case CODEC_ID_PCM_MULAW:
414 case CODEC_ID_PCM_ALAW:
415 case CODEC_ID_PCM_U8:
416 case CODEC_ID_PCM_S8:
417 rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
419 case CODEC_ID_PCM_U16BE:
420 case CODEC_ID_PCM_U16LE:
421 case CODEC_ID_PCM_S16BE:
422 case CODEC_ID_PCM_S16LE:
423 rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
425 case CODEC_ID_ADPCM_G722:
426 /* The actual sample size is half a byte per sample, but since the
427 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
428 * the correct parameter for send_samples_bits is 8 bits per stream
430 rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
432 case CODEC_ID_ADPCM_G726:
433 rtp_send_samples(s1, pkt->data, size,
434 st->codec->bits_per_coded_sample * st->codec->channels);
438 rtp_send_mpegaudio(s1, pkt->data, size);
440 case CODEC_ID_MPEG1VIDEO:
441 case CODEC_ID_MPEG2VIDEO:
442 ff_rtp_send_mpegvideo(s1, pkt->data, size);
445 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
446 ff_rtp_send_latm(s1, pkt->data, size);
448 ff_rtp_send_aac(s1, pkt->data, size);
450 case CODEC_ID_AMR_NB:
451 case CODEC_ID_AMR_WB:
452 ff_rtp_send_amr(s1, pkt->data, size);
454 case CODEC_ID_MPEG2TS:
455 rtp_send_mpegts_raw(s1, pkt->data, size);
458 ff_rtp_send_h264(s1, pkt->data, size);
461 if (s->flags & FF_RTP_FLAG_RFC2190) {
462 int mb_info_size = 0;
463 const uint8_t *mb_info =
464 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
466 ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
471 ff_rtp_send_h263(s1, pkt->data, size);
473 case CODEC_ID_VORBIS:
474 case CODEC_ID_THEORA:
475 ff_rtp_send_xiph(s1, pkt->data, size);
478 ff_rtp_send_vp8(s1, pkt->data, size);
481 /* better than nothing : send the codec raw data */
482 rtp_send_raw(s1, pkt->data, size);
488 static int rtp_write_trailer(AVFormatContext *s1)
490 RTPMuxContext *s = s1->priv_data;
497 AVOutputFormat ff_rtp_muxer = {
499 .long_name = NULL_IF_CONFIG_SMALL("RTP output format"),
500 .priv_data_size = sizeof(RTPMuxContext),
501 .audio_codec = CODEC_ID_PCM_MULAW,
502 .video_codec = CODEC_ID_MPEG4,
503 .write_header = rtp_write_header,
504 .write_packet = rtp_write_packet,
505 .write_trailer = rtp_write_trailer,
506 .priv_class = &rtp_muxer_class,