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"
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 { "max_packet_size", "Max packet size", offsetof(RTPMuxContext, max_packet_size), AV_OPT_TYPE_INT, {.dbl = 0 }, 0, 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:
83 static int rtp_write_header(AVFormatContext *s1)
85 RTPMuxContext *s = s1->priv_data;
89 if (s1->nb_streams != 1) {
90 av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
91 return AVERROR(EINVAL);
94 if (!is_supported(st->codec->codec_id)) {
95 av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
100 if (s->payload_type < 0)
101 s->payload_type = ff_rtp_get_payload_type(s1, st->codec);
102 s->base_timestamp = av_get_random_seed();
103 s->timestamp = s->base_timestamp;
104 s->cur_timestamp = 0;
105 s->ssrc = av_get_random_seed();
107 s->first_rtcp_ntp_time = ff_ntp_time();
108 if (s1->start_time_realtime)
109 /* Round the NTP time to whole milliseconds. */
110 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
113 if (s->max_packet_size) {
114 if (s1->pb->max_packet_size)
115 s->max_packet_size = FFMIN(s->max_payload_size,
116 s1->pb->max_packet_size);
118 s->max_packet_size = s1->pb->max_packet_size;
119 if (s->max_packet_size <= 12) {
120 av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s->max_packet_size);
123 s->buf = av_malloc(s->max_packet_size);
124 if (s->buf == NULL) {
125 return AVERROR(ENOMEM);
127 s->max_payload_size = s->max_packet_size - 12;
129 s->max_frames_per_packet = 0;
131 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
132 if (st->codec->frame_size == 0) {
133 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
135 s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * (int64_t)st->codec->frame_size, AV_ROUND_DOWN);
138 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
139 /* FIXME: We should round down here... */
140 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
144 avpriv_set_pts_info(st, 32, 1, 90000);
145 switch(st->codec->codec_id) {
148 s->buf_ptr = s->buf + 4;
150 case CODEC_ID_MPEG1VIDEO:
151 case CODEC_ID_MPEG2VIDEO:
153 case CODEC_ID_MPEG2TS:
154 n = s->max_payload_size / TS_PACKET_SIZE;
157 s->max_payload_size = n * TS_PACKET_SIZE;
161 /* check for H.264 MP4 syntax */
162 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
163 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
166 case CODEC_ID_VORBIS:
167 case CODEC_ID_THEORA:
168 if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
169 s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
170 s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
174 av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
175 "incompatible with the latest spec drafts.\n");
177 case CODEC_ID_ADPCM_G722:
178 /* Due to a historical error, the clock rate for G722 in RTP is
179 * 8000, even if the sample rate is 16000. See RFC 3551. */
180 avpriv_set_pts_info(st, 32, 1, 8000);
182 case CODEC_ID_AMR_NB:
183 case CODEC_ID_AMR_WB:
184 if (!s->max_frames_per_packet)
185 s->max_frames_per_packet = 12;
186 if (st->codec->codec_id == CODEC_ID_AMR_NB)
190 /* max_header_toc_size + the largest AMR payload must fit */
191 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
192 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
195 if (st->codec->channels != 1) {
196 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
203 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
204 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
213 /* send an rtcp sender report packet */
214 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
216 RTPMuxContext *s = s1->priv_data;
219 av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
221 s->last_rtcp_ntp_time = ntp_time;
222 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
223 s1->streams[0]->time_base) + s->base_timestamp;
224 avio_w8(s1->pb, (RTP_VERSION << 6));
225 avio_w8(s1->pb, RTCP_SR);
226 avio_wb16(s1->pb, 6); /* length in words - 1 */
227 avio_wb32(s1->pb, s->ssrc);
228 avio_wb32(s1->pb, ntp_time / 1000000);
229 avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
230 avio_wb32(s1->pb, rtp_ts);
231 avio_wb32(s1->pb, s->packet_count);
232 avio_wb32(s1->pb, s->octet_count);
236 /* send an rtp packet. sequence number is incremented, but the caller
237 must update the timestamp itself */
238 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
240 RTPMuxContext *s = s1->priv_data;
242 av_dlog(s1, "rtp_send_data size=%d\n", len);
244 /* build the RTP header */
245 avio_w8(s1->pb, (RTP_VERSION << 6));
246 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
247 avio_wb16(s1->pb, s->seq);
248 avio_wb32(s1->pb, s->timestamp);
249 avio_wb32(s1->pb, s->ssrc);
251 avio_write(s1->pb, buf1, len);
255 s->octet_count += len;
259 /* send an integer number of samples and compute time stamp and fill
260 the rtp send buffer before sending. */
261 static void rtp_send_samples(AVFormatContext *s1,
262 const uint8_t *buf1, int size, int sample_size_bits)
264 RTPMuxContext *s = s1->priv_data;
265 int len, max_packet_size, n;
266 /* Calculate the number of bytes to get samples aligned on a byte border */
267 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
269 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
270 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
271 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
276 len = FFMIN(max_packet_size, size);
279 memcpy(s->buf_ptr, buf1, len);
283 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
284 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
285 n += (s->buf_ptr - s->buf);
289 static void rtp_send_mpegaudio(AVFormatContext *s1,
290 const uint8_t *buf1, int size)
292 RTPMuxContext *s = s1->priv_data;
293 int len, count, max_packet_size;
295 max_packet_size = s->max_payload_size;
297 /* test if we must flush because not enough space */
298 len = (s->buf_ptr - s->buf);
299 if ((len + size) > max_packet_size) {
301 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
302 s->buf_ptr = s->buf + 4;
305 if (s->buf_ptr == s->buf + 4) {
306 s->timestamp = s->cur_timestamp;
310 if (size > max_packet_size) {
311 /* big packet: fragment */
314 len = max_packet_size - 4;
317 /* build fragmented packet */
320 s->buf[2] = count >> 8;
322 memcpy(s->buf + 4, buf1, len);
323 ff_rtp_send_data(s1, s->buf, len + 4, 0);
329 if (s->buf_ptr == s->buf + 4) {
330 /* no fragmentation possible */
336 memcpy(s->buf_ptr, buf1, size);
341 static void rtp_send_raw(AVFormatContext *s1,
342 const uint8_t *buf1, int size)
344 RTPMuxContext *s = s1->priv_data;
345 int len, max_packet_size;
347 max_packet_size = s->max_payload_size;
350 len = max_packet_size;
354 s->timestamp = s->cur_timestamp;
355 ff_rtp_send_data(s1, buf1, len, (len == size));
362 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
363 static void rtp_send_mpegts_raw(AVFormatContext *s1,
364 const uint8_t *buf1, int size)
366 RTPMuxContext *s = s1->priv_data;
369 while (size >= TS_PACKET_SIZE) {
370 len = s->max_payload_size - (s->buf_ptr - s->buf);
373 memcpy(s->buf_ptr, buf1, len);
378 out_len = s->buf_ptr - s->buf;
379 if (out_len >= s->max_payload_size) {
380 ff_rtp_send_data(s1, s->buf, out_len, 0);
386 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
388 RTPMuxContext *s = s1->priv_data;
389 AVStream *st = s1->streams[0];
393 av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
395 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
397 if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
398 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
399 !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
400 rtcp_send_sr(s1, ff_ntp_time());
401 s->last_octet_count = s->octet_count;
404 s->cur_timestamp = s->base_timestamp + pkt->pts;
406 switch(st->codec->codec_id) {
407 case CODEC_ID_PCM_MULAW:
408 case CODEC_ID_PCM_ALAW:
409 case CODEC_ID_PCM_U8:
410 case CODEC_ID_PCM_S8:
411 rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
413 case CODEC_ID_PCM_U16BE:
414 case CODEC_ID_PCM_U16LE:
415 case CODEC_ID_PCM_S16BE:
416 case CODEC_ID_PCM_S16LE:
417 rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
419 case CODEC_ID_ADPCM_G722:
420 /* The actual sample size is half a byte per sample, but since the
421 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
422 * the correct parameter for send_samples_bits is 8 bits per stream
424 rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
426 case CODEC_ID_ADPCM_G726:
427 rtp_send_samples(s1, pkt->data, size,
428 st->codec->bits_per_coded_sample * st->codec->channels);
432 rtp_send_mpegaudio(s1, pkt->data, size);
434 case CODEC_ID_MPEG1VIDEO:
435 case CODEC_ID_MPEG2VIDEO:
436 ff_rtp_send_mpegvideo(s1, pkt->data, size);
439 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
440 ff_rtp_send_latm(s1, pkt->data, size);
442 ff_rtp_send_aac(s1, pkt->data, size);
444 case CODEC_ID_AMR_NB:
445 case CODEC_ID_AMR_WB:
446 ff_rtp_send_amr(s1, pkt->data, size);
448 case CODEC_ID_MPEG2TS:
449 rtp_send_mpegts_raw(s1, pkt->data, size);
452 ff_rtp_send_h264(s1, pkt->data, size);
455 if (s->flags & FF_RTP_FLAG_RFC2190) {
456 ff_rtp_send_h263_rfc2190(s1, pkt->data, size);
461 ff_rtp_send_h263(s1, pkt->data, size);
463 case CODEC_ID_VORBIS:
464 case CODEC_ID_THEORA:
465 ff_rtp_send_xiph(s1, pkt->data, size);
468 ff_rtp_send_vp8(s1, pkt->data, size);
471 /* better than nothing : send the codec raw data */
472 rtp_send_raw(s1, pkt->data, size);
478 static int rtp_write_trailer(AVFormatContext *s1)
480 RTPMuxContext *s = s1->priv_data;
487 AVOutputFormat ff_rtp_muxer = {
489 .long_name = NULL_IF_CONFIG_SMALL("RTP output format"),
490 .priv_data_size = sizeof(RTPMuxContext),
491 .audio_codec = CODEC_ID_PCM_MULAW,
492 .video_codec = CODEC_ID_MPEG4,
493 .write_header = rtp_write_header,
494 .write_packet = rtp_write_packet,
495 .write_trailer = rtp_write_trailer,
496 .priv_class = &rtp_muxer_class,