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),
38 static const AVClass rtp_muxer_class = {
39 .class_name = "RTP muxer",
40 .item_name = av_default_item_name,
42 .version = LIBAVUTIL_VERSION_INT,
45 #define RTCP_SR_SIZE 28
47 static int is_supported(enum CodecID id)
53 case CODEC_ID_MPEG1VIDEO:
54 case CODEC_ID_MPEG2VIDEO:
59 case CODEC_ID_PCM_ALAW:
60 case CODEC_ID_PCM_MULAW:
62 case CODEC_ID_PCM_S16BE:
63 case CODEC_ID_PCM_S16LE:
64 case CODEC_ID_PCM_U16BE:
65 case CODEC_ID_PCM_U16LE:
67 case CODEC_ID_MPEG2TS:
73 case CODEC_ID_ADPCM_G722:
80 static int rtp_write_header(AVFormatContext *s1)
82 RTPMuxContext *s = s1->priv_data;
83 int max_packet_size, n;
86 if (s1->nb_streams != 1)
89 if (!is_supported(st->codec->codec_id)) {
90 av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));
95 s->payload_type = ff_rtp_get_payload_type(st->codec);
96 if (s->payload_type < 0)
97 s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == AVMEDIA_TYPE_AUDIO);
99 s->base_timestamp = av_get_random_seed();
100 s->timestamp = s->base_timestamp;
101 s->cur_timestamp = 0;
102 s->ssrc = av_get_random_seed();
104 s->first_rtcp_ntp_time = ff_ntp_time();
105 if (s1->start_time_realtime)
106 /* Round the NTP time to whole milliseconds. */
107 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
110 max_packet_size = s1->pb->max_packet_size;
111 if (max_packet_size <= 12)
113 s->buf = av_malloc(max_packet_size);
114 if (s->buf == NULL) {
115 return AVERROR(ENOMEM);
117 s->max_payload_size = max_packet_size - 12;
119 s->max_frames_per_packet = 0;
121 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
122 if (st->codec->frame_size == 0) {
123 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
125 s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN);
128 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
129 /* FIXME: We should round down here... */
130 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
134 av_set_pts_info(st, 32, 1, 90000);
135 switch(st->codec->codec_id) {
138 s->buf_ptr = s->buf + 4;
140 case CODEC_ID_MPEG1VIDEO:
141 case CODEC_ID_MPEG2VIDEO:
143 case CODEC_ID_MPEG2TS:
144 n = s->max_payload_size / TS_PACKET_SIZE;
147 s->max_payload_size = n * TS_PACKET_SIZE;
151 /* check for H.264 MP4 syntax */
152 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
153 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
156 case CODEC_ID_VORBIS:
157 case CODEC_ID_THEORA:
158 if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
159 s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
160 s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
164 av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
165 "incompatible with the latest spec drafts.\n");
167 case CODEC_ID_ADPCM_G722:
168 /* Due to a historical error, the clock rate for G722 in RTP is
169 * 8000, even if the sample rate is 16000. See RFC 3551. */
170 av_set_pts_info(st, 32, 1, 8000);
172 case CODEC_ID_AMR_NB:
173 case CODEC_ID_AMR_WB:
174 if (!s->max_frames_per_packet)
175 s->max_frames_per_packet = 12;
176 if (st->codec->codec_id == CODEC_ID_AMR_NB)
180 /* max_header_toc_size + the largest AMR payload must fit */
181 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
182 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
185 if (st->codec->channels != 1) {
186 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
193 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
194 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
203 /* send an rtcp sender report packet */
204 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
206 RTPMuxContext *s = s1->priv_data;
209 av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
211 s->last_rtcp_ntp_time = ntp_time;
212 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
213 s1->streams[0]->time_base) + s->base_timestamp;
214 avio_w8(s1->pb, (RTP_VERSION << 6));
215 avio_w8(s1->pb, RTCP_SR);
216 avio_wb16(s1->pb, 6); /* length in words - 1 */
217 avio_wb32(s1->pb, s->ssrc);
218 avio_wb32(s1->pb, ntp_time / 1000000);
219 avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
220 avio_wb32(s1->pb, rtp_ts);
221 avio_wb32(s1->pb, s->packet_count);
222 avio_wb32(s1->pb, s->octet_count);
226 /* send an rtp packet. sequence number is incremented, but the caller
227 must update the timestamp itself */
228 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
230 RTPMuxContext *s = s1->priv_data;
232 av_dlog(s1, "rtp_send_data size=%d\n", len);
234 /* build the RTP header */
235 avio_w8(s1->pb, (RTP_VERSION << 6));
236 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
237 avio_wb16(s1->pb, s->seq);
238 avio_wb32(s1->pb, s->timestamp);
239 avio_wb32(s1->pb, s->ssrc);
241 avio_write(s1->pb, buf1, len);
245 s->octet_count += len;
249 /* send an integer number of samples and compute time stamp and fill
250 the rtp send buffer before sending. */
251 static void rtp_send_samples(AVFormatContext *s1,
252 const uint8_t *buf1, int size, int sample_size)
254 RTPMuxContext *s = s1->priv_data;
255 int len, max_packet_size, n;
257 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
258 /* not needed, but who nows */
259 if ((size % sample_size) != 0)
264 len = FFMIN(max_packet_size, size);
267 memcpy(s->buf_ptr, buf1, len);
271 s->timestamp = s->cur_timestamp + n / sample_size;
272 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
273 n += (s->buf_ptr - s->buf);
277 static void rtp_send_mpegaudio(AVFormatContext *s1,
278 const uint8_t *buf1, int size)
280 RTPMuxContext *s = s1->priv_data;
281 int len, count, max_packet_size;
283 max_packet_size = s->max_payload_size;
285 /* test if we must flush because not enough space */
286 len = (s->buf_ptr - s->buf);
287 if ((len + size) > max_packet_size) {
289 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
290 s->buf_ptr = s->buf + 4;
293 if (s->buf_ptr == s->buf + 4) {
294 s->timestamp = s->cur_timestamp;
298 if (size > max_packet_size) {
299 /* big packet: fragment */
302 len = max_packet_size - 4;
305 /* build fragmented packet */
308 s->buf[2] = count >> 8;
310 memcpy(s->buf + 4, buf1, len);
311 ff_rtp_send_data(s1, s->buf, len + 4, 0);
317 if (s->buf_ptr == s->buf + 4) {
318 /* no fragmentation possible */
324 memcpy(s->buf_ptr, buf1, size);
329 static void rtp_send_raw(AVFormatContext *s1,
330 const uint8_t *buf1, int size)
332 RTPMuxContext *s = s1->priv_data;
333 int len, max_packet_size;
335 max_packet_size = s->max_payload_size;
338 len = max_packet_size;
342 s->timestamp = s->cur_timestamp;
343 ff_rtp_send_data(s1, buf1, len, (len == size));
350 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
351 static void rtp_send_mpegts_raw(AVFormatContext *s1,
352 const uint8_t *buf1, int size)
354 RTPMuxContext *s = s1->priv_data;
357 while (size >= TS_PACKET_SIZE) {
358 len = s->max_payload_size - (s->buf_ptr - s->buf);
361 memcpy(s->buf_ptr, buf1, len);
366 out_len = s->buf_ptr - s->buf;
367 if (out_len >= s->max_payload_size) {
368 ff_rtp_send_data(s1, s->buf, out_len, 0);
374 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
376 RTPMuxContext *s = s1->priv_data;
377 AVStream *st = s1->streams[0];
381 av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
383 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
385 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
386 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
387 rtcp_send_sr(s1, ff_ntp_time());
388 s->last_octet_count = s->octet_count;
391 s->cur_timestamp = s->base_timestamp + pkt->pts;
393 switch(st->codec->codec_id) {
394 case CODEC_ID_PCM_MULAW:
395 case CODEC_ID_PCM_ALAW:
396 case CODEC_ID_PCM_U8:
397 case CODEC_ID_PCM_S8:
398 rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
400 case CODEC_ID_PCM_U16BE:
401 case CODEC_ID_PCM_U16LE:
402 case CODEC_ID_PCM_S16BE:
403 case CODEC_ID_PCM_S16LE:
404 rtp_send_samples(s1, pkt->data, size, 2 * st->codec->channels);
406 case CODEC_ID_ADPCM_G722:
407 /* The actual sample size is half a byte per sample, but since the
408 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
409 * the correct parameter for send_samples is 1 byte per stream clock. */
410 rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
414 rtp_send_mpegaudio(s1, pkt->data, size);
416 case CODEC_ID_MPEG1VIDEO:
417 case CODEC_ID_MPEG2VIDEO:
418 ff_rtp_send_mpegvideo(s1, pkt->data, size);
421 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
422 ff_rtp_send_latm(s1, pkt->data, size);
424 ff_rtp_send_aac(s1, pkt->data, size);
426 case CODEC_ID_AMR_NB:
427 case CODEC_ID_AMR_WB:
428 ff_rtp_send_amr(s1, pkt->data, size);
430 case CODEC_ID_MPEG2TS:
431 rtp_send_mpegts_raw(s1, pkt->data, size);
434 ff_rtp_send_h264(s1, pkt->data, size);
438 ff_rtp_send_h263(s1, pkt->data, size);
440 case CODEC_ID_VORBIS:
441 case CODEC_ID_THEORA:
442 ff_rtp_send_xiph(s1, pkt->data, size);
445 ff_rtp_send_vp8(s1, pkt->data, size);
448 /* better than nothing : send the codec raw data */
449 rtp_send_raw(s1, pkt->data, size);
455 static int rtp_write_trailer(AVFormatContext *s1)
457 RTPMuxContext *s = s1->priv_data;
464 AVOutputFormat ff_rtp_muxer = {
466 .long_name = NULL_IF_CONFIG_SMALL("RTP output format"),
467 .priv_data_size = sizeof(RTPMuxContext),
468 .audio_codec = CODEC_ID_PCM_MULAW,
469 .video_codec = CODEC_ID_NONE,
470 .write_header = rtp_write_header,
471 .write_packet = rtp_write_packet,
472 .write_trailer = rtp_write_trailer,
473 .priv_class = &rtp_muxer_class,