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 },
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;
85 int max_packet_size, n;
88 if (s1->nb_streams != 1)
91 if (!is_supported(st->codec->codec_id)) {
92 av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
97 if (s->payload_type < 0)
98 s->payload_type = ff_rtp_get_payload_type(s1, st->codec);
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) {
112 av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", max_packet_size);
115 s->buf = av_malloc(max_packet_size);
116 if (s->buf == NULL) {
117 return AVERROR(ENOMEM);
119 s->max_payload_size = max_packet_size - 12;
121 s->max_frames_per_packet = 0;
123 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
124 if (st->codec->frame_size == 0) {
125 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
127 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);
130 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
131 /* FIXME: We should round down here... */
132 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
136 avpriv_set_pts_info(st, 32, 1, 90000);
137 switch(st->codec->codec_id) {
140 s->buf_ptr = s->buf + 4;
142 case CODEC_ID_MPEG1VIDEO:
143 case CODEC_ID_MPEG2VIDEO:
145 case CODEC_ID_MPEG2TS:
146 n = s->max_payload_size / TS_PACKET_SIZE;
149 s->max_payload_size = n * TS_PACKET_SIZE;
153 /* check for H.264 MP4 syntax */
154 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
155 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
158 case CODEC_ID_VORBIS:
159 case CODEC_ID_THEORA:
160 if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
161 s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
162 s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
166 av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
167 "incompatible with the latest spec drafts.\n");
169 case CODEC_ID_ADPCM_G722:
170 /* Due to a historical error, the clock rate for G722 in RTP is
171 * 8000, even if the sample rate is 16000. See RFC 3551. */
172 avpriv_set_pts_info(st, 32, 1, 8000);
174 case CODEC_ID_AMR_NB:
175 case CODEC_ID_AMR_WB:
176 if (!s->max_frames_per_packet)
177 s->max_frames_per_packet = 12;
178 if (st->codec->codec_id == CODEC_ID_AMR_NB)
182 /* max_header_toc_size + the largest AMR payload must fit */
183 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
184 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
187 if (st->codec->channels != 1) {
188 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
195 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
196 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
205 /* send an rtcp sender report packet */
206 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
208 RTPMuxContext *s = s1->priv_data;
211 av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
213 s->last_rtcp_ntp_time = ntp_time;
214 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
215 s1->streams[0]->time_base) + s->base_timestamp;
216 avio_w8(s1->pb, (RTP_VERSION << 6));
217 avio_w8(s1->pb, RTCP_SR);
218 avio_wb16(s1->pb, 6); /* length in words - 1 */
219 avio_wb32(s1->pb, s->ssrc);
220 avio_wb32(s1->pb, ntp_time / 1000000);
221 avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
222 avio_wb32(s1->pb, rtp_ts);
223 avio_wb32(s1->pb, s->packet_count);
224 avio_wb32(s1->pb, s->octet_count);
228 /* send an rtp packet. sequence number is incremented, but the caller
229 must update the timestamp itself */
230 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
232 RTPMuxContext *s = s1->priv_data;
234 av_dlog(s1, "rtp_send_data size=%d\n", len);
236 /* build the RTP header */
237 avio_w8(s1->pb, (RTP_VERSION << 6));
238 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
239 avio_wb16(s1->pb, s->seq);
240 avio_wb32(s1->pb, s->timestamp);
241 avio_wb32(s1->pb, s->ssrc);
243 avio_write(s1->pb, buf1, len);
247 s->octet_count += len;
251 /* send an integer number of samples and compute time stamp and fill
252 the rtp send buffer before sending. */
253 static void rtp_send_samples(AVFormatContext *s1,
254 const uint8_t *buf1, int size, int sample_size_bits)
256 RTPMuxContext *s = s1->priv_data;
257 int len, max_packet_size, n;
258 /* Calculate the number of bytes to get samples aligned on a byte border */
259 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
261 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
262 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
263 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
268 len = FFMIN(max_packet_size, size);
271 memcpy(s->buf_ptr, buf1, len);
275 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
276 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
277 n += (s->buf_ptr - s->buf);
281 static void rtp_send_mpegaudio(AVFormatContext *s1,
282 const uint8_t *buf1, int size)
284 RTPMuxContext *s = s1->priv_data;
285 int len, count, max_packet_size;
287 max_packet_size = s->max_payload_size;
289 /* test if we must flush because not enough space */
290 len = (s->buf_ptr - s->buf);
291 if ((len + size) > max_packet_size) {
293 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
294 s->buf_ptr = s->buf + 4;
297 if (s->buf_ptr == s->buf + 4) {
298 s->timestamp = s->cur_timestamp;
302 if (size > max_packet_size) {
303 /* big packet: fragment */
306 len = max_packet_size - 4;
309 /* build fragmented packet */
312 s->buf[2] = count >> 8;
314 memcpy(s->buf + 4, buf1, len);
315 ff_rtp_send_data(s1, s->buf, len + 4, 0);
321 if (s->buf_ptr == s->buf + 4) {
322 /* no fragmentation possible */
328 memcpy(s->buf_ptr, buf1, size);
333 static void rtp_send_raw(AVFormatContext *s1,
334 const uint8_t *buf1, int size)
336 RTPMuxContext *s = s1->priv_data;
337 int len, max_packet_size;
339 max_packet_size = s->max_payload_size;
342 len = max_packet_size;
346 s->timestamp = s->cur_timestamp;
347 ff_rtp_send_data(s1, buf1, len, (len == size));
354 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
355 static void rtp_send_mpegts_raw(AVFormatContext *s1,
356 const uint8_t *buf1, int size)
358 RTPMuxContext *s = s1->priv_data;
361 while (size >= TS_PACKET_SIZE) {
362 len = s->max_payload_size - (s->buf_ptr - s->buf);
365 memcpy(s->buf_ptr, buf1, len);
370 out_len = s->buf_ptr - s->buf;
371 if (out_len >= s->max_payload_size) {
372 ff_rtp_send_data(s1, s->buf, out_len, 0);
378 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
380 RTPMuxContext *s = s1->priv_data;
381 AVStream *st = s1->streams[0];
385 av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
387 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
389 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
390 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
391 rtcp_send_sr(s1, ff_ntp_time());
392 s->last_octet_count = s->octet_count;
395 s->cur_timestamp = s->base_timestamp + pkt->pts;
397 switch(st->codec->codec_id) {
398 case CODEC_ID_PCM_MULAW:
399 case CODEC_ID_PCM_ALAW:
400 case CODEC_ID_PCM_U8:
401 case CODEC_ID_PCM_S8:
402 rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
404 case CODEC_ID_PCM_U16BE:
405 case CODEC_ID_PCM_U16LE:
406 case CODEC_ID_PCM_S16BE:
407 case CODEC_ID_PCM_S16LE:
408 rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
410 case CODEC_ID_ADPCM_G722:
411 /* The actual sample size is half a byte per sample, but since the
412 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
413 * the correct parameter for send_samples_bits is 8 bits per stream
415 rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
417 case CODEC_ID_ADPCM_G726:
418 rtp_send_samples(s1, pkt->data, size,
419 st->codec->bits_per_coded_sample * st->codec->channels);
423 rtp_send_mpegaudio(s1, pkt->data, size);
425 case CODEC_ID_MPEG1VIDEO:
426 case CODEC_ID_MPEG2VIDEO:
427 ff_rtp_send_mpegvideo(s1, pkt->data, size);
430 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
431 ff_rtp_send_latm(s1, pkt->data, size);
433 ff_rtp_send_aac(s1, pkt->data, size);
435 case CODEC_ID_AMR_NB:
436 case CODEC_ID_AMR_WB:
437 ff_rtp_send_amr(s1, pkt->data, size);
439 case CODEC_ID_MPEG2TS:
440 rtp_send_mpegts_raw(s1, pkt->data, size);
443 ff_rtp_send_h264(s1, pkt->data, size);
447 ff_rtp_send_h263(s1, pkt->data, size);
449 case CODEC_ID_VORBIS:
450 case CODEC_ID_THEORA:
451 ff_rtp_send_xiph(s1, pkt->data, size);
454 ff_rtp_send_vp8(s1, pkt->data, size);
457 /* better than nothing : send the codec raw data */
458 rtp_send_raw(s1, pkt->data, size);
464 static int rtp_write_trailer(AVFormatContext *s1)
466 RTPMuxContext *s = s1->priv_data;
473 AVOutputFormat ff_rtp_muxer = {
475 .long_name = NULL_IF_CONFIG_SMALL("RTP output format"),
476 .priv_data_size = sizeof(RTPMuxContext),
477 .audio_codec = CODEC_ID_PCM_MULAW,
478 .video_codec = CODEC_ID_MPEG4,
479 .write_header = rtp_write_header,
480 .write_packet = rtp_write_packet,
481 .write_trailer = rtp_write_trailer,
482 .priv_class = &rtp_muxer_class,