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:
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 %s\n", avcodec_get_name(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;
106 s->ssrc = av_get_random_seed();
108 s->first_rtcp_ntp_time = ff_ntp_time();
109 if (s1->start_time_realtime)
110 /* Round the NTP time to whole milliseconds. */
111 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
114 if (s1->packet_size) {
115 if (s1->pb->max_packet_size)
116 s1->packet_size = FFMIN(s1->packet_size,
117 s1->pb->max_packet_size);
119 s1->packet_size = s1->pb->max_packet_size;
120 if (s1->packet_size <= 12) {
121 av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
124 s->buf = av_malloc(s1->packet_size);
125 if (s->buf == NULL) {
126 return AVERROR(ENOMEM);
128 s->max_payload_size = s1->packet_size - 12;
130 s->max_frames_per_packet = 0;
131 if (s1->max_delay > 0) {
132 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
133 int frame_size = av_get_audio_frame_duration(st->codec, 0);
135 frame_size = st->codec->frame_size;
136 if (frame_size == 0) {
137 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
139 s->max_frames_per_packet =
140 av_rescale_q_rnd(s1->max_delay,
142 (AVRational){ frame_size, st->codec->sample_rate },
146 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
147 /* FIXME: We should round down here... */
148 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
152 avpriv_set_pts_info(st, 32, 1, 90000);
153 switch(st->codec->codec_id) {
156 s->buf_ptr = s->buf + 4;
158 case CODEC_ID_MPEG1VIDEO:
159 case CODEC_ID_MPEG2VIDEO:
161 case CODEC_ID_MPEG2TS:
162 n = s->max_payload_size / TS_PACKET_SIZE;
165 s->max_payload_size = n * TS_PACKET_SIZE;
169 /* check for H.264 MP4 syntax */
170 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
171 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
174 case CODEC_ID_VORBIS:
175 case CODEC_ID_THEORA:
176 if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
177 s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
178 s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
182 av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
183 "incompatible with the latest spec drafts.\n");
185 case CODEC_ID_ADPCM_G722:
186 /* Due to a historical error, the clock rate for G722 in RTP is
187 * 8000, even if the sample rate is 16000. See RFC 3551. */
188 avpriv_set_pts_info(st, 32, 1, 8000);
190 case CODEC_ID_AMR_NB:
191 case CODEC_ID_AMR_WB:
192 if (!s->max_frames_per_packet)
193 s->max_frames_per_packet = 12;
194 if (st->codec->codec_id == CODEC_ID_AMR_NB)
198 /* max_header_toc_size + the largest AMR payload must fit */
199 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
200 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
203 if (st->codec->channels != 1) {
204 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
211 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
212 avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
221 /* send an rtcp sender report packet */
222 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
224 RTPMuxContext *s = s1->priv_data;
227 av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
229 s->last_rtcp_ntp_time = ntp_time;
230 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
231 s1->streams[0]->time_base) + s->base_timestamp;
232 avio_w8(s1->pb, (RTP_VERSION << 6));
233 avio_w8(s1->pb, RTCP_SR);
234 avio_wb16(s1->pb, 6); /* length in words - 1 */
235 avio_wb32(s1->pb, s->ssrc);
236 avio_wb32(s1->pb, ntp_time / 1000000);
237 avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
238 avio_wb32(s1->pb, rtp_ts);
239 avio_wb32(s1->pb, s->packet_count);
240 avio_wb32(s1->pb, s->octet_count);
244 /* send an rtp packet. sequence number is incremented, but the caller
245 must update the timestamp itself */
246 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
248 RTPMuxContext *s = s1->priv_data;
250 av_dlog(s1, "rtp_send_data size=%d\n", len);
252 /* build the RTP header */
253 avio_w8(s1->pb, (RTP_VERSION << 6));
254 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
255 avio_wb16(s1->pb, s->seq);
256 avio_wb32(s1->pb, s->timestamp);
257 avio_wb32(s1->pb, s->ssrc);
259 avio_write(s1->pb, buf1, len);
263 s->octet_count += len;
267 /* send an integer number of samples and compute time stamp and fill
268 the rtp send buffer before sending. */
269 static void rtp_send_samples(AVFormatContext *s1,
270 const uint8_t *buf1, int size, int sample_size_bits)
272 RTPMuxContext *s = s1->priv_data;
273 int len, max_packet_size, n;
274 /* Calculate the number of bytes to get samples aligned on a byte border */
275 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
277 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
278 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
279 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
284 len = FFMIN(max_packet_size, size);
287 memcpy(s->buf_ptr, buf1, len);
291 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
292 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
293 n += (s->buf_ptr - s->buf);
297 static void rtp_send_mpegaudio(AVFormatContext *s1,
298 const uint8_t *buf1, int size)
300 RTPMuxContext *s = s1->priv_data;
301 int len, count, max_packet_size;
303 max_packet_size = s->max_payload_size;
305 /* test if we must flush because not enough space */
306 len = (s->buf_ptr - s->buf);
307 if ((len + size) > max_packet_size) {
309 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
310 s->buf_ptr = s->buf + 4;
313 if (s->buf_ptr == s->buf + 4) {
314 s->timestamp = s->cur_timestamp;
318 if (size > max_packet_size) {
319 /* big packet: fragment */
322 len = max_packet_size - 4;
325 /* build fragmented packet */
328 s->buf[2] = count >> 8;
330 memcpy(s->buf + 4, buf1, len);
331 ff_rtp_send_data(s1, s->buf, len + 4, 0);
337 if (s->buf_ptr == s->buf + 4) {
338 /* no fragmentation possible */
344 memcpy(s->buf_ptr, buf1, size);
349 static void rtp_send_raw(AVFormatContext *s1,
350 const uint8_t *buf1, int size)
352 RTPMuxContext *s = s1->priv_data;
353 int len, max_packet_size;
355 max_packet_size = s->max_payload_size;
358 len = max_packet_size;
362 s->timestamp = s->cur_timestamp;
363 ff_rtp_send_data(s1, buf1, len, (len == size));
370 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
371 static void rtp_send_mpegts_raw(AVFormatContext *s1,
372 const uint8_t *buf1, int size)
374 RTPMuxContext *s = s1->priv_data;
377 while (size >= TS_PACKET_SIZE) {
378 len = s->max_payload_size - (s->buf_ptr - s->buf);
381 memcpy(s->buf_ptr, buf1, len);
386 out_len = s->buf_ptr - s->buf;
387 if (out_len >= s->max_payload_size) {
388 ff_rtp_send_data(s1, s->buf, out_len, 0);
394 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
396 RTPMuxContext *s = s1->priv_data;
397 AVStream *st = s1->streams[0];
401 av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
403 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
405 if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
406 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
407 !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
408 rtcp_send_sr(s1, ff_ntp_time());
409 s->last_octet_count = s->octet_count;
412 s->cur_timestamp = s->base_timestamp + pkt->pts;
414 switch(st->codec->codec_id) {
415 case CODEC_ID_PCM_MULAW:
416 case CODEC_ID_PCM_ALAW:
417 case CODEC_ID_PCM_U8:
418 case CODEC_ID_PCM_S8:
419 rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
421 case CODEC_ID_PCM_U16BE:
422 case CODEC_ID_PCM_U16LE:
423 case CODEC_ID_PCM_S16BE:
424 case CODEC_ID_PCM_S16LE:
425 rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
427 case CODEC_ID_ADPCM_G722:
428 /* The actual sample size is half a byte per sample, but since the
429 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
430 * the correct parameter for send_samples_bits is 8 bits per stream
432 rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
434 case CODEC_ID_ADPCM_G726:
435 rtp_send_samples(s1, pkt->data, size,
436 st->codec->bits_per_coded_sample * st->codec->channels);
440 rtp_send_mpegaudio(s1, pkt->data, size);
442 case CODEC_ID_MPEG1VIDEO:
443 case CODEC_ID_MPEG2VIDEO:
444 ff_rtp_send_mpegvideo(s1, pkt->data, size);
447 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
448 ff_rtp_send_latm(s1, pkt->data, size);
450 ff_rtp_send_aac(s1, pkt->data, size);
452 case CODEC_ID_AMR_NB:
453 case CODEC_ID_AMR_WB:
454 ff_rtp_send_amr(s1, pkt->data, size);
456 case CODEC_ID_MPEG2TS:
457 rtp_send_mpegts_raw(s1, pkt->data, size);
460 ff_rtp_send_h264(s1, pkt->data, size);
463 if (s->flags & FF_RTP_FLAG_RFC2190) {
464 int mb_info_size = 0;
465 const uint8_t *mb_info =
466 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
468 ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
473 ff_rtp_send_h263(s1, pkt->data, size);
475 case CODEC_ID_VORBIS:
476 case CODEC_ID_THEORA:
477 ff_rtp_send_xiph(s1, pkt->data, size);
480 ff_rtp_send_vp8(s1, pkt->data, size);
483 /* better than nothing : send the codec raw data */
484 rtp_send_raw(s1, pkt->data, size);
490 static int rtp_write_trailer(AVFormatContext *s1)
492 RTPMuxContext *s = s1->priv_data;
499 AVOutputFormat ff_rtp_muxer = {
501 .long_name = NULL_IF_CONFIG_SMALL("RTP output format"),
502 .priv_data_size = sizeof(RTPMuxContext),
503 .audio_codec = CODEC_ID_PCM_MULAW,
504 .video_codec = CODEC_ID_MPEG4,
505 .write_header = rtp_write_header,
506 .write_packet = rtp_write_packet,
507 .write_trailer = rtp_write_trailer,
508 .priv_class = &rtp_muxer_class,