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:
81 static int rtp_write_header(AVFormatContext *s1)
83 RTPMuxContext *s = s1->priv_data;
84 int max_packet_size, n;
87 if (s1->nb_streams != 1)
90 if (!is_supported(st->codec->codec_id)) {
91 av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));
96 if (s->payload_type < 0)
97 s->payload_type = ff_rtp_get_payload_type(s1, st->codec);
98 s->base_timestamp = av_get_random_seed();
99 s->timestamp = s->base_timestamp;
100 s->cur_timestamp = 0;
101 s->ssrc = av_get_random_seed();
103 s->first_rtcp_ntp_time = ff_ntp_time();
104 if (s1->start_time_realtime)
105 /* Round the NTP time to whole milliseconds. */
106 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
109 max_packet_size = s1->pb->max_packet_size;
110 if (max_packet_size <= 12)
112 s->buf = av_malloc(max_packet_size);
113 if (s->buf == NULL) {
114 return AVERROR(ENOMEM);
116 s->max_payload_size = max_packet_size - 12;
118 s->max_frames_per_packet = 0;
120 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
121 if (st->codec->frame_size == 0) {
122 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
124 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);
127 if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
128 /* FIXME: We should round down here... */
129 s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
133 av_set_pts_info(st, 32, 1, 90000);
134 switch(st->codec->codec_id) {
137 s->buf_ptr = s->buf + 4;
139 case CODEC_ID_MPEG1VIDEO:
140 case CODEC_ID_MPEG2VIDEO:
142 case CODEC_ID_MPEG2TS:
143 n = s->max_payload_size / TS_PACKET_SIZE;
146 s->max_payload_size = n * TS_PACKET_SIZE;
150 /* check for H.264 MP4 syntax */
151 if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
152 s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
155 case CODEC_ID_VORBIS:
156 case CODEC_ID_THEORA:
157 if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
158 s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
159 s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
163 av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
164 "incompatible with the latest spec drafts.\n");
166 case CODEC_ID_ADPCM_G722:
167 /* Due to a historical error, the clock rate for G722 in RTP is
168 * 8000, even if the sample rate is 16000. See RFC 3551. */
169 av_set_pts_info(st, 32, 1, 8000);
171 case CODEC_ID_AMR_NB:
172 case CODEC_ID_AMR_WB:
173 if (!s->max_frames_per_packet)
174 s->max_frames_per_packet = 12;
175 if (st->codec->codec_id == CODEC_ID_AMR_NB)
179 /* max_header_toc_size + the largest AMR payload must fit */
180 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
181 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
184 if (st->codec->channels != 1) {
185 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
192 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
193 av_set_pts_info(st, 32, 1, st->codec->sample_rate);
202 /* send an rtcp sender report packet */
203 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
205 RTPMuxContext *s = s1->priv_data;
208 av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
210 s->last_rtcp_ntp_time = ntp_time;
211 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
212 s1->streams[0]->time_base) + s->base_timestamp;
213 avio_w8(s1->pb, (RTP_VERSION << 6));
214 avio_w8(s1->pb, RTCP_SR);
215 avio_wb16(s1->pb, 6); /* length in words - 1 */
216 avio_wb32(s1->pb, s->ssrc);
217 avio_wb32(s1->pb, ntp_time / 1000000);
218 avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
219 avio_wb32(s1->pb, rtp_ts);
220 avio_wb32(s1->pb, s->packet_count);
221 avio_wb32(s1->pb, s->octet_count);
225 /* send an rtp packet. sequence number is incremented, but the caller
226 must update the timestamp itself */
227 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
229 RTPMuxContext *s = s1->priv_data;
231 av_dlog(s1, "rtp_send_data size=%d\n", len);
233 /* build the RTP header */
234 avio_w8(s1->pb, (RTP_VERSION << 6));
235 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
236 avio_wb16(s1->pb, s->seq);
237 avio_wb32(s1->pb, s->timestamp);
238 avio_wb32(s1->pb, s->ssrc);
240 avio_write(s1->pb, buf1, len);
244 s->octet_count += len;
248 /* send an integer number of samples and compute time stamp and fill
249 the rtp send buffer before sending. */
250 static void rtp_send_samples(AVFormatContext *s1,
251 const uint8_t *buf1, int size, int sample_size)
253 RTPMuxContext *s = s1->priv_data;
254 int len, max_packet_size, n;
256 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
257 /* not needed, but who nows */
258 if ((size % sample_size) != 0)
263 len = FFMIN(max_packet_size, size);
266 memcpy(s->buf_ptr, buf1, len);
270 s->timestamp = s->cur_timestamp + n / sample_size;
271 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
272 n += (s->buf_ptr - s->buf);
276 static void rtp_send_mpegaudio(AVFormatContext *s1,
277 const uint8_t *buf1, int size)
279 RTPMuxContext *s = s1->priv_data;
280 int len, count, max_packet_size;
282 max_packet_size = s->max_payload_size;
284 /* test if we must flush because not enough space */
285 len = (s->buf_ptr - s->buf);
286 if ((len + size) > max_packet_size) {
288 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
289 s->buf_ptr = s->buf + 4;
292 if (s->buf_ptr == s->buf + 4) {
293 s->timestamp = s->cur_timestamp;
297 if (size > max_packet_size) {
298 /* big packet: fragment */
301 len = max_packet_size - 4;
304 /* build fragmented packet */
307 s->buf[2] = count >> 8;
309 memcpy(s->buf + 4, buf1, len);
310 ff_rtp_send_data(s1, s->buf, len + 4, 0);
316 if (s->buf_ptr == s->buf + 4) {
317 /* no fragmentation possible */
323 memcpy(s->buf_ptr, buf1, size);
328 static void rtp_send_raw(AVFormatContext *s1,
329 const uint8_t *buf1, int size)
331 RTPMuxContext *s = s1->priv_data;
332 int len, max_packet_size;
334 max_packet_size = s->max_payload_size;
337 len = max_packet_size;
341 s->timestamp = s->cur_timestamp;
342 ff_rtp_send_data(s1, buf1, len, (len == size));
349 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
350 static void rtp_send_mpegts_raw(AVFormatContext *s1,
351 const uint8_t *buf1, int size)
353 RTPMuxContext *s = s1->priv_data;
356 while (size >= TS_PACKET_SIZE) {
357 len = s->max_payload_size - (s->buf_ptr - s->buf);
360 memcpy(s->buf_ptr, buf1, len);
365 out_len = s->buf_ptr - s->buf;
366 if (out_len >= s->max_payload_size) {
367 ff_rtp_send_data(s1, s->buf, out_len, 0);
373 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
375 RTPMuxContext *s = s1->priv_data;
376 AVStream *st = s1->streams[0];
380 av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
382 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
384 if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
385 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
386 rtcp_send_sr(s1, ff_ntp_time());
387 s->last_octet_count = s->octet_count;
390 s->cur_timestamp = s->base_timestamp + pkt->pts;
392 switch(st->codec->codec_id) {
393 case CODEC_ID_PCM_MULAW:
394 case CODEC_ID_PCM_ALAW:
395 case CODEC_ID_PCM_U8:
396 case CODEC_ID_PCM_S8:
397 rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
399 case CODEC_ID_PCM_U16BE:
400 case CODEC_ID_PCM_U16LE:
401 case CODEC_ID_PCM_S16BE:
402 case CODEC_ID_PCM_S16LE:
403 rtp_send_samples(s1, pkt->data, size, 2 * st->codec->channels);
405 case CODEC_ID_ADPCM_G722:
406 /* The actual sample size is half a byte per sample, but since the
407 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
408 * the correct parameter for send_samples is 1 byte per stream clock. */
409 rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
413 rtp_send_mpegaudio(s1, pkt->data, size);
415 case CODEC_ID_MPEG1VIDEO:
416 case CODEC_ID_MPEG2VIDEO:
417 ff_rtp_send_mpegvideo(s1, pkt->data, size);
420 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
421 ff_rtp_send_latm(s1, pkt->data, size);
423 ff_rtp_send_aac(s1, pkt->data, size);
425 case CODEC_ID_AMR_NB:
426 case CODEC_ID_AMR_WB:
427 ff_rtp_send_amr(s1, pkt->data, size);
429 case CODEC_ID_MPEG2TS:
430 rtp_send_mpegts_raw(s1, pkt->data, size);
433 ff_rtp_send_h264(s1, pkt->data, size);
437 ff_rtp_send_h263(s1, pkt->data, size);
439 case CODEC_ID_VORBIS:
440 case CODEC_ID_THEORA:
441 ff_rtp_send_xiph(s1, pkt->data, size);
444 ff_rtp_send_vp8(s1, pkt->data, size);
447 /* better than nothing : send the codec raw data */
448 rtp_send_raw(s1, pkt->data, size);
454 static int rtp_write_trailer(AVFormatContext *s1)
456 RTPMuxContext *s = s1->priv_data;
463 AVOutputFormat ff_rtp_muxer = {
465 .long_name = NULL_IF_CONFIG_SMALL("RTP output format"),
466 .priv_data_size = sizeof(RTPMuxContext),
467 .audio_codec = CODEC_ID_PCM_MULAW,
468 .video_codec = CODEC_ID_MPEG4,
469 .write_header = rtp_write_header,
470 .write_packet = rtp_write_packet,
471 .write_trailer = rtp_write_trailer,
472 .priv_class = &rtp_muxer_class,