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1/*
2 * RTP output format
3 * Copyright (c) 2002 Fabrice Bellard
4 *
5 * This file is part of FFmpeg.
6 *
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.
11 *
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.
16 *
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
20 */
21
22#include "avformat.h"
23#include "mpegts.h"
24#include "internal.h"
25#include "libavutil/mathematics.h"
26#include "libavutil/random_seed.h"
27#include "libavutil/opt.h"
28
29#include "rtpenc.h"
30
31static const AVOption options[] = {
32 FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
33 { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
34 { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
35 { "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM },
36 { "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM },
37 { NULL },
38};
39
40static const AVClass rtp_muxer_class = {
41 .class_name = "RTP muxer",
42 .item_name = av_default_item_name,
43 .option = options,
44 .version = LIBAVUTIL_VERSION_INT,
45};
46
47#define RTCP_SR_SIZE 28
48
49static int is_supported(enum AVCodecID id)
50{
51 switch(id) {
52 case AV_CODEC_ID_DIRAC:
53 case AV_CODEC_ID_H261:
54 case AV_CODEC_ID_H263:
55 case AV_CODEC_ID_H263P:
56 case AV_CODEC_ID_H264:
57 case AV_CODEC_ID_HEVC:
58 case AV_CODEC_ID_MPEG1VIDEO:
59 case AV_CODEC_ID_MPEG2VIDEO:
60 case AV_CODEC_ID_MPEG4:
61 case AV_CODEC_ID_AAC:
62 case AV_CODEC_ID_MP2:
63 case AV_CODEC_ID_MP3:
64 case AV_CODEC_ID_PCM_ALAW:
65 case AV_CODEC_ID_PCM_MULAW:
66 case AV_CODEC_ID_PCM_S8:
67 case AV_CODEC_ID_PCM_S16BE:
68 case AV_CODEC_ID_PCM_S16LE:
69 case AV_CODEC_ID_PCM_U16BE:
70 case AV_CODEC_ID_PCM_U16LE:
71 case AV_CODEC_ID_PCM_U8:
72 case AV_CODEC_ID_MPEG2TS:
73 case AV_CODEC_ID_AMR_NB:
74 case AV_CODEC_ID_AMR_WB:
75 case AV_CODEC_ID_VORBIS:
76 case AV_CODEC_ID_THEORA:
77 case AV_CODEC_ID_VP8:
78 case AV_CODEC_ID_VP9:
79 case AV_CODEC_ID_ADPCM_G722:
80 case AV_CODEC_ID_ADPCM_G726:
81 case AV_CODEC_ID_ILBC:
82 case AV_CODEC_ID_MJPEG:
83 case AV_CODEC_ID_SPEEX:
84 case AV_CODEC_ID_OPUS:
85 return 1;
86 default:
87 return 0;
88 }
89}
90
91static int rtp_write_header(AVFormatContext *s1)
92{
93 RTPMuxContext *s = s1->priv_data;
94 int n, ret = AVERROR(EINVAL);
95 AVStream *st;
96
97 if (s1->nb_streams != 1) {
98 av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
99 return AVERROR(EINVAL);
100 }
101 st = s1->streams[0];
102 if (!is_supported(st->codecpar->codec_id)) {
103 av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codecpar->codec_id));
104
105 return -1;
106 }
107
108 if (s->payload_type < 0) {
109 /* Re-validate non-dynamic payload types */
110 if (st->id < RTP_PT_PRIVATE)
111 st->id = ff_rtp_get_payload_type(s1, st->codecpar, -1);
112
113 s->payload_type = st->id;
114 } else {
115 /* private option takes priority */
116 st->id = s->payload_type;
117 }
118
119 s->base_timestamp = av_get_random_seed();
120 s->timestamp = s->base_timestamp;
121 s->cur_timestamp = 0;
122 if (!s->ssrc)
123 s->ssrc = av_get_random_seed();
124 s->first_packet = 1;
125 s->first_rtcp_ntp_time = ff_ntp_time();
126 if (s1->start_time_realtime != 0 && s1->start_time_realtime != AV_NOPTS_VALUE)
127 /* Round the NTP time to whole milliseconds. */
128 s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
129 NTP_OFFSET_US;
130 // Pick a random sequence start number, but in the lower end of the
131 // available range, so that any wraparound doesn't happen immediately.
132 // (Immediate wraparound would be an issue for SRTP.)
133 if (s->seq < 0) {
134 if (s1->flags & AVFMT_FLAG_BITEXACT) {
135 s->seq = 0;
136 } else
137 s->seq = av_get_random_seed() & 0x0fff;
138 } else
139 s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
140
141 if (s1->packet_size) {
142 if (s1->pb->max_packet_size)
143 s1->packet_size = FFMIN(s1->packet_size,
144 s1->pb->max_packet_size);
145 } else
146 s1->packet_size = s1->pb->max_packet_size;
147 if (s1->packet_size <= 12) {
148 av_log(s1, AV_LOG_ERROR, "Max packet size %u too low\n", s1->packet_size);
149 return AVERROR(EIO);
150 }
151 s->buf = av_malloc(s1->packet_size);
152 if (!s->buf) {
153 return AVERROR(ENOMEM);
154 }
155 s->max_payload_size = s1->packet_size - 12;
156
157 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
158 avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
159 } else {
160 avpriv_set_pts_info(st, 32, 1, 90000);
161 }
162 s->buf_ptr = s->buf;
163 switch(st->codecpar->codec_id) {
164 case AV_CODEC_ID_MP2:
165 case AV_CODEC_ID_MP3:
166 s->buf_ptr = s->buf + 4;
167 avpriv_set_pts_info(st, 32, 1, 90000);
168 break;
169 case AV_CODEC_ID_MPEG1VIDEO:
170 case AV_CODEC_ID_MPEG2VIDEO:
171 break;
172 case AV_CODEC_ID_MPEG2TS:
173 n = s->max_payload_size / TS_PACKET_SIZE;
174 if (n < 1)
175 n = 1;
176 s->max_payload_size = n * TS_PACKET_SIZE;
177 break;
178 case AV_CODEC_ID_DIRAC:
179 if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
180 av_log(s, AV_LOG_ERROR,
181 "Packetizing VC-2 is experimental and does not use all values "
182 "of the specification "
183 "(even though most receivers may handle it just fine). "
184 "Please set -strict experimental in order to enable it.\n");
185 ret = AVERROR_EXPERIMENTAL;
186 goto fail;
187 }
188 break;
189 case AV_CODEC_ID_H261:
190 if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
191 av_log(s, AV_LOG_ERROR,
192 "Packetizing H.261 is experimental and produces incorrect "
193 "packetization for cases where GOBs don't fit into packets "
194 "(even though most receivers may handle it just fine). "
195 "Please set -f_strict experimental in order to enable it.\n");
196 ret = AVERROR_EXPERIMENTAL;
197 goto fail;
198 }
199 break;
200 case AV_CODEC_ID_H264:
201 /* check for H.264 MP4 syntax */
202 if (st->codecpar->extradata_size > 4 && st->codecpar->extradata[0] == 1) {
203 s->nal_length_size = (st->codecpar->extradata[4] & 0x03) + 1;
204 }
205 break;
206 case AV_CODEC_ID_HEVC:
207 /* Only check for the standardized hvcC version of extradata, keeping
208 * things simple and similar to the avcC/H.264 case above, instead
209 * of trying to handle the pre-standardization versions (as in
210 * libavcodec/hevc.c). */
211 if (st->codecpar->extradata_size > 21 && st->codecpar->extradata[0] == 1) {
212 s->nal_length_size = (st->codecpar->extradata[21] & 0x03) + 1;
213 }
214 break;
215 case AV_CODEC_ID_VP9:
216 if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
217 av_log(s, AV_LOG_ERROR,
218 "Packetizing VP9 is experimental and its specification is "
219 "still in draft state. "
220 "Please set -strict experimental in order to enable it.\n");
221 ret = AVERROR_EXPERIMENTAL;
222 goto fail;
223 }
224 break;
225 case AV_CODEC_ID_VORBIS:
226 case AV_CODEC_ID_THEORA:
227 s->max_frames_per_packet = 15;
228 break;
229 case AV_CODEC_ID_ADPCM_G722:
230 /* Due to a historical error, the clock rate for G722 in RTP is
231 * 8000, even if the sample rate is 16000. See RFC 3551. */
232 avpriv_set_pts_info(st, 32, 1, 8000);
233 break;
234 case AV_CODEC_ID_OPUS:
235 if (st->codecpar->channels > 2) {
236 av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
237 goto fail;
238 }
239 /* The opus RTP RFC says that all opus streams should use 48000 Hz
240 * as clock rate, since all opus sample rates can be expressed in
241 * this clock rate, and sample rate changes on the fly are supported. */
242 avpriv_set_pts_info(st, 32, 1, 48000);
243 break;
244 case AV_CODEC_ID_ILBC:
245 if (st->codecpar->block_align != 38 && st->codecpar->block_align != 50) {
246 av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
247 goto fail;
248 }
249 s->max_frames_per_packet = s->max_payload_size / st->codecpar->block_align;
250 break;
251 case AV_CODEC_ID_AMR_NB:
252 case AV_CODEC_ID_AMR_WB:
253 s->max_frames_per_packet = 50;
254 if (st->codecpar->codec_id == AV_CODEC_ID_AMR_NB)
255 n = 31;
256 else
257 n = 61;
258 /* max_header_toc_size + the largest AMR payload must fit */
259 if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
260 av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
261 goto fail;
262 }
263 if (st->codecpar->channels != 1) {
264 av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
265 goto fail;
266 }
267 break;
268 case AV_CODEC_ID_AAC:
269 s->max_frames_per_packet = 50;
270 break;
271 default:
272 break;
273 }
274
275 return 0;
276
277fail:
278 av_freep(&s->buf);
279 return ret;
280}
281
282/* send an rtcp sender report packet */
283static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
284{
285 RTPMuxContext *s = s1->priv_data;
286 uint32_t rtp_ts;
287
288 av_log(s1, AV_LOG_TRACE, "RTCP: %02x %"PRIx64" %"PRIx32"\n", s->payload_type, ntp_time, s->timestamp);
289
290 s->last_rtcp_ntp_time = ntp_time;
291 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
292 s1->streams[0]->time_base) + s->base_timestamp;
293 avio_w8(s1->pb, RTP_VERSION << 6);
294 avio_w8(s1->pb, RTCP_SR);
295 avio_wb16(s1->pb, 6); /* length in words - 1 */
296 avio_wb32(s1->pb, s->ssrc);
297 avio_wb32(s1->pb, ntp_time / 1000000);
298 avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
299 avio_wb32(s1->pb, rtp_ts);
300 avio_wb32(s1->pb, s->packet_count);
301 avio_wb32(s1->pb, s->octet_count);
302
303 if (s->cname) {
304 int len = FFMIN(strlen(s->cname), 255);
305 avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
306 avio_w8(s1->pb, RTCP_SDES);
307 avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
308
309 avio_wb32(s1->pb, s->ssrc);
310 avio_w8(s1->pb, 0x01); /* CNAME */
311 avio_w8(s1->pb, len);
312 avio_write(s1->pb, s->cname, len);
313 avio_w8(s1->pb, 0); /* END */
314 for (len = (7 + len) % 4; len % 4; len++)
315 avio_w8(s1->pb, 0);
316 }
317
318 if (bye) {
319 avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
320 avio_w8(s1->pb, RTCP_BYE);
321 avio_wb16(s1->pb, 1); /* length in words - 1 */
322 avio_wb32(s1->pb, s->ssrc);
323 }
324
325 avio_flush(s1->pb);
326}
327
328/* send an rtp packet. sequence number is incremented, but the caller
329 must update the timestamp itself */
330void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
331{
332 RTPMuxContext *s = s1->priv_data;
333
334 av_log(s1, AV_LOG_TRACE, "rtp_send_data size=%d\n", len);
335
336 /* build the RTP header */
337 avio_w8(s1->pb, RTP_VERSION << 6);
338 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
339 avio_wb16(s1->pb, s->seq);
340 avio_wb32(s1->pb, s->timestamp);
341 avio_wb32(s1->pb, s->ssrc);
342
343 avio_write(s1->pb, buf1, len);
344 avio_flush(s1->pb);
345
346 s->seq = (s->seq + 1) & 0xffff;
347 s->octet_count += len;
348 s->packet_count++;
349}
350
351/* send an integer number of samples and compute time stamp and fill
352 the rtp send buffer before sending. */
353static int rtp_send_samples(AVFormatContext *s1,
354 const uint8_t *buf1, int size, int sample_size_bits)
355{
356 RTPMuxContext *s = s1->priv_data;
357 int len, max_packet_size, n;
358 /* Calculate the number of bytes to get samples aligned on a byte border */
359 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
360
361 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
362 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
363 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
364 return AVERROR(EINVAL);
365 n = 0;
366 while (size > 0) {
367 s->buf_ptr = s->buf;
368 len = FFMIN(max_packet_size, size);
369
370 /* copy data */
371 memcpy(s->buf_ptr, buf1, len);
372 s->buf_ptr += len;
373 buf1 += len;
374 size -= len;
375 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
376 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
377 n += (s->buf_ptr - s->buf);
378 }
379 return 0;
380}
381
382static void rtp_send_mpegaudio(AVFormatContext *s1,
383 const uint8_t *buf1, int size)
384{
385 RTPMuxContext *s = s1->priv_data;
386 int len, count, max_packet_size;
387
388 max_packet_size = s->max_payload_size;
389
390 /* test if we must flush because not enough space */
391 len = (s->buf_ptr - s->buf);
392 if ((len + size) > max_packet_size) {
393 if (len > 4) {
394 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
395 s->buf_ptr = s->buf + 4;
396 }
397 }
398 if (s->buf_ptr == s->buf + 4) {
399 s->timestamp = s->cur_timestamp;
400 }
401
402 /* add the packet */
403 if (size > max_packet_size) {
404 /* big packet: fragment */
405 count = 0;
406 while (size > 0) {
407 len = max_packet_size - 4;
408 if (len > size)
409 len = size;
410 /* build fragmented packet */
411 s->buf[0] = 0;
412 s->buf[1] = 0;
413 s->buf[2] = count >> 8;
414 s->buf[3] = count;
415 memcpy(s->buf + 4, buf1, len);
416 ff_rtp_send_data(s1, s->buf, len + 4, 0);
417 size -= len;
418 buf1 += len;
419 count += len;
420 }
421 } else {
422 if (s->buf_ptr == s->buf + 4) {
423 /* no fragmentation possible */
424 s->buf[0] = 0;
425 s->buf[1] = 0;
426 s->buf[2] = 0;
427 s->buf[3] = 0;
428 }
429 memcpy(s->buf_ptr, buf1, size);
430 s->buf_ptr += size;
431 }
432}
433
434static void rtp_send_raw(AVFormatContext *s1,
435 const uint8_t *buf1, int size)
436{
437 RTPMuxContext *s = s1->priv_data;
438 int len, max_packet_size;
439
440 max_packet_size = s->max_payload_size;
441
442 while (size > 0) {
443 len = max_packet_size;
444 if (len > size)
445 len = size;
446
447 s->timestamp = s->cur_timestamp;
448 ff_rtp_send_data(s1, buf1, len, (len == size));
449
450 buf1 += len;
451 size -= len;
452 }
453}
454
455/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
456static void rtp_send_mpegts_raw(AVFormatContext *s1,
457 const uint8_t *buf1, int size)
458{
459 RTPMuxContext *s = s1->priv_data;
460 int len, out_len;
461
462 s->timestamp = s->cur_timestamp;
463 while (size >= TS_PACKET_SIZE) {
464 len = s->max_payload_size - (s->buf_ptr - s->buf);
465 if (len > size)
466 len = size;
467 memcpy(s->buf_ptr, buf1, len);
468 buf1 += len;
469 size -= len;
470 s->buf_ptr += len;
471
472 out_len = s->buf_ptr - s->buf;
473 if (out_len >= s->max_payload_size) {
474 ff_rtp_send_data(s1, s->buf, out_len, 0);
475 s->buf_ptr = s->buf;
476 }
477 }
478}
479
480static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
481{
482 RTPMuxContext *s = s1->priv_data;
483 AVStream *st = s1->streams[0];
484 int frame_duration = av_get_audio_frame_duration2(st->codecpar, 0);
485 int frame_size = st->codecpar->block_align;
486 int frames = size / frame_size;
487
488 while (frames > 0) {
489 if (s->num_frames > 0 &&
490 av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
491 s1->max_delay, AV_TIME_BASE_Q) >= 0) {
492 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
493 s->num_frames = 0;
494 }
495
496 if (!s->num_frames) {
497 s->buf_ptr = s->buf;
498 s->timestamp = s->cur_timestamp;
499 }
500 memcpy(s->buf_ptr, buf, frame_size);
501 frames--;
502 s->num_frames++;
503 s->buf_ptr += frame_size;
504 buf += frame_size;
505 s->cur_timestamp += frame_duration;
506
507 if (s->num_frames == s->max_frames_per_packet) {
508 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
509 s->num_frames = 0;
510 }
511 }
512 return 0;
513}
514
515static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
516{
517 RTPMuxContext *s = s1->priv_data;
518 AVStream *st = s1->streams[0];
519 int rtcp_bytes;
520 int size= pkt->size;
521
522 av_log(s1, AV_LOG_TRACE, "%d: write len=%d\n", pkt->stream_index, size);
523
524 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
525 RTCP_TX_RATIO_DEN;
526 if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
527 (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
528 !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
529 rtcp_send_sr(s1, ff_ntp_time(), 0);
530 s->last_octet_count = s->octet_count;
531 s->first_packet = 0;
532 }
533 s->cur_timestamp = s->base_timestamp + pkt->pts;
534
535 switch(st->codecpar->codec_id) {
536 case AV_CODEC_ID_PCM_MULAW:
537 case AV_CODEC_ID_PCM_ALAW:
538 case AV_CODEC_ID_PCM_U8:
539 case AV_CODEC_ID_PCM_S8:
540 return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
541 case AV_CODEC_ID_PCM_U16BE:
542 case AV_CODEC_ID_PCM_U16LE:
543 case AV_CODEC_ID_PCM_S16BE:
544 case AV_CODEC_ID_PCM_S16LE:
545 return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->channels);
546 case AV_CODEC_ID_ADPCM_G722:
547 /* The actual sample size is half a byte per sample, but since the
548 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
549 * the correct parameter for send_samples_bits is 8 bits per stream
550 * clock. */
551 return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels);
552 case AV_CODEC_ID_ADPCM_G726:
553 return rtp_send_samples(s1, pkt->data, size,
554 st->codecpar->bits_per_coded_sample * st->codecpar->channels);
555 case AV_CODEC_ID_MP2:
556 case AV_CODEC_ID_MP3:
557 rtp_send_mpegaudio(s1, pkt->data, size);
558 break;
559 case AV_CODEC_ID_MPEG1VIDEO:
560 case AV_CODEC_ID_MPEG2VIDEO:
561 ff_rtp_send_mpegvideo(s1, pkt->data, size);
562 break;
563 case AV_CODEC_ID_AAC:
564 if (s->flags & FF_RTP_FLAG_MP4A_LATM)
565 ff_rtp_send_latm(s1, pkt->data, size);
566 else
567 ff_rtp_send_aac(s1, pkt->data, size);
568 break;
569 case AV_CODEC_ID_AMR_NB:
570 case AV_CODEC_ID_AMR_WB:
571 ff_rtp_send_amr(s1, pkt->data, size);
572 break;
573 case AV_CODEC_ID_MPEG2TS:
574 rtp_send_mpegts_raw(s1, pkt->data, size);
575 break;
576 case AV_CODEC_ID_DIRAC:
577 ff_rtp_send_vc2hq(s1, pkt->data, size, st->codecpar->field_order != AV_FIELD_PROGRESSIVE ? 1 : 0);
578 break;
579 case AV_CODEC_ID_H264:
580 ff_rtp_send_h264_hevc(s1, pkt->data, size);
581 break;
582 case AV_CODEC_ID_H261:
583 ff_rtp_send_h261(s1, pkt->data, size);
584 break;
585 case AV_CODEC_ID_H263:
586 if (s->flags & FF_RTP_FLAG_RFC2190) {
587 int mb_info_size = 0;
588 const uint8_t *mb_info =
589 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
590 &mb_info_size);
591 ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
592 break;
593 }
594 /* Fallthrough */
595 case AV_CODEC_ID_H263P:
596 ff_rtp_send_h263(s1, pkt->data, size);
597 break;
598 case AV_CODEC_ID_HEVC:
599 ff_rtp_send_h264_hevc(s1, pkt->data, size);
600 break;
601 case AV_CODEC_ID_VORBIS:
602 case AV_CODEC_ID_THEORA:
603 ff_rtp_send_xiph(s1, pkt->data, size);
604 break;
605 case AV_CODEC_ID_VP8:
606 ff_rtp_send_vp8(s1, pkt->data, size);
607 break;
608 case AV_CODEC_ID_VP9:
609 ff_rtp_send_vp9(s1, pkt->data, size);
610 break;
611 case AV_CODEC_ID_ILBC:
612 rtp_send_ilbc(s1, pkt->data, size);
613 break;
614 case AV_CODEC_ID_MJPEG:
615 ff_rtp_send_jpeg(s1, pkt->data, size);
616 break;
617 case AV_CODEC_ID_OPUS:
618 if (size > s->max_payload_size) {
619 av_log(s1, AV_LOG_ERROR,
620 "Packet size %d too large for max RTP payload size %d\n",
621 size, s->max_payload_size);
622 return AVERROR(EINVAL);
623 }
624 /* Intentional fallthrough */
625 default:
626 /* better than nothing : send the codec raw data */
627 rtp_send_raw(s1, pkt->data, size);
628 break;
629 }
630 return 0;
631}
632
633static int rtp_write_trailer(AVFormatContext *s1)
634{
635 RTPMuxContext *s = s1->priv_data;
636
637 /* If the caller closes and recreates ->pb, this might actually
638 * be NULL here even if it was successfully allocated at the start. */
639 if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
640 rtcp_send_sr(s1, ff_ntp_time(), 1);
641 av_freep(&s->buf);
642
643 return 0;
644}
645
646AVOutputFormat ff_rtp_muxer = {
647 .name = "rtp",
648 .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
649 .priv_data_size = sizeof(RTPMuxContext),
650 .audio_codec = AV_CODEC_ID_PCM_MULAW,
651 .video_codec = AV_CODEC_ID_MPEG4,
652 .write_header = rtp_write_header,
653 .write_packet = rtp_write_packet,
654 .write_trailer = rtp_write_trailer,
655 .priv_class = &rtp_muxer_class,
656 .flags = AVFMT_TS_NONSTRICT,
657};
658
generated by cgit at 2024-05-17 05:01:29 (GMT)