FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavformat/rtpenc.c
Date: 2026-04-30 02:42:23
Exec Total Coverage
Lines: 127 394 32.2%
Functions: 9 12 75.0%
Branches: 36 174 20.7%
Line Branch Exec Source
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 "libavutil/attributes.h"
24 #include "mpegts.h"
25 #include "internal.h"
26 #include "mux.h"
27 #include "libavutil/mathematics.h"
28 #include "libavutil/mem.h"
29 #include "libavutil/random_seed.h"
30 #include "libavutil/opt.h"
31
32 #include "rtpenc.h"
33
34 static const AVOption options[] = {
35 FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
36 { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
37 { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_UINT, { .i64 = 0 }, 0, UINT32_MAX, AV_OPT_FLAG_ENCODING_PARAM },
38 { "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM },
39 { "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM },
40 { NULL },
41 };
42
43 static const AVClass rtp_muxer_class = {
44 .class_name = "RTP muxer",
45 .item_name = av_default_item_name,
46 .option = options,
47 .version = LIBAVUTIL_VERSION_INT,
48 };
49
50 #define RTCP_SR_SIZE 28
51
52 2 static int is_supported(enum AVCodecID id)
53 {
54
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 2 switch(id) {
55 2 case AV_CODEC_ID_DIRAC:
56 case AV_CODEC_ID_H261:
57 case AV_CODEC_ID_H263:
58 case AV_CODEC_ID_H263P:
59 case AV_CODEC_ID_H264:
60 case AV_CODEC_ID_HEVC:
61 case AV_CODEC_ID_MPEG1VIDEO:
62 case AV_CODEC_ID_MPEG2VIDEO:
63 case AV_CODEC_ID_MPEG4:
64 case AV_CODEC_ID_AAC:
65 case AV_CODEC_ID_MP2:
66 case AV_CODEC_ID_MP3:
67 case AV_CODEC_ID_PCM_ALAW:
68 case AV_CODEC_ID_PCM_MULAW:
69 case AV_CODEC_ID_PCM_S8:
70 case AV_CODEC_ID_PCM_S16BE:
71 case AV_CODEC_ID_PCM_S16LE:
72 case AV_CODEC_ID_PCM_S24BE:
73 case AV_CODEC_ID_PCM_U16BE:
74 case AV_CODEC_ID_PCM_U16LE:
75 case AV_CODEC_ID_PCM_U8:
76 case AV_CODEC_ID_MPEG2TS:
77 case AV_CODEC_ID_AMR_NB:
78 case AV_CODEC_ID_AMR_WB:
79 case AV_CODEC_ID_VORBIS:
80 case AV_CODEC_ID_THEORA:
81 case AV_CODEC_ID_VP8:
82 case AV_CODEC_ID_VP9:
83 case AV_CODEC_ID_AV1:
84 case AV_CODEC_ID_ADPCM_G722:
85 case AV_CODEC_ID_ADPCM_G726:
86 case AV_CODEC_ID_ADPCM_G726LE:
87 case AV_CODEC_ID_ILBC:
88 case AV_CODEC_ID_MJPEG:
89 case AV_CODEC_ID_SPEEX:
90 case AV_CODEC_ID_OPUS:
91 case AV_CODEC_ID_RAWVIDEO:
92 case AV_CODEC_ID_BITPACKED:
93 case AV_CODEC_ID_G728:
94 2 return 1;
95 ✗ default:
96 ✗ return 0;
97 }
98 }
99
100 2 static int rtp_write_header(AVFormatContext *s1)
101 {
102 2 RTPMuxContext *s = s1->priv_data;
103 2 int n, ret = AVERROR(EINVAL);
104 AVStream *st;
105
106
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 2 if (s1->nb_streams != 1) {
107 ✗ av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
108 ✗ return AVERROR(EINVAL);
109 }
110 2 st = s1->streams[0];
111
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 2 if (!is_supported(st->codecpar->codec_id)) {
112 ✗ av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codecpar->codec_id));
113
114 ✗ return -1;
115 }
116
117
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 2 if (s->payload_type < 0) {
118 /* Re-validate non-dynamic payload types */
119
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 2 if (st->id < RTP_PT_PRIVATE)
120 ✗ st->id = ff_rtp_get_payload_type(s1, st->codecpar, -1);
121
122 2 s->payload_type = st->id;
123 } else {
124 /* private option takes priority */
125 ✗ st->id = s->payload_type;
126 }
127
128 2 s->base_timestamp = av_get_random_seed();
129 2 s->timestamp = s->base_timestamp;
130 2 s->cur_timestamp = 0;
131
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 2 if (!s->ssrc)
132 2 s->ssrc = av_get_random_seed();
133 2 s->first_packet = 1;
134 2 s->first_rtcp_ntp_time = ff_ntp_time();
135
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 2 if (s1->start_time_realtime != 0 && s1->start_time_realtime != AV_NOPTS_VALUE)
136 /* Round the NTP time to whole milliseconds. */
137 ✗ s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
138 NTP_OFFSET_US;
139 // Pick a random sequence start number, but in the lower end of the
140 // available range, so that any wraparound doesn't happen immediately.
141 // (Immediate wraparound would be an issue for SRTP.)
142
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 2 if (s->seq < 0) {
143
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 2 if (s1->flags & AVFMT_FLAG_BITEXACT) {
144 2 s->seq = 0;
145 } else
146 ✗ s->seq = av_get_random_seed() & 0x0fff;
147 } else
148 ✗ s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
149
150
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 2 if (s1->packet_size) {
151 ✗ if (s1->pb->max_packet_size)
152 ✗ s1->packet_size = FFMIN(s1->packet_size,
153 s1->pb->max_packet_size);
154 } else
155 2 s1->packet_size = s1->pb->max_packet_size;
156
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 2 if (s1->packet_size <= 12) {
157 ✗ av_log(s1, AV_LOG_ERROR, "Max packet size %u too low\n", s1->packet_size);
158 ✗ return AVERROR(EIO);
159 }
160 2 s->buf = av_malloc(s1->packet_size);
161
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 2 if (!s->buf) {
162 ✗ return AVERROR(ENOMEM);
163 }
164 2 s->max_payload_size = s1->packet_size - 12;
165
166
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 2 if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
167 1 avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate);
168 } else {
169 1 avpriv_set_pts_info(st, 32, 1, 90000);
170 }
171 2 s->buf_ptr = s->buf;
172
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 2 switch(st->codecpar->codec_id) {
173 ✗ case AV_CODEC_ID_MP2:
174 case AV_CODEC_ID_MP3:
175 ✗ s->buf_ptr = s->buf + 4;
176 ✗ avpriv_set_pts_info(st, 32, 1, 90000);
177 ✗ break;
178 ✗ case AV_CODEC_ID_MPEG1VIDEO:
179 case AV_CODEC_ID_MPEG2VIDEO:
180 ✗ break;
181 ✗ case AV_CODEC_ID_MPEG2TS:
182 ✗ if (s->max_payload_size < TS_PACKET_SIZE) {
183 ✗ av_log(s1, AV_LOG_ERROR,
184 "RTP payload size %u too small for MPEG-TS "
185 "(minimum %d bytes required)\n",
186 s->max_payload_size, TS_PACKET_SIZE);
187 ✗ ret = AVERROR(EINVAL);
188 ✗ goto fail;
189 }
190
191 ✗ n = s->max_payload_size / TS_PACKET_SIZE;
192 ✗ s->max_payload_size = n * TS_PACKET_SIZE;
193 ✗ break;
194 ✗ case AV_CODEC_ID_DIRAC:
195 ✗ if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
196 ✗ av_log(s, AV_LOG_ERROR,
197 "Packetizing VC-2 is experimental and does not use all values "
198 "of the specification "
199 "(even though most receivers may handle it just fine). "
200 "Please set -strict experimental in order to enable it.\n");
201 ✗ ret = AVERROR_EXPERIMENTAL;
202 ✗ goto fail;
203 }
204 ✗ break;
205 ✗ case AV_CODEC_ID_H261:
206 ✗ if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
207 ✗ av_log(s, AV_LOG_ERROR,
208 "Packetizing H.261 is experimental and produces incorrect "
209 "packetization for cases where GOBs don't fit into packets "
210 "(even though most receivers may handle it just fine). "
211 "Please set -f_strict experimental in order to enable it.\n");
212 ✗ ret = AVERROR_EXPERIMENTAL;
213 ✗ goto fail;
214 }
215 ✗ break;
216 ✗ case AV_CODEC_ID_H264:
217 /* check for H.264 MP4 syntax */
218 ✗ if (st->codecpar->extradata_size > 4 && st->codecpar->extradata[0] == 1) {
219 ✗ s->nal_length_size = (st->codecpar->extradata[4] & 0x03) + 1;
220 }
221 ✗ break;
222 ✗ case AV_CODEC_ID_HEVC:
223 /* Only check for the standardized hvcC version of extradata, keeping
224 * things simple and similar to the avcC/H.264 case above, instead
225 * of trying to handle the pre-standardization versions (as in
226 * libavcodec/hevc.c). */
227 ✗ if (st->codecpar->extradata_size > 21 && st->codecpar->extradata[0] == 1) {
228 ✗ s->nal_length_size = (st->codecpar->extradata[21] & 0x03) + 1;
229 }
230 ✗ break;
231 ✗ case AV_CODEC_ID_MJPEG:
232 case AV_CODEC_ID_BITPACKED:
233 case AV_CODEC_ID_RAWVIDEO:
234 ✗ if (st->codecpar->width <= 0 || st->codecpar->height <= 0) {
235 ✗ av_log(s1, AV_LOG_ERROR, "dimensions not set\n");
236 ✗ return AVERROR(EINVAL);
237 }
238 ✗ break;
239 ✗ case AV_CODEC_ID_VP9:
240 ✗ if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
241 ✗ av_log(s, AV_LOG_ERROR,
242 "Packetizing VP9 is experimental and its specification is "
243 "still in draft state. "
244 "Please set -strict experimental in order to enable it.\n");
245 ✗ ret = AVERROR_EXPERIMENTAL;
246 ✗ goto fail;
247 }
248 ✗ break;
249 ✗ case AV_CODEC_ID_AV1:
250 ✗ if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
251 ✗ av_log(s, AV_LOG_ERROR,
252 "Packetizing AV1 is experimental and its specification is "
253 "still in draft state. "
254 "Please set -strict experimental in order to enable it.\n");
255 ✗ ret = AVERROR_EXPERIMENTAL;
256 ✗ goto fail;
257 }
258 ✗ break;
259 ✗ case AV_CODEC_ID_VORBIS:
260 case AV_CODEC_ID_THEORA:
261 ✗ s->max_frames_per_packet = 15;
262 ✗ break;
263 ✗ case AV_CODEC_ID_ADPCM_G722:
264 /* Due to a historical error, the clock rate for G722 in RTP is
265 * 8000, even if the sample rate is 16000. See RFC 3551. */
266 ✗ avpriv_set_pts_info(st, 32, 1, 8000);
267 ✗ break;
268 ✗ case AV_CODEC_ID_OPUS:
269 ✗ if (st->codecpar->ch_layout.nb_channels > 2) {
270 ✗ av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
271 ✗ goto fail;
272 }
273 /* The opus RTP RFC says that all opus streams should use 48000 Hz
274 * as clock rate, since all opus sample rates can be expressed in
275 * this clock rate, and sample rate changes on the fly are supported. */
276 ✗ avpriv_set_pts_info(st, 32, 1, 48000);
277 ✗ break;
278 ✗ case AV_CODEC_ID_ILBC:
279 ✗ if (st->codecpar->block_align != 38 && st->codecpar->block_align != 50) {
280 ✗ av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
281 ✗ goto fail;
282 }
283 ✗ s->max_frames_per_packet = s->max_payload_size / st->codecpar->block_align;
284 ✗ break;
285 ✗ case AV_CODEC_ID_AMR_NB:
286 case AV_CODEC_ID_AMR_WB:
287 ✗ s->max_frames_per_packet = 50;
288 ✗ if (st->codecpar->codec_id == AV_CODEC_ID_AMR_NB)
289 ✗ n = 31;
290 else
291 ✗ n = 61;
292 /* max_header_toc_size + the largest AMR payload must fit */
293 ✗ if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
294 ✗ av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
295 ✗ goto fail;
296 }
297 ✗ if (st->codecpar->ch_layout.nb_channels != 1) {
298 ✗ av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
299 ✗ goto fail;
300 }
301 ✗ break;
302 ✗ case AV_CODEC_ID_AAC:
303 ✗ s->max_frames_per_packet = 50;
304 ✗ break;
305 2 default:
306 2 break;
307 }
308
309 2 return 0;
310
311 ✗ fail:
312 ✗ av_freep(&s->buf);
313 ✗ return ret;
314 }
315
316 /* send an rtcp sender report packet */
317 2 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
318 {
319 2 RTPMuxContext *s = s1->priv_data;
320 uint32_t rtp_ts;
321
322 2 av_log(s1, AV_LOG_TRACE, "RTCP: %02x %"PRIx64" %"PRIx32"\n", s->payload_type, ntp_time, s->timestamp);
323
324 2 s->last_rtcp_ntp_time = ntp_time;
325 2 rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
326 2 s1->streams[0]->time_base) + s->base_timestamp;
327 2 avio_w8(s1->pb, RTP_VERSION << 6);
328 2 avio_w8(s1->pb, RTCP_SR);
329 2 avio_wb16(s1->pb, 6); /* length in words - 1 */
330 2 avio_wb32(s1->pb, s->ssrc);
331 2 avio_wb32(s1->pb, ntp_time / 1000000);
332 2 avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
333 2 avio_wb32(s1->pb, rtp_ts);
334 2 avio_wb32(s1->pb, s->packet_count);
335 2 avio_wb32(s1->pb, s->octet_count);
336
337
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 2 if (s->cname) {
338 ✗ int len = FFMIN(strlen(s->cname), 255);
339 ✗ avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
340 ✗ avio_w8(s1->pb, RTCP_SDES);
341 ✗ avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
342
343 ✗ avio_wb32(s1->pb, s->ssrc);
344 ✗ avio_w8(s1->pb, 0x01); /* CNAME */
345 ✗ avio_w8(s1->pb, len);
346 ✗ avio_write(s1->pb, s->cname, len);
347 ✗ avio_w8(s1->pb, 0); /* END */
348 ✗ for (len = (7 + len) % 4; len % 4; len++)
349 ✗ avio_w8(s1->pb, 0);
350 }
351
352
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 2 if (bye) {
353 ✗ avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
354 ✗ avio_w8(s1->pb, RTCP_BYE);
355 ✗ avio_wb16(s1->pb, 1); /* length in words - 1 */
356 ✗ avio_wb32(s1->pb, s->ssrc);
357 }
358
359 2 avio_flush(s1->pb);
360 2 }
361
362 /* send an rtp packet. sequence number is incremented, but the caller
363 must update the timestamp itself */
364 263 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
365 {
366 263 RTPMuxContext *s = s1->priv_data;
367
368 263 av_log(s1, AV_LOG_TRACE, "rtp_send_data size=%d\n", len);
369
370 /* build the RTP header */
371 263 avio_w8(s1->pb, RTP_VERSION << 6);
372 263 avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
373 263 avio_wb16(s1->pb, s->seq);
374 263 avio_wb32(s1->pb, s->timestamp);
375 263 avio_wb32(s1->pb, s->ssrc);
376
377 263 avio_write(s1->pb, buf1, len);
378 263 avio_flush(s1->pb);
379
380 263 s->seq = (s->seq + 1) & 0xffff;
381 263 s->octet_count += len;
382 263 s->packet_count++;
383 263 }
384
385 /* send an integer number of samples and compute time stamp and fill
386 the rtp send buffer before sending. */
387 11 static int rtp_send_samples(AVFormatContext *s1,
388 const uint8_t *buf1, int size, int sample_size_bits)
389 {
390 11 RTPMuxContext *s = s1->priv_data;
391 int len, max_packet_size, n;
392 /* Calculate the number of bytes to get samples aligned on a byte border */
393 11 int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
394
395 11 max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
396 /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
397
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 11 if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
398 ✗ return AVERROR(EINVAL);
399 11 n = 0;
400
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 44 while (size > 0) {
401 33 s->buf_ptr = s->buf;
402 33 len = FFMIN(max_packet_size, size);
403
404 /* copy data */
405 33 memcpy(s->buf_ptr, buf1, len);
406 33 s->buf_ptr += len;
407 33 buf1 += len;
408 33 size -= len;
409 33 s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
410 33 ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
411 33 n += (s->buf_ptr - s->buf);
412 }
413 11 return 0;
414 }
415
416 ✗ static void rtp_send_mpegaudio(AVFormatContext *s1,
417 const uint8_t *buf1, int size)
418 {
419 ✗ RTPMuxContext *s = s1->priv_data;
420 int len, count, max_packet_size;
421
422 ✗ max_packet_size = s->max_payload_size;
423
424 /* test if we must flush because not enough space */
425 ✗ len = (s->buf_ptr - s->buf);
426 ✗ if ((len + size) > max_packet_size) {
427 ✗ if (len > 4) {
428 ✗ ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
429 ✗ s->buf_ptr = s->buf + 4;
430 }
431 }
432 ✗ if (s->buf_ptr == s->buf + 4) {
433 ✗ s->timestamp = s->cur_timestamp;
434 }
435
436 /* add the packet */
437 ✗ if (size > max_packet_size) {
438 /* big packet: fragment */
439 ✗ count = 0;
440 ✗ while (size > 0) {
441 ✗ len = max_packet_size - 4;
442 ✗ if (len > size)
443 ✗ len = size;
444 /* build fragmented packet */
445 ✗ s->buf[0] = 0;
446 ✗ s->buf[1] = 0;
447 ✗ s->buf[2] = count >> 8;
448 ✗ s->buf[3] = count;
449 ✗ memcpy(s->buf + 4, buf1, len);
450 ✗ ff_rtp_send_data(s1, s->buf, len + 4, 0);
451 ✗ size -= len;
452 ✗ buf1 += len;
453 ✗ count += len;
454 }
455 } else {
456 ✗ if (s->buf_ptr == s->buf + 4) {
457 /* no fragmentation possible */
458 ✗ s->buf[0] = 0;
459 ✗ s->buf[1] = 0;
460 ✗ s->buf[2] = 0;
461 ✗ s->buf[3] = 0;
462 }
463 ✗ memcpy(s->buf_ptr, buf1, size);
464 ✗ s->buf_ptr += size;
465 }
466 ✗ }
467
468 25 static void rtp_send_raw(AVFormatContext *s1,
469 const uint8_t *buf1, int size)
470 {
471 25 RTPMuxContext *s = s1->priv_data;
472 int len, max_packet_size;
473
474 25 max_packet_size = s->max_payload_size;
475
476
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 255 while (size > 0) {
477 230 len = max_packet_size;
478
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 230 if (len > size)
479 25 len = size;
480
481 230 s->timestamp = s->cur_timestamp;
482 230 ff_rtp_send_data(s1, buf1, len, (len == size));
483
484 230 buf1 += len;
485 230 size -= len;
486 }
487 25 }
488
489 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
490 ✗ static void rtp_send_mpegts_raw(AVFormatContext *s1,
491 const uint8_t *buf1, int size)
492 {
493 ✗ RTPMuxContext *s = s1->priv_data;
494 int len, out_len;
495
496 ✗ s->timestamp = s->cur_timestamp;
497 ✗ while (size >= TS_PACKET_SIZE) {
498 ✗ len = s->max_payload_size - (s->buf_ptr - s->buf);
499 ✗ if (len > size)
500 ✗ len = size;
501 ✗ memcpy(s->buf_ptr, buf1, len);
502 ✗ buf1 += len;
503 ✗ size -= len;
504 ✗ s->buf_ptr += len;
505
506 ✗ out_len = s->buf_ptr - s->buf;
507 ✗ if (out_len >= s->max_payload_size) {
508 ✗ ff_rtp_send_data(s1, s->buf, out_len, 0);
509 ✗ s->buf_ptr = s->buf;
510 }
511 }
512 ✗ }
513
514 ✗ static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
515 {
516 ✗ RTPMuxContext *s = s1->priv_data;
517 ✗ AVStream *st = s1->streams[0];
518 ✗ int frame_duration = av_get_audio_frame_duration2(st->codecpar, 0);
519 ✗ int frame_size = st->codecpar->block_align;
520 ✗ int frames = size / frame_size;
521
522 ✗ while (frames > 0) {
523 ✗ if (s->num_frames > 0 &&
524 ✗ av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
525 ✗ s1->max_delay, AV_TIME_BASE_Q) >= 0) {
526 ✗ ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
527 ✗ s->num_frames = 0;
528 }
529
530 ✗ if (!s->num_frames) {
531 ✗ s->buf_ptr = s->buf;
532 ✗ s->timestamp = s->cur_timestamp;
533 }
534 ✗ memcpy(s->buf_ptr, buf, frame_size);
535 ✗ frames--;
536 ✗ s->num_frames++;
537 ✗ s->buf_ptr += frame_size;
538 ✗ buf += frame_size;
539 ✗ s->cur_timestamp += frame_duration;
540
541 ✗ if (s->num_frames == s->max_frames_per_packet) {
542 ✗ ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
543 ✗ s->num_frames = 0;
544 }
545 }
546 ✗ return 0;
547 }
548
549 36 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
550 {
551 36 RTPMuxContext *s = s1->priv_data;
552 36 AVStream *st = s1->streams[0];
553 int rtcp_bytes;
554 36 int size= pkt->size;
555
556 36 av_log(s1, AV_LOG_TRACE, "%d: write len=%d\n", pkt->stream_index, size);
557
558 36 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
559 RTCP_TX_RATIO_DEN;
560
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561
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562
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 2 !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
563 2 rtcp_send_sr(s1, ff_ntp_time(), 0);
564 2 s->last_octet_count = s->octet_count;
565 2 s->first_packet = 0;
566 }
567 36 s->cur_timestamp = s->base_timestamp + pkt->pts;
568
569
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 36 switch(st->codecpar->codec_id) {
570 11 case AV_CODEC_ID_PCM_MULAW:
571 case AV_CODEC_ID_PCM_ALAW:
572 case AV_CODEC_ID_PCM_U8:
573 case AV_CODEC_ID_PCM_S8:
574 11 return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->ch_layout.nb_channels);
575 ✗ case AV_CODEC_ID_PCM_U16BE:
576 case AV_CODEC_ID_PCM_U16LE:
577 case AV_CODEC_ID_PCM_S16BE:
578 case AV_CODEC_ID_PCM_S16LE:
579 ✗ return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->ch_layout.nb_channels);
580 ✗ case AV_CODEC_ID_PCM_S24BE:
581 ✗ return rtp_send_samples(s1, pkt->data, size, 24 * st->codecpar->ch_layout.nb_channels);
582 ✗ case AV_CODEC_ID_ADPCM_G722:
583 /* The actual sample size is half a byte per sample, but since the
584 * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
585 * the correct parameter for send_samples_bits is 8 bits per stream
586 * clock. */
587 ✗ return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->ch_layout.nb_channels);
588 ✗ case AV_CODEC_ID_ADPCM_G726:
589 case AV_CODEC_ID_ADPCM_G726LE:
590 ✗ return rtp_send_samples(s1, pkt->data, size,
591 ✗ st->codecpar->bits_per_coded_sample * st->codecpar->ch_layout.nb_channels);
592 ✗ case AV_CODEC_ID_MP2:
593 case AV_CODEC_ID_MP3:
594 ✗ rtp_send_mpegaudio(s1, pkt->data, size);
595 ✗ break;
596 ✗ case AV_CODEC_ID_MPEG1VIDEO:
597 case AV_CODEC_ID_MPEG2VIDEO:
598 ✗ ff_rtp_send_mpegvideo(s1, pkt->data, size);
599 ✗ break;
600 ✗ case AV_CODEC_ID_AAC:
601 ✗ if (s->flags & FF_RTP_FLAG_MP4A_LATM)
602 ✗ ff_rtp_send_latm(s1, pkt->data, size);
603 else
604 ✗ ff_rtp_send_aac(s1, pkt->data, size);
605 ✗ break;
606 ✗ case AV_CODEC_ID_AMR_NB:
607 case AV_CODEC_ID_AMR_WB:
608 ✗ ff_rtp_send_amr(s1, pkt->data, size);
609 ✗ break;
610 ✗ case AV_CODEC_ID_AV1:
611 ✗ ff_rtp_send_av1(s1, pkt->data, size, (pkt->flags & AV_PKT_FLAG_KEY) ? 1 : 0);
612 ✗ break;
613 ✗ case AV_CODEC_ID_MPEG2TS:
614 ✗ rtp_send_mpegts_raw(s1, pkt->data, size);
615 ✗ break;
616 ✗ case AV_CODEC_ID_DIRAC:
617 ✗ ff_rtp_send_vc2hq(s1, pkt->data, size, st->codecpar->field_order != AV_FIELD_PROGRESSIVE ? 1 : 0);
618 ✗ break;
619 ✗ case AV_CODEC_ID_H264:
620 ✗ ff_rtp_send_h264_hevc(s1, pkt->data, size);
621 ✗ break;
622 ✗ case AV_CODEC_ID_H261:
623 ✗ ff_rtp_send_h261(s1, pkt->data, size);
624 ✗ break;
625 ✗ case AV_CODEC_ID_H263:
626 ✗ if (s->flags & FF_RTP_FLAG_RFC2190) {
627 size_t mb_info_size;
628 const uint8_t *mb_info =
629 ✗ av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
630 &mb_info_size);
631 ✗ ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
632 ✗ break;
633 }
634 av_fallthrough;
635 case AV_CODEC_ID_H263P:
636 ✗ ff_rtp_send_h263(s1, pkt->data, size);
637 ✗ break;
638 ✗ case AV_CODEC_ID_HEVC:
639 ✗ ff_rtp_send_h264_hevc(s1, pkt->data, size);
640 ✗ break;
641 ✗ case AV_CODEC_ID_VORBIS:
642 case AV_CODEC_ID_THEORA:
643 ✗ ff_rtp_send_xiph(s1, pkt->data, size);
644 ✗ break;
645 ✗ case AV_CODEC_ID_VP8:
646 ✗ ff_rtp_send_vp8(s1, pkt->data, size);
647 ✗ break;
648 ✗ case AV_CODEC_ID_VP9:
649 ✗ ff_rtp_send_vp9(s1, pkt->data, size);
650 ✗ break;
651 ✗ case AV_CODEC_ID_ILBC:
652 ✗ rtp_send_ilbc(s1, pkt->data, size);
653 ✗ break;
654 ✗ case AV_CODEC_ID_MJPEG:
655 ✗ ff_rtp_send_jpeg(s1, pkt->data, size);
656 ✗ break;
657 ✗ case AV_CODEC_ID_BITPACKED:
658 case AV_CODEC_ID_RAWVIDEO: {
659 ✗ int interlaced = st->codecpar->field_order != AV_FIELD_PROGRESSIVE;
660
661 ✗ ff_rtp_send_raw_rfc4175(s1, pkt->data, size, interlaced, 0);
662 ✗ if (interlaced)
663 ✗ ff_rtp_send_raw_rfc4175(s1, pkt->data, size, interlaced, 1);
664 ✗ break;
665 }
666 ✗ case AV_CODEC_ID_OPUS:
667 ✗ if (size > s->max_payload_size) {
668 ✗ av_log(s1, AV_LOG_ERROR,
669 "Packet size %d too large for max RTP payload size %d\n",
670 size, s->max_payload_size);
671 ✗ return AVERROR(EINVAL);
672 }
673 av_fallthrough;
674 default:
675 /* better than nothing : send the codec raw data */
676 25 rtp_send_raw(s1, pkt->data, size);
677 25 break;
678 }
679 25 return 0;
680 }
681
682 2 static int rtp_write_trailer(AVFormatContext *s1)
683 {
684 2 RTPMuxContext *s = s1->priv_data;
685
686 /* If the caller closes and recreates ->pb, this might actually
687 * be NULL here even if it was successfully allocated at the start. */
688
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 2 if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
689 ✗ rtcp_send_sr(s1, ff_ntp_time(), 1);
690
691 2 return 0;
692 }
693
694 2 static void rtp_deinit(AVFormatContext *s1)
695 {
696 2 RTPMuxContext *s = s1->priv_data;
697
698 2 av_freep(&s->buf);
699 2 }
700
701 const FFOutputFormat ff_rtp_muxer = {
702 .p.name = "rtp",
703 .p.long_name = NULL_IF_CONFIG_SMALL("RTP output"),
704 .priv_data_size = sizeof(RTPMuxContext),
705 .p.audio_codec = AV_CODEC_ID_PCM_MULAW,
706 .p.video_codec = AV_CODEC_ID_MPEG4,
707 .write_header = rtp_write_header,
708 .write_packet = rtp_write_packet,
709 .write_trailer = rtp_write_trailer,
710 .deinit = rtp_deinit,
711 .p.priv_class = &rtp_muxer_class,
712 .p.flags = AVFMT_NODIMENSIONS | AVFMT_TS_NONSTRICT,
713 };
714