| 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 | |
| 31 | static 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 | |
| 40 | static 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 | |
| 49 | static int is_supported(enum AVCodecID id) |
| 50 | { |
| 51 | switch(id) { |
| 52 | case AV_CODEC_ID_H261: |
| 53 | case AV_CODEC_ID_H263: |
| 54 | case AV_CODEC_ID_H263P: |
| 55 | case AV_CODEC_ID_H264: |
| 56 | case AV_CODEC_ID_MPEG1VIDEO: |
| 57 | case AV_CODEC_ID_MPEG2VIDEO: |
| 58 | case AV_CODEC_ID_MPEG4: |
| 59 | case AV_CODEC_ID_AAC: |
| 60 | case AV_CODEC_ID_MP2: |
| 61 | case AV_CODEC_ID_MP3: |
| 62 | case AV_CODEC_ID_PCM_ALAW: |
| 63 | case AV_CODEC_ID_PCM_MULAW: |
| 64 | case AV_CODEC_ID_PCM_S8: |
| 65 | case AV_CODEC_ID_PCM_S16BE: |
| 66 | case AV_CODEC_ID_PCM_S16LE: |
| 67 | case AV_CODEC_ID_PCM_U16BE: |
| 68 | case AV_CODEC_ID_PCM_U16LE: |
| 69 | case AV_CODEC_ID_PCM_U8: |
| 70 | case AV_CODEC_ID_MPEG2TS: |
| 71 | case AV_CODEC_ID_AMR_NB: |
| 72 | case AV_CODEC_ID_AMR_WB: |
| 73 | case AV_CODEC_ID_VORBIS: |
| 74 | case AV_CODEC_ID_THEORA: |
| 75 | case AV_CODEC_ID_VP8: |
| 76 | case AV_CODEC_ID_ADPCM_G722: |
| 77 | case AV_CODEC_ID_ADPCM_G726: |
| 78 | case AV_CODEC_ID_ILBC: |
| 79 | case AV_CODEC_ID_MJPEG: |
| 80 | case AV_CODEC_ID_SPEEX: |
| 81 | case AV_CODEC_ID_OPUS: |
| 82 | return 1; |
| 83 | default: |
| 84 | return 0; |
| 85 | } |
| 86 | } |
| 87 | |
| 88 | static int rtp_write_header(AVFormatContext *s1) |
| 89 | { |
| 90 | RTPMuxContext *s = s1->priv_data; |
| 91 | int n; |
| 92 | AVStream *st; |
| 93 | |
| 94 | if (s1->nb_streams != 1) { |
| 95 | av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n"); |
| 96 | return AVERROR(EINVAL); |
| 97 | } |
| 98 | st = s1->streams[0]; |
| 99 | if (!is_supported(st->codec->codec_id)) { |
| 100 | av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id)); |
| 101 | |
| 102 | return -1; |
| 103 | } |
| 104 | |
| 105 | if (s->payload_type < 0) { |
| 106 | /* Re-validate non-dynamic payload types */ |
| 107 | if (st->id < RTP_PT_PRIVATE) |
| 108 | st->id = ff_rtp_get_payload_type(s1, st->codec, -1); |
| 109 | |
| 110 | s->payload_type = st->id; |
| 111 | } else { |
| 112 | /* private option takes priority */ |
| 113 | st->id = s->payload_type; |
| 114 | } |
| 115 | |
| 116 | s->base_timestamp = av_get_random_seed(); |
| 117 | s->timestamp = s->base_timestamp; |
| 118 | s->cur_timestamp = 0; |
| 119 | if (!s->ssrc) |
| 120 | s->ssrc = av_get_random_seed(); |
| 121 | s->first_packet = 1; |
| 122 | s->first_rtcp_ntp_time = ff_ntp_time(); |
| 123 | if (s1->start_time_realtime != 0 && s1->start_time_realtime != AV_NOPTS_VALUE) |
| 124 | /* Round the NTP time to whole milliseconds. */ |
| 125 | s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 + |
| 126 | NTP_OFFSET_US; |
| 127 | // Pick a random sequence start number, but in the lower end of the |
| 128 | // available range, so that any wraparound doesn't happen immediately. |
| 129 | // (Immediate wraparound would be an issue for SRTP.) |
| 130 | if (s->seq < 0) { |
| 131 | if (s1->flags & AVFMT_FLAG_BITEXACT) { |
| 132 | s->seq = 0; |
| 133 | } else |
| 134 | s->seq = av_get_random_seed() & 0x0fff; |
| 135 | } else |
| 136 | s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval |
| 137 | |
| 138 | if (s1->packet_size) { |
| 139 | if (s1->pb->max_packet_size) |
| 140 | s1->packet_size = FFMIN(s1->packet_size, |
| 141 | s1->pb->max_packet_size); |
| 142 | } else |
| 143 | s1->packet_size = s1->pb->max_packet_size; |
| 144 | if (s1->packet_size <= 12) { |
| 145 | av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size); |
| 146 | return AVERROR(EIO); |
| 147 | } |
| 148 | s->buf = av_malloc(s1->packet_size); |
| 149 | if (!s->buf) { |
| 150 | return AVERROR(ENOMEM); |
| 151 | } |
| 152 | s->max_payload_size = s1->packet_size - 12; |
| 153 | |
| 154 | s->max_frames_per_packet = 0; |
| 155 | if (s1->max_delay > 0) { |
| 156 | if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { |
| 157 | int frame_size = av_get_audio_frame_duration(st->codec, 0); |
| 158 | if (!frame_size) |
| 159 | frame_size = st->codec->frame_size; |
| 160 | if (frame_size == 0) { |
| 161 | av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n"); |
| 162 | } else { |
| 163 | s->max_frames_per_packet = |
| 164 | av_rescale_q_rnd(s1->max_delay, |
| 165 | AV_TIME_BASE_Q, |
| 166 | (AVRational){ frame_size, st->codec->sample_rate }, |
| 167 | AV_ROUND_DOWN); |
| 168 | } |
| 169 | } |
| 170 | if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) { |
| 171 | /* FIXME: We should round down here... */ |
| 172 | if (st->avg_frame_rate.num > 0 && st->avg_frame_rate.den > 0) { |
| 173 | s->max_frames_per_packet = av_rescale_q(s1->max_delay, |
| 174 | (AVRational){1, 1000000}, |
| 175 | av_inv_q(st->avg_frame_rate)); |
| 176 | } else |
| 177 | s->max_frames_per_packet = 1; |
| 178 | } |
| 179 | } |
| 180 | |
| 181 | avpriv_set_pts_info(st, 32, 1, 90000); |
| 182 | switch(st->codec->codec_id) { |
| 183 | case AV_CODEC_ID_MP2: |
| 184 | case AV_CODEC_ID_MP3: |
| 185 | s->buf_ptr = s->buf + 4; |
| 186 | break; |
| 187 | case AV_CODEC_ID_MPEG1VIDEO: |
| 188 | case AV_CODEC_ID_MPEG2VIDEO: |
| 189 | break; |
| 190 | case AV_CODEC_ID_MPEG2TS: |
| 191 | n = s->max_payload_size / TS_PACKET_SIZE; |
| 192 | if (n < 1) |
| 193 | n = 1; |
| 194 | s->max_payload_size = n * TS_PACKET_SIZE; |
| 195 | s->buf_ptr = s->buf; |
| 196 | break; |
| 197 | case AV_CODEC_ID_H264: |
| 198 | /* check for H.264 MP4 syntax */ |
| 199 | if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) { |
| 200 | s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1; |
| 201 | } |
| 202 | break; |
| 203 | case AV_CODEC_ID_VORBIS: |
| 204 | case AV_CODEC_ID_THEORA: |
| 205 | if (!s->max_frames_per_packet) s->max_frames_per_packet = 15; |
| 206 | s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15); |
| 207 | s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length |
| 208 | s->num_frames = 0; |
| 209 | goto defaultcase; |
| 210 | case AV_CODEC_ID_ADPCM_G722: |
| 211 | /* Due to a historical error, the clock rate for G722 in RTP is |
| 212 | * 8000, even if the sample rate is 16000. See RFC 3551. */ |
| 213 | avpriv_set_pts_info(st, 32, 1, 8000); |
| 214 | break; |
| 215 | case AV_CODEC_ID_OPUS: |
| 216 | if (st->codec->channels > 2) { |
| 217 | av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n"); |
| 218 | goto fail; |
| 219 | } |
| 220 | /* The opus RTP RFC says that all opus streams should use 48000 Hz |
| 221 | * as clock rate, since all opus sample rates can be expressed in |
| 222 | * this clock rate, and sample rate changes on the fly are supported. */ |
| 223 | avpriv_set_pts_info(st, 32, 1, 48000); |
| 224 | break; |
| 225 | case AV_CODEC_ID_ILBC: |
| 226 | if (st->codec->block_align != 38 && st->codec->block_align != 50) { |
| 227 | av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n"); |
| 228 | goto fail; |
| 229 | } |
| 230 | if (!s->max_frames_per_packet) |
| 231 | s->max_frames_per_packet = 1; |
| 232 | s->max_frames_per_packet = FFMIN(s->max_frames_per_packet, |
| 233 | s->max_payload_size / st->codec->block_align); |
| 234 | goto defaultcase; |
| 235 | case AV_CODEC_ID_AMR_NB: |
| 236 | case AV_CODEC_ID_AMR_WB: |
| 237 | if (!s->max_frames_per_packet) |
| 238 | s->max_frames_per_packet = 12; |
| 239 | if (st->codec->codec_id == AV_CODEC_ID_AMR_NB) |
| 240 | n = 31; |
| 241 | else |
| 242 | n = 61; |
| 243 | /* max_header_toc_size + the largest AMR payload must fit */ |
| 244 | if (1 + s->max_frames_per_packet + n > s->max_payload_size) { |
| 245 | av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n"); |
| 246 | goto fail; |
| 247 | } |
| 248 | if (st->codec->channels != 1) { |
| 249 | av_log(s1, AV_LOG_ERROR, "Only mono is supported\n"); |
| 250 | goto fail; |
| 251 | } |
| 252 | case AV_CODEC_ID_AAC: |
| 253 | s->num_frames = 0; |
| 254 | default: |
| 255 | defaultcase: |
| 256 | if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { |
| 257 | avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate); |
| 258 | } |
| 259 | s->buf_ptr = s->buf; |
| 260 | break; |
| 261 | } |
| 262 | |
| 263 | return 0; |
| 264 | |
| 265 | fail: |
| 266 | av_freep(&s->buf); |
| 267 | return AVERROR(EINVAL); |
| 268 | } |
| 269 | |
| 270 | /* send an rtcp sender report packet */ |
| 271 | static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye) |
| 272 | { |
| 273 | RTPMuxContext *s = s1->priv_data; |
| 274 | uint32_t rtp_ts; |
| 275 | |
| 276 | av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); |
| 277 | |
| 278 | s->last_rtcp_ntp_time = ntp_time; |
| 279 | rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000}, |
| 280 | s1->streams[0]->time_base) + s->base_timestamp; |
| 281 | avio_w8(s1->pb, RTP_VERSION << 6); |
| 282 | avio_w8(s1->pb, RTCP_SR); |
| 283 | avio_wb16(s1->pb, 6); /* length in words - 1 */ |
| 284 | avio_wb32(s1->pb, s->ssrc); |
| 285 | avio_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time)); |
| 286 | avio_wb32(s1->pb, rtp_ts); |
| 287 | avio_wb32(s1->pb, s->packet_count); |
| 288 | avio_wb32(s1->pb, s->octet_count); |
| 289 | |
| 290 | if (s->cname) { |
| 291 | int len = FFMIN(strlen(s->cname), 255); |
| 292 | avio_w8(s1->pb, (RTP_VERSION << 6) + 1); |
| 293 | avio_w8(s1->pb, RTCP_SDES); |
| 294 | avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */ |
| 295 | |
| 296 | avio_wb32(s1->pb, s->ssrc); |
| 297 | avio_w8(s1->pb, 0x01); /* CNAME */ |
| 298 | avio_w8(s1->pb, len); |
| 299 | avio_write(s1->pb, s->cname, len); |
| 300 | avio_w8(s1->pb, 0); /* END */ |
| 301 | for (len = (7 + len) % 4; len % 4; len++) |
| 302 | avio_w8(s1->pb, 0); |
| 303 | } |
| 304 | |
| 305 | if (bye) { |
| 306 | avio_w8(s1->pb, (RTP_VERSION << 6) | 1); |
| 307 | avio_w8(s1->pb, RTCP_BYE); |
| 308 | avio_wb16(s1->pb, 1); /* length in words - 1 */ |
| 309 | avio_wb32(s1->pb, s->ssrc); |
| 310 | } |
| 311 | |
| 312 | avio_flush(s1->pb); |
| 313 | } |
| 314 | |
| 315 | /* send an rtp packet. sequence number is incremented, but the caller |
| 316 | must update the timestamp itself */ |
| 317 | void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) |
| 318 | { |
| 319 | RTPMuxContext *s = s1->priv_data; |
| 320 | |
| 321 | av_dlog(s1, "rtp_send_data size=%d\n", len); |
| 322 | |
| 323 | /* build the RTP header */ |
| 324 | avio_w8(s1->pb, RTP_VERSION << 6); |
| 325 | avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); |
| 326 | avio_wb16(s1->pb, s->seq); |
| 327 | avio_wb32(s1->pb, s->timestamp); |
| 328 | avio_wb32(s1->pb, s->ssrc); |
| 329 | |
| 330 | avio_write(s1->pb, buf1, len); |
| 331 | avio_flush(s1->pb); |
| 332 | |
| 333 | s->seq = (s->seq + 1) & 0xffff; |
| 334 | s->octet_count += len; |
| 335 | s->packet_count++; |
| 336 | } |
| 337 | |
| 338 | /* send an integer number of samples and compute time stamp and fill |
| 339 | the rtp send buffer before sending. */ |
| 340 | static int rtp_send_samples(AVFormatContext *s1, |
| 341 | const uint8_t *buf1, int size, int sample_size_bits) |
| 342 | { |
| 343 | RTPMuxContext *s = s1->priv_data; |
| 344 | int len, max_packet_size, n; |
| 345 | /* Calculate the number of bytes to get samples aligned on a byte border */ |
| 346 | int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8); |
| 347 | |
| 348 | max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size; |
| 349 | /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */ |
| 350 | if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0) |
| 351 | return AVERROR(EINVAL); |
| 352 | n = 0; |
| 353 | while (size > 0) { |
| 354 | s->buf_ptr = s->buf; |
| 355 | len = FFMIN(max_packet_size, size); |
| 356 | |
| 357 | /* copy data */ |
| 358 | memcpy(s->buf_ptr, buf1, len); |
| 359 | s->buf_ptr += len; |
| 360 | buf1 += len; |
| 361 | size -= len; |
| 362 | s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits; |
| 363 | ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
| 364 | n += (s->buf_ptr - s->buf); |
| 365 | } |
| 366 | return 0; |
| 367 | } |
| 368 | |
| 369 | static void rtp_send_mpegaudio(AVFormatContext *s1, |
| 370 | const uint8_t *buf1, int size) |
| 371 | { |
| 372 | RTPMuxContext *s = s1->priv_data; |
| 373 | int len, count, max_packet_size; |
| 374 | |
| 375 | max_packet_size = s->max_payload_size; |
| 376 | |
| 377 | /* test if we must flush because not enough space */ |
| 378 | len = (s->buf_ptr - s->buf); |
| 379 | if ((len + size) > max_packet_size) { |
| 380 | if (len > 4) { |
| 381 | ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
| 382 | s->buf_ptr = s->buf + 4; |
| 383 | } |
| 384 | } |
| 385 | if (s->buf_ptr == s->buf + 4) { |
| 386 | s->timestamp = s->cur_timestamp; |
| 387 | } |
| 388 | |
| 389 | /* add the packet */ |
| 390 | if (size > max_packet_size) { |
| 391 | /* big packet: fragment */ |
| 392 | count = 0; |
| 393 | while (size > 0) { |
| 394 | len = max_packet_size - 4; |
| 395 | if (len > size) |
| 396 | len = size; |
| 397 | /* build fragmented packet */ |
| 398 | s->buf[0] = 0; |
| 399 | s->buf[1] = 0; |
| 400 | s->buf[2] = count >> 8; |
| 401 | s->buf[3] = count; |
| 402 | memcpy(s->buf + 4, buf1, len); |
| 403 | ff_rtp_send_data(s1, s->buf, len + 4, 0); |
| 404 | size -= len; |
| 405 | buf1 += len; |
| 406 | count += len; |
| 407 | } |
| 408 | } else { |
| 409 | if (s->buf_ptr == s->buf + 4) { |
| 410 | /* no fragmentation possible */ |
| 411 | s->buf[0] = 0; |
| 412 | s->buf[1] = 0; |
| 413 | s->buf[2] = 0; |
| 414 | s->buf[3] = 0; |
| 415 | } |
| 416 | memcpy(s->buf_ptr, buf1, size); |
| 417 | s->buf_ptr += size; |
| 418 | } |
| 419 | } |
| 420 | |
| 421 | static void rtp_send_raw(AVFormatContext *s1, |
| 422 | const uint8_t *buf1, int size) |
| 423 | { |
| 424 | RTPMuxContext *s = s1->priv_data; |
| 425 | int len, max_packet_size; |
| 426 | |
| 427 | max_packet_size = s->max_payload_size; |
| 428 | |
| 429 | while (size > 0) { |
| 430 | len = max_packet_size; |
| 431 | if (len > size) |
| 432 | len = size; |
| 433 | |
| 434 | s->timestamp = s->cur_timestamp; |
| 435 | ff_rtp_send_data(s1, buf1, len, (len == size)); |
| 436 | |
| 437 | buf1 += len; |
| 438 | size -= len; |
| 439 | } |
| 440 | } |
| 441 | |
| 442 | /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ |
| 443 | static void rtp_send_mpegts_raw(AVFormatContext *s1, |
| 444 | const uint8_t *buf1, int size) |
| 445 | { |
| 446 | RTPMuxContext *s = s1->priv_data; |
| 447 | int len, out_len; |
| 448 | |
| 449 | while (size >= TS_PACKET_SIZE) { |
| 450 | len = s->max_payload_size - (s->buf_ptr - s->buf); |
| 451 | if (len > size) |
| 452 | len = size; |
| 453 | memcpy(s->buf_ptr, buf1, len); |
| 454 | buf1 += len; |
| 455 | size -= len; |
| 456 | s->buf_ptr += len; |
| 457 | |
| 458 | out_len = s->buf_ptr - s->buf; |
| 459 | if (out_len >= s->max_payload_size) { |
| 460 | ff_rtp_send_data(s1, s->buf, out_len, 0); |
| 461 | s->buf_ptr = s->buf; |
| 462 | } |
| 463 | } |
| 464 | } |
| 465 | |
| 466 | static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size) |
| 467 | { |
| 468 | RTPMuxContext *s = s1->priv_data; |
| 469 | AVStream *st = s1->streams[0]; |
| 470 | int frame_duration = av_get_audio_frame_duration(st->codec, 0); |
| 471 | int frame_size = st->codec->block_align; |
| 472 | int frames = size / frame_size; |
| 473 | |
| 474 | while (frames > 0) { |
| 475 | int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames); |
| 476 | |
| 477 | if (!s->num_frames) { |
| 478 | s->buf_ptr = s->buf; |
| 479 | s->timestamp = s->cur_timestamp; |
| 480 | } |
| 481 | memcpy(s->buf_ptr, buf, n * frame_size); |
| 482 | frames -= n; |
| 483 | s->num_frames += n; |
| 484 | s->buf_ptr += n * frame_size; |
| 485 | buf += n * frame_size; |
| 486 | s->cur_timestamp += n * frame_duration; |
| 487 | |
| 488 | if (s->num_frames == s->max_frames_per_packet) { |
| 489 | ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1); |
| 490 | s->num_frames = 0; |
| 491 | } |
| 492 | } |
| 493 | return 0; |
| 494 | } |
| 495 | |
| 496 | static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) |
| 497 | { |
| 498 | RTPMuxContext *s = s1->priv_data; |
| 499 | AVStream *st = s1->streams[0]; |
| 500 | int rtcp_bytes; |
| 501 | int size= pkt->size; |
| 502 | |
| 503 | av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size); |
| 504 | |
| 505 | rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
| 506 | RTCP_TX_RATIO_DEN; |
| 507 | if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) && |
| 508 | (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) && |
| 509 | !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) { |
| 510 | rtcp_send_sr(s1, ff_ntp_time(), 0); |
| 511 | s->last_octet_count = s->octet_count; |
| 512 | s->first_packet = 0; |
| 513 | } |
| 514 | s->cur_timestamp = s->base_timestamp + pkt->pts; |
| 515 | |
| 516 | switch(st->codec->codec_id) { |
| 517 | case AV_CODEC_ID_PCM_MULAW: |
| 518 | case AV_CODEC_ID_PCM_ALAW: |
| 519 | case AV_CODEC_ID_PCM_U8: |
| 520 | case AV_CODEC_ID_PCM_S8: |
| 521 | return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels); |
| 522 | case AV_CODEC_ID_PCM_U16BE: |
| 523 | case AV_CODEC_ID_PCM_U16LE: |
| 524 | case AV_CODEC_ID_PCM_S16BE: |
| 525 | case AV_CODEC_ID_PCM_S16LE: |
| 526 | return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels); |
| 527 | case AV_CODEC_ID_ADPCM_G722: |
| 528 | /* The actual sample size is half a byte per sample, but since the |
| 529 | * stream clock rate is 8000 Hz while the sample rate is 16000 Hz, |
| 530 | * the correct parameter for send_samples_bits is 8 bits per stream |
| 531 | * clock. */ |
| 532 | return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels); |
| 533 | case AV_CODEC_ID_ADPCM_G726: |
| 534 | return rtp_send_samples(s1, pkt->data, size, |
| 535 | st->codec->bits_per_coded_sample * st->codec->channels); |
| 536 | case AV_CODEC_ID_MP2: |
| 537 | case AV_CODEC_ID_MP3: |
| 538 | rtp_send_mpegaudio(s1, pkt->data, size); |
| 539 | break; |
| 540 | case AV_CODEC_ID_MPEG1VIDEO: |
| 541 | case AV_CODEC_ID_MPEG2VIDEO: |
| 542 | ff_rtp_send_mpegvideo(s1, pkt->data, size); |
| 543 | break; |
| 544 | case AV_CODEC_ID_AAC: |
| 545 | if (s->flags & FF_RTP_FLAG_MP4A_LATM) |
| 546 | ff_rtp_send_latm(s1, pkt->data, size); |
| 547 | else |
| 548 | ff_rtp_send_aac(s1, pkt->data, size); |
| 549 | break; |
| 550 | case AV_CODEC_ID_AMR_NB: |
| 551 | case AV_CODEC_ID_AMR_WB: |
| 552 | ff_rtp_send_amr(s1, pkt->data, size); |
| 553 | break; |
| 554 | case AV_CODEC_ID_MPEG2TS: |
| 555 | rtp_send_mpegts_raw(s1, pkt->data, size); |
| 556 | break; |
| 557 | case AV_CODEC_ID_H264: |
| 558 | ff_rtp_send_h264(s1, pkt->data, size); |
| 559 | break; |
| 560 | case AV_CODEC_ID_H261: |
| 561 | ff_rtp_send_h261(s1, pkt->data, size); |
| 562 | break; |
| 563 | case AV_CODEC_ID_H263: |
| 564 | if (s->flags & FF_RTP_FLAG_RFC2190) { |
| 565 | int mb_info_size = 0; |
| 566 | const uint8_t *mb_info = |
| 567 | av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO, |
| 568 | &mb_info_size); |
| 569 | if (!mb_info) { |
| 570 | av_log(s1, AV_LOG_ERROR, "failed to allocate side data\n"); |
| 571 | return AVERROR(ENOMEM); |
| 572 | } |
| 573 | ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size); |
| 574 | break; |
| 575 | } |
| 576 | /* Fallthrough */ |
| 577 | case AV_CODEC_ID_H263P: |
| 578 | ff_rtp_send_h263(s1, pkt->data, size); |
| 579 | break; |
| 580 | case AV_CODEC_ID_VORBIS: |
| 581 | case AV_CODEC_ID_THEORA: |
| 582 | ff_rtp_send_xiph(s1, pkt->data, size); |
| 583 | break; |
| 584 | case AV_CODEC_ID_VP8: |
| 585 | ff_rtp_send_vp8(s1, pkt->data, size); |
| 586 | break; |
| 587 | case AV_CODEC_ID_ILBC: |
| 588 | rtp_send_ilbc(s1, pkt->data, size); |
| 589 | break; |
| 590 | case AV_CODEC_ID_MJPEG: |
| 591 | ff_rtp_send_jpeg(s1, pkt->data, size); |
| 592 | break; |
| 593 | case AV_CODEC_ID_OPUS: |
| 594 | if (size > s->max_payload_size) { |
| 595 | av_log(s1, AV_LOG_ERROR, |
| 596 | "Packet size %d too large for max RTP payload size %d\n", |
| 597 | size, s->max_payload_size); |
| 598 | return AVERROR(EINVAL); |
| 599 | } |
| 600 | /* Intentional fallthrough */ |
| 601 | default: |
| 602 | /* better than nothing : send the codec raw data */ |
| 603 | rtp_send_raw(s1, pkt->data, size); |
| 604 | break; |
| 605 | } |
| 606 | return 0; |
| 607 | } |
| 608 | |
| 609 | static int rtp_write_trailer(AVFormatContext *s1) |
| 610 | { |
| 611 | RTPMuxContext *s = s1->priv_data; |
| 612 | |
| 613 | /* If the caller closes and recreates ->pb, this might actually |
| 614 | * be NULL here even if it was successfully allocated at the start. */ |
| 615 | if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE)) |
| 616 | rtcp_send_sr(s1, ff_ntp_time(), 1); |
| 617 | av_freep(&s->buf); |
| 618 | |
| 619 | return 0; |
| 620 | } |
| 621 | |
| 622 | AVOutputFormat ff_rtp_muxer = { |
| 623 | .name = "rtp", |
| 624 | .long_name = NULL_IF_CONFIG_SMALL("RTP output"), |
| 625 | .priv_data_size = sizeof(RTPMuxContext), |
| 626 | .audio_codec = AV_CODEC_ID_PCM_MULAW, |
| 627 | .video_codec = AV_CODEC_ID_MPEG4, |
| 628 | .write_header = rtp_write_header, |
| 629 | .write_packet = rtp_write_packet, |
| 630 | .write_trailer = rtp_write_trailer, |
| 631 | .priv_class = &rtp_muxer_class, |
| 632 | }; |