libavformat/rtpenc.c

   1 /*
   2  * RTP output format
   3  * Copyright (c) 2002 Fabrice Bellard
   4  *
   5  * This file is part of Libav.
   6  *
   7  * Libav 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  * Libav 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 Libav; 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 //#define DEBUG
  32
  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     { NULL },
  37 };
  38
  39 static const AVClass rtp_muxer_class = {
  40     .class_name = "RTP muxer",
  41     .item_name  = av_default_item_name,
  42     .option     = options,
  43     .version    = LIBAVUTIL_VERSION_INT,
  44 };
  45
  46 #define RTCP_SR_SIZE 28
  47
  48 static int is_supported(enum CodecID id)
  49 {
  50     switch(id) {
  51     case CODEC_ID_H263:
  52     case CODEC_ID_H263P:
  53     case CODEC_ID_H264:
  54     case CODEC_ID_MPEG1VIDEO:
  55     case CODEC_ID_MPEG2VIDEO:
  56     case CODEC_ID_MPEG4:
  57     case CODEC_ID_AAC:
  58     case CODEC_ID_MP2:
  59     case CODEC_ID_MP3:
  60     case CODEC_ID_PCM_ALAW:
  61     case CODEC_ID_PCM_MULAW:
  62     case CODEC_ID_PCM_S8:
  63     case CODEC_ID_PCM_S16BE:
  64     case CODEC_ID_PCM_S16LE:
  65     case CODEC_ID_PCM_U16BE:
  66     case CODEC_ID_PCM_U16LE:
  67     case CODEC_ID_PCM_U8:
  68     case CODEC_ID_MPEG2TS:
  69     case CODEC_ID_AMR_NB:
  70     case CODEC_ID_AMR_WB:
  71     case CODEC_ID_VORBIS:
  72     case CODEC_ID_THEORA:
  73     case CODEC_ID_VP8:
  74     case CODEC_ID_ADPCM_G722:
  75         return 1;
  76     default:
  77         return 0;
  78     }
  79 }
  80
  81 static int rtp_write_header(AVFormatContext *s1)
  82 {
  83     RTPMuxContext *s = s1->priv_data;
  84     int max_packet_size, n;
  85     AVStream *st;
  86
  87     if (s1->nb_streams != 1)
  88         return -1;
  89     st = s1->streams[0];
  90     if (!is_supported(st->codec->codec_id)) {
  91         av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
  92
  93         return -1;
  94     }
  95
  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();
 102     s->first_packet = 1;
 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 +
 107                                  NTP_OFFSET_US;
 108
 109     max_packet_size = s1->pb->max_packet_size;
 110     if (max_packet_size <= 12)
 111         return AVERROR(EIO);
 112     s->buf = av_malloc(max_packet_size);
 113     if (s->buf == NULL) {
 114         return AVERROR(ENOMEM);
 115     }
 116     s->max_payload_size = max_packet_size - 12;
 117
 118     s->max_frames_per_packet = 0;
 119     if (s1->max_delay) {
 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");
 123             } else {
 124                 s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * (int64_t)st->codec->frame_size, AV_ROUND_DOWN);
 125             }
 126         }
 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);
 130         }
 131     }
 132
 133     avpriv_set_pts_info(st, 32, 1, 90000);
 134     switch(st->codec->codec_id) {
 135     case CODEC_ID_MP2:
 136     case CODEC_ID_MP3:
 137         s->buf_ptr = s->buf + 4;
 138         break;
 139     case CODEC_ID_MPEG1VIDEO:
 140     case CODEC_ID_MPEG2VIDEO:
 141         break;
 142     case CODEC_ID_MPEG2TS:
 143         n = s->max_payload_size / TS_PACKET_SIZE;
 144         if (n < 1)
 145             n = 1;
 146         s->max_payload_size = n * TS_PACKET_SIZE;
 147         s->buf_ptr = s->buf;
 148         break;
 149     case CODEC_ID_H264:
 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;
 153         }
 154         break;
 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
 160         s->num_frames = 0;
 161         goto defaultcase;
 162     case CODEC_ID_VP8:
 163         av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
 164                                  "incompatible with the latest spec drafts.\n");
 165         break;
 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         avpriv_set_pts_info(st, 32, 1, 8000);
 170         break;
 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)
 176             n = 31;
 177         else
 178             n = 61;
 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");
 182             return -1;
 183         }
 184         if (st->codec->channels != 1) {
 185             av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
 186             return -1;
 187         }
 188     case CODEC_ID_AAC:
 189         s->num_frames = 0;
 190     default:
 191 defaultcase:
 192         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
 193             avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
 194         }
 195         s->buf_ptr = s->buf;
 196         break;
 197     }
 198
 199     return 0;
 200 }
 201
 202 /* send an rtcp sender report packet */
 203 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
 204 {
 205     RTPMuxContext *s = s1->priv_data;
 206     uint32_t rtp_ts;
 207
 208     av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
 209
 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);
 222     avio_flush(s1->pb);
 223 }
 224
 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)
 228 {
 229     RTPMuxContext *s = s1->priv_data;
 230
 231     av_dlog(s1, "rtp_send_data size=%d\n", len);
 232
 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);
 239
 240     avio_write(s1->pb, buf1, len);
 241     avio_flush(s1->pb);
 242
 243     s->seq++;
 244     s->octet_count += len;
 245     s->packet_count++;
 246 }
 247
 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)
 252 {
 253     RTPMuxContext *s = s1->priv_data;
 254     int len, max_packet_size, n;
 255
 256     max_packet_size = (s->max_payload_size / sample_size) * sample_size;
 257     /* not needed, but who nows */
 258     if ((size % sample_size) != 0)
 259         av_abort();
 260     n = 0;
 261     while (size > 0) {
 262         s->buf_ptr = s->buf;
 263         len = FFMIN(max_packet_size, size);
 264
 265         /* copy data */
 266         memcpy(s->buf_ptr, buf1, len);
 267         s->buf_ptr += len;
 268         buf1 += len;
 269         size -= 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);
 273     }
 274 }
 275
 276 static void rtp_send_mpegaudio(AVFormatContext *s1,
 277                                const uint8_t *buf1, int size)
 278 {
 279     RTPMuxContext *s = s1->priv_data;
 280     int len, count, max_packet_size;
 281
 282     max_packet_size = s->max_payload_size;
 283
 284     /* test if we must flush because not enough space */
 285     len = (s->buf_ptr - s->buf);
 286     if ((len + size) > max_packet_size) {
 287         if (len > 4) {
 288             ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
 289             s->buf_ptr = s->buf + 4;
 290         }
 291     }
 292     if (s->buf_ptr == s->buf + 4) {
 293         s->timestamp = s->cur_timestamp;
 294     }
 295
 296     /* add the packet */
 297     if (size > max_packet_size) {
 298         /* big packet: fragment */
 299         count = 0;
 300         while (size > 0) {
 301             len = max_packet_size - 4;
 302             if (len > size)
 303                 len = size;
 304             /* build fragmented packet */
 305             s->buf[0] = 0;
 306             s->buf[1] = 0;
 307             s->buf[2] = count >> 8;
 308             s->buf[3] = count;
 309             memcpy(s->buf + 4, buf1, len);
 310             ff_rtp_send_data(s1, s->buf, len + 4, 0);
 311             size -= len;
 312             buf1 += len;
 313             count += len;
 314         }
 315     } else {
 316         if (s->buf_ptr == s->buf + 4) {
 317             /* no fragmentation possible */
 318             s->buf[0] = 0;
 319             s->buf[1] = 0;
 320             s->buf[2] = 0;
 321             s->buf[3] = 0;
 322         }
 323         memcpy(s->buf_ptr, buf1, size);
 324         s->buf_ptr += size;
 325     }
 326 }
 327
 328 static void rtp_send_raw(AVFormatContext *s1,
 329                          const uint8_t *buf1, int size)
 330 {
 331     RTPMuxContext *s = s1->priv_data;
 332     int len, max_packet_size;
 333
 334     max_packet_size = s->max_payload_size;
 335
 336     while (size > 0) {
 337         len = max_packet_size;
 338         if (len > size)
 339             len = size;
 340
 341         s->timestamp = s->cur_timestamp;
 342         ff_rtp_send_data(s1, buf1, len, (len == size));
 343
 344         buf1 += len;
 345         size -= len;
 346     }
 347 }
 348
 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)
 352 {
 353     RTPMuxContext *s = s1->priv_data;
 354     int len, out_len;
 355
 356     while (size >= TS_PACKET_SIZE) {
 357         len = s->max_payload_size - (s->buf_ptr - s->buf);
 358         if (len > size)
 359             len = size;
 360         memcpy(s->buf_ptr, buf1, len);
 361         buf1 += len;
 362         size -= len;
 363         s->buf_ptr += len;
 364
 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);
 368             s->buf_ptr = s->buf;
 369         }
 370     }
 371 }
 372
 373 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
 374 {
 375     RTPMuxContext *s = s1->priv_data;
 376     AVStream *st = s1->streams[0];
 377     int rtcp_bytes;
 378     int size= pkt->size;
 379
 380     av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
 381
 382     rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
 383         RTCP_TX_RATIO_DEN;
 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;
 388         s->first_packet = 0;
 389     }
 390     s->cur_timestamp = s->base_timestamp + pkt->pts;
 391
 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);
 398         break;
 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);
 404         break;
 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);
 410         break;
 411     case CODEC_ID_MP2:
 412     case CODEC_ID_MP3:
 413         rtp_send_mpegaudio(s1, pkt->data, size);
 414         break;
 415     case CODEC_ID_MPEG1VIDEO:
 416     case CODEC_ID_MPEG2VIDEO:
 417         ff_rtp_send_mpegvideo(s1, pkt->data, size);
 418         break;
 419     case CODEC_ID_AAC:
 420         if (s->flags & FF_RTP_FLAG_MP4A_LATM)
 421             ff_rtp_send_latm(s1, pkt->data, size);
 422         else
 423             ff_rtp_send_aac(s1, pkt->data, size);
 424         break;
 425     case CODEC_ID_AMR_NB:
 426     case CODEC_ID_AMR_WB:
 427         ff_rtp_send_amr(s1, pkt->data, size);
 428         break;
 429     case CODEC_ID_MPEG2TS:
 430         rtp_send_mpegts_raw(s1, pkt->data, size);
 431         break;
 432     case CODEC_ID_H264:
 433         ff_rtp_send_h264(s1, pkt->data, size);
 434         break;
 435     case CODEC_ID_H263:
 436     case CODEC_ID_H263P:
 437         ff_rtp_send_h263(s1, pkt->data, size);
 438         break;
 439     case CODEC_ID_VORBIS:
 440     case CODEC_ID_THEORA:
 441         ff_rtp_send_xiph(s1, pkt->data, size);
 442         break;
 443     case CODEC_ID_VP8:
 444         ff_rtp_send_vp8(s1, pkt->data, size);
 445         break;
 446     default:
 447         /* better than nothing : send the codec raw data */
 448         rtp_send_raw(s1, pkt->data, size);
 449         break;
 450     }
 451     return 0;
 452 }
 453
 454 static int rtp_write_trailer(AVFormatContext *s1)
 455 {
 456     RTPMuxContext *s = s1->priv_data;
 457
 458     av_freep(&s->buf);
 459
 460     return 0;
 461 }
 462
 463 AVOutputFormat ff_rtp_muxer = {
 464     .name              = "rtp",
 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,
 473 };