libavformat/rtpenc.c

   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 "libavcodec/get_bits.h"
  23 #include "avformat.h"
  24 #include "mpegts.h"
  25
  26 #include <unistd.h>
  27 #include "network.h"
  28
  29 #include "rtpenc.h"
  30
  31 //#define DEBUG
  32
  33 #define RTCP_SR_SIZE 28
  34 #define NTP_OFFSET 2208988800ULL
  35 #define NTP_OFFSET_US (NTP_OFFSET * 1000000ULL)
  36
  37 static uint64_t ntp_time(void)
  38 {
  39   return (av_gettime() / 1000) * 1000 + NTP_OFFSET_US;
  40 }
  41
  42 static int is_supported(enum CodecID id)
  43 {
  44     switch(id) {
  45     case CODEC_ID_H263:
  46     case CODEC_ID_H263P:
  47     case CODEC_ID_H264:
  48     case CODEC_ID_MPEG1VIDEO:
  49     case CODEC_ID_MPEG2VIDEO:
  50     case CODEC_ID_MPEG4:
  51     case CODEC_ID_AAC:
  52     case CODEC_ID_MP2:
  53     case CODEC_ID_MP3:
  54     case CODEC_ID_PCM_ALAW:
  55     case CODEC_ID_PCM_MULAW:
  56     case CODEC_ID_PCM_S8:
  57     case CODEC_ID_PCM_S16BE:
  58     case CODEC_ID_PCM_S16LE:
  59     case CODEC_ID_PCM_U16BE:
  60     case CODEC_ID_PCM_U16LE:
  61     case CODEC_ID_PCM_U8:
  62     case CODEC_ID_MPEG2TS:
  63     case CODEC_ID_AMR_NB:
  64     case CODEC_ID_AMR_WB:
  65         return 1;
  66     default:
  67         return 0;
  68     }
  69 }
  70
  71 static int rtp_write_header(AVFormatContext *s1)
  72 {
  73     RTPMuxContext *s = s1->priv_data;
  74     int payload_type, max_packet_size, n;
  75     AVStream *st;
  76
  77     if (s1->nb_streams != 1)
  78         return -1;
  79     st = s1->streams[0];
  80     if (!is_supported(st->codec->codec_id)) {
  81         av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
  82
  83         return -1;
  84     }
  85
  86     payload_type = ff_rtp_get_payload_type(st->codec);
  87     if (payload_type < 0)
  88         payload_type = RTP_PT_PRIVATE; /* private payload type */
  89     s->payload_type = payload_type;
  90
  91 // following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
  92     s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
  93     s->timestamp = s->base_timestamp;
  94     s->cur_timestamp = 0;
  95     s->ssrc = 0; /* FIXME: was random(), what should this be? */
  96     s->first_packet = 1;
  97     s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
  98
  99     max_packet_size = url_fget_max_packet_size(s1->pb);
 100     if (max_packet_size <= 12)
 101         return AVERROR(EIO);
 102     s->buf = av_malloc(max_packet_size);
 103     if (s->buf == NULL) {
 104         return AVERROR(ENOMEM);
 105     }
 106     s->max_payload_size = max_packet_size - 12;
 107
 108     s->max_frames_per_packet = 0;
 109     if (s1->max_delay) {
 110         if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
 111             if (st->codec->frame_size == 0) {
 112                 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
 113             } else {
 114                 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);
 115             }
 116         }
 117         if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
 118             /* FIXME: We should round down here... */
 119             s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
 120         }
 121     }
 122
 123     av_set_pts_info(st, 32, 1, 90000);
 124     switch(st->codec->codec_id) {
 125     case CODEC_ID_MP2:
 126     case CODEC_ID_MP3:
 127         s->buf_ptr = s->buf + 4;
 128         break;
 129     case CODEC_ID_MPEG1VIDEO:
 130     case CODEC_ID_MPEG2VIDEO:
 131         break;
 132     case CODEC_ID_MPEG2TS:
 133         n = s->max_payload_size / TS_PACKET_SIZE;
 134         if (n < 1)
 135             n = 1;
 136         s->max_payload_size = n * TS_PACKET_SIZE;
 137         s->buf_ptr = s->buf;
 138         break;
 139     case CODEC_ID_AMR_NB:
 140     case CODEC_ID_AMR_WB:
 141         if (!s->max_frames_per_packet)
 142             s->max_frames_per_packet = 12;
 143         if (st->codec->codec_id == CODEC_ID_AMR_NB)
 144             n = 31;
 145         else
 146             n = 61;
 147         /* max_header_toc_size + the largest AMR payload must fit */
 148         if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
 149             av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
 150             return -1;
 151         }
 152         if (st->codec->channels != 1) {
 153             av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
 154             return -1;
 155         }
 156     case CODEC_ID_AAC:
 157         s->num_frames = 0;
 158     default:
 159         if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
 160             av_set_pts_info(st, 32, 1, st->codec->sample_rate);
 161         }
 162         s->buf_ptr = s->buf;
 163         break;
 164     }
 165
 166     return 0;
 167 }
 168
 169 /* send an rtcp sender report packet */
 170 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
 171 {
 172     RTPMuxContext *s = s1->priv_data;
 173     uint32_t rtp_ts;
 174
 175     dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
 176
 177     if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
 178     s->last_rtcp_ntp_time = ntp_time;
 179     rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
 180                           s1->streams[0]->time_base) + s->base_timestamp;
 181     put_byte(s1->pb, (RTP_VERSION << 6));
 182     put_byte(s1->pb, 200);
 183     put_be16(s1->pb, 6); /* length in words - 1 */
 184     put_be32(s1->pb, s->ssrc);
 185     put_be32(s1->pb, ntp_time / 1000000);
 186     put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
 187     put_be32(s1->pb, rtp_ts);
 188     put_be32(s1->pb, s->packet_count);
 189     put_be32(s1->pb, s->octet_count);
 190     put_flush_packet(s1->pb);
 191 }
 192
 193 /* send an rtp packet. sequence number is incremented, but the caller
 194    must update the timestamp itself */
 195 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
 196 {
 197     RTPMuxContext *s = s1->priv_data;
 198
 199     dprintf(s1, "rtp_send_data size=%d\n", len);
 200
 201     /* build the RTP header */
 202     put_byte(s1->pb, (RTP_VERSION << 6));
 203     put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
 204     put_be16(s1->pb, s->seq);
 205     put_be32(s1->pb, s->timestamp);
 206     put_be32(s1->pb, s->ssrc);
 207
 208     put_buffer(s1->pb, buf1, len);
 209     put_flush_packet(s1->pb);
 210
 211     s->seq++;
 212     s->octet_count += len;
 213     s->packet_count++;
 214 }
 215
 216 /* send an integer number of samples and compute time stamp and fill
 217    the rtp send buffer before sending. */
 218 static void rtp_send_samples(AVFormatContext *s1,
 219                              const uint8_t *buf1, int size, int sample_size)
 220 {
 221     RTPMuxContext *s = s1->priv_data;
 222     int len, max_packet_size, n;
 223
 224     max_packet_size = (s->max_payload_size / sample_size) * sample_size;
 225     /* not needed, but who nows */
 226     if ((size % sample_size) != 0)
 227         av_abort();
 228     n = 0;
 229     while (size > 0) {
 230         s->buf_ptr = s->buf;
 231         len = FFMIN(max_packet_size, size);
 232
 233         /* copy data */
 234         memcpy(s->buf_ptr, buf1, len);
 235         s->buf_ptr += len;
 236         buf1 += len;
 237         size -= len;
 238         s->timestamp = s->cur_timestamp + n / sample_size;
 239         ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
 240         n += (s->buf_ptr - s->buf);
 241     }
 242 }
 243
 244 /* NOTE: we suppose that exactly one frame is given as argument here */
 245 /* XXX: test it */
 246 static void rtp_send_mpegaudio(AVFormatContext *s1,
 247                                const uint8_t *buf1, int size)
 248 {
 249     RTPMuxContext *s = s1->priv_data;
 250     int len, count, max_packet_size;
 251
 252     max_packet_size = s->max_payload_size;
 253
 254     /* test if we must flush because not enough space */
 255     len = (s->buf_ptr - s->buf);
 256     if ((len + size) > max_packet_size) {
 257         if (len > 4) {
 258             ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
 259             s->buf_ptr = s->buf + 4;
 260         }
 261     }
 262     if (s->buf_ptr == s->buf + 4) {
 263         s->timestamp = s->cur_timestamp;
 264     }
 265
 266     /* add the packet */
 267     if (size > max_packet_size) {
 268         /* big packet: fragment */
 269         count = 0;
 270         while (size > 0) {
 271             len = max_packet_size - 4;
 272             if (len > size)
 273                 len = size;
 274             /* build fragmented packet */
 275             s->buf[0] = 0;
 276             s->buf[1] = 0;
 277             s->buf[2] = count >> 8;
 278             s->buf[3] = count;
 279             memcpy(s->buf + 4, buf1, len);
 280             ff_rtp_send_data(s1, s->buf, len + 4, 0);
 281             size -= len;
 282             buf1 += len;
 283             count += len;
 284         }
 285     } else {
 286         if (s->buf_ptr == s->buf + 4) {
 287             /* no fragmentation possible */
 288             s->buf[0] = 0;
 289             s->buf[1] = 0;
 290             s->buf[2] = 0;
 291             s->buf[3] = 0;
 292         }
 293         memcpy(s->buf_ptr, buf1, size);
 294         s->buf_ptr += size;
 295     }
 296 }
 297
 298 static void rtp_send_raw(AVFormatContext *s1,
 299                          const uint8_t *buf1, int size)
 300 {
 301     RTPMuxContext *s = s1->priv_data;
 302     int len, max_packet_size;
 303
 304     max_packet_size = s->max_payload_size;
 305
 306     while (size > 0) {
 307         len = max_packet_size;
 308         if (len > size)
 309             len = size;
 310
 311         s->timestamp = s->cur_timestamp;
 312         ff_rtp_send_data(s1, buf1, len, (len == size));
 313
 314         buf1 += len;
 315         size -= len;
 316     }
 317 }
 318
 319 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
 320 static void rtp_send_mpegts_raw(AVFormatContext *s1,
 321                                 const uint8_t *buf1, int size)
 322 {
 323     RTPMuxContext *s = s1->priv_data;
 324     int len, out_len;
 325
 326     while (size >= TS_PACKET_SIZE) {
 327         len = s->max_payload_size - (s->buf_ptr - s->buf);
 328         if (len > size)
 329             len = size;
 330         memcpy(s->buf_ptr, buf1, len);
 331         buf1 += len;
 332         size -= len;
 333         s->buf_ptr += len;
 334
 335         out_len = s->buf_ptr - s->buf;
 336         if (out_len >= s->max_payload_size) {
 337             ff_rtp_send_data(s1, s->buf, out_len, 0);
 338             s->buf_ptr = s->buf;
 339         }
 340     }
 341 }
 342
 343 /* write an RTP packet. 'buf1' must contain a single specific frame. */
 344 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
 345 {
 346     RTPMuxContext *s = s1->priv_data;
 347     AVStream *st = s1->streams[0];
 348     int rtcp_bytes;
 349     int size= pkt->size;
 350     uint8_t *buf1= pkt->data;
 351
 352     dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size);
 353
 354     rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
 355         RTCP_TX_RATIO_DEN;
 356     if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
 357                            (ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
 358         rtcp_send_sr(s1, ntp_time());
 359         s->last_octet_count = s->octet_count;
 360         s->first_packet = 0;
 361     }
 362     s->cur_timestamp = s->base_timestamp + pkt->pts;
 363
 364     switch(st->codec->codec_id) {
 365     case CODEC_ID_PCM_MULAW:
 366     case CODEC_ID_PCM_ALAW:
 367     case CODEC_ID_PCM_U8:
 368     case CODEC_ID_PCM_S8:
 369         rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
 370         break;
 371     case CODEC_ID_PCM_U16BE:
 372     case CODEC_ID_PCM_U16LE:
 373     case CODEC_ID_PCM_S16BE:
 374     case CODEC_ID_PCM_S16LE:
 375         rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
 376         break;
 377     case CODEC_ID_MP2:
 378     case CODEC_ID_MP3:
 379         rtp_send_mpegaudio(s1, buf1, size);
 380         break;
 381     case CODEC_ID_MPEG1VIDEO:
 382     case CODEC_ID_MPEG2VIDEO:
 383         ff_rtp_send_mpegvideo(s1, buf1, size);
 384         break;
 385     case CODEC_ID_AAC:
 386         ff_rtp_send_aac(s1, buf1, size);
 387         break;
 388     case CODEC_ID_AMR_NB:
 389     case CODEC_ID_AMR_WB:
 390         ff_rtp_send_amr(s1, buf1, size);
 391         break;
 392     case CODEC_ID_MPEG2TS:
 393         rtp_send_mpegts_raw(s1, buf1, size);
 394         break;
 395     case CODEC_ID_H264:
 396         ff_rtp_send_h264(s1, buf1, size);
 397         break;
 398     case CODEC_ID_H263:
 399     case CODEC_ID_H263P:
 400         ff_rtp_send_h263(s1, buf1, size);
 401         break;
 402     default:
 403         /* better than nothing : send the codec raw data */
 404         rtp_send_raw(s1, buf1, size);
 405         break;
 406     }
 407     return 0;
 408 }
 409
 410 static int rtp_write_trailer(AVFormatContext *s1)
 411 {
 412     RTPMuxContext *s = s1->priv_data;
 413
 414     av_freep(&s->buf);
 415
 416     return 0;
 417 }
 418
 419 AVOutputFormat rtp_muxer = {
 420     "rtp",
 421     NULL_IF_CONFIG_SMALL("RTP output format"),
 422     NULL,
 423     NULL,
 424     sizeof(RTPMuxContext),
 425     CODEC_ID_PCM_MULAW,
 426     CODEC_ID_NONE,
 427     rtp_write_header,
 428     rtp_write_packet,
 429     rtp_write_trailer,
 430 };