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