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 "rtp_internal.h"
  30 #include "rtp_mpv.h"
  31 #include "rtp_aac.h"
  32 #include "rtp_h264.h"
  33
  34 //#define DEBUG
  35
  36 #define RTCP_SR_SIZE 28
  37 #define NTP_OFFSET 2208988800ULL
  38 #define NTP_OFFSET_US (NTP_OFFSET * 1000000ULL)
  39
  40 static uint64_t ntp_time(void)
  41 {
  42   return (av_gettime() / 1000) * 1000 + NTP_OFFSET_US;
  43 }
  44
  45 static int rtp_write_header(AVFormatContext *s1)
  46 {
  47     RTPDemuxContext *s = s1->priv_data;
  48     int payload_type, max_packet_size, n;
  49     AVStream *st;
  50
  51     if (s1->nb_streams != 1)
  52         return -1;
  53     st = s1->streams[0];
  54
  55     payload_type = rtp_get_payload_type(st->codec);
  56     if (payload_type < 0)
  57         payload_type = RTP_PT_PRIVATE; /* private payload type */
  58     s->payload_type = payload_type;
  59
  60 // following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
  61     s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
  62     s->timestamp = s->base_timestamp;
  63     s->cur_timestamp = 0;
  64     s->ssrc = 0; /* FIXME: was random(), what should this be? */
  65     s->first_packet = 1;
  66     s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
  67
  68     max_packet_size = url_fget_max_packet_size(s1->pb);
  69     if (max_packet_size <= 12)
  70         return AVERROR(EIO);
  71     s->max_payload_size = max_packet_size - 12;
  72
  73     s->max_frames_per_packet = 0;
  74     if (s1->max_delay) {
  75         if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
  76             if (st->codec->frame_size == 0) {
  77                 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
  78             } else {
  79                 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);
  80             }
  81         }
  82         if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
  83             /* FIXME: We should round down here... */
  84             s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base);
  85         }
  86     }
  87
  88     av_set_pts_info(st, 32, 1, 90000);
  89     switch(st->codec->codec_id) {
  90     case CODEC_ID_MP2:
  91     case CODEC_ID_MP3:
  92         s->buf_ptr = s->buf + 4;
  93         break;
  94     case CODEC_ID_MPEG1VIDEO:
  95     case CODEC_ID_MPEG2VIDEO:
  96         break;
  97     case CODEC_ID_MPEG2TS:
  98         n = s->max_payload_size / TS_PACKET_SIZE;
  99         if (n < 1)
 100             n = 1;
 101         s->max_payload_size = n * TS_PACKET_SIZE;
 102         s->buf_ptr = s->buf;
 103         break;
 104     case CODEC_ID_AAC:
 105         s->read_buf_index = 0;
 106     default:
 107         if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
 108             av_set_pts_info(st, 32, 1, st->codec->sample_rate);
 109         }
 110         s->buf_ptr = s->buf;
 111         break;
 112     }
 113
 114     return 0;
 115 }
 116
 117 /* send an rtcp sender report packet */
 118 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
 119 {
 120     RTPDemuxContext *s = s1->priv_data;
 121     uint32_t rtp_ts;
 122
 123 #if defined(DEBUG)
 124     printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
 125 #endif
 126
 127     if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
 128     s->last_rtcp_ntp_time = ntp_time;
 129     rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q,
 130                           s1->streams[0]->time_base) + s->base_timestamp;
 131     put_byte(s1->pb, (RTP_VERSION << 6));
 132     put_byte(s1->pb, 200);
 133     put_be16(s1->pb, 6); /* length in words - 1 */
 134     put_be32(s1->pb, s->ssrc);
 135     put_be32(s1->pb, ntp_time / 1000000);
 136     put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
 137     put_be32(s1->pb, rtp_ts);
 138     put_be32(s1->pb, s->packet_count);
 139     put_be32(s1->pb, s->octet_count);
 140     put_flush_packet(s1->pb);
 141 }
 142
 143 /* send an rtp packet. sequence number is incremented, but the caller
 144    must update the timestamp itself */
 145 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
 146 {
 147     RTPDemuxContext *s = s1->priv_data;
 148
 149 #ifdef DEBUG
 150     printf("rtp_send_data size=%d\n", len);
 151 #endif
 152
 153     /* build the RTP header */
 154     put_byte(s1->pb, (RTP_VERSION << 6));
 155     put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
 156     put_be16(s1->pb, s->seq);
 157     put_be32(s1->pb, s->timestamp);
 158     put_be32(s1->pb, s->ssrc);
 159
 160     put_buffer(s1->pb, buf1, len);
 161     put_flush_packet(s1->pb);
 162
 163     s->seq++;
 164     s->octet_count += len;
 165     s->packet_count++;
 166 }
 167
 168 /* send an integer number of samples and compute time stamp and fill
 169    the rtp send buffer before sending. */
 170 static void rtp_send_samples(AVFormatContext *s1,
 171                              const uint8_t *buf1, int size, int sample_size)
 172 {
 173     RTPDemuxContext *s = s1->priv_data;
 174     int len, max_packet_size, n;
 175
 176     max_packet_size = (s->max_payload_size / sample_size) * sample_size;
 177     /* not needed, but who nows */
 178     if ((size % sample_size) != 0)
 179         av_abort();
 180     n = 0;
 181     while (size > 0) {
 182         s->buf_ptr = s->buf;
 183         len = FFMIN(max_packet_size, size);
 184
 185         /* copy data */
 186         memcpy(s->buf_ptr, buf1, len);
 187         s->buf_ptr += len;
 188         buf1 += len;
 189         size -= len;
 190         s->timestamp = s->cur_timestamp + n / sample_size;
 191         ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
 192         n += (s->buf_ptr - s->buf);
 193     }
 194 }
 195
 196 /* NOTE: we suppose that exactly one frame is given as argument here */
 197 /* XXX: test it */
 198 static void rtp_send_mpegaudio(AVFormatContext *s1,
 199                                const uint8_t *buf1, int size)
 200 {
 201     RTPDemuxContext *s = s1->priv_data;
 202     int len, count, max_packet_size;
 203
 204     max_packet_size = s->max_payload_size;
 205
 206     /* test if we must flush because not enough space */
 207     len = (s->buf_ptr - s->buf);
 208     if ((len + size) > max_packet_size) {
 209         if (len > 4) {
 210             ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
 211             s->buf_ptr = s->buf + 4;
 212         }
 213     }
 214     if (s->buf_ptr == s->buf + 4) {
 215         s->timestamp = s->cur_timestamp;
 216     }
 217
 218     /* add the packet */
 219     if (size > max_packet_size) {
 220         /* big packet: fragment */
 221         count = 0;
 222         while (size > 0) {
 223             len = max_packet_size - 4;
 224             if (len > size)
 225                 len = size;
 226             /* build fragmented packet */
 227             s->buf[0] = 0;
 228             s->buf[1] = 0;
 229             s->buf[2] = count >> 8;
 230             s->buf[3] = count;
 231             memcpy(s->buf + 4, buf1, len);
 232             ff_rtp_send_data(s1, s->buf, len + 4, 0);
 233             size -= len;
 234             buf1 += len;
 235             count += len;
 236         }
 237     } else {
 238         if (s->buf_ptr == s->buf + 4) {
 239             /* no fragmentation possible */
 240             s->buf[0] = 0;
 241             s->buf[1] = 0;
 242             s->buf[2] = 0;
 243             s->buf[3] = 0;
 244         }
 245         memcpy(s->buf_ptr, buf1, size);
 246         s->buf_ptr += size;
 247     }
 248 }
 249
 250 static void rtp_send_raw(AVFormatContext *s1,
 251                          const uint8_t *buf1, int size)
 252 {
 253     RTPDemuxContext *s = s1->priv_data;
 254     int len, max_packet_size;
 255
 256     max_packet_size = s->max_payload_size;
 257
 258     while (size > 0) {
 259         len = max_packet_size;
 260         if (len > size)
 261             len = size;
 262
 263         s->timestamp = s->cur_timestamp;
 264         ff_rtp_send_data(s1, buf1, len, (len == size));
 265
 266         buf1 += len;
 267         size -= len;
 268     }
 269 }
 270
 271 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
 272 static void rtp_send_mpegts_raw(AVFormatContext *s1,
 273                                 const uint8_t *buf1, int size)
 274 {
 275     RTPDemuxContext *s = s1->priv_data;
 276     int len, out_len;
 277
 278     while (size >= TS_PACKET_SIZE) {
 279         len = s->max_payload_size - (s->buf_ptr - s->buf);
 280         if (len > size)
 281             len = size;
 282         memcpy(s->buf_ptr, buf1, len);
 283         buf1 += len;
 284         size -= len;
 285         s->buf_ptr += len;
 286
 287         out_len = s->buf_ptr - s->buf;
 288         if (out_len >= s->max_payload_size) {
 289             ff_rtp_send_data(s1, s->buf, out_len, 0);
 290             s->buf_ptr = s->buf;
 291         }
 292     }
 293 }
 294
 295 /* write an RTP packet. 'buf1' must contain a single specific frame. */
 296 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
 297 {
 298     RTPDemuxContext *s = s1->priv_data;
 299     AVStream *st = s1->streams[0];
 300     int rtcp_bytes;
 301     int size= pkt->size;
 302     uint8_t *buf1= pkt->data;
 303
 304 #ifdef DEBUG
 305     printf("%d: write len=%d\n", pkt->stream_index, size);
 306 #endif
 307
 308     /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
 309     rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
 310         RTCP_TX_RATIO_DEN;
 311     if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
 312                            (ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
 313         rtcp_send_sr(s1, ntp_time());
 314         s->last_octet_count = s->octet_count;
 315         s->first_packet = 0;
 316     }
 317     s->cur_timestamp = s->base_timestamp + pkt->pts;
 318
 319     switch(st->codec->codec_id) {
 320     case CODEC_ID_PCM_MULAW:
 321     case CODEC_ID_PCM_ALAW:
 322     case CODEC_ID_PCM_U8:
 323     case CODEC_ID_PCM_S8:
 324         rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
 325         break;
 326     case CODEC_ID_PCM_U16BE:
 327     case CODEC_ID_PCM_U16LE:
 328     case CODEC_ID_PCM_S16BE:
 329     case CODEC_ID_PCM_S16LE:
 330         rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
 331         break;
 332     case CODEC_ID_MP2:
 333     case CODEC_ID_MP3:
 334         rtp_send_mpegaudio(s1, buf1, size);
 335         break;
 336     case CODEC_ID_MPEG1VIDEO:
 337     case CODEC_ID_MPEG2VIDEO:
 338         ff_rtp_send_mpegvideo(s1, buf1, size);
 339         break;
 340     case CODEC_ID_AAC:
 341         ff_rtp_send_aac(s1, buf1, size);
 342         break;
 343     case CODEC_ID_MPEG2TS:
 344         rtp_send_mpegts_raw(s1, buf1, size);
 345         break;
 346     case CODEC_ID_H264:
 347         ff_rtp_send_h264(s1, buf1, size);
 348         break;
 349     default:
 350         /* better than nothing : send the codec raw data */
 351         rtp_send_raw(s1, buf1, size);
 352         break;
 353     }
 354     return 0;
 355 }
 356
 357 AVOutputFormat rtp_muxer = {
 358     "rtp",
 359     "RTP output format",
 360     NULL,
 361     NULL,
 362     sizeof(RTPDemuxContext),
 363     CODEC_ID_PCM_MULAW,
 364     CODEC_ID_NONE,
 365     rtp_write_header,
 366     rtp_write_packet,
 367 };