rtp.c 37.4 KB
Newer Older
1 2 3 4
/*
 * RTP input/output format
 * Copyright (c) 2002 Fabrice Bellard.
 *
5 6 7
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
8 9
 * modify it under the terms of the GNU Lesser General Public
 * 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 14 15 16 17
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * 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
 */
#include "avformat.h"
22
#include "mpegts.h"
Romain Degez's avatar
Romain Degez committed
23
#include "bitstream.h"
24 25

#include <unistd.h>
26
#include "network.h"
27

Ryan Martell's avatar
Ryan Martell committed
28
#include "rtp_internal.h"
29
#include "rtp_h264.h"
Ryan Martell's avatar
Ryan Martell committed
30

31 32 33 34 35 36 37 38 39
//#define DEBUG


/* TODO: - add RTCP statistics reporting (should be optional).

         - add support for h263/mpeg4 packetized output : IDEA: send a
         buffer to 'rtp_write_packet' contains all the packets for ONE
         frame. Each packet should have a four byte header containing
         the length in big endian format (same trick as
40
         'url_open_dyn_packet_buf')
41 42
*/

Romain Degez's avatar
Romain Degez committed
43 44 45 46 47 48 49 50 51 52 53 54 55 56 57
/* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
AVRtpPayloadType_t AVRtpPayloadTypes[]=
{
  {0, "PCMU",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_MULAW, 8000, 1},
  {1, "Reserved",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {2, "Reserved",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {3, "GSM",         CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
  {4, "G723",        CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
  {5, "DVI4",        CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
  {6, "DVI4",        CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 16000, 1},
  {7, "LPC",         CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
  {8, "PCMA",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_ALAW, 8000, 1},
  {9, "G722",        CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
  {10, "L16",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_S16BE, 44100, 2},
  {11, "L16",        CODEC_TYPE_AUDIO,   CODEC_ID_PCM_S16BE, 44100, 1},
Benjamin Larsson's avatar
Benjamin Larsson committed
58
  {12, "QCELP",      CODEC_TYPE_AUDIO,   CODEC_ID_QCELP, 8000, 1},
Romain Degez's avatar
Romain Degez committed
59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176
  {13, "CN",         CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
  {14, "MPA",        CODEC_TYPE_AUDIO,   CODEC_ID_MP2, 90000, -1},
  {15, "G728",       CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
  {16, "DVI4",       CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 11025, 1},
  {17, "DVI4",       CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 22050, 1},
  {18, "G729",       CODEC_TYPE_AUDIO,   CODEC_ID_NONE, 8000, 1},
  {19, "reserved",   CODEC_TYPE_AUDIO,   CODEC_ID_NONE, -1, -1},
  {20, "unassigned", CODEC_TYPE_AUDIO,   CODEC_ID_NONE, -1, -1},
  {21, "unassigned", CODEC_TYPE_AUDIO,   CODEC_ID_NONE, -1, -1},
  {22, "unassigned", CODEC_TYPE_AUDIO,   CODEC_ID_NONE, -1, -1},
  {23, "unassigned", CODEC_TYPE_AUDIO,   CODEC_ID_NONE, -1, -1},
  {24, "unassigned", CODEC_TYPE_VIDEO,   CODEC_ID_NONE, -1, -1},
  {25, "CelB",       CODEC_TYPE_VIDEO,   CODEC_ID_NONE, 90000, -1},
  {26, "JPEG",       CODEC_TYPE_VIDEO,   CODEC_ID_MJPEG, 90000, -1},
  {27, "unassigned", CODEC_TYPE_VIDEO,   CODEC_ID_NONE, -1, -1},
  {28, "nv",         CODEC_TYPE_VIDEO,   CODEC_ID_NONE, 90000, -1},
  {29, "unassigned", CODEC_TYPE_VIDEO,   CODEC_ID_NONE, -1, -1},
  {30, "unassigned", CODEC_TYPE_VIDEO,   CODEC_ID_NONE, -1, -1},
  {31, "H261",       CODEC_TYPE_VIDEO,   CODEC_ID_H261, 90000, -1},
  {32, "MPV",        CODEC_TYPE_VIDEO,   CODEC_ID_MPEG1VIDEO, 90000, -1},
  {33, "MP2T",       CODEC_TYPE_DATA,    CODEC_ID_MPEG2TS, 90000, -1},
  {34, "H263",       CODEC_TYPE_VIDEO,   CODEC_ID_H263, 90000, -1},
  {35, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {36, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {37, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {38, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {39, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {40, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {41, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {42, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {43, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {44, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {45, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {46, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {47, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {48, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {49, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {50, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {51, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {52, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {53, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {54, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {55, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {56, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {57, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {58, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {59, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {60, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {61, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {62, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {63, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {64, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {65, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {66, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {67, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {68, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {69, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {70, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {71, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {72, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {73, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {74, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {75, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {76, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {77, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {78, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {79, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {80, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {81, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {82, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {83, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {84, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {85, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {86, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {87, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {88, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {89, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {90, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {91, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {92, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {93, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {94, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {95, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {96, "dynamic",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {97, "dynamic",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {98, "dynamic",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {99, "dynamic",    CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {100, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {101, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {102, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {103, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {104, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {105, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {106, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {107, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {108, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {109, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {110, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {111, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {112, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {113, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {114, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {115, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {116, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {117, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {118, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {119, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {120, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {121, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {122, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {123, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {124, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {125, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {126, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {127, "dynamic",   CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
  {-1, "",           CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}
};

Ryan Martell's avatar
Ryan Martell committed
177 178 179 180
/* statistics functions */
RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;

static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
181
static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
Ryan Martell's avatar
Ryan Martell committed
182 183

static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
Romain Degez's avatar
Romain Degez committed
184
{
Ryan Martell's avatar
Ryan Martell committed
185 186 187
    handler->next= RTPFirstDynamicPayloadHandler;
    RTPFirstDynamicPayloadHandler= handler;
}
188

189
void av_register_rtp_dynamic_payload_handlers(void)
Ryan Martell's avatar
Ryan Martell committed
190 191 192 193 194
{
    register_dynamic_payload_handler(&mp4v_es_handler);
    register_dynamic_payload_handler(&mpeg4_generic_handler);
    register_dynamic_payload_handler(&ff_h264_dynamic_handler);
}
195 196 197

int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
{
Romain Degez's avatar
Romain Degez committed
198 199
    if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) {
        codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type;
Luca Abeni's avatar
Luca Abeni committed
200
        codec->codec_id = AVRtpPayloadTypes[payload_type].codec_id;
Romain Degez's avatar
Romain Degez committed
201 202 203 204 205
        if (AVRtpPayloadTypes[payload_type].audio_channels > 0)
            codec->channels = AVRtpPayloadTypes[payload_type].audio_channels;
        if (AVRtpPayloadTypes[payload_type].clock_rate > 0)
            codec->sample_rate = AVRtpPayloadTypes[payload_type].clock_rate;
        return 0;
206
    }
Romain Degez's avatar
Romain Degez committed
207
    return -1;
208 209 210 211
}

int rtp_get_payload_type(AVCodecContext *codec)
{
Romain Degez's avatar
Romain Degez committed
212
    int i, payload_type;
213 214

    /* compute the payload type */
Romain Degez's avatar
Romain Degez committed
215 216 217 218 219 220
    for (payload_type = -1, i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
        if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) {
            if (codec->codec_id == CODEC_ID_PCM_S16BE)
                if (codec->channels != AVRtpPayloadTypes[i].audio_channels)
                    continue;
            payload_type = AVRtpPayloadTypes[i].pt;
221 222 223 224
        }
    return payload_type;
}

225
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
226 227 228
{
    if (buf[1] != 200)
        return -1;
229
    s->last_rtcp_ntp_time = AV_RB64(buf + 8);
230 231
    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
        s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
232
    s->last_rtcp_timestamp = AV_RB32(buf + 16);
233 234 235
    return 0;
}

236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327
#define RTP_SEQ_MOD (1<<16)

/**
* called on parse open packet
*/
static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
{
    memset(s, 0, sizeof(RTPStatistics));
    s->max_seq= base_sequence;
    s->probation= 1;
}

/**
* called whenever there is a large jump in sequence numbers, or when they get out of probation...
*/
static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
{
    s->max_seq= seq;
    s->cycles= 0;
    s->base_seq= seq -1;
    s->bad_seq= RTP_SEQ_MOD + 1;
    s->received= 0;
    s->expected_prior= 0;
    s->received_prior= 0;
    s->jitter= 0;
    s->transit= 0;
}

/**
* returns 1 if we should handle this packet.
*/
static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
{
    uint16_t udelta= seq - s->max_seq;
    const int MAX_DROPOUT= 3000;
    const int MAX_MISORDER = 100;
    const int MIN_SEQUENTIAL = 2;

    /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
    if(s->probation)
    {
        if(seq==s->max_seq + 1) {
            s->probation--;
            s->max_seq= seq;
            if(s->probation==0) {
                rtp_init_sequence(s, seq);
                s->received++;
                return 1;
            }
        } else {
            s->probation= MIN_SEQUENTIAL - 1;
            s->max_seq = seq;
        }
    } else if (udelta < MAX_DROPOUT) {
        // in order, with permissible gap
        if(seq < s->max_seq) {
            //sequence number wrapped; count antother 64k cycles
            s->cycles += RTP_SEQ_MOD;
        }
        s->max_seq= seq;
    } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
        // sequence made a large jump...
        if(seq==s->bad_seq) {
            // two sequential packets-- assume that the other side restarted without telling us; just resync.
            rtp_init_sequence(s, seq);
        } else {
            s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
            return 0;
        }
    } else {
        // duplicate or reordered packet...
    }
    s->received++;
    return 1;
}

#if 0
/**
* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
* difference between the arrival and sent timestamp.  As a result, the jitter and transit statistics values
* never change.  I left this in in case someone else can see a way. (rdm)
*/
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
{
    uint32_t transit= arrival_timestamp - sent_timestamp;
    int d;
    s->transit= transit;
    d= FFABS(transit - s->transit);
    s->jitter += d - ((s->jitter + 8)>>4);
}
#endif

328 329 330 331 332 333
int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
{
    ByteIOContext pb;
    uint8_t *buf;
    int len;
    int rtcp_bytes;
334 335 336 337 338 339 340 341 342
    RTPStatistics *stats= &s->statistics;
    uint32_t lost;
    uint32_t extended_max;
    uint32_t expected_interval;
    uint32_t received_interval;
    uint32_t lost_interval;
    uint32_t expected;
    uint32_t fraction;
    uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
343 344 345 346

    if (!s->rtp_ctx || (count < 1))
        return -1;

347
    /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
348 349 350 351 352 353 354 355 356 357 358 359 360 361 362 363 364 365 366
    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
    s->octet_count += count;
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
        RTCP_TX_RATIO_DEN;
    rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
    if (rtcp_bytes < 28)
        return -1;
    s->last_octet_count = s->octet_count;

    if (url_open_dyn_buf(&pb) < 0)
        return -1;

    // Receiver Report
    put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
    put_byte(&pb, 201);
    put_be16(&pb, 7); /* length in words - 1 */
    put_be32(&pb, s->ssrc); // our own SSRC
    put_be32(&pb, s->ssrc); // XXX: should be the server's here!
    // some placeholders we should really fill...
367 368 369 370 371 372 373 374 375 376 377 378 379 380 381 382 383 384 385 386 387 388 389 390 391 392 393 394 395 396
    // RFC 1889/p64
    extended_max= stats->cycles + stats->max_seq;
    expected= extended_max - stats->base_seq + 1;
    lost= expected - stats->received;
    lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
    expected_interval= expected - stats->expected_prior;
    stats->expected_prior= expected;
    received_interval= stats->received - stats->received_prior;
    stats->received_prior= stats->received;
    lost_interval= expected_interval - received_interval;
    if (expected_interval==0 || lost_interval<=0) fraction= 0;
    else fraction = (lost_interval<<8)/expected_interval;

    fraction= (fraction<<24) | lost;

    put_be32(&pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
    put_be32(&pb, extended_max); /* max sequence received */
    put_be32(&pb, stats->jitter>>4); /* jitter */

    if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
    {
        put_be32(&pb, 0); /* last SR timestamp */
        put_be32(&pb, 0); /* delay since last SR */
    } else {
        uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
        uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;

        put_be32(&pb, middle_32_bits); /* last SR timestamp */
        put_be32(&pb, delay_since_last); /* delay since last SR */
    }
397 398 399 400 401 402 403 404 405 406 407 408 409 410 411 412 413 414

    // CNAME
    put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
    put_byte(&pb, 202);
    len = strlen(s->hostname);
    put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */
    put_be32(&pb, s->ssrc);
    put_byte(&pb, 0x01);
    put_byte(&pb, len);
    put_buffer(&pb, s->hostname, len);
    // padding
    for (len = (6 + len) % 4; len % 4; len++) {
        put_byte(&pb, 0);
    }

    put_flush_packet(&pb);
    len = url_close_dyn_buf(&pb, &buf);
    if ((len > 0) && buf) {
415
        int result;
416 417 418
#if defined(DEBUG)
        printf("sending %d bytes of RR\n", len);
#endif
419 420 421 422
        result= url_write(s->rtp_ctx, buf, len);
#if defined(DEBUG)
        printf("result from url_write: %d\n", result);
#endif
423 424 425 426 427
        av_free(buf);
    }
    return 0;
}

428
/**
429 430
 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
 * MPEG2TS streams to indicate that they should be demuxed inside the
431
 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
432
 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
433
 */
434
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
435 436 437 438 439 440 441 442 443 444 445
{
    RTPDemuxContext *s;

    s = av_mallocz(sizeof(RTPDemuxContext));
    if (!s)
        return NULL;
    s->payload_type = payload_type;
    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
    s->ic = s1;
    s->st = st;
Romain Degez's avatar
Romain Degez committed
446
    s->rtp_payload_data = rtp_payload_data;
447
    rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
Romain Degez's avatar
Romain Degez committed
448
    if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) {
449 450 451 452 453
        s->ts = mpegts_parse_open(s->ic);
        if (s->ts == NULL) {
            av_free(s);
            return NULL;
        }
Fabrice Bellard's avatar
Fabrice Bellard committed
454
    } else {
455
        switch(st->codec->codec_id) {
Fabrice Bellard's avatar
Fabrice Bellard committed
456 457 458 459 460
        case CODEC_ID_MPEG1VIDEO:
        case CODEC_ID_MPEG2VIDEO:
        case CODEC_ID_MP2:
        case CODEC_ID_MP3:
        case CODEC_ID_MPEG4:
Ryan Martell's avatar
Ryan Martell committed
461
        case CODEC_ID_H264:
Aurelien Jacobs's avatar
Aurelien Jacobs committed
462
            st->need_parsing = AVSTREAM_PARSE_FULL;
Fabrice Bellard's avatar
Fabrice Bellard committed
463 464 465 466
            break;
        default:
            break;
        }
467
    }
468 469 470
    // needed to send back RTCP RR in RTSP sessions
    s->rtp_ctx = rtpc;
    gethostname(s->hostname, sizeof(s->hostname));
471 472 473
    return s;
}

Romain Degez's avatar
Romain Degez committed
474 475 476 477 478 479 480 481 482 483 484 485 486
static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
{
    int au_headers_length, au_header_size, i;
    GetBitContext getbitcontext;
    rtp_payload_data_t *infos;

    infos = s->rtp_payload_data;

    if (infos == NULL)
        return -1;

    /* decode the first 2 bytes where are stored the AUHeader sections
       length in bits */
487
    au_headers_length = AV_RB16(buf);
Romain Degez's avatar
Romain Degez committed
488 489 490 491 492 493 494 495 496 497 498 499 500 501 502 503 504 505 506 507

    if (au_headers_length > RTP_MAX_PACKET_LENGTH)
      return -1;

    infos->au_headers_length_bytes = (au_headers_length + 7) / 8;

    /* skip AU headers length section (2 bytes) */
    buf += 2;

    init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);

    /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
    au_header_size = infos->sizelength + infos->indexlength;
    if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
        return -1;

    infos->nb_au_headers = au_headers_length / au_header_size;
    infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);

    /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
Diego Biurrun's avatar
Diego Biurrun committed
508
       In my test, the FAAD decoder does not behave correctly when sending each AU one by one
Romain Degez's avatar
Romain Degez committed
509 510 511 512 513 514 515 516 517 518 519 520 521
       but does when sending the whole as one big packet...  */
    infos->au_headers[0].size = 0;
    infos->au_headers[0].index = 0;
    for (i = 0; i < infos->nb_au_headers; ++i) {
        infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
        infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
    }

    infos->nb_au_headers = 1;

    return 0;
}

522 523 524 525 526 527 528 529 530 531 532 533 534 535 536 537 538 539 540 541 542
/**
 * This was the second switch in rtp_parse packet.  Normalizes time, if required, sets stream_index, etc.
 */
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
{
    switch(s->st->codec->codec_id) {
        case CODEC_ID_MP2:
        case CODEC_ID_MPEG1VIDEO:
            if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
                int64_t addend;

                int delta_timestamp;
                /* XXX: is it really necessary to unify the timestamp base ? */
                /* compute pts from timestamp with received ntp_time */
                delta_timestamp = timestamp - s->last_rtcp_timestamp;
                /* convert to 90 kHz without overflow */
                addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
                addend = (addend * 5625) >> 14;
                pkt->pts = addend + delta_timestamp;
            }
            break;
543
        case CODEC_ID_AAC:
544 545 546 547 548 549 550 551 552 553 554
        case CODEC_ID_H264:
        case CODEC_ID_MPEG4:
            pkt->pts = timestamp;
            break;
        default:
            /* no timestamp info yet */
            break;
    }
    pkt->stream_index = s->st->index;
}

555
/**
556
 * Parse an RTP or RTCP packet directly sent as a buffer.
557
 * @param s RTP parse context.
558
 * @param pkt returned packet
559
 * @param buf input buffer or NULL to read the next packets
560
 * @param len buffer len
561
 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
562
 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
563
 */
564
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
565
                     const uint8_t *buf, int len)
566 567
{
    unsigned int ssrc, h;
568
    int payload_type, seq, ret;
569
    AVStream *st;
570
    uint32_t timestamp;
571
    int rv= 0;
572

573 574
    if (!buf) {
        /* return the next packets, if any */
Ryan Martell's avatar
Ryan Martell committed
575
        if(s->st && s->parse_packet) {
576 577 578 579
            timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
            rv= s->parse_packet(s, pkt, &timestamp, NULL, 0);
            finalize_packet(s, pkt, timestamp);
            return rv;
Ryan Martell's avatar
Ryan Martell committed
580
        } else {
581
            // TODO: Move to a dynamic packet handler (like above)
582 583 584 585 586 587 588 589 590 591 592
            if (s->read_buf_index >= s->read_buf_size)
                return -1;
            ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
                                      s->read_buf_size - s->read_buf_index);
            if (ret < 0)
                return -1;
            s->read_buf_index += ret;
            if (s->read_buf_index < s->read_buf_size)
                return 1;
            else
                return 0;
Ryan Martell's avatar
Ryan Martell committed
593
        }
594 595
    }

596 597 598 599 600 601
    if (len < 12)
        return -1;

    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
        return -1;
    if (buf[1] >= 200 && buf[1] <= 204) {
602
        rtcp_parse_packet(s, buf, len);
603 604 605
        return -1;
    }
    payload_type = buf[1] & 0x7f;
606
    seq  = AV_RB16(buf + 2);
607 608
    timestamp = AV_RB32(buf + 4);
    ssrc = AV_RB32(buf + 8);
609 610
    /* store the ssrc in the RTPDemuxContext */
    s->ssrc = ssrc;
611

612 613 614
    /* NOTE: we can handle only one payload type */
    if (s->payload_type != payload_type)
        return -1;
615 616

    st = s->st;
617 618 619
    // only do something with this if all the rtp checks pass...
    if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
    {
620
        av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
621
               payload_type, seq, ((s->seq + 1) & 0xffff));
622
        return -1;
623
    }
624

Romain Degez's avatar
Romain Degez committed
625
    s->seq = seq;
626 627
    len -= 12;
    buf += 12;
628 629 630 631 632

    if (!st) {
        /* specific MPEG2TS demux support */
        ret = mpegts_parse_packet(s->ts, pkt, buf, len);
        if (ret < 0)
633
            return -1;
634 635 636 637 638 639 640
        if (ret < len) {
            s->read_buf_size = len - ret;
            memcpy(s->buf, buf + ret, s->read_buf_size);
            s->read_buf_index = 0;
            return 1;
        }
    } else {
Ryan Martell's avatar
Ryan Martell committed
641
        // at this point, the RTP header has been stripped;  This is ASSUMING that there is only 1 CSRC, which in't wise.
642
        switch(st->codec->codec_id) {
643 644 645 646
        case CODEC_ID_MP2:
            /* better than nothing: skip mpeg audio RTP header */
            if (len <= 4)
                return -1;
647
            h = AV_RB32(buf);
648 649 650 651 652 653
            len -= 4;
            buf += 4;
            av_new_packet(pkt, len);
            memcpy(pkt->data, buf, len);
            break;
        case CODEC_ID_MPEG1VIDEO:
Fabrice Bellard's avatar
Fabrice Bellard committed
654
            /* better than nothing: skip mpeg video RTP header */
655 656
            if (len <= 4)
                return -1;
657
            h = AV_RB32(buf);
658 659
            buf += 4;
            len -= 4;
660 661 662 663 664 665 666 667 668 669
            if (h & (1 << 26)) {
                /* mpeg2 */
                if (len <= 4)
                    return -1;
                buf += 4;
                len -= 4;
            }
            av_new_packet(pkt, len);
            memcpy(pkt->data, buf, len);
            break;
670 671 672
            // moved from below, verbatim.  this is because this section handles packets, and the lower switch handles
            // timestamps.
            // TODO: Put this into a dynamic packet handler...
673
        case CODEC_ID_AAC:
674 675 676 677 678 679 680 681 682 683 684 685 686 687 688 689 690 691
            if (rtp_parse_mp4_au(s, buf))
                return -1;
            {
                rtp_payload_data_t *infos = s->rtp_payload_data;
                if (infos == NULL)
                    return -1;
                buf += infos->au_headers_length_bytes + 2;
                len -= infos->au_headers_length_bytes + 2;

                /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
                    one au_header */
                av_new_packet(pkt, infos->au_headers[0].size);
                memcpy(pkt->data, buf, infos->au_headers[0].size);
                buf += infos->au_headers[0].size;
                len -= infos->au_headers[0].size;
            }
            s->read_buf_size = len;
            s->buf_ptr = buf;
692
            rv= 0;
693
            break;
694
        default:
Ryan Martell's avatar
Ryan Martell committed
695
            if(s->parse_packet) {
696
                rv= s->parse_packet(s, pkt, &timestamp, buf, len);
Ryan Martell's avatar
Ryan Martell committed
697
            } else {
698 699
                av_new_packet(pkt, len);
                memcpy(pkt->data, buf, len);
Ryan Martell's avatar
Ryan Martell committed
700
            }
701
            break;
702
        }
703

704 705
        // now perform timestamp things....
        finalize_packet(s, pkt, timestamp);
706
    }
707
    return rv;
708 709
}

710
void rtp_parse_close(RTPDemuxContext *s)
711
{
Ryan Martell's avatar
Ryan Martell committed
712
    // TODO: fold this into the protocol specific data fields.
Romain Degez's avatar
Romain Degez committed
713
    if (!strcmp(AVRtpPayloadTypes[s->payload_type].enc_name, "MP2T")) {
714
        mpegts_parse_close(s->ts);
715
    }
716
    av_free(s);
717 718 719 720 721 722
}

/* rtp output */

static int rtp_write_header(AVFormatContext *s1)
{
723 724
    RTPDemuxContext *s = s1->priv_data;
    int payload_type, max_packet_size, n;
725 726 727 728 729 730
    AVStream *st;

    if (s1->nb_streams != 1)
        return -1;
    st = s1->streams[0];

731
    payload_type = rtp_get_payload_type(st->codec);
732
    if (payload_type < 0)
733
        payload_type = RTP_PT_PRIVATE; /* private payload type */
734 735
    s->payload_type = payload_type;

736
// following 2 FIXMies could be set based on the current time, theres normaly no info leak, as rtp will likely be transmitted immedeatly
737
    s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
738
    s->timestamp = s->base_timestamp;
739
    s->ssrc = 0; /* FIXME: was random(), what should this be? */
740 741 742 743 744 745 746
    s->first_packet = 1;

    max_packet_size = url_fget_max_packet_size(&s1->pb);
    if (max_packet_size <= 12)
        return AVERROR_IO;
    s->max_payload_size = max_packet_size - 12;

747
    switch(st->codec->codec_id) {
748
    case CODEC_ID_MP2:
Fabrice Bellard's avatar
Fabrice Bellard committed
749
    case CODEC_ID_MP3:
750 751 752 753 754 755
        s->buf_ptr = s->buf + 4;
        s->cur_timestamp = 0;
        break;
    case CODEC_ID_MPEG1VIDEO:
        s->cur_timestamp = 0;
        break;
756 757 758 759 760 761 762
    case CODEC_ID_MPEG2TS:
        n = s->max_payload_size / TS_PACKET_SIZE;
        if (n < 1)
            n = 1;
        s->max_payload_size = n * TS_PACKET_SIZE;
        s->buf_ptr = s->buf;
        break;
763 764 765 766 767 768 769 770 771
    default:
        s->buf_ptr = s->buf;
        break;
    }

    return 0;
}

/* send an rtcp sender report packet */
772
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
773
{
774
    RTPDemuxContext *s = s1->priv_data;
775
#if defined(DEBUG)
776
    printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
777 778 779 780 781 782 783 784 785 786 787 788 789 790
#endif
    put_byte(&s1->pb, (RTP_VERSION << 6));
    put_byte(&s1->pb, 200);
    put_be16(&s1->pb, 6); /* length in words - 1 */
    put_be32(&s1->pb, s->ssrc);
    put_be64(&s1->pb, ntp_time);
    put_be32(&s1->pb, s->timestamp);
    put_be32(&s1->pb, s->packet_count);
    put_be32(&s1->pb, s->octet_count);
    put_flush_packet(&s1->pb);
}

/* send an rtp packet. sequence number is incremented, but the caller
   must update the timestamp itself */
791
static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
792
{
793
    RTPDemuxContext *s = s1->priv_data;
794 795 796 797 798 799 800

#ifdef DEBUG
    printf("rtp_send_data size=%d\n", len);
#endif

    /* build the RTP header */
    put_byte(&s1->pb, (RTP_VERSION << 6));
801
    put_byte(&s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
802 803 804
    put_be16(&s1->pb, s->seq);
    put_be32(&s1->pb, s->timestamp);
    put_be32(&s1->pb, s->ssrc);
805

806 807
    put_buffer(&s1->pb, buf1, len);
    put_flush_packet(&s1->pb);
808

809 810 811 812 813 814 815 816
    s->seq++;
    s->octet_count += len;
    s->packet_count++;
}

/* send an integer number of samples and compute time stamp and fill
   the rtp send buffer before sending. */
static void rtp_send_samples(AVFormatContext *s1,
817
                             const uint8_t *buf1, int size, int sample_size)
818
{
819
    RTPDemuxContext *s = s1->priv_data;
820 821 822 823 824 825 826 827 828 829 830 831 832 833 834 835 836 837 838
    int len, max_packet_size, n;

    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
    /* not needed, but who nows */
    if ((size % sample_size) != 0)
        av_abort();
    while (size > 0) {
        len = (max_packet_size - (s->buf_ptr - s->buf));
        if (len > size)
            len = size;

        /* copy data */
        memcpy(s->buf_ptr, buf1, len);
        s->buf_ptr += len;
        buf1 += len;
        size -= len;
        n = (s->buf_ptr - s->buf);
        /* if buffer full, then send it */
        if (n >= max_packet_size) {
839
            rtp_send_data(s1, s->buf, n, 0);
840 841 842 843 844
            s->buf_ptr = s->buf;
            /* update timestamp */
            s->timestamp += n / sample_size;
        }
    }
845
}
846 847 848 849

/* NOTE: we suppose that exactly one frame is given as argument here */
/* XXX: test it */
static void rtp_send_mpegaudio(AVFormatContext *s1,
850
                               const uint8_t *buf1, int size)
851
{
852
    RTPDemuxContext *s = s1->priv_data;
853 854 855 856 857 858 859 860 861
    AVStream *st = s1->streams[0];
    int len, count, max_packet_size;

    max_packet_size = s->max_payload_size;

    /* test if we must flush because not enough space */
    len = (s->buf_ptr - s->buf);
    if ((len + size) > max_packet_size) {
        if (len > 4) {
862
            rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
863 864
            s->buf_ptr = s->buf + 4;
            /* 90 KHz time stamp */
865
            s->timestamp = s->base_timestamp +
866
                (s->cur_timestamp * 90000LL) / st->codec->sample_rate;
867 868 869 870 871 872 873 874 875 876 877 878 879 880 881 882 883
        }
    }

    /* add the packet */
    if (size > max_packet_size) {
        /* big packet: fragment */
        count = 0;
        while (size > 0) {
            len = max_packet_size - 4;
            if (len > size)
                len = size;
            /* build fragmented packet */
            s->buf[0] = 0;
            s->buf[1] = 0;
            s->buf[2] = count >> 8;
            s->buf[3] = count;
            memcpy(s->buf + 4, buf1, len);
884
            rtp_send_data(s1, s->buf, len + 4, 0);
885 886 887 888 889 890 891 892 893 894 895 896 897 898 899
            size -= len;
            buf1 += len;
            count += len;
        }
    } else {
        if (s->buf_ptr == s->buf + 4) {
            /* no fragmentation possible */
            s->buf[0] = 0;
            s->buf[1] = 0;
            s->buf[2] = 0;
            s->buf[3] = 0;
        }
        memcpy(s->buf_ptr, buf1, size);
        s->buf_ptr += size;
    }
900
    s->cur_timestamp += st->codec->frame_size;
901 902 903 904 905
}

/* NOTE: a single frame must be passed with sequence header if
   needed. XXX: use slices. */
static void rtp_send_mpegvideo(AVFormatContext *s1,
906
                               const uint8_t *buf1, int size)
907
{
908
    RTPDemuxContext *s = s1->priv_data;
909 910
    AVStream *st = s1->streams[0];
    int len, h, max_packet_size;
911
    uint8_t *q;
912 913 914 915 916 917

    max_packet_size = s->max_payload_size;

    while (size > 0) {
        /* XXX: more correct headers */
        h = 0;
918
        if (st->codec->sub_id == 2)
919 920 921 922 923 924 925
            h |= 1 << 26; /* mpeg 2 indicator */
        q = s->buf;
        *q++ = h >> 24;
        *q++ = h >> 16;
        *q++ = h >> 8;
        *q++ = h;

926
        if (st->codec->sub_id == 2) {
927 928 929 930 931 932
            h = 0;
            *q++ = h >> 24;
            *q++ = h >> 16;
            *q++ = h >> 8;
            *q++ = h;
        }
933

934 935 936 937 938 939 940 941
        len = max_packet_size - (q - s->buf);
        if (len > size)
            len = size;

        memcpy(q, buf1, len);
        q += len;

        /* 90 KHz time stamp */
942
        s->timestamp = s->base_timestamp +
943
            av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
944
        rtp_send_data(s1, s->buf, q - s->buf, (len == size));
945 946 947 948 949 950 951

        buf1 += len;
        size -= len;
    }
    s->cur_timestamp++;
}

952
static void rtp_send_raw(AVFormatContext *s1,
953
                         const uint8_t *buf1, int size)
954
{
955
    RTPDemuxContext *s = s1->priv_data;
956 957 958 959 960 961 962 963 964 965 966
    AVStream *st = s1->streams[0];
    int len, max_packet_size;

    max_packet_size = s->max_payload_size;

    while (size > 0) {
        len = max_packet_size;
        if (len > size)
            len = size;

        /* 90 KHz time stamp */
967
        s->timestamp = s->base_timestamp +
968
            av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
969
        rtp_send_data(s1, buf1, len, (len == size));
970 971 972 973 974 975 976

        buf1 += len;
        size -= len;
    }
    s->cur_timestamp++;
}

977 978 979 980 981 982 983 984 985 986 987 988 989 990 991
/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
static void rtp_send_mpegts_raw(AVFormatContext *s1,
                                const uint8_t *buf1, int size)
{
    RTPDemuxContext *s = s1->priv_data;
    int len, out_len;

    while (size >= TS_PACKET_SIZE) {
        len = s->max_payload_size - (s->buf_ptr - s->buf);
        if (len > size)
            len = size;
        memcpy(s->buf_ptr, buf1, len);
        buf1 += len;
        size -= len;
        s->buf_ptr += len;
992

993 994
        out_len = s->buf_ptr - s->buf;
        if (out_len >= s->max_payload_size) {
995
            rtp_send_data(s1, s->buf, out_len, 0);
996 997 998 999 1000
            s->buf_ptr = s->buf;
        }
    }
}

1001
/* write an RTP packet. 'buf1' must contain a single specific frame. */
1002
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
1003
{
1004
    RTPDemuxContext *s = s1->priv_data;
1005 1006
    AVStream *st = s1->streams[0];
    int rtcp_bytes;
1007
    int64_t ntp_time;
1008 1009
    int size= pkt->size;
    uint8_t *buf1= pkt->data;
1010

1011
#ifdef DEBUG
1012
    printf("%d: write len=%d\n", pkt->stream_index, size);
1013 1014 1015
#endif

    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
1016
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
1017 1018 1019
        RTCP_TX_RATIO_DEN;
    if (s->first_packet || rtcp_bytes >= 28) {
        /* compute NTP time */
1020
        /* XXX: 90 kHz timestamp hardcoded */
1021
        ntp_time = (pkt->pts << 28) / 5625;
1022
        rtcp_send_sr(s1, ntp_time);
1023 1024 1025 1026
        s->last_octet_count = s->octet_count;
        s->first_packet = 0;
    }

1027
    switch(st->codec->codec_id) {
1028 1029 1030 1031
    case CODEC_ID_PCM_MULAW:
    case CODEC_ID_PCM_ALAW:
    case CODEC_ID_PCM_U8:
    case CODEC_ID_PCM_S8:
1032
        rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
1033 1034 1035 1036 1037
        break;
    case CODEC_ID_PCM_U16BE:
    case CODEC_ID_PCM_U16LE:
    case CODEC_ID_PCM_S16BE:
    case CODEC_ID_PCM_S16LE:
1038
        rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
1039 1040
        break;
    case CODEC_ID_MP2:
Fabrice Bellard's avatar
Fabrice Bellard committed
1041
    case CODEC_ID_MP3:
1042 1043 1044 1045 1046
        rtp_send_mpegaudio(s1, buf1, size);
        break;
    case CODEC_ID_MPEG1VIDEO:
        rtp_send_mpegvideo(s1, buf1, size);
        break;
1047 1048 1049
    case CODEC_ID_MPEG2TS:
        rtp_send_mpegts_raw(s1, buf1, size);
        break;
1050
    default:
1051 1052 1053
        /* better than nothing : send the codec raw data */
        rtp_send_raw(s1, buf1, size);
        break;
1054 1055 1056 1057
    }
    return 0;
}

1058
AVOutputFormat rtp_muxer = {
1059 1060 1061 1062
    "rtp",
    "RTP output format",
    NULL,
    NULL,
1063
    sizeof(RTPDemuxContext),
1064 1065 1066 1067 1068
    CODEC_ID_PCM_MULAW,
    CODEC_ID_NONE,
    rtp_write_header,
    rtp_write_packet,
};