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ffmpeg / libavformat / rtpdec.c @ 72415b2a

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1
/*
2
 * RTP input format
3
 * Copyright (c) 2002 Fabrice Bellard
4
 *
5
 * This file is part of FFmpeg.
6
 *
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 * FFmpeg is free software; you can redistribute it and/or
8
 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
10
 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * 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
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
21

    
22
/* needed for gethostname() */
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#define _XOPEN_SOURCE 600
24

    
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#include "libavcodec/get_bits.h"
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#include "avformat.h"
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#include "mpegts.h"
28

    
29
#include <unistd.h>
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#include "network.h"
31

    
32
#include "rtpdec.h"
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#include "rtpdec_amr.h"
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#include "rtpdec_asf.h"
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#include "rtpdec_h263.h"
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#include "rtpdec_h264.h"
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#include "rtpdec_vorbis.h"
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#include "rtpdec_theora.h"
39

    
40
//#define DEBUG
41

    
42
/* TODO: - add RTCP statistics reporting (should be optional).
43

44
         - add support for h263/mpeg4 packetized output : IDEA: send a
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         buffer to 'rtp_write_packet' contains all the packets for ONE
46
         frame. Each packet should have a four byte header containing
47
         the length in big endian format (same trick as
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         'url_open_dyn_packet_buf')
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*/
50

    
51
/* statistics functions */
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RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
53

    
54
static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", AVMEDIA_TYPE_VIDEO, CODEC_ID_MPEG4};
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static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", AVMEDIA_TYPE_AUDIO, CODEC_ID_AAC};
56

    
57
void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
58
{
59
    handler->next= RTPFirstDynamicPayloadHandler;
60
    RTPFirstDynamicPayloadHandler= handler;
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}
62

    
63
void av_register_rtp_dynamic_payload_handlers(void)
64
{
65
    ff_register_dynamic_payload_handler(&mp4v_es_handler);
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    ff_register_dynamic_payload_handler(&mpeg4_generic_handler);
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    ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
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    ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
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    ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
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    ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
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    ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
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    ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
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    ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
74

    
75
    ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
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    ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
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}
78

    
79
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
80
{
81
    if (buf[1] != 200)
82
        return -1;
83
    s->last_rtcp_ntp_time = AV_RB64(buf + 8);
84
    s->last_rtcp_timestamp = AV_RB32(buf + 16);
85
    return 0;
86
}
87

    
88
#define RTP_SEQ_MOD (1<<16)
89

    
90
/**
91
* called on parse open packet
92
*/
93
static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
94
{
95
    memset(s, 0, sizeof(RTPStatistics));
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    s->max_seq= base_sequence;
97
    s->probation= 1;
98
}
99

    
100
/**
101
* called whenever there is a large jump in sequence numbers, or when they get out of probation...
102
*/
103
static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
104
{
105
    s->max_seq= seq;
106
    s->cycles= 0;
107
    s->base_seq= seq -1;
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    s->bad_seq= RTP_SEQ_MOD + 1;
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    s->received= 0;
110
    s->expected_prior= 0;
111
    s->received_prior= 0;
112
    s->jitter= 0;
113
    s->transit= 0;
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}
115

    
116
/**
117
* returns 1 if we should handle this packet.
118
*/
119
static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
120
{
121
    uint16_t udelta= seq - s->max_seq;
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    const int MAX_DROPOUT= 3000;
123
    const int MAX_MISORDER = 100;
124
    const int MIN_SEQUENTIAL = 2;
125

    
126
    /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
127
    if(s->probation)
128
    {
129
        if(seq==s->max_seq + 1) {
130
            s->probation--;
131
            s->max_seq= seq;
132
            if(s->probation==0) {
133
                rtp_init_sequence(s, seq);
134
                s->received++;
135
                return 1;
136
            }
137
        } else {
138
            s->probation= MIN_SEQUENTIAL - 1;
139
            s->max_seq = seq;
140
        }
141
    } else if (udelta < MAX_DROPOUT) {
142
        // in order, with permissible gap
143
        if(seq < s->max_seq) {
144
            //sequence number wrapped; count antother 64k cycles
145
            s->cycles += RTP_SEQ_MOD;
146
        }
147
        s->max_seq= seq;
148
    } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
149
        // sequence made a large jump...
150
        if(seq==s->bad_seq) {
151
            // two sequential packets-- assume that the other side restarted without telling us; just resync.
152
            rtp_init_sequence(s, seq);
153
        } else {
154
            s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
155
            return 0;
156
        }
157
    } else {
158
        // duplicate or reordered packet...
159
    }
160
    s->received++;
161
    return 1;
162
}
163

    
164
#if 0
165
/**
166
* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
167
* difference between the arrival and sent timestamp.  As a result, the jitter and transit statistics values
168
* never change.  I left this in in case someone else can see a way. (rdm)
169
*/
170
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
171
{
172
    uint32_t transit= arrival_timestamp - sent_timestamp;
173
    int d;
174
    s->transit= transit;
175
    d= FFABS(transit - s->transit);
176
    s->jitter += d - ((s->jitter + 8)>>4);
177
}
178
#endif
179

    
180
int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
181
{
182
    ByteIOContext *pb;
183
    uint8_t *buf;
184
    int len;
185
    int rtcp_bytes;
186
    RTPStatistics *stats= &s->statistics;
187
    uint32_t lost;
188
    uint32_t extended_max;
189
    uint32_t expected_interval;
190
    uint32_t received_interval;
191
    uint32_t lost_interval;
192
    uint32_t expected;
193
    uint32_t fraction;
194
    uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
195

    
196
    if (!s->rtp_ctx || (count < 1))
197
        return -1;
198

    
199
    /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
200
    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
201
    s->octet_count += count;
202
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
203
        RTCP_TX_RATIO_DEN;
204
    rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
205
    if (rtcp_bytes < 28)
206
        return -1;
207
    s->last_octet_count = s->octet_count;
208

    
209
    if (url_open_dyn_buf(&pb) < 0)
210
        return -1;
211

    
212
    // Receiver Report
213
    put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
214
    put_byte(pb, 201);
215
    put_be16(pb, 7); /* length in words - 1 */
216
    put_be32(pb, s->ssrc); // our own SSRC
217
    put_be32(pb, s->ssrc); // XXX: should be the server's here!
218
    // some placeholders we should really fill...
219
    // RFC 1889/p64
220
    extended_max= stats->cycles + stats->max_seq;
221
    expected= extended_max - stats->base_seq + 1;
222
    lost= expected - stats->received;
223
    lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
224
    expected_interval= expected - stats->expected_prior;
225
    stats->expected_prior= expected;
226
    received_interval= stats->received - stats->received_prior;
227
    stats->received_prior= stats->received;
228
    lost_interval= expected_interval - received_interval;
229
    if (expected_interval==0 || lost_interval<=0) fraction= 0;
230
    else fraction = (lost_interval<<8)/expected_interval;
231

    
232
    fraction= (fraction<<24) | lost;
233

    
234
    put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
235
    put_be32(pb, extended_max); /* max sequence received */
236
    put_be32(pb, stats->jitter>>4); /* jitter */
237

    
238
    if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
239
    {
240
        put_be32(pb, 0); /* last SR timestamp */
241
        put_be32(pb, 0); /* delay since last SR */
242
    } else {
243
        uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
244
        uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
245

    
246
        put_be32(pb, middle_32_bits); /* last SR timestamp */
247
        put_be32(pb, delay_since_last); /* delay since last SR */
248
    }
249

    
250
    // CNAME
251
    put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
252
    put_byte(pb, 202);
253
    len = strlen(s->hostname);
254
    put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
255
    put_be32(pb, s->ssrc);
256
    put_byte(pb, 0x01);
257
    put_byte(pb, len);
258
    put_buffer(pb, s->hostname, len);
259
    // padding
260
    for (len = (6 + len) % 4; len % 4; len++) {
261
        put_byte(pb, 0);
262
    }
263

    
264
    put_flush_packet(pb);
265
    len = url_close_dyn_buf(pb, &buf);
266
    if ((len > 0) && buf) {
267
        int result;
268
        dprintf(s->ic, "sending %d bytes of RR\n", len);
269
        result= url_write(s->rtp_ctx, buf, len);
270
        dprintf(s->ic, "result from url_write: %d\n", result);
271
        av_free(buf);
272
    }
273
    return 0;
274
}
275

    
276
void rtp_send_punch_packets(URLContext* rtp_handle)
277
{
278
    ByteIOContext *pb;
279
    uint8_t *buf;
280
    int len;
281

    
282
    /* Send a small RTP packet */
283
    if (url_open_dyn_buf(&pb) < 0)
284
        return;
285

    
286
    put_byte(pb, (RTP_VERSION << 6));
287
    put_byte(pb, 0); /* Payload type */
288
    put_be16(pb, 0); /* Seq */
289
    put_be32(pb, 0); /* Timestamp */
290
    put_be32(pb, 0); /* SSRC */
291

    
292
    put_flush_packet(pb);
293
    len = url_close_dyn_buf(pb, &buf);
294
    if ((len > 0) && buf)
295
        url_write(rtp_handle, buf, len);
296
    av_free(buf);
297

    
298
    /* Send a minimal RTCP RR */
299
    if (url_open_dyn_buf(&pb) < 0)
300
        return;
301

    
302
    put_byte(pb, (RTP_VERSION << 6));
303
    put_byte(pb, 201); /* receiver report */
304
    put_be16(pb, 1); /* length in words - 1 */
305
    put_be32(pb, 0); /* our own SSRC */
306

    
307
    put_flush_packet(pb);
308
    len = url_close_dyn_buf(pb, &buf);
309
    if ((len > 0) && buf)
310
        url_write(rtp_handle, buf, len);
311
    av_free(buf);
312
}
313

    
314

    
315
/**
316
 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
317
 * MPEG2TS streams to indicate that they should be demuxed inside the
318
 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
319
 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
320
 */
321
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data)
322
{
323
    RTPDemuxContext *s;
324

    
325
    s = av_mallocz(sizeof(RTPDemuxContext));
326
    if (!s)
327
        return NULL;
328
    s->payload_type = payload_type;
329
    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
330
    s->ic = s1;
331
    s->st = st;
332
    s->rtp_payload_data = rtp_payload_data;
333
    rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
334
    if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
335
        s->ts = ff_mpegts_parse_open(s->ic);
336
        if (s->ts == NULL) {
337
            av_free(s);
338
            return NULL;
339
        }
340
    } else {
341
        av_set_pts_info(st, 32, 1, 90000);
342
        switch(st->codec->codec_id) {
343
        case CODEC_ID_MPEG1VIDEO:
344
        case CODEC_ID_MPEG2VIDEO:
345
        case CODEC_ID_MP2:
346
        case CODEC_ID_MP3:
347
        case CODEC_ID_MPEG4:
348
        case CODEC_ID_H263:
349
        case CODEC_ID_H264:
350
            st->need_parsing = AVSTREAM_PARSE_FULL;
351
            break;
352
        default:
353
            if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
354
                av_set_pts_info(st, 32, 1, st->codec->sample_rate);
355
            }
356
            break;
357
        }
358
    }
359
    // needed to send back RTCP RR in RTSP sessions
360
    s->rtp_ctx = rtpc;
361
    gethostname(s->hostname, sizeof(s->hostname));
362
    return s;
363
}
364

    
365
void
366
rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
367
                               RTPDynamicProtocolHandler *handler)
368
{
369
    s->dynamic_protocol_context = ctx;
370
    s->parse_packet = handler->parse_packet;
371
}
372

    
373
static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
374
{
375
    int au_headers_length, au_header_size, i;
376
    GetBitContext getbitcontext;
377
    RTPPayloadData *infos;
378

    
379
    infos = s->rtp_payload_data;
380

    
381
    if (infos == NULL)
382
        return -1;
383

    
384
    /* decode the first 2 bytes where the AUHeader sections are stored
385
       length in bits */
386
    au_headers_length = AV_RB16(buf);
387

    
388
    if (au_headers_length > RTP_MAX_PACKET_LENGTH)
389
      return -1;
390

    
391
    infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
392

    
393
    /* skip AU headers length section (2 bytes) */
394
    buf += 2;
395

    
396
    init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
397

    
398
    /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
399
    au_header_size = infos->sizelength + infos->indexlength;
400
    if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
401
        return -1;
402

    
403
    infos->nb_au_headers = au_headers_length / au_header_size;
404
    infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
405

    
406
    /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
407
       In my test, the FAAD decoder does not behave correctly when sending each AU one by one
408
       but does when sending the whole as one big packet...  */
409
    infos->au_headers[0].size = 0;
410
    infos->au_headers[0].index = 0;
411
    for (i = 0; i < infos->nb_au_headers; ++i) {
412
        infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
413
        infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
414
    }
415

    
416
    infos->nb_au_headers = 1;
417

    
418
    return 0;
419
}
420

    
421
/**
422
 * This was the second switch in rtp_parse packet.  Normalizes time, if required, sets stream_index, etc.
423
 */
424
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
425
{
426
    if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
427
        int64_t addend;
428
        int delta_timestamp;
429

    
430
        /* compute pts from timestamp with received ntp_time */
431
        delta_timestamp = timestamp - s->last_rtcp_timestamp;
432
        /* convert to the PTS timebase */
433
        addend = av_rescale(s->last_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);
434
        pkt->pts = addend + delta_timestamp;
435
    }
436
}
437

    
438
/**
439
 * Parse an RTP or RTCP packet directly sent as a buffer.
440
 * @param s RTP parse context.
441
 * @param pkt returned packet
442
 * @param buf input buffer or NULL to read the next packets
443
 * @param len buffer len
444
 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
445
 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
446
 */
447
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
448
                     const uint8_t *buf, int len)
449
{
450
    unsigned int ssrc, h;
451
    int payload_type, seq, ret, flags = 0;
452
    AVStream *st;
453
    uint32_t timestamp;
454
    int rv= 0;
455

    
456
    if (!buf) {
457
        /* return the next packets, if any */
458
        if(s->st && s->parse_packet) {
459
            timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
460
            rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
461
                                s->st, pkt, &timestamp, NULL, 0, flags);
462
            finalize_packet(s, pkt, timestamp);
463
            return rv;
464
        } else {
465
            // TODO: Move to a dynamic packet handler (like above)
466
            if (s->read_buf_index >= s->read_buf_size)
467
                return -1;
468
            ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
469
                                      s->read_buf_size - s->read_buf_index);
470
            if (ret < 0)
471
                return -1;
472
            s->read_buf_index += ret;
473
            if (s->read_buf_index < s->read_buf_size)
474
                return 1;
475
            else
476
                return 0;
477
        }
478
    }
479

    
480
    if (len < 12)
481
        return -1;
482

    
483
    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
484
        return -1;
485
    if (buf[1] >= 200 && buf[1] <= 204) {
486
        rtcp_parse_packet(s, buf, len);
487
        return -1;
488
    }
489
    payload_type = buf[1] & 0x7f;
490
    if (buf[1] & 0x80)
491
        flags |= RTP_FLAG_MARKER;
492
    seq  = AV_RB16(buf + 2);
493
    timestamp = AV_RB32(buf + 4);
494
    ssrc = AV_RB32(buf + 8);
495
    /* store the ssrc in the RTPDemuxContext */
496
    s->ssrc = ssrc;
497

    
498
    /* NOTE: we can handle only one payload type */
499
    if (s->payload_type != payload_type)
500
        return -1;
501

    
502
    st = s->st;
503
    // only do something with this if all the rtp checks pass...
504
    if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
505
    {
506
        av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
507
               payload_type, seq, ((s->seq + 1) & 0xffff));
508
        return -1;
509
    }
510

    
511
    s->seq = seq;
512
    len -= 12;
513
    buf += 12;
514

    
515
    if (!st) {
516
        /* specific MPEG2TS demux support */
517
        ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
518
        if (ret < 0)
519
            return -1;
520
        if (ret < len) {
521
            s->read_buf_size = len - ret;
522
            memcpy(s->buf, buf + ret, s->read_buf_size);
523
            s->read_buf_index = 0;
524
            return 1;
525
        }
526
        return 0;
527
    } else if (s->parse_packet) {
528
        rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
529
                             s->st, pkt, &timestamp, buf, len, flags);
530
    } else {
531
        // at this point, the RTP header has been stripped;  This is ASSUMING that there is only 1 CSRC, which in't wise.
532
        switch(st->codec->codec_id) {
533
        case CODEC_ID_MP2:
534
        case CODEC_ID_MP3:
535
            /* better than nothing: skip mpeg audio RTP header */
536
            if (len <= 4)
537
                return -1;
538
            h = AV_RB32(buf);
539
            len -= 4;
540
            buf += 4;
541
            av_new_packet(pkt, len);
542
            memcpy(pkt->data, buf, len);
543
            break;
544
        case CODEC_ID_MPEG1VIDEO:
545
        case CODEC_ID_MPEG2VIDEO:
546
            /* better than nothing: skip mpeg video RTP header */
547
            if (len <= 4)
548
                return -1;
549
            h = AV_RB32(buf);
550
            buf += 4;
551
            len -= 4;
552
            if (h & (1 << 26)) {
553
                /* mpeg2 */
554
                if (len <= 4)
555
                    return -1;
556
                buf += 4;
557
                len -= 4;
558
            }
559
            av_new_packet(pkt, len);
560
            memcpy(pkt->data, buf, len);
561
            break;
562
            // moved from below, verbatim.  this is because this section handles packets, and the lower switch handles
563
            // timestamps.
564
            // TODO: Put this into a dynamic packet handler...
565
        case CODEC_ID_AAC:
566
            if (rtp_parse_mp4_au(s, buf))
567
                return -1;
568
            {
569
                RTPPayloadData *infos = s->rtp_payload_data;
570
                if (infos == NULL)
571
                    return -1;
572
                buf += infos->au_headers_length_bytes + 2;
573
                len -= infos->au_headers_length_bytes + 2;
574

    
575
                /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
576
                    one au_header */
577
                av_new_packet(pkt, infos->au_headers[0].size);
578
                memcpy(pkt->data, buf, infos->au_headers[0].size);
579
                buf += infos->au_headers[0].size;
580
                len -= infos->au_headers[0].size;
581
            }
582
            s->read_buf_size = len;
583
            rv= 0;
584
            break;
585
        default:
586
            av_new_packet(pkt, len);
587
            memcpy(pkt->data, buf, len);
588
            break;
589
        }
590

    
591
        pkt->stream_index = st->index;
592
    }
593

    
594
    // now perform timestamp things....
595
    finalize_packet(s, pkt, timestamp);
596

    
597
    return rv;
598
}
599

    
600
void rtp_parse_close(RTPDemuxContext *s)
601
{
602
    // TODO: fold this into the protocol specific data fields.
603
    if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
604
        ff_mpegts_parse_close(s->ts);
605
    }
606
    av_free(s);
607
}