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1
/*
2
 * RTP input format
3
 * Copyright (c) 2002 Fabrice Bellard
4
 *
5
 * This file is part of FFmpeg.
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 *
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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
9
 * 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

    
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/* 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 <strings.h>
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#include "network.h"
32

    
33
#include "rtpdec.h"
34
#include "rtpdec_formats.h"
35

    
36
//#define DEBUG
37

    
38
/* TODO: - add RTCP statistics reporting (should be optional).
39

40
         - 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
42
         frame. Each packet should have a four byte header containing
43
         the length in big endian format (same trick as
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         'url_open_dyn_packet_buf')
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*/
46

    
47
RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
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    .enc_name           = "X-MP3-draft-00",
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    .codec_type         = AVMEDIA_TYPE_AUDIO,
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    .codec_id           = CODEC_ID_MP3ADU,
51
};
52

    
53
/* statistics functions */
54
RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
55

    
56
void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
57
{
58
    handler->next= RTPFirstDynamicPayloadHandler;
59
    RTPFirstDynamicPayloadHandler= handler;
60
}
61

    
62
void av_register_rtp_dynamic_payload_handlers(void)
63
{
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    ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
65
    ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_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);
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    ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
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    ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
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    ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
76
    ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
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    ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
78
    ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);
79

    
80
    ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
81
    ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
82

    
83
    ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
84
    ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
85
    ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
86
    ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
87
}
88

    
89
RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
90
                                                  enum AVMediaType codec_type)
91
{
92
    RTPDynamicProtocolHandler *handler;
93
    for (handler = RTPFirstDynamicPayloadHandler;
94
         handler; handler = handler->next)
95
        if (!strcasecmp(name, handler->enc_name) &&
96
            codec_type == handler->codec_type)
97
            return handler;
98
    return NULL;
99
}
100

    
101
RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
102
                                                enum AVMediaType codec_type)
103
{
104
    RTPDynamicProtocolHandler *handler;
105
    for (handler = RTPFirstDynamicPayloadHandler;
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         handler; handler = handler->next)
107
        if (handler->static_payload_id && handler->static_payload_id == id &&
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            codec_type == handler->codec_type)
109
            return handler;
110
    return NULL;
111
}
112

    
113
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
114
{
115
    int payload_len;
116
    while (len >= 2) {
117
        switch (buf[1]) {
118
        case RTCP_SR:
119
            if (len < 16) {
120
                av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
121
                return AVERROR_INVALIDDATA;
122
            }
123
            payload_len = (AV_RB16(buf + 2) + 1) * 4;
124

    
125
            s->last_rtcp_ntp_time = AV_RB64(buf + 8);
126
            s->last_rtcp_timestamp = AV_RB32(buf + 16);
127
            if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
128
                s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
129
                if (!s->base_timestamp)
130
                    s->base_timestamp = s->last_rtcp_timestamp;
131
                s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
132
            }
133

    
134
            buf += payload_len;
135
            len -= payload_len;
136
            break;
137
        case RTCP_BYE:
138
            return -RTCP_BYE;
139
        default:
140
            return -1;
141
        }
142
    }
143
    return -1;
144
}
145

    
146
#define RTP_SEQ_MOD (1<<16)
147

    
148
/**
149
* called on parse open packet
150
*/
151
static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
152
{
153
    memset(s, 0, sizeof(RTPStatistics));
154
    s->max_seq= base_sequence;
155
    s->probation= 1;
156
}
157

    
158
/**
159
* called whenever there is a large jump in sequence numbers, or when they get out of probation...
160
*/
161
static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
162
{
163
    s->max_seq= seq;
164
    s->cycles= 0;
165
    s->base_seq= seq -1;
166
    s->bad_seq= RTP_SEQ_MOD + 1;
167
    s->received= 0;
168
    s->expected_prior= 0;
169
    s->received_prior= 0;
170
    s->jitter= 0;
171
    s->transit= 0;
172
}
173

    
174
/**
175
* returns 1 if we should handle this packet.
176
*/
177
static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
178
{
179
    uint16_t udelta= seq - s->max_seq;
180
    const int MAX_DROPOUT= 3000;
181
    const int MAX_MISORDER = 100;
182
    const int MIN_SEQUENTIAL = 2;
183

    
184
    /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
185
    if(s->probation)
186
    {
187
        if(seq==s->max_seq + 1) {
188
            s->probation--;
189
            s->max_seq= seq;
190
            if(s->probation==0) {
191
                rtp_init_sequence(s, seq);
192
                s->received++;
193
                return 1;
194
            }
195
        } else {
196
            s->probation= MIN_SEQUENTIAL - 1;
197
            s->max_seq = seq;
198
        }
199
    } else if (udelta < MAX_DROPOUT) {
200
        // in order, with permissible gap
201
        if(seq < s->max_seq) {
202
            //sequence number wrapped; count antother 64k cycles
203
            s->cycles += RTP_SEQ_MOD;
204
        }
205
        s->max_seq= seq;
206
    } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
207
        // sequence made a large jump...
208
        if(seq==s->bad_seq) {
209
            // two sequential packets-- assume that the other side restarted without telling us; just resync.
210
            rtp_init_sequence(s, seq);
211
        } else {
212
            s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
213
            return 0;
214
        }
215
    } else {
216
        // duplicate or reordered packet...
217
    }
218
    s->received++;
219
    return 1;
220
}
221

    
222
#if 0
223
/**
224
* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
225
* difference between the arrival and sent timestamp.  As a result, the jitter and transit statistics values
226
* never change.  I left this in in case someone else can see a way. (rdm)
227
*/
228
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
229
{
230
    uint32_t transit= arrival_timestamp - sent_timestamp;
231
    int d;
232
    s->transit= transit;
233
    d= FFABS(transit - s->transit);
234
    s->jitter += d - ((s->jitter + 8)>>4);
235
}
236
#endif
237

    
238
int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
239
{
240
    ByteIOContext *pb;
241
    uint8_t *buf;
242
    int len;
243
    int rtcp_bytes;
244
    RTPStatistics *stats= &s->statistics;
245
    uint32_t lost;
246
    uint32_t extended_max;
247
    uint32_t expected_interval;
248
    uint32_t received_interval;
249
    uint32_t lost_interval;
250
    uint32_t expected;
251
    uint32_t fraction;
252
    uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
253

    
254
    if (!s->rtp_ctx || (count < 1))
255
        return -1;
256

    
257
    /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
258
    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
259
    s->octet_count += count;
260
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
261
        RTCP_TX_RATIO_DEN;
262
    rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
263
    if (rtcp_bytes < 28)
264
        return -1;
265
    s->last_octet_count = s->octet_count;
266

    
267
    if (url_open_dyn_buf(&pb) < 0)
268
        return -1;
269

    
270
    // Receiver Report
271
    put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
272
    put_byte(pb, RTCP_RR);
273
    put_be16(pb, 7); /* length in words - 1 */
274
    // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
275
    put_be32(pb, s->ssrc + 1);
276
    put_be32(pb, s->ssrc); // server SSRC
277
    // some placeholders we should really fill...
278
    // RFC 1889/p64
279
    extended_max= stats->cycles + stats->max_seq;
280
    expected= extended_max - stats->base_seq + 1;
281
    lost= expected - stats->received;
282
    lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
283
    expected_interval= expected - stats->expected_prior;
284
    stats->expected_prior= expected;
285
    received_interval= stats->received - stats->received_prior;
286
    stats->received_prior= stats->received;
287
    lost_interval= expected_interval - received_interval;
288
    if (expected_interval==0 || lost_interval<=0) fraction= 0;
289
    else fraction = (lost_interval<<8)/expected_interval;
290

    
291
    fraction= (fraction<<24) | lost;
292

    
293
    put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
294
    put_be32(pb, extended_max); /* max sequence received */
295
    put_be32(pb, stats->jitter>>4); /* jitter */
296

    
297
    if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
298
    {
299
        put_be32(pb, 0); /* last SR timestamp */
300
        put_be32(pb, 0); /* delay since last SR */
301
    } else {
302
        uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
303
        uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
304

    
305
        put_be32(pb, middle_32_bits); /* last SR timestamp */
306
        put_be32(pb, delay_since_last); /* delay since last SR */
307
    }
308

    
309
    // CNAME
310
    put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
311
    put_byte(pb, RTCP_SDES);
312
    len = strlen(s->hostname);
313
    put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
314
    put_be32(pb, s->ssrc);
315
    put_byte(pb, 0x01);
316
    put_byte(pb, len);
317
    put_buffer(pb, s->hostname, len);
318
    // padding
319
    for (len = (6 + len) % 4; len % 4; len++) {
320
        put_byte(pb, 0);
321
    }
322

    
323
    put_flush_packet(pb);
324
    len = url_close_dyn_buf(pb, &buf);
325
    if ((len > 0) && buf) {
326
        int result;
327
        dprintf(s->ic, "sending %d bytes of RR\n", len);
328
        result= url_write(s->rtp_ctx, buf, len);
329
        dprintf(s->ic, "result from url_write: %d\n", result);
330
        av_free(buf);
331
    }
332
    return 0;
333
}
334

    
335
void rtp_send_punch_packets(URLContext* rtp_handle)
336
{
337
    ByteIOContext *pb;
338
    uint8_t *buf;
339
    int len;
340

    
341
    /* Send a small RTP packet */
342
    if (url_open_dyn_buf(&pb) < 0)
343
        return;
344

    
345
    put_byte(pb, (RTP_VERSION << 6));
346
    put_byte(pb, 0); /* Payload type */
347
    put_be16(pb, 0); /* Seq */
348
    put_be32(pb, 0); /* Timestamp */
349
    put_be32(pb, 0); /* SSRC */
350

    
351
    put_flush_packet(pb);
352
    len = url_close_dyn_buf(pb, &buf);
353
    if ((len > 0) && buf)
354
        url_write(rtp_handle, buf, len);
355
    av_free(buf);
356

    
357
    /* Send a minimal RTCP RR */
358
    if (url_open_dyn_buf(&pb) < 0)
359
        return;
360

    
361
    put_byte(pb, (RTP_VERSION << 6));
362
    put_byte(pb, RTCP_RR); /* receiver report */
363
    put_be16(pb, 1); /* length in words - 1 */
364
    put_be32(pb, 0); /* our own SSRC */
365

    
366
    put_flush_packet(pb);
367
    len = url_close_dyn_buf(pb, &buf);
368
    if ((len > 0) && buf)
369
        url_write(rtp_handle, buf, len);
370
    av_free(buf);
371
}
372

    
373

    
374
/**
375
 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
376
 * MPEG2TS streams to indicate that they should be demuxed inside the
377
 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
378
 */
379
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
380
{
381
    RTPDemuxContext *s;
382

    
383
    s = av_mallocz(sizeof(RTPDemuxContext));
384
    if (!s)
385
        return NULL;
386
    s->payload_type = payload_type;
387
    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
388
    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
389
    s->ic = s1;
390
    s->st = st;
391
    s->queue_size = queue_size;
392
    rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
393
    if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
394
        s->ts = ff_mpegts_parse_open(s->ic);
395
        if (s->ts == NULL) {
396
            av_free(s);
397
            return NULL;
398
        }
399
    } else {
400
        switch(st->codec->codec_id) {
401
        case CODEC_ID_MPEG1VIDEO:
402
        case CODEC_ID_MPEG2VIDEO:
403
        case CODEC_ID_MP2:
404
        case CODEC_ID_MP3:
405
        case CODEC_ID_MPEG4:
406
        case CODEC_ID_H263:
407
        case CODEC_ID_H264:
408
            st->need_parsing = AVSTREAM_PARSE_FULL;
409
            break;
410
        case CODEC_ID_ADPCM_G722:
411
            /* According to RFC 3551, the stream clock rate is 8000
412
             * even if the sample rate is 16000. */
413
            if (st->codec->sample_rate == 8000)
414
                st->codec->sample_rate = 16000;
415
            break;
416
        default:
417
            break;
418
        }
419
    }
420
    // needed to send back RTCP RR in RTSP sessions
421
    s->rtp_ctx = rtpc;
422
    gethostname(s->hostname, sizeof(s->hostname));
423
    return s;
424
}
425

    
426
void
427
rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
428
                               RTPDynamicProtocolHandler *handler)
429
{
430
    s->dynamic_protocol_context = ctx;
431
    s->parse_packet = handler->parse_packet;
432
}
433

    
434
/**
435
 * This was the second switch in rtp_parse packet.  Normalizes time, if required, sets stream_index, etc.
436
 */
437
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
438
{
439
    if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
440
        int64_t addend;
441
        int delta_timestamp;
442

    
443
        /* compute pts from timestamp with received ntp_time */
444
        delta_timestamp = timestamp - s->last_rtcp_timestamp;
445
        /* convert to the PTS timebase */
446
        addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);
447
        pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
448
                   delta_timestamp;
449
        return;
450
    }
451
    if (timestamp == RTP_NOTS_VALUE)
452
        return;
453
    if (!s->base_timestamp)
454
        s->base_timestamp = timestamp;
455
    pkt->pts = s->range_start_offset + timestamp - s->base_timestamp;
456
}
457

    
458
static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
459
                                     const uint8_t *buf, int len)
460
{
461
    unsigned int ssrc, h;
462
    int payload_type, seq, ret, flags = 0;
463
    int ext;
464
    AVStream *st;
465
    uint32_t timestamp;
466
    int rv= 0;
467

    
468
    ext = buf[0] & 0x10;
469
    payload_type = buf[1] & 0x7f;
470
    if (buf[1] & 0x80)
471
        flags |= RTP_FLAG_MARKER;
472
    seq  = AV_RB16(buf + 2);
473
    timestamp = AV_RB32(buf + 4);
474
    ssrc = AV_RB32(buf + 8);
475
    /* store the ssrc in the RTPDemuxContext */
476
    s->ssrc = ssrc;
477

    
478
    /* NOTE: we can handle only one payload type */
479
    if (s->payload_type != payload_type)
480
        return -1;
481

    
482
    st = s->st;
483
    // only do something with this if all the rtp checks pass...
484
    if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
485
    {
486
        av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
487
               payload_type, seq, ((s->seq + 1) & 0xffff));
488
        return -1;
489
    }
490

    
491
    if (buf[0] & 0x20) {
492
        int padding = buf[len - 1];
493
        if (len >= 12 + padding)
494
            len -= padding;
495
    }
496

    
497
    s->seq = seq;
498
    len -= 12;
499
    buf += 12;
500

    
501
    /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
502
    if (ext) {
503
        if (len < 4)
504
            return -1;
505
        /* calculate the header extension length (stored as number
506
         * of 32-bit words) */
507
        ext = (AV_RB16(buf + 2) + 1) << 2;
508

    
509
        if (len < ext)
510
            return -1;
511
        // skip past RTP header extension
512
        len -= ext;
513
        buf += ext;
514
    }
515

    
516
    if (!st) {
517
        /* specific MPEG2TS demux support */
518
        ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
519
        /* The only error that can be returned from ff_mpegts_parse_packet
520
         * is "no more data to return from the provided buffer", so return
521
         * AVERROR(EAGAIN) for all errors */
522
        if (ret < 0)
523
            return AVERROR(EAGAIN);
524
        if (ret < len) {
525
            s->read_buf_size = len - ret;
526
            memcpy(s->buf, buf + ret, s->read_buf_size);
527
            s->read_buf_index = 0;
528
            return 1;
529
        }
530
        return 0;
531
    } else if (s->parse_packet) {
532
        rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
533
                             s->st, pkt, &timestamp, buf, len, flags);
534
    } else {
535
        // at this point, the RTP header has been stripped;  This is ASSUMING that there is only 1 CSRC, which in't wise.
536
        switch(st->codec->codec_id) {
537
        case CODEC_ID_MP2:
538
        case CODEC_ID_MP3:
539
            /* better than nothing: skip mpeg audio RTP header */
540
            if (len <= 4)
541
                return -1;
542
            h = AV_RB32(buf);
543
            len -= 4;
544
            buf += 4;
545
            av_new_packet(pkt, len);
546
            memcpy(pkt->data, buf, len);
547
            break;
548
        case CODEC_ID_MPEG1VIDEO:
549
        case CODEC_ID_MPEG2VIDEO:
550
            /* better than nothing: skip mpeg video RTP header */
551
            if (len <= 4)
552
                return -1;
553
            h = AV_RB32(buf);
554
            buf += 4;
555
            len -= 4;
556
            if (h & (1 << 26)) {
557
                /* mpeg2 */
558
                if (len <= 4)
559
                    return -1;
560
                buf += 4;
561
                len -= 4;
562
            }
563
            av_new_packet(pkt, len);
564
            memcpy(pkt->data, buf, len);
565
            break;
566
        default:
567
            av_new_packet(pkt, len);
568
            memcpy(pkt->data, buf, len);
569
            break;
570
        }
571

    
572
        pkt->stream_index = st->index;
573
    }
574

    
575
    // now perform timestamp things....
576
    finalize_packet(s, pkt, timestamp);
577

    
578
    return rv;
579
}
580

    
581
void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
582
{
583
    while (s->queue) {
584
        RTPPacket *next = s->queue->next;
585
        av_free(s->queue->buf);
586
        av_free(s->queue);
587
        s->queue = next;
588
    }
589
    s->seq       = 0;
590
    s->queue_len = 0;
591
    s->prev_ret  = 0;
592
}
593

    
594
static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
595
{
596
    uint16_t seq = AV_RB16(buf + 2);
597
    RTPPacket *cur = s->queue, *prev = NULL, *packet;
598

    
599
    /* Find the correct place in the queue to insert the packet */
600
    while (cur) {
601
        int16_t diff = seq - cur->seq;
602
        if (diff < 0)
603
            break;
604
        prev = cur;
605
        cur = cur->next;
606
    }
607

    
608
    packet = av_mallocz(sizeof(*packet));
609
    if (!packet)
610
        return;
611
    packet->recvtime = av_gettime();
612
    packet->seq = seq;
613
    packet->len = len;
614
    packet->buf = buf;
615
    packet->next = cur;
616
    if (prev)
617
        prev->next = packet;
618
    else
619
        s->queue = packet;
620
    s->queue_len++;
621
}
622

    
623
static int has_next_packet(RTPDemuxContext *s)
624
{
625
    return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
626
}
627

    
628
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
629
{
630
    return s->queue ? s->queue->recvtime : 0;
631
}
632

    
633
static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
634
{
635
    int rv;
636
    RTPPacket *next;
637

    
638
    if (s->queue_len <= 0)
639
        return -1;
640

    
641
    if (!has_next_packet(s))
642
        av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
643
               "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
644

    
645
    /* Parse the first packet in the queue, and dequeue it */
646
    rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
647
    next = s->queue->next;
648
    av_free(s->queue->buf);
649
    av_free(s->queue);
650
    s->queue = next;
651
    s->queue_len--;
652
    return rv;
653
}
654

    
655
static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
656
                     uint8_t **bufptr, int len)
657
{
658
    uint8_t* buf = bufptr ? *bufptr : NULL;
659
    int ret, flags = 0;
660
    uint32_t timestamp;
661
    int rv= 0;
662

    
663
    if (!buf) {
664
        /* If parsing of the previous packet actually returned 0 or an error,
665
         * there's nothing more to be parsed from that packet, but we may have
666
         * indicated that we can return the next enqueued packet. */
667
        if (s->prev_ret <= 0)
668
            return rtp_parse_queued_packet(s, pkt);
669
        /* return the next packets, if any */
670
        if(s->st && s->parse_packet) {
671
            /* timestamp should be overwritten by parse_packet, if not,
672
             * the packet is left with pts == AV_NOPTS_VALUE */
673
            timestamp = RTP_NOTS_VALUE;
674
            rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
675
                                s->st, pkt, &timestamp, NULL, 0, flags);
676
            finalize_packet(s, pkt, timestamp);
677
            return rv;
678
        } else {
679
            // TODO: Move to a dynamic packet handler (like above)
680
            if (s->read_buf_index >= s->read_buf_size)
681
                return AVERROR(EAGAIN);
682
            ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
683
                                      s->read_buf_size - s->read_buf_index);
684
            if (ret < 0)
685
                return AVERROR(EAGAIN);
686
            s->read_buf_index += ret;
687
            if (s->read_buf_index < s->read_buf_size)
688
                return 1;
689
            else
690
                return 0;
691
        }
692
    }
693

    
694
    if (len < 12)
695
        return -1;
696

    
697
    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
698
        return -1;
699
    if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
700
        return rtcp_parse_packet(s, buf, len);
701
    }
702

    
703
    if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
704
        /* First packet, or no reordering */
705
        return rtp_parse_packet_internal(s, pkt, buf, len);
706
    } else {
707
        uint16_t seq = AV_RB16(buf + 2);
708
        int16_t diff = seq - s->seq;
709
        if (diff < 0) {
710
            /* Packet older than the previously emitted one, drop */
711
            av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
712
                   "RTP: dropping old packet received too late\n");
713
            return -1;
714
        } else if (diff <= 1) {
715
            /* Correct packet */
716
            rv = rtp_parse_packet_internal(s, pkt, buf, len);
717
            return rv;
718
        } else {
719
            /* Still missing some packet, enqueue this one. */
720
            enqueue_packet(s, buf, len);
721
            *bufptr = NULL;
722
            /* Return the first enqueued packet if the queue is full,
723
             * even if we're missing something */
724
            if (s->queue_len >= s->queue_size)
725
                return rtp_parse_queued_packet(s, pkt);
726
            return -1;
727
        }
728
    }
729
}
730

    
731
/**
732
 * Parse an RTP or RTCP packet directly sent as a buffer.
733
 * @param s RTP parse context.
734
 * @param pkt returned packet
735
 * @param bufptr pointer to the input buffer or NULL to read the next packets
736
 * @param len buffer len
737
 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
738
 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
739
 */
740
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
741
                     uint8_t **bufptr, int len)
742
{
743
    int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
744
    s->prev_ret = rv;
745
    while (rv == AVERROR(EAGAIN) && has_next_packet(s))
746
        rv = rtp_parse_queued_packet(s, pkt);
747
    return rv ? rv : has_next_packet(s);
748
}
749

    
750
void rtp_parse_close(RTPDemuxContext *s)
751
{
752
    ff_rtp_reset_packet_queue(s);
753
    if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
754
        ff_mpegts_parse_close(s->ts);
755
    }
756
    av_free(s);
757
}
758

    
759
int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
760
                  int (*parse_fmtp)(AVStream *stream,
761
                                    PayloadContext *data,
762
                                    char *attr, char *value))
763
{
764
    char attr[256];
765
    char *value;
766
    int res;
767
    int value_size = strlen(p) + 1;
768

    
769
    if (!(value = av_malloc(value_size))) {
770
        av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
771
        return AVERROR(ENOMEM);
772
    }
773

    
774
    // remove protocol identifier
775
    while (*p && *p == ' ') p++; // strip spaces
776
    while (*p && *p != ' ') p++; // eat protocol identifier
777
    while (*p && *p == ' ') p++; // strip trailing spaces
778

    
779
    while (ff_rtsp_next_attr_and_value(&p,
780
                                       attr, sizeof(attr),
781
                                       value, value_size)) {
782

    
783
        res = parse_fmtp(stream, data, attr, value);
784
        if (res < 0 && res != AVERROR_PATCHWELCOME) {
785
            av_free(value);
786
            return res;
787
        }
788
    }
789
    av_free(value);
790
    return 0;
791
}