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ffmpeg / libavformat / rtpdec.c @ dfd2a005

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1 8eb793c4 Luca Abeni
/*
2
 * RTP input format
3 406792e7 Diego Biurrun
 * Copyright (c) 2002 Fabrice Bellard
4 8eb793c4 Luca Abeni
 *
5
 * This file is part of FFmpeg.
6
 *
7
 * FFmpeg is free software; you can redistribute it and/or
8
 * modify it under the terms of the GNU Lesser General Public
9
 * License as published by the Free Software Foundation; either
10
 * version 2.1 of the License, or (at your option) any later version.
11
 *
12
 * FFmpeg is distributed in the hope that it will be useful,
13
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
15
 * Lesser General Public License for more details.
16
 *
17
 * You should have received a copy of the GNU Lesser General Public
18
 * License along with FFmpeg; if not, write to the Free Software
19
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20
 */
21 245976da Diego Biurrun
22 7246177d Aurelien Jacobs
/* needed for gethostname() */
23 d0feff2a Diego Biurrun
#define _XOPEN_SOURCE 600
24 7246177d Aurelien Jacobs
25 9106a698 Stefano Sabatini
#include "libavcodec/get_bits.h"
26 8eb793c4 Luca Abeni
#include "avformat.h"
27
#include "mpegts.h"
28
29
#include <unistd.h>
30 1e515c42 Martin Storsjö
#include <strings.h>
31 8eb793c4 Luca Abeni
#include "network.h"
32
33 302879cb Luca Abeni
#include "rtpdec.h"
34 965a3ddb Martin Storsjö
#include "rtpdec_formats.h"
35 8eb793c4 Luca Abeni
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
41
         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
44
         'url_open_dyn_packet_buf')
45
*/
46
47 69ad22c7 Diego Elio Pettenò
static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
48 2eeefe20 Martin Storsjö
    .enc_name           = "X-MP3-draft-00",
49
    .codec_type         = AVMEDIA_TYPE_AUDIO,
50
    .codec_id           = CODEC_ID_MP3ADU,
51
};
52
53 8eb793c4 Luca Abeni
/* statistics functions */
54 119cc033 Diego Elio Pettenò
static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
55 8eb793c4 Luca Abeni
56 0369d2b0 Ronald S. Bultje
void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
57 8eb793c4 Luca Abeni
{
58
    handler->next= RTPFirstDynamicPayloadHandler;
59
    RTPFirstDynamicPayloadHandler= handler;
60
}
61
62
void av_register_rtp_dynamic_payload_handlers(void)
63
{
64 9b3788ef Josh Allmann
    ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
65
    ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
66 556aa7a1 Ronald S. Bultje
    ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
67
    ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
68 45aa9080 Ronald S. Bultje
    ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
69
    ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
70 0369d2b0 Ronald S. Bultje
    ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
71 e6327fba Ronald S. Bultje
    ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
72 887af2aa Josh Allmann
    ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
73 a59096e4 Josh Allmann
    ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
74 4449df6b Josh Allmann
    ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
75 1ddc176e Martin Storsjö
    ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
76 51291e60 Josh Allmann
    ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
77 35014efc Martin Storsjö
    ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
78 2eeefe20 Martin Storsjö
    ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);
79 e9fce261 Ronald S. Bultje
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 3ece3e4c Martin Storsjö
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 8eb793c4 Luca Abeni
}
88
89 1e515c42 Martin Storsjö
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;
106
         handler; handler = handler->next)
107
        if (handler->static_payload_id && handler->static_payload_id == id &&
108
            codec_type == handler->codec_type)
109
            return handler;
110
    return NULL;
111
}
112
113 8eb793c4 Luca Abeni
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
114
{
115 ff328c02 Josh Allmann
    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 682d28a9 Josh Allmann
            s->last_rtcp_ntp_time = AV_RB64(buf + 8);
126
            s->last_rtcp_timestamp = AV_RB32(buf + 16);
127 3a1cdcc7 Martin Storsjö
            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 ff328c02 Josh Allmann
134
            buf += payload_len;
135
            len -= payload_len;
136
            break;
137 b20359f5 Josh Allmann
        case RTCP_BYE:
138
            return -RTCP_BYE;
139 ff328c02 Josh Allmann
        default:
140
            return -1;
141
        }
142
    }
143 b20359f5 Josh Allmann
    return -1;
144 8eb793c4 Luca Abeni
}
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 7f3468d3 Josh Allmann
    put_byte(pb, RTCP_RR);
273 8eb793c4 Luca Abeni
    put_be16(pb, 7); /* length in words - 1 */
274 952139a3 Luca Abeni
    // 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 8eb793c4 Luca Abeni
    // 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 7f3468d3 Josh Allmann
    put_byte(pb, RTCP_SDES);
312 8eb793c4 Luca Abeni
    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 dfd2a005 Luca Barbato
        av_dlog(s->ic, "sending %d bytes of RR\n", len);
328 8eb793c4 Luca Abeni
        result= url_write(s->rtp_ctx, buf, len);
329 dfd2a005 Luca Barbato
        av_dlog(s->ic, "result from url_write: %d\n", result);
330 8eb793c4 Luca Abeni
        av_free(buf);
331
    }
332
    return 0;
333
}
334
335 9c8fa20d Martin Storsjö
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 7f3468d3 Josh Allmann
    put_byte(pb, RTCP_RR); /* receiver report */
363 9c8fa20d Martin Storsjö
    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 8eb793c4 Luca Abeni
/**
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 58ee0991 Martin Storsjö
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
380 8eb793c4 Luca Abeni
{
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 2cab6b48 Martin Storsjö
    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
389 8eb793c4 Luca Abeni
    s->ic = s1;
390
    s->st = st;
391 58ee0991 Martin Storsjö
    s->queue_size = queue_size;
392 8eb793c4 Luca Abeni
    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 9125806e Alexis Ballier
        s->ts = ff_mpegts_parse_open(s->ic);
395 8eb793c4 Luca Abeni
        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 45aa9080 Ronald S. Bultje
        case CODEC_ID_H263:
407 8eb793c4 Luca Abeni
        case CODEC_ID_H264:
408
            st->need_parsing = AVSTREAM_PARSE_FULL;
409
            break;
410 0048a2a8 Martin Storsjö
        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 8eb793c4 Luca Abeni
        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 99a1d191 Ronald S. Bultje
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 8eb793c4 Luca Abeni
/**
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 79d482b1 Martin Storsjö
    if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
440
        return; /* Timestamp already set by depacketizer */
441 d74c6145 Martin Storsjö
    if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
442 fba7815d Luca Abeni
        int64_t addend;
443
        int delta_timestamp;
444
445
        /* compute pts from timestamp with received ntp_time */
446
        delta_timestamp = timestamp - s->last_rtcp_timestamp;
447
        /* convert to the PTS timebase */
448 2cab6b48 Martin Storsjö
        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);
449 3a1cdcc7 Martin Storsjö
        pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
450
                   delta_timestamp;
451
        return;
452 fba7815d Luca Abeni
    }
453 3a1cdcc7 Martin Storsjö
    if (timestamp == RTP_NOTS_VALUE)
454
        return;
455
    if (!s->base_timestamp)
456
        s->base_timestamp = timestamp;
457
    pkt->pts = s->range_start_offset + timestamp - s->base_timestamp;
458 8eb793c4 Luca Abeni
}
459
460 02607418 Martin Storsjö
static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
461
                                     const uint8_t *buf, int len)
462 8eb793c4 Luca Abeni
{
463
    unsigned int ssrc, h;
464 f841a0fc Ronald S. Bultje
    int payload_type, seq, ret, flags = 0;
465 9446b4bb Robert Schlabbach
    int ext;
466 8eb793c4 Luca Abeni
    AVStream *st;
467
    uint32_t timestamp;
468
    int rv= 0;
469
470 9446b4bb Robert Schlabbach
    ext = buf[0] & 0x10;
471 8eb793c4 Luca Abeni
    payload_type = buf[1] & 0x7f;
472 144ae29d Ronald S. Bultje
    if (buf[1] & 0x80)
473
        flags |= RTP_FLAG_MARKER;
474 8eb793c4 Luca Abeni
    seq  = AV_RB16(buf + 2);
475
    timestamp = AV_RB32(buf + 4);
476
    ssrc = AV_RB32(buf + 8);
477
    /* store the ssrc in the RTPDemuxContext */
478
    s->ssrc = ssrc;
479
480
    /* NOTE: we can handle only one payload type */
481
    if (s->payload_type != payload_type)
482
        return -1;
483
484
    st = s->st;
485
    // only do something with this if all the rtp checks pass...
486
    if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
487
    {
488
        av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
489
               payload_type, seq, ((s->seq + 1) & 0xffff));
490
        return -1;
491
    }
492
493 4838cdab Martin Storsjö
    if (buf[0] & 0x20) {
494
        int padding = buf[len - 1];
495
        if (len >= 12 + padding)
496
            len -= padding;
497
    }
498
499 8eb793c4 Luca Abeni
    s->seq = seq;
500
    len -= 12;
501
    buf += 12;
502
503 9446b4bb Robert Schlabbach
    /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
504
    if (ext) {
505
        if (len < 4)
506
            return -1;
507
        /* calculate the header extension length (stored as number
508
         * of 32-bit words) */
509
        ext = (AV_RB16(buf + 2) + 1) << 2;
510
511
        if (len < ext)
512
            return -1;
513
        // skip past RTP header extension
514
        len -= ext;
515
        buf += ext;
516
    }
517
518 8eb793c4 Luca Abeni
    if (!st) {
519
        /* specific MPEG2TS demux support */
520 9125806e Alexis Ballier
        ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
521 946df059 Martin Storsjö
        /* The only error that can be returned from ff_mpegts_parse_packet
522
         * is "no more data to return from the provided buffer", so return
523
         * AVERROR(EAGAIN) for all errors */
524 4ffff367 Martin Storsjö
        if (ret < 0)
525 946df059 Martin Storsjö
            return AVERROR(EAGAIN);
526 8eb793c4 Luca Abeni
        if (ret < len) {
527
            s->read_buf_size = len - ret;
528
            memcpy(s->buf, buf + ret, s->read_buf_size);
529
            s->read_buf_index = 0;
530
            return 1;
531
        }
532 f3e71942 Ronald S. Bultje
        return 0;
533 b4e3330c Ronald S. Bultje
    } else if (s->parse_packet) {
534 1a45a9f4 Ronald S. Bultje
        rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
535 9b932b8a Ronald S. Bultje
                             s->st, pkt, &timestamp, buf, len, flags);
536 8eb793c4 Luca Abeni
    } else {
537
        // at this point, the RTP header has been stripped;  This is ASSUMING that there is only 1 CSRC, which in't wise.
538
        switch(st->codec->codec_id) {
539
        case CODEC_ID_MP2:
540 76faff6e Luca Abeni
        case CODEC_ID_MP3:
541 8eb793c4 Luca Abeni
            /* better than nothing: skip mpeg audio RTP header */
542
            if (len <= 4)
543
                return -1;
544
            h = AV_RB32(buf);
545
            len -= 4;
546
            buf += 4;
547
            av_new_packet(pkt, len);
548
            memcpy(pkt->data, buf, len);
549
            break;
550
        case CODEC_ID_MPEG1VIDEO:
551
        case CODEC_ID_MPEG2VIDEO:
552
            /* better than nothing: skip mpeg video RTP header */
553
            if (len <= 4)
554
                return -1;
555
            h = AV_RB32(buf);
556
            buf += 4;
557
            len -= 4;
558
            if (h & (1 << 26)) {
559
                /* mpeg2 */
560
                if (len <= 4)
561
                    return -1;
562
                buf += 4;
563
                len -= 4;
564
            }
565
            av_new_packet(pkt, len);
566
            memcpy(pkt->data, buf, len);
567
            break;
568
        default:
569 f739b36d Ronald S. Bultje
            av_new_packet(pkt, len);
570
            memcpy(pkt->data, buf, len);
571 8eb793c4 Luca Abeni
            break;
572
        }
573 eafb17d1 Ronald S. Bultje
574
        pkt->stream_index = st->index;
575 f3e71942 Ronald S. Bultje
    }
576 8eb793c4 Luca Abeni
577 95f03cf3 Ronald S. Bultje
    // now perform timestamp things....
578
    finalize_packet(s, pkt, timestamp);
579 f3e71942 Ronald S. Bultje
580 8eb793c4 Luca Abeni
    return rv;
581
}
582
583 58ee0991 Martin Storsjö
void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
584
{
585
    while (s->queue) {
586
        RTPPacket *next = s->queue->next;
587
        av_free(s->queue->buf);
588
        av_free(s->queue);
589
        s->queue = next;
590
    }
591
    s->seq       = 0;
592
    s->queue_len = 0;
593
    s->prev_ret  = 0;
594
}
595
596
static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
597
{
598
    uint16_t seq = AV_RB16(buf + 2);
599
    RTPPacket *cur = s->queue, *prev = NULL, *packet;
600
601
    /* Find the correct place in the queue to insert the packet */
602
    while (cur) {
603
        int16_t diff = seq - cur->seq;
604
        if (diff < 0)
605
            break;
606
        prev = cur;
607
        cur = cur->next;
608
    }
609
610
    packet = av_mallocz(sizeof(*packet));
611
    if (!packet)
612
        return;
613
    packet->recvtime = av_gettime();
614
    packet->seq = seq;
615
    packet->len = len;
616
    packet->buf = buf;
617
    packet->next = cur;
618
    if (prev)
619
        prev->next = packet;
620
    else
621
        s->queue = packet;
622
    s->queue_len++;
623
}
624
625
static int has_next_packet(RTPDemuxContext *s)
626
{
627 ddcf8411 Martin Storsjö
    return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
628 58ee0991 Martin Storsjö
}
629
630
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
631
{
632
    return s->queue ? s->queue->recvtime : 0;
633
}
634
635
static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
636
{
637
    int rv;
638
    RTPPacket *next;
639
640
    if (s->queue_len <= 0)
641
        return -1;
642
643
    if (!has_next_packet(s))
644
        av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
645
               "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
646
647
    /* Parse the first packet in the queue, and dequeue it */
648
    rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
649
    next = s->queue->next;
650
    av_free(s->queue->buf);
651
    av_free(s->queue);
652
    s->queue = next;
653
    s->queue_len--;
654 4ffff367 Martin Storsjö
    return rv;
655 58ee0991 Martin Storsjö
}
656
657 4ffff367 Martin Storsjö
static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
658 02607418 Martin Storsjö
                     uint8_t **bufptr, int len)
659
{
660
    uint8_t* buf = bufptr ? *bufptr : NULL;
661
    int ret, flags = 0;
662
    uint32_t timestamp;
663
    int rv= 0;
664
665
    if (!buf) {
666 f6e138b4 Martin Storsjö
        /* If parsing of the previous packet actually returned 0 or an error,
667
         * there's nothing more to be parsed from that packet, but we may have
668 58ee0991 Martin Storsjö
         * indicated that we can return the next enqueued packet. */
669 f6e138b4 Martin Storsjö
        if (s->prev_ret <= 0)
670 58ee0991 Martin Storsjö
            return rtp_parse_queued_packet(s, pkt);
671 02607418 Martin Storsjö
        /* return the next packets, if any */
672
        if(s->st && s->parse_packet) {
673
            /* timestamp should be overwritten by parse_packet, if not,
674
             * the packet is left with pts == AV_NOPTS_VALUE */
675
            timestamp = RTP_NOTS_VALUE;
676
            rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
677
                                s->st, pkt, &timestamp, NULL, 0, flags);
678
            finalize_packet(s, pkt, timestamp);
679 4ffff367 Martin Storsjö
            return rv;
680 02607418 Martin Storsjö
        } else {
681
            // TODO: Move to a dynamic packet handler (like above)
682 4ffff367 Martin Storsjö
            if (s->read_buf_index >= s->read_buf_size)
683 91ec7aea Martin Storsjö
                return AVERROR(EAGAIN);
684 02607418 Martin Storsjö
            ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
685
                                      s->read_buf_size - s->read_buf_index);
686 4ffff367 Martin Storsjö
            if (ret < 0)
687 946df059 Martin Storsjö
                return AVERROR(EAGAIN);
688 02607418 Martin Storsjö
            s->read_buf_index += ret;
689
            if (s->read_buf_index < s->read_buf_size)
690
                return 1;
691 4ffff367 Martin Storsjö
            else
692
                return 0;
693 02607418 Martin Storsjö
        }
694
    }
695
696
    if (len < 12)
697
        return -1;
698
699
    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
700
        return -1;
701
    if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
702
        return rtcp_parse_packet(s, buf, len);
703
    }
704
705 65cdee9c Martin Storsjö
    if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
706 58ee0991 Martin Storsjö
        /* First packet, or no reordering */
707
        return rtp_parse_packet_internal(s, pkt, buf, len);
708
    } else {
709
        uint16_t seq = AV_RB16(buf + 2);
710
        int16_t diff = seq - s->seq;
711
        if (diff < 0) {
712
            /* Packet older than the previously emitted one, drop */
713
            av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
714
                   "RTP: dropping old packet received too late\n");
715
            return -1;
716
        } else if (diff <= 1) {
717
            /* Correct packet */
718
            rv = rtp_parse_packet_internal(s, pkt, buf, len);
719 4ffff367 Martin Storsjö
            return rv;
720 58ee0991 Martin Storsjö
        } else {
721
            /* Still missing some packet, enqueue this one. */
722
            enqueue_packet(s, buf, len);
723
            *bufptr = NULL;
724
            /* Return the first enqueued packet if the queue is full,
725
             * even if we're missing something */
726
            if (s->queue_len >= s->queue_size)
727
                return rtp_parse_queued_packet(s, pkt);
728
            return -1;
729
        }
730
    }
731 02607418 Martin Storsjö
}
732
733 4ffff367 Martin Storsjö
/**
734
 * Parse an RTP or RTCP packet directly sent as a buffer.
735
 * @param s RTP parse context.
736
 * @param pkt returned packet
737
 * @param bufptr pointer to the input buffer or NULL to read the next packets
738
 * @param len buffer len
739
 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
740
 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
741
 */
742
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
743
                     uint8_t **bufptr, int len)
744
{
745
    int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
746
    s->prev_ret = rv;
747 d678a6fd Martin Storsjö
    while (rv == AVERROR(EAGAIN) && has_next_packet(s))
748
        rv = rtp_parse_queued_packet(s, pkt);
749 4ffff367 Martin Storsjö
    return rv ? rv : has_next_packet(s);
750
}
751
752 8eb793c4 Luca Abeni
void rtp_parse_close(RTPDemuxContext *s)
753
{
754 58ee0991 Martin Storsjö
    ff_rtp_reset_packet_queue(s);
755 8eb793c4 Luca Abeni
    if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
756 9125806e Alexis Ballier
        ff_mpegts_parse_close(s->ts);
757 8eb793c4 Luca Abeni
    }
758
    av_free(s);
759
}
760 016bc031 Josh Allmann
761
int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
762
                  int (*parse_fmtp)(AVStream *stream,
763
                                    PayloadContext *data,
764
                                    char *attr, char *value))
765
{
766
    char attr[256];
767 824535e3 Josh Allmann
    char *value;
768 016bc031 Josh Allmann
    int res;
769 824535e3 Josh Allmann
    int value_size = strlen(p) + 1;
770
771
    if (!(value = av_malloc(value_size))) {
772
        av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
773
        return AVERROR(ENOMEM);
774
    }
775 016bc031 Josh Allmann
776
    // remove protocol identifier
777
    while (*p && *p == ' ') p++; // strip spaces
778
    while (*p && *p != ' ') p++; // eat protocol identifier
779
    while (*p && *p == ' ') p++; // strip trailing spaces
780
781
    while (ff_rtsp_next_attr_and_value(&p,
782
                                       attr, sizeof(attr),
783 824535e3 Josh Allmann
                                       value, value_size)) {
784 016bc031 Josh Allmann
785
        res = parse_fmtp(stream, data, attr, value);
786 824535e3 Josh Allmann
        if (res < 0 && res != AVERROR_PATCHWELCOME) {
787
            av_free(value);
788 016bc031 Josh Allmann
            return res;
789 824535e3 Josh Allmann
        }
790 016bc031 Josh Allmann
    }
791 824535e3 Josh Allmann
    av_free(value);
792 016bc031 Josh Allmann
    return 0;
793
}