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1
/*
2
 * Realmedia RTSP protocol (RDT) support.
3
 * Copyright (c) 2007 Ronald S. Bultje
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

    
22
/**
23
 * @file libavformat/rdt.c
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 * @brief Realmedia RTSP protocol (RDT) support
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 * @author Ronald S. Bultje <rbultje@ronald.bitfreak.net>
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 */
27

    
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#include "avformat.h"
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#include "libavutil/avstring.h"
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#include "rtp.h"
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#include "rdt.h"
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#include "libavutil/base64.h"
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#include "libavutil/md5.h"
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#include "rm.h"
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#include "internal.h"
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#include "libavcodec/bitstream.h"
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38
struct RDTDemuxContext {
39
    AVFormatContext *ic; /**< the containing (RTSP) demux context */
40
    /** Each RDT stream-set (represented by one RTSPStream) can contain
41
     * multiple streams (of the same content, but with possibly different
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     * codecs/bitrates). Each such stream is represented by one AVStream
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     * in the AVFormatContext, and this variable points to the offset in
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     * that array such that the first is the first stream of this set. */
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    AVStream **streams;
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    int n_streams; /**< streams with identifical content in this set */
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    void *dynamic_protocol_context;
48
    DynamicPayloadPacketHandlerProc parse_packet;
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    uint32_t prev_timestamp;
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    int prev_set_id, prev_stream_id;
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};
52

    
53
RDTDemuxContext *
54
ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx,
55
                  void *priv_data, RTPDynamicProtocolHandler *handler)
56
{
57
    RDTDemuxContext *s = av_mallocz(sizeof(RDTDemuxContext));
58
    if (!s)
59
        return NULL;
60

    
61
    s->ic = ic;
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    s->streams = &ic->streams[first_stream_of_set_idx];
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    do {
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        s->n_streams++;
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    } while (first_stream_of_set_idx + s->n_streams < ic->nb_streams &&
66
             s->streams[s->n_streams]->priv_data == s->streams[0]->priv_data);
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    s->prev_set_id    = -1;
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    s->prev_stream_id = -1;
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    s->prev_timestamp = -1;
70
    s->parse_packet = handler->parse_packet;
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    s->dynamic_protocol_context = priv_data;
72

    
73
    return s;
74
}
75

    
76
void
77
ff_rdt_parse_close(RDTDemuxContext *s)
78
{
79
    int i;
80

    
81
    for (i = 1; i < s->n_streams; i++)
82
        s->streams[i]->priv_data = NULL;
83

    
84
    av_free(s);
85
}
86

    
87
struct PayloadContext {
88
    AVFormatContext *rmctx;
89
    RMStream *rmst[MAX_STREAMS];
90
    uint8_t *mlti_data;
91
    unsigned int mlti_data_size;
92
    char buffer[RTP_MAX_PACKET_LENGTH + FF_INPUT_BUFFER_PADDING_SIZE];
93
    int audio_pkt_cnt; /**< remaining audio packets in rmdec */
94
};
95

    
96
void
97
ff_rdt_calc_response_and_checksum(char response[41], char chksum[9],
98
                                  const char *challenge)
99
{
100
    int ch_len = strlen (challenge), i;
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    unsigned char zres[16],
102
        buf[64] = { 0xa1, 0xe9, 0x14, 0x9d, 0x0e, 0x6b, 0x3b, 0x59 };
103
#define XOR_TABLE_SIZE 37
104
    const unsigned char xor_table[XOR_TABLE_SIZE] = {
105
        0x05, 0x18, 0x74, 0xd0, 0x0d, 0x09, 0x02, 0x53,
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        0xc0, 0x01, 0x05, 0x05, 0x67, 0x03, 0x19, 0x70,
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        0x08, 0x27, 0x66, 0x10, 0x10, 0x72, 0x08, 0x09,
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        0x63, 0x11, 0x03, 0x71, 0x08, 0x08, 0x70, 0x02,
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        0x10, 0x57, 0x05, 0x18, 0x54 };
110

    
111
    /* some (length) checks */
112
    if (ch_len == 40) /* what a hack... */
113
        ch_len = 32;
114
    else if (ch_len > 56)
115
        ch_len = 56;
116
    memcpy(buf + 8, challenge, ch_len);
117

    
118
    /* xor challenge bytewise with xor_table */
119
    for (i = 0; i < XOR_TABLE_SIZE; i++)
120
        buf[8 + i] ^= xor_table[i];
121

    
122
    av_md5_sum(zres, buf, 64);
123
    ff_data_to_hex(response, zres, 16);
124
    for (i=0;i<32;i++) response[i] = tolower(response[i]);
125

    
126
    /* add tail */
127
    strcpy (response + 32, "01d0a8e3");
128

    
129
    /* calculate checksum */
130
    for (i = 0; i < 8; i++)
131
        chksum[i] = response[i * 4];
132
    chksum[8] = 0;
133
}
134

    
135
static int
136
rdt_load_mdpr (PayloadContext *rdt, AVStream *st, int rule_nr)
137
{
138
    ByteIOContext pb;
139
    int size;
140
    uint32_t tag;
141

    
142
    /**
143
     * Layout of the MLTI chunk:
144
     * 4:MLTI
145
     * 2:<number of streams>
146
     * Then for each stream ([number_of_streams] times):
147
     *     2:<mdpr index>
148
     * 2:<number of mdpr chunks>
149
     * Then for each mdpr chunk ([number_of_mdpr_chunks] times):
150
     *     4:<size>
151
     *     [size]:<data>
152
     * we skip MDPR chunks until we reach the one of the stream
153
     * we're interested in, and forward that ([size]+[data]) to
154
     * the RM demuxer to parse the stream-specific header data.
155
     */
156
    if (!rdt->mlti_data)
157
        return -1;
158
    init_put_byte(&pb, rdt->mlti_data, rdt->mlti_data_size, 0,
159
                  NULL, NULL, NULL, NULL);
160
    tag = get_le32(&pb);
161
    if (tag == MKTAG('M', 'L', 'T', 'I')) {
162
        int num, chunk_nr;
163

    
164
        /* read index of MDPR chunk numbers */
165
        num = get_be16(&pb);
166
        if (rule_nr < 0 || rule_nr >= num)
167
            return -1;
168
        url_fskip(&pb, rule_nr * 2);
169
        chunk_nr = get_be16(&pb);
170
        url_fskip(&pb, (num - 1 - rule_nr) * 2);
171

    
172
        /* read MDPR chunks */
173
        num = get_be16(&pb);
174
        if (chunk_nr >= num)
175
            return -1;
176
        while (chunk_nr--)
177
            url_fskip(&pb, get_be32(&pb));
178
        size = get_be32(&pb);
179
    } else {
180
        size = rdt->mlti_data_size;
181
        url_fseek(&pb, 0, SEEK_SET);
182
    }
183
    if (ff_rm_read_mdpr_codecdata(rdt->rmctx, &pb, st, rdt->rmst[st->index], size) < 0)
184
        return -1;
185

    
186
    return 0;
187
}
188

    
189
/**
190
 * Actual data handling.
191
 */
192

    
193
int
194
ff_rdt_parse_header(const uint8_t *buf, int len,
195
                    int *pset_id, int *pseq_no, int *pstream_id,
196
                    int *pis_keyframe, uint32_t *ptimestamp)
197
{
198
    GetBitContext gb;
199
    int consumed = 0, set_id, seq_no, stream_id, is_keyframe,
200
        len_included, need_reliable;
201
    uint32_t timestamp;
202

    
203
    /* skip status packets */
204
    while (len >= 5 && buf[1] == 0xFF /* status packet */) {
205
        int pkt_len;
206

    
207
        if (!(buf[0] & 0x80))
208
            return -1; /* not followed by a data packet */
209

    
210
        pkt_len = AV_RB16(buf+3);
211
        buf += pkt_len;
212
        len -= pkt_len;
213
        consumed += pkt_len;
214
    }
215
    if (len < 16)
216
        return -1;
217
    /**
218
     * Layout of the header (in bits):
219
     * 1:  len_included
220
     *     Flag indicating whether this header includes a length field;
221
     *     this can be used to concatenate multiple RDT packets in a
222
     *     single UDP/TCP data frame and is used to precede RDT data
223
     *     by stream status packets
224
     * 1:  need_reliable
225
     *     Flag indicating whether this header includes a "reliable
226
     *     sequence number"; these are apparently sequence numbers of
227
     *     data packets alone. For data packets, this flag is always
228
     *     set, according to the Real documentation [1]
229
     * 5:  set_id
230
     *     ID of a set of streams of identical content, possibly with
231
     *     different codecs or bitrates
232
     * 1:  is_reliable
233
     *     Flag set for certain streams deemed less tolerable for packet
234
     *     loss
235
     * 16: seq_no
236
     *     Packet sequence number; if >=0xFF00, this is a non-data packet
237
     *     containing stream status info, the second byte indicates the
238
     *     type of status packet (see wireshark docs / source code [2])
239
     * if (len_included) {
240
     *     16: packet_len
241
     * } else {
242
     *     packet_len = remainder of UDP/TCP frame
243
     * }
244
     * 1:  is_back_to_back
245
     *     Back-to-Back flag; used for timing, set for one in every 10
246
     *     packets, according to the Real documentation [1]
247
     * 1:  is_slow_data
248
     *     Slow-data flag; currently unused, according to Real docs [1]
249
     * 5:  stream_id
250
     *     ID of the stream within this particular set of streams
251
     * 1:  is_no_keyframe
252
     *     Non-keyframe flag (unset if packet belongs to a keyframe)
253
     * 32: timestamp (PTS)
254
     * if (set_id == 0x1F) {
255
     *     16: set_id (extended set-of-streams ID; see set_id)
256
     * }
257
     * if (need_reliable) {
258
     *     16: reliable_seq_no
259
     *         Reliable sequence number (see need_reliable)
260
     * }
261
     * if (stream_id == 0x3F) {
262
     *     16: stream_id (extended stream ID; see stream_id)
263
     * }
264
     * [1] https://protocol.helixcommunity.org/files/2005/devdocs/RDT_Feature_Level_20.txt
265
     * [2] http://www.wireshark.org/docs/dfref/r/rdt.html and
266
     *     http://anonsvn.wireshark.org/viewvc/trunk/epan/dissectors/packet-rdt.c
267
     */
268
    init_get_bits(&gb, buf, len << 3);
269
    len_included  = get_bits1(&gb);
270
    need_reliable = get_bits1(&gb);
271
    set_id        = get_bits(&gb, 5);
272
    skip_bits(&gb, 1);
273
    seq_no        = get_bits(&gb, 16);
274
    if (len_included)
275
        skip_bits(&gb, 16);
276
    skip_bits(&gb, 2);
277
    stream_id     = get_bits(&gb, 5);
278
    is_keyframe   = !get_bits1(&gb);
279
    timestamp     = get_bits_long(&gb, 32);
280
    if (set_id == 0x1f)
281
        set_id    = get_bits(&gb, 16);
282
    if (need_reliable)
283
        skip_bits(&gb, 16);
284
    if (stream_id == 0x1f)
285
        stream_id = get_bits(&gb, 16);
286

    
287
    if (pset_id)      *pset_id      = set_id;
288
    if (pseq_no)      *pseq_no      = seq_no;
289
    if (pstream_id)   *pstream_id   = stream_id;
290
    if (pis_keyframe) *pis_keyframe = is_keyframe;
291
    if (ptimestamp)   *ptimestamp   = timestamp;
292

    
293
    return consumed + (get_bits_count(&gb) >> 3);
294
}
295

    
296
/**< return 0 on packet, no more left, 1 on packet, 1 on partial packet... */
297
static int
298
rdt_parse_packet (AVFormatContext *ctx, PayloadContext *rdt, AVStream *st,
299
                  AVPacket *pkt, uint32_t *timestamp,
300
                  const uint8_t *buf, int len, int flags)
301
{
302
    int seq = 1, res;
303
    ByteIOContext pb;
304

    
305
    if (rdt->audio_pkt_cnt == 0) {
306
        int pos;
307

    
308
        init_put_byte(&pb, buf, len, 0, NULL, NULL, NULL, NULL);
309
        flags = (flags & PKT_FLAG_KEY) ? 2 : 0;
310
        res = ff_rm_parse_packet (rdt->rmctx, &pb, st, rdt->rmst[st->index], len, pkt,
311
                                  &seq, &flags, timestamp);
312
        pos = url_ftell(&pb);
313
        if (res < 0)
314
            return res;
315
        rdt->audio_pkt_cnt = res;
316
        if (rdt->audio_pkt_cnt > 0 &&
317
            st->codec->codec_id == CODEC_ID_AAC) {
318
            memcpy (rdt->buffer, buf + pos, len - pos);
319
            rdt->rmctx->pb = av_alloc_put_byte (rdt->buffer, len - pos, 0,
320
                                                NULL, NULL, NULL, NULL);
321
        }
322
    } else {
323
        rdt->audio_pkt_cnt =
324
            ff_rm_retrieve_cache (rdt->rmctx, rdt->rmctx->pb,
325
                                  st, rdt->rmst[st->index], pkt);
326
        if (rdt->audio_pkt_cnt == 0 &&
327
            st->codec->codec_id == CODEC_ID_AAC)
328
            av_freep(&rdt->rmctx->pb);
329
    }
330
    pkt->stream_index = st->index;
331
    pkt->pts = *timestamp;
332

    
333
    return rdt->audio_pkt_cnt > 0;
334
}
335

    
336
int
337
ff_rdt_parse_packet(RDTDemuxContext *s, AVPacket *pkt,
338
                    const uint8_t *buf, int len)
339
{
340
    int seq_no, flags = 0, stream_id, set_id, is_keyframe;
341
    uint32_t timestamp;
342
    int rv= 0;
343

    
344
    if (!s->parse_packet)
345
        return -1;
346

    
347
    if (!buf && s->prev_stream_id != -1) {
348
        /* return the next packets, if any */
349
        timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
350
        rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
351
                            s->streams[s->prev_stream_id],
352
                            pkt, &timestamp, NULL, 0, flags);
353
        return rv;
354
    }
355

    
356
    if (len < 12)
357
        return -1;
358
    rv = ff_rdt_parse_header(buf, len, &set_id, &seq_no, &stream_id, &is_keyframe, &timestamp);
359
    if (rv < 0)
360
        return rv;
361
    if (is_keyframe &&
362
        (set_id != s->prev_set_id || timestamp != s->prev_timestamp ||
363
         stream_id != s->prev_stream_id)) {
364
        flags |= PKT_FLAG_KEY;
365
        s->prev_set_id    = set_id;
366
        s->prev_timestamp = timestamp;
367
    }
368
    s->prev_stream_id = stream_id;
369
    buf += rv;
370
    len -= rv;
371

    
372
     if (s->prev_stream_id >= s->n_streams) {
373
         s->prev_stream_id = -1;
374
         return -1;
375
     }
376

    
377
    rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
378
                         s->streams[s->prev_stream_id],
379
                         pkt, &timestamp, buf, len, flags);
380

    
381
    return rv;
382
}
383

    
384
void
385
ff_rdt_subscribe_rule (char *cmd, int size,
386
                       int stream_nr, int rule_nr)
387
{
388
    av_strlcatf(cmd, size, "stream=%d;rule=%d,stream=%d;rule=%d",
389
                stream_nr, rule_nr * 2, stream_nr, rule_nr * 2 + 1);
390
}
391

    
392
static unsigned char *
393
rdt_parse_b64buf (unsigned int *target_len, const char *p)
394
{
395
    unsigned char *target;
396
    int len = strlen(p);
397
    if (*p == '\"') {
398
        p++;
399
        len -= 2; /* skip embracing " at start/end */
400
    }
401
    *target_len = len * 3 / 4;
402
    target = av_mallocz(*target_len + FF_INPUT_BUFFER_PADDING_SIZE);
403
    av_base64_decode(target, p, *target_len);
404
    return target;
405
}
406

    
407
static int
408
rdt_parse_sdp_line (AVFormatContext *s, int st_index,
409
                    PayloadContext *rdt, const char *line)
410
{
411
    AVStream *stream = s->streams[st_index];
412
    const char *p = line;
413

    
414
    if (av_strstart(p, "OpaqueData:buffer;", &p)) {
415
        rdt->mlti_data = rdt_parse_b64buf(&rdt->mlti_data_size, p);
416
    } else if (av_strstart(p, "StartTime:integer;", &p))
417
        stream->first_dts = atoi(p);
418
    else if (av_strstart(p, "ASMRuleBook:string;", &p)) {
419
        int n = st_index, first = -1;
420

    
421
        for (n = 0; n < s->nb_streams; n++)
422
            if (s->streams[n]->priv_data == stream->priv_data) {
423
                if (first == -1) first = n;
424
                rdt->rmst[s->streams[n]->index] = ff_rm_alloc_rmstream();
425
                rdt_load_mdpr(rdt, s->streams[n], (n - first) * 2);
426

    
427
                if (s->streams[n]->codec->codec_id == CODEC_ID_AAC)
428
                    s->streams[n]->codec->frame_size = 1; // FIXME
429
           }
430
    }
431

    
432
    return 0;
433
}
434

    
435
static void
436
real_parse_asm_rule(AVStream *st, const char *p, const char *end)
437
{
438
    do {
439
        /* can be either averagebandwidth= or AverageBandwidth= */
440
        if (sscanf(p, " %*1[Aa]verage%*1[Bb]andwidth=%d", &st->codec->bit_rate) == 1)
441
            break;
442
        if (!(p = strchr(p, ',')) || p > end)
443
            p = end;
444
        p++;
445
    } while (p < end);
446
}
447

    
448
static AVStream *
449
add_dstream(AVFormatContext *s, AVStream *orig_st)
450
{
451
    AVStream *st;
452

    
453
    if (!(st = av_new_stream(s, 0)))
454
        return NULL;
455
    st->codec->codec_type = orig_st->codec->codec_type;
456
    st->priv_data         = orig_st->priv_data;
457
    st->first_dts         = orig_st->first_dts;
458

    
459
    return st;
460
}
461

    
462
static void
463
real_parse_asm_rulebook(AVFormatContext *s, AVStream *orig_st,
464
                        const char *p)
465
{
466
    const char *end;
467
    int n_rules, odd = 0;
468
    AVStream *st;
469

    
470
    /**
471
     * The ASMRuleBook contains a list of comma-separated strings per rule,
472
     * and each rule is separated by a ;. The last one also has a ; at the
473
     * end so we can use it as delimiter.
474
     * Every rule occurs twice, once for when the RTSP packet header marker
475
     * is set and once for if it isn't. We only read the first because we
476
     * don't care much (that's what the "odd" variable is for).
477
     * Each rule contains a set of one or more statements, optionally
478
     * preceeded by a single condition. If there's a condition, the rule
479
     * starts with a '#'. Multiple conditions are merged between brackets,
480
     * so there are never multiple conditions spread out over separate
481
     * statements. Generally, these conditions are bitrate limits (min/max)
482
     * for multi-bitrate streams.
483
     */
484
    if (*p == '\"') p++;
485
    for (n_rules = 0; s->nb_streams < MAX_STREAMS;) {
486
        if (!(end = strchr(p, ';')))
487
            break;
488
        if (!odd && end != p) {
489
            if (n_rules > 0)
490
                st = add_dstream(s, orig_st);
491
            else
492
                st = orig_st;
493
            real_parse_asm_rule(st, p, end);
494
            n_rules++;
495
        }
496
        p = end + 1;
497
        odd ^= 1;
498
    }
499
}
500

    
501
void
502
ff_real_parse_sdp_a_line (AVFormatContext *s, int stream_index,
503
                          const char *line)
504
{
505
    const char *p = line;
506

    
507
    if (av_strstart(p, "ASMRuleBook:string;", &p))
508
        real_parse_asm_rulebook(s, s->streams[stream_index], p);
509
}
510

    
511
static PayloadContext *
512
rdt_new_extradata (void)
513
{
514
    PayloadContext *rdt = av_mallocz(sizeof(PayloadContext));
515

    
516
    av_open_input_stream(&rdt->rmctx, NULL, "", &rdt_demuxer, NULL);
517

    
518
    return rdt;
519
}
520

    
521
static void
522
rdt_free_extradata (PayloadContext *rdt)
523
{
524
    int i;
525

    
526
    for (i = 0; i < MAX_STREAMS; i++)
527
        if (rdt->rmst[i]) {
528
            ff_rm_free_rmstream(rdt->rmst[i]);
529
            av_freep(&rdt->rmst[i]);
530
        }
531
    if (rdt->rmctx)
532
        av_close_input_stream(rdt->rmctx);
533
    av_freep(&rdt->mlti_data);
534
    av_free(rdt);
535
}
536

    
537
#define RDT_HANDLER(n, s, t) \
538
static RTPDynamicProtocolHandler ff_rdt_ ## n ## _handler = { \
539
    s, \
540
    t, \
541
    CODEC_ID_NONE, \
542
    rdt_parse_sdp_line, \
543
    rdt_new_extradata, \
544
    rdt_free_extradata, \
545
    rdt_parse_packet \
546
};
547

    
548
RDT_HANDLER(live_video, "x-pn-multirate-realvideo-live", CODEC_TYPE_VIDEO);
549
RDT_HANDLER(live_audio, "x-pn-multirate-realaudio-live", CODEC_TYPE_AUDIO);
550
RDT_HANDLER(video,      "x-pn-realvideo",                CODEC_TYPE_VIDEO);
551
RDT_HANDLER(audio,      "x-pn-realaudio",                CODEC_TYPE_AUDIO);
552

    
553
void av_register_rdt_dynamic_payload_handlers(void)
554
{
555
    ff_register_dynamic_payload_handler(&ff_rdt_video_handler);
556
    ff_register_dynamic_payload_handler(&ff_rdt_audio_handler);
557
    ff_register_dynamic_payload_handler(&ff_rdt_live_video_handler);
558
    ff_register_dynamic_payload_handler(&ff_rdt_live_audio_handler);
559
}