2 * Realmedia RTSP protocol (RDT) support.
3 * Copyright (c) 2007 Ronald S. Bultje
5 * This file is part of FFmpeg.
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.
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.
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
23 * @file libavformat/rdt.c
24 * @brief Realmedia RTSP protocol (RDT) support
25 * @author Ronald S. Bultje <rbultje@ronald.bitfreak.net>
29 #include "libavutil/avstring.h"
32 #include "libavutil/base64.h"
33 #include "libavutil/md5.h"
36 #include "libavcodec/get_bits.h"
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
42 * codecs/bitrates). Each such stream is represented by one AVStream
43 * in the AVFormatContext, and this variable points to the offset in
44 * that array such that the first is the first stream of this set. */
46 int n_streams; /**< streams with identifical content in this set */
47 void *dynamic_protocol_context;
48 DynamicPayloadPacketHandlerProc parse_packet;
49 uint32_t prev_timestamp;
50 int prev_set_id, prev_stream_id;
54 ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx,
55 void *priv_data, RTPDynamicProtocolHandler *handler)
57 RDTDemuxContext *s = av_mallocz(sizeof(RDTDemuxContext));
62 s->streams = &ic->streams[first_stream_of_set_idx];
65 } 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);
68 s->prev_stream_id = -1;
69 s->prev_timestamp = -1;
70 s->parse_packet = handler ? handler->parse_packet : NULL;
71 s->dynamic_protocol_context = priv_data;
77 ff_rdt_parse_close(RDTDemuxContext *s)
81 for (i = 1; i < s->n_streams; i++)
82 s->streams[i]->priv_data = NULL;
87 struct PayloadContext {
88 AVFormatContext *rmctx;
89 RMStream *rmst[MAX_STREAMS];
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 */
97 ff_rdt_calc_response_and_checksum(char response[41], char chksum[9],
98 const char *challenge)
100 int ch_len = strlen (challenge), i;
101 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,
106 0xc0, 0x01, 0x05, 0x05, 0x67, 0x03, 0x19, 0x70,
107 0x08, 0x27, 0x66, 0x10, 0x10, 0x72, 0x08, 0x09,
108 0x63, 0x11, 0x03, 0x71, 0x08, 0x08, 0x70, 0x02,
109 0x10, 0x57, 0x05, 0x18, 0x54 };
111 /* some (length) checks */
112 if (ch_len == 40) /* what a hack... */
114 else if (ch_len > 56)
116 memcpy(buf + 8, challenge, ch_len);
118 /* xor challenge bytewise with xor_table */
119 for (i = 0; i < XOR_TABLE_SIZE; i++)
120 buf[8 + i] ^= xor_table[i];
122 av_md5_sum(zres, buf, 64);
123 ff_data_to_hex(response, zres, 16, 1);
126 strcpy (response + 32, "01d0a8e3");
128 /* calculate checksum */
129 for (i = 0; i < 8; i++)
130 chksum[i] = response[i * 4];
135 rdt_load_mdpr (PayloadContext *rdt, AVStream *st, int rule_nr)
142 * Layout of the MLTI chunk:
144 * 2:<number of streams>
145 * Then for each stream ([number_of_streams] times):
147 * 2:<number of mdpr chunks>
148 * Then for each mdpr chunk ([number_of_mdpr_chunks] times):
151 * we skip MDPR chunks until we reach the one of the stream
152 * we're interested in, and forward that ([size]+[data]) to
153 * the RM demuxer to parse the stream-specific header data.
157 init_put_byte(&pb, rdt->mlti_data, rdt->mlti_data_size, 0,
158 NULL, NULL, NULL, NULL);
160 if (tag == MKTAG('M', 'L', 'T', 'I')) {
163 /* read index of MDPR chunk numbers */
165 if (rule_nr < 0 || rule_nr >= num)
167 url_fskip(&pb, rule_nr * 2);
168 chunk_nr = get_be16(&pb);
169 url_fskip(&pb, (num - 1 - rule_nr) * 2);
171 /* read MDPR chunks */
176 url_fskip(&pb, get_be32(&pb));
177 size = get_be32(&pb);
179 size = rdt->mlti_data_size;
180 url_fseek(&pb, 0, SEEK_SET);
182 if (ff_rm_read_mdpr_codecdata(rdt->rmctx, &pb, st, rdt->rmst[st->index], size) < 0)
189 * Actual data handling.
193 ff_rdt_parse_header(const uint8_t *buf, int len,
194 int *pset_id, int *pseq_no, int *pstream_id,
195 int *pis_keyframe, uint32_t *ptimestamp)
198 int consumed = 0, set_id, seq_no, stream_id, is_keyframe,
199 len_included, need_reliable;
202 /* skip status packets */
203 while (len >= 5 && buf[1] == 0xFF /* status packet */) {
206 if (!(buf[0] & 0x80))
207 return -1; /* not followed by a data packet */
209 pkt_len = AV_RB16(buf+3);
217 * Layout of the header (in bits):
219 * Flag indicating whether this header includes a length field;
220 * this can be used to concatenate multiple RDT packets in a
221 * single UDP/TCP data frame and is used to precede RDT data
222 * by stream status packets
224 * Flag indicating whether this header includes a "reliable
225 * sequence number"; these are apparently sequence numbers of
226 * data packets alone. For data packets, this flag is always
227 * set, according to the Real documentation [1]
229 * ID of a set of streams of identical content, possibly with
230 * different codecs or bitrates
232 * Flag set for certain streams deemed less tolerable for packet
235 * Packet sequence number; if >=0xFF00, this is a non-data packet
236 * containing stream status info, the second byte indicates the
237 * type of status packet (see wireshark docs / source code [2])
238 * if (len_included) {
241 * packet_len = remainder of UDP/TCP frame
244 * Back-to-Back flag; used for timing, set for one in every 10
245 * packets, according to the Real documentation [1]
247 * Slow-data flag; currently unused, according to Real docs [1]
249 * ID of the stream within this particular set of streams
251 * Non-keyframe flag (unset if packet belongs to a keyframe)
252 * 32: timestamp (PTS)
253 * if (set_id == 0x1F) {
254 * 16: set_id (extended set-of-streams ID; see set_id)
256 * if (need_reliable) {
257 * 16: reliable_seq_no
258 * Reliable sequence number (see need_reliable)
260 * if (stream_id == 0x3F) {
261 * 16: stream_id (extended stream ID; see stream_id)
263 * [1] https://protocol.helixcommunity.org/files/2005/devdocs/RDT_Feature_Level_20.txt
264 * [2] http://www.wireshark.org/docs/dfref/r/rdt.html and
265 * http://anonsvn.wireshark.org/viewvc/trunk/epan/dissectors/packet-rdt.c
267 init_get_bits(&gb, buf, len << 3);
268 len_included = get_bits1(&gb);
269 need_reliable = get_bits1(&gb);
270 set_id = get_bits(&gb, 5);
272 seq_no = get_bits(&gb, 16);
276 stream_id = get_bits(&gb, 5);
277 is_keyframe = !get_bits1(&gb);
278 timestamp = get_bits_long(&gb, 32);
280 set_id = get_bits(&gb, 16);
283 if (stream_id == 0x1f)
284 stream_id = get_bits(&gb, 16);
286 if (pset_id) *pset_id = set_id;
287 if (pseq_no) *pseq_no = seq_no;
288 if (pstream_id) *pstream_id = stream_id;
289 if (pis_keyframe) *pis_keyframe = is_keyframe;
290 if (ptimestamp) *ptimestamp = timestamp;
292 return consumed + (get_bits_count(&gb) >> 3);
295 /**< return 0 on packet, no more left, 1 on packet, 1 on partial packet... */
297 rdt_parse_packet (AVFormatContext *ctx, PayloadContext *rdt, AVStream *st,
298 AVPacket *pkt, uint32_t *timestamp,
299 const uint8_t *buf, int len, int flags)
304 if (rdt->audio_pkt_cnt == 0) {
307 init_put_byte(&pb, buf, len, 0, NULL, NULL, NULL, NULL);
308 flags = (flags & RTP_FLAG_KEY) ? 2 : 0;
309 res = ff_rm_parse_packet (rdt->rmctx, &pb, st, rdt->rmst[st->index], len, pkt,
310 &seq, flags, *timestamp);
311 pos = url_ftell(&pb);
315 if (st->codec->codec_id == CODEC_ID_AAC) {
316 memcpy (rdt->buffer, buf + pos, len - pos);
317 rdt->rmctx->pb = av_alloc_put_byte (rdt->buffer, len - pos, 0,
318 NULL, NULL, NULL, NULL);
325 ff_rm_retrieve_cache (rdt->rmctx, rdt->rmctx->pb,
326 st, rdt->rmst[st->index], pkt);
327 if (rdt->audio_pkt_cnt == 0 &&
328 st->codec->codec_id == CODEC_ID_AAC)
329 av_freep(&rdt->rmctx->pb);
331 pkt->stream_index = st->index;
332 pkt->pts = *timestamp;
334 return rdt->audio_pkt_cnt > 0;
338 ff_rdt_parse_packet(RDTDemuxContext *s, AVPacket *pkt,
339 const uint8_t *buf, int len)
341 int seq_no, flags = 0, stream_id, set_id, is_keyframe;
345 if (!s->parse_packet)
348 if (!buf && s->prev_stream_id != -1) {
349 /* return the next packets, if any */
350 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
351 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
352 s->streams[s->prev_stream_id],
353 pkt, ×tamp, NULL, 0, flags);
359 rv = ff_rdt_parse_header(buf, len, &set_id, &seq_no, &stream_id, &is_keyframe, ×tamp);
363 (set_id != s->prev_set_id || timestamp != s->prev_timestamp ||
364 stream_id != s->prev_stream_id)) {
365 flags |= RTP_FLAG_KEY;
366 s->prev_set_id = set_id;
367 s->prev_timestamp = timestamp;
369 s->prev_stream_id = stream_id;
373 if (s->prev_stream_id >= s->n_streams) {
374 s->prev_stream_id = -1;
378 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
379 s->streams[s->prev_stream_id],
380 pkt, ×tamp, buf, len, flags);
386 ff_rdt_subscribe_rule (char *cmd, int size,
387 int stream_nr, int rule_nr)
389 av_strlcatf(cmd, size, "stream=%d;rule=%d,stream=%d;rule=%d",
390 stream_nr, rule_nr * 2, stream_nr, rule_nr * 2 + 1);
393 static unsigned char *
394 rdt_parse_b64buf (unsigned int *target_len, const char *p)
396 unsigned char *target;
400 len -= 2; /* skip embracing " at start/end */
402 *target_len = len * 3 / 4;
403 target = av_mallocz(*target_len + FF_INPUT_BUFFER_PADDING_SIZE);
404 av_base64_decode(target, p, *target_len);
409 rdt_parse_sdp_line (AVFormatContext *s, int st_index,
410 PayloadContext *rdt, const char *line)
412 AVStream *stream = s->streams[st_index];
413 const char *p = line;
415 if (av_strstart(p, "OpaqueData:buffer;", &p)) {
416 rdt->mlti_data = rdt_parse_b64buf(&rdt->mlti_data_size, p);
417 } else if (av_strstart(p, "StartTime:integer;", &p))
418 stream->first_dts = atoi(p);
419 else if (av_strstart(p, "ASMRuleBook:string;", &p)) {
422 for (n = 0; n < s->nb_streams; n++)
423 if (s->streams[n]->priv_data == stream->priv_data) {
424 if (first == -1) first = n;
425 rdt->rmst[s->streams[n]->index] = ff_rm_alloc_rmstream();
426 rdt_load_mdpr(rdt, s->streams[n], (n - first) * 2);
428 if (s->streams[n]->codec->codec_id == CODEC_ID_AAC)
429 s->streams[n]->codec->frame_size = 1; // FIXME
437 real_parse_asm_rule(AVStream *st, const char *p, const char *end)
440 /* can be either averagebandwidth= or AverageBandwidth= */
441 if (sscanf(p, " %*1[Aa]verage%*1[Bb]andwidth=%d", &st->codec->bit_rate) == 1)
443 if (!(p = strchr(p, ',')) || p > end)
450 add_dstream(AVFormatContext *s, AVStream *orig_st)
454 if (!(st = av_new_stream(s, 0)))
456 st->codec->codec_type = orig_st->codec->codec_type;
457 st->priv_data = orig_st->priv_data;
458 st->first_dts = orig_st->first_dts;
464 real_parse_asm_rulebook(AVFormatContext *s, AVStream *orig_st,
468 int n_rules, odd = 0;
472 * The ASMRuleBook contains a list of comma-separated strings per rule,
473 * and each rule is separated by a ;. The last one also has a ; at the
474 * end so we can use it as delimiter.
475 * Every rule occurs twice, once for when the RTSP packet header marker
476 * is set and once for if it isn't. We only read the first because we
477 * don't care much (that's what the "odd" variable is for).
478 * Each rule contains a set of one or more statements, optionally
479 * preceeded by a single condition. If there's a condition, the rule
480 * starts with a '#'. Multiple conditions are merged between brackets,
481 * so there are never multiple conditions spread out over separate
482 * statements. Generally, these conditions are bitrate limits (min/max)
483 * for multi-bitrate streams.
486 for (n_rules = 0; s->nb_streams < MAX_STREAMS;) {
487 if (!(end = strchr(p, ';')))
489 if (!odd && end != p) {
491 st = add_dstream(s, orig_st);
494 real_parse_asm_rule(st, p, end);
503 ff_real_parse_sdp_a_line (AVFormatContext *s, int stream_index,
506 const char *p = line;
508 if (av_strstart(p, "ASMRuleBook:string;", &p))
509 real_parse_asm_rulebook(s, s->streams[stream_index], p);
512 static PayloadContext *
513 rdt_new_context (void)
515 PayloadContext *rdt = av_mallocz(sizeof(PayloadContext));
517 av_open_input_stream(&rdt->rmctx, NULL, "", &rdt_demuxer, NULL);
523 rdt_free_context (PayloadContext *rdt)
527 for (i = 0; i < MAX_STREAMS; i++)
529 ff_rm_free_rmstream(rdt->rmst[i]);
530 av_freep(&rdt->rmst[i]);
533 av_close_input_stream(rdt->rmctx);
534 av_freep(&rdt->mlti_data);
538 #define RDT_HANDLER(n, s, t) \
539 static RTPDynamicProtocolHandler ff_rdt_ ## n ## _handler = { \
542 .codec_id = CODEC_ID_NONE, \
543 .parse_sdp_a_line = rdt_parse_sdp_line, \
544 .open = rdt_new_context, \
545 .close = rdt_free_context, \
546 .parse_packet = rdt_parse_packet \
549 RDT_HANDLER(live_video, "x-pn-multirate-realvideo-live", AVMEDIA_TYPE_VIDEO);
550 RDT_HANDLER(live_audio, "x-pn-multirate-realaudio-live", AVMEDIA_TYPE_AUDIO);
551 RDT_HANDLER(video, "x-pn-realvideo", AVMEDIA_TYPE_VIDEO);
552 RDT_HANDLER(audio, "x-pn-realaudio", AVMEDIA_TYPE_AUDIO);
554 void av_register_rdt_dynamic_payload_handlers(void)
556 ff_register_dynamic_payload_handler(&ff_rdt_video_handler);
557 ff_register_dynamic_payload_handler(&ff_rdt_audio_handler);
558 ff_register_dynamic_payload_handler(&ff_rdt_live_video_handler);
559 ff_register_dynamic_payload_handler(&ff_rdt_live_audio_handler);