/* * Realmedia RTSP protocol (RDT) support. * Copyright (c) 2007 Ronald S. Bultje * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * @brief Realmedia RTSP protocol (RDT) support * @author Ronald S. Bultje <rbultje@ronald.bitfreak.net> */ #include "avformat.h" #include "libavutil/avstring.h" #include "rtpdec.h" #include "rdt.h" #include "libavutil/base64.h" #include "libavutil/md5.h" #include "rm.h" #include "internal.h" #include "libavcodec/get_bits.h" struct RDTDemuxContext { AVFormatContext *ic; /**< the containing (RTSP) demux context */ /** Each RDT stream-set (represented by one RTSPStream) can contain * multiple streams (of the same content, but with possibly different * codecs/bitrates). Each such stream is represented by one AVStream * in the AVFormatContext, and this variable points to the offset in * that array such that the first is the first stream of this set. */ AVStream **streams; int n_streams; /**< streams with identifical content in this set */ void *dynamic_protocol_context; DynamicPayloadPacketHandlerProc parse_packet; uint32_t prev_timestamp; int prev_set_id, prev_stream_id; }; RDTDemuxContext * ff_rdt_parse_open(AVFormatContext *ic, int first_stream_of_set_idx, void *priv_data, RTPDynamicProtocolHandler *handler) { RDTDemuxContext *s = av_mallocz(sizeof(RDTDemuxContext)); if (!s) return NULL; s->ic = ic; s->streams = &ic->streams[first_stream_of_set_idx]; do { s->n_streams++; } while (first_stream_of_set_idx + s->n_streams < ic->nb_streams && s->streams[s->n_streams]->priv_data == s->streams[0]->priv_data); s->prev_set_id = -1; s->prev_stream_id = -1; s->prev_timestamp = -1; s->parse_packet = handler ? handler->parse_packet : NULL; s->dynamic_protocol_context = priv_data; return s; } void ff_rdt_parse_close(RDTDemuxContext *s) { int i; for (i = 1; i < s->n_streams; i++) s->streams[i]->priv_data = NULL; av_free(s); } struct PayloadContext { AVFormatContext *rmctx; int nb_rmst; RMStream **rmst; uint8_t *mlti_data; unsigned int mlti_data_size; char buffer[RTP_MAX_PACKET_LENGTH + FF_INPUT_BUFFER_PADDING_SIZE]; int audio_pkt_cnt; /**< remaining audio packets in rmdec */ }; void ff_rdt_calc_response_and_checksum(char response[41], char chksum[9], const char *challenge) { int ch_len = strlen (challenge), i; unsigned char zres[16], buf[64] = { 0xa1, 0xe9, 0x14, 0x9d, 0x0e, 0x6b, 0x3b, 0x59 }; #define XOR_TABLE_SIZE 37 const unsigned char xor_table[XOR_TABLE_SIZE] = { 0x05, 0x18, 0x74, 0xd0, 0x0d, 0x09, 0x02, 0x53, 0xc0, 0x01, 0x05, 0x05, 0x67, 0x03, 0x19, 0x70, 0x08, 0x27, 0x66, 0x10, 0x10, 0x72, 0x08, 0x09, 0x63, 0x11, 0x03, 0x71, 0x08, 0x08, 0x70, 0x02, 0x10, 0x57, 0x05, 0x18, 0x54 }; /* some (length) checks */ if (ch_len == 40) /* what a hack... */ ch_len = 32; else if (ch_len > 56) ch_len = 56; memcpy(buf + 8, challenge, ch_len); /* xor challenge bytewise with xor_table */ for (i = 0; i < XOR_TABLE_SIZE; i++) buf[8 + i] ^= xor_table[i]; av_md5_sum(zres, buf, 64); ff_data_to_hex(response, zres, 16, 1); /* add tail */ strcpy (response + 32, "01d0a8e3"); /* calculate checksum */ for (i = 0; i < 8; i++) chksum[i] = response[i * 4]; chksum[8] = 0; } static int rdt_load_mdpr (PayloadContext *rdt, AVStream *st, int rule_nr) { ByteIOContext pb; int size; uint32_t tag; /** * Layout of the MLTI chunk: * 4: MLTI * 2: number of streams * Then for each stream ([number_of_streams] times): * 2: mdpr index * 2: number of mdpr chunks * Then for each mdpr chunk ([number_of_mdpr_chunks] times): * 4: size * [size]: data * we skip MDPR chunks until we reach the one of the stream * we're interested in, and forward that ([size]+[data]) to * the RM demuxer to parse the stream-specific header data. */ if (!rdt->mlti_data) return -1; init_put_byte(&pb, rdt->mlti_data, rdt->mlti_data_size, 0, NULL, NULL, NULL, NULL); tag = get_le32(&pb); if (tag == MKTAG('M', 'L', 'T', 'I')) { int num, chunk_nr; /* read index of MDPR chunk numbers */ num = get_be16(&pb); if (rule_nr < 0 || rule_nr >= num) return -1; url_fskip(&pb, rule_nr * 2); chunk_nr = get_be16(&pb); url_fskip(&pb, (num - 1 - rule_nr) * 2); /* read MDPR chunks */ num = get_be16(&pb); if (chunk_nr >= num) return -1; while (chunk_nr--) url_fskip(&pb, get_be32(&pb)); size = get_be32(&pb); } else { size = rdt->mlti_data_size; url_fseek(&pb, 0, SEEK_SET); } if (ff_rm_read_mdpr_codecdata(rdt->rmctx, &pb, st, rdt->rmst[st->index], size) < 0) return -1; return 0; } /** * Actual data handling. */ int ff_rdt_parse_header(const uint8_t *buf, int len, int *pset_id, int *pseq_no, int *pstream_id, int *pis_keyframe, uint32_t *ptimestamp) { GetBitContext gb; int consumed = 0, set_id, seq_no, stream_id, is_keyframe, len_included, need_reliable; uint32_t timestamp; /* skip status packets */ while (len >= 5 && buf[1] == 0xFF /* status packet */) { int pkt_len; if (!(buf[0] & 0x80)) return -1; /* not followed by a data packet */ pkt_len = AV_RB16(buf+3); buf += pkt_len; len -= pkt_len; consumed += pkt_len; } if (len < 16) return -1; /** * Layout of the header (in bits): * 1: len_included * Flag indicating whether this header includes a length field; * this can be used to concatenate multiple RDT packets in a * single UDP/TCP data frame and is used to precede RDT data * by stream status packets * 1: need_reliable * Flag indicating whether this header includes a "reliable * sequence number"; these are apparently sequence numbers of * data packets alone. For data packets, this flag is always * set, according to the Real documentation [1] * 5: set_id * ID of a set of streams of identical content, possibly with * different codecs or bitrates * 1: is_reliable * Flag set for certain streams deemed less tolerable for packet * loss * 16: seq_no * Packet sequence number; if >=0xFF00, this is a non-data packet * containing stream status info, the second byte indicates the * type of status packet (see wireshark docs / source code [2]) * if (len_included) { * 16: packet_len * } else { * packet_len = remainder of UDP/TCP frame * } * 1: is_back_to_back * Back-to-Back flag; used for timing, set for one in every 10 * packets, according to the Real documentation [1] * 1: is_slow_data * Slow-data flag; currently unused, according to Real docs [1] * 5: stream_id * ID of the stream within this particular set of streams * 1: is_no_keyframe * Non-keyframe flag (unset if packet belongs to a keyframe) * 32: timestamp (PTS) * if (set_id == 0x1F) { * 16: set_id (extended set-of-streams ID; see set_id) * } * if (need_reliable) { * 16: reliable_seq_no * Reliable sequence number (see need_reliable) * } * if (stream_id == 0x3F) { * 16: stream_id (extended stream ID; see stream_id) * } * [1] https://protocol.helixcommunity.org/files/2005/devdocs/RDT_Feature_Level_20.txt * [2] http://www.wireshark.org/docs/dfref/r/rdt.html and * http://anonsvn.wireshark.org/viewvc/trunk/epan/dissectors/packet-rdt.c */ init_get_bits(&gb, buf, len << 3); len_included = get_bits1(&gb); need_reliable = get_bits1(&gb); set_id = get_bits(&gb, 5); skip_bits(&gb, 1); seq_no = get_bits(&gb, 16); if (len_included) skip_bits(&gb, 16); skip_bits(&gb, 2); stream_id = get_bits(&gb, 5); is_keyframe = !get_bits1(&gb); timestamp = get_bits_long(&gb, 32); if (set_id == 0x1f) set_id = get_bits(&gb, 16); if (need_reliable) skip_bits(&gb, 16); if (stream_id == 0x1f) stream_id = get_bits(&gb, 16); if (pset_id) *pset_id = set_id; if (pseq_no) *pseq_no = seq_no; if (pstream_id) *pstream_id = stream_id; if (pis_keyframe) *pis_keyframe = is_keyframe; if (ptimestamp) *ptimestamp = timestamp; return consumed + (get_bits_count(&gb) >> 3); } /**< return 0 on packet, no more left, 1 on packet, 1 on partial packet... */ static int rdt_parse_packet (AVFormatContext *ctx, PayloadContext *rdt, AVStream *st, AVPacket *pkt, uint32_t *timestamp, const uint8_t *buf, int len, int flags) { int seq = 1, res; ByteIOContext pb; if (rdt->audio_pkt_cnt == 0) { int pos; init_put_byte(&pb, buf, len, 0, NULL, NULL, NULL, NULL); flags = (flags & RTP_FLAG_KEY) ? 2 : 0; res = ff_rm_parse_packet (rdt->rmctx, &pb, st, rdt->rmst[st->index], len, pkt, &seq, flags, *timestamp); pos = url_ftell(&pb); if (res < 0) return res; if (res > 0) { if (st->codec->codec_id == CODEC_ID_AAC) { memcpy (rdt->buffer, buf + pos, len - pos); rdt->rmctx->pb = av_alloc_put_byte (rdt->buffer, len - pos, 0, NULL, NULL, NULL, NULL); } goto get_cache; } } else { get_cache: rdt->audio_pkt_cnt = ff_rm_retrieve_cache (rdt->rmctx, rdt->rmctx->pb, st, rdt->rmst[st->index], pkt); if (rdt->audio_pkt_cnt == 0 && st->codec->codec_id == CODEC_ID_AAC) av_freep(&rdt->rmctx->pb); } pkt->stream_index = st->index; pkt->pts = *timestamp; return rdt->audio_pkt_cnt > 0; } int ff_rdt_parse_packet(RDTDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len) { uint8_t *buf = bufptr ? *bufptr : NULL; int seq_no, flags = 0, stream_id, set_id, is_keyframe; uint32_t timestamp; int rv= 0; if (!s->parse_packet) return -1; if (!buf && s->prev_stream_id != -1) { /* return the next packets, if any */ timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned.... rv= s->parse_packet(s->ic, s->dynamic_protocol_context, s->streams[s->prev_stream_id], pkt, ×tamp, NULL, 0, flags); return rv; } if (len < 12) return -1; rv = ff_rdt_parse_header(buf, len, &set_id, &seq_no, &stream_id, &is_keyframe, ×tamp); if (rv < 0) return rv; if (is_keyframe && (set_id != s->prev_set_id || timestamp != s->prev_timestamp || stream_id != s->prev_stream_id)) { flags |= RTP_FLAG_KEY; s->prev_set_id = set_id; s->prev_timestamp = timestamp; } s->prev_stream_id = stream_id; buf += rv; len -= rv; if (s->prev_stream_id >= s->n_streams) { s->prev_stream_id = -1; return -1; } rv = s->parse_packet(s->ic, s->dynamic_protocol_context, s->streams[s->prev_stream_id], pkt, ×tamp, buf, len, flags); return rv; } void ff_rdt_subscribe_rule (char *cmd, int size, int stream_nr, int rule_nr) { av_strlcatf(cmd, size, "stream=%d;rule=%d,stream=%d;rule=%d", stream_nr, rule_nr * 2, stream_nr, rule_nr * 2 + 1); } static unsigned char * rdt_parse_b64buf (unsigned int *target_len, const char *p) { unsigned char *target; int len = strlen(p); if (*p == '\"') { p++; len -= 2; /* skip embracing " at start/end */ } *target_len = len * 3 / 4; target = av_mallocz(*target_len + FF_INPUT_BUFFER_PADDING_SIZE); av_base64_decode(target, p, *target_len); return target; } static int rdt_parse_sdp_line (AVFormatContext *s, int st_index, PayloadContext *rdt, const char *line) { AVStream *stream = s->streams[st_index]; const char *p = line; if (av_strstart(p, "OpaqueData:buffer;", &p)) { rdt->mlti_data = rdt_parse_b64buf(&rdt->mlti_data_size, p); } else if (av_strstart(p, "StartTime:integer;", &p)) stream->first_dts = atoi(p); else if (av_strstart(p, "ASMRuleBook:string;", &p)) { int n, first = -1; for (n = 0; n < s->nb_streams; n++) if (s->streams[n]->priv_data == stream->priv_data) { int count = s->streams[n]->index + 1; if (first == -1) first = n; if (rdt->nb_rmst < count) { RMStream **rmst= av_realloc(rdt->rmst, count*sizeof(*rmst)); if (!rmst) return AVERROR(ENOMEM); memset(rmst + rdt->nb_rmst, 0, (count - rdt->nb_rmst) * sizeof(*rmst)); rdt->rmst = rmst; rdt->nb_rmst = count; } rdt->rmst[s->streams[n]->index] = ff_rm_alloc_rmstream(); rdt_load_mdpr(rdt, s->streams[n], (n - first) * 2); if (s->streams[n]->codec->codec_id == CODEC_ID_AAC) s->streams[n]->codec->frame_size = 1; // FIXME } } return 0; } static void real_parse_asm_rule(AVStream *st, const char *p, const char *end) { do { /* can be either averagebandwidth= or AverageBandwidth= */ if (sscanf(p, " %*1[Aa]verage%*1[Bb]andwidth=%d", &st->codec->bit_rate) == 1) break; if (!(p = strchr(p, ',')) || p > end) p = end; p++; } while (p < end); } static AVStream * add_dstream(AVFormatContext *s, AVStream *orig_st) { AVStream *st; if (!(st = av_new_stream(s, 0))) return NULL; st->codec->codec_type = orig_st->codec->codec_type; st->priv_data = orig_st->priv_data; st->first_dts = orig_st->first_dts; return st; } static void real_parse_asm_rulebook(AVFormatContext *s, AVStream *orig_st, const char *p) { const char *end; int n_rules = 0, odd = 0; AVStream *st; /** * The ASMRuleBook contains a list of comma-separated strings per rule, * and each rule is separated by a ;. The last one also has a ; at the * end so we can use it as delimiter. * Every rule occurs twice, once for when the RTSP packet header marker * is set and once for if it isn't. We only read the first because we * don't care much (that's what the "odd" variable is for). * Each rule contains a set of one or more statements, optionally * preceeded by a single condition. If there's a condition, the rule * starts with a '#'. Multiple conditions are merged between brackets, * so there are never multiple conditions spread out over separate * statements. Generally, these conditions are bitrate limits (min/max) * for multi-bitrate streams. */ if (*p == '\"') p++; while (1) { if (!(end = strchr(p, ';'))) break; if (!odd && end != p) { if (n_rules > 0) st = add_dstream(s, orig_st); else st = orig_st; if (!st) break; real_parse_asm_rule(st, p, end); n_rules++; } p = end + 1; odd ^= 1; } } void ff_real_parse_sdp_a_line (AVFormatContext *s, int stream_index, const char *line) { const char *p = line; if (av_strstart(p, "ASMRuleBook:string;", &p)) real_parse_asm_rulebook(s, s->streams[stream_index], p); } static PayloadContext * rdt_new_context (void) { PayloadContext *rdt = av_mallocz(sizeof(PayloadContext)); av_open_input_stream(&rdt->rmctx, NULL, "", &rdt_demuxer, NULL); return rdt; } static void rdt_free_context (PayloadContext *rdt) { int i; for (i = 0; i < rdt->nb_rmst; i++) if (rdt->rmst[i]) { ff_rm_free_rmstream(rdt->rmst[i]); av_freep(&rdt->rmst[i]); } if (rdt->rmctx) av_close_input_stream(rdt->rmctx); av_freep(&rdt->mlti_data); av_freep(&rdt->rmst); av_free(rdt); } #define RDT_HANDLER(n, s, t) \ static RTPDynamicProtocolHandler ff_rdt_ ## n ## _handler = { \ .enc_name = s, \ .codec_type = t, \ .codec_id = CODEC_ID_NONE, \ .parse_sdp_a_line = rdt_parse_sdp_line, \ .open = rdt_new_context, \ .close = rdt_free_context, \ .parse_packet = rdt_parse_packet \ }; RDT_HANDLER(live_video, "x-pn-multirate-realvideo-live", AVMEDIA_TYPE_VIDEO); RDT_HANDLER(live_audio, "x-pn-multirate-realaudio-live", AVMEDIA_TYPE_AUDIO); RDT_HANDLER(video, "x-pn-realvideo", AVMEDIA_TYPE_VIDEO); RDT_HANDLER(audio, "x-pn-realaudio", AVMEDIA_TYPE_AUDIO); void av_register_rdt_dynamic_payload_handlers(void) { ff_register_dynamic_payload_handler(&ff_rdt_video_handler); ff_register_dynamic_payload_handler(&ff_rdt_audio_handler); ff_register_dynamic_payload_handler(&ff_rdt_live_video_handler); ff_register_dynamic_payload_handler(&ff_rdt_live_audio_handler); }