Mercurial > libavformat.hg
view rdt.c @ 3916:4a7174c2b34e libavformat
matroskadec: demux relevant subtitle packets after a seek
only works for subtitle tracks which were properly indexed up to the seek point
author | aurel |
---|---|
date | Tue, 09 Sep 2008 12:10:25 +0000 |
parents | 91987686113d |
children | ed6f6aa977ec |
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/* * 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 rdt.c * @brief Realmedia RTSP protocol (RDT) support * @author Ronald S. Bultje <rbultje@ronald.bitfreak.net> */ #include "avformat.h" #include "libavutil/avstring.h" #include "rtp_internal.h" #include "rdt.h" #include "libavutil/base64.h" #include "libavutil/md5.h" #include "rm.h" #include "internal.h" typedef struct rdt_data { AVFormatContext *rmctx; uint8_t *mlti_data; unsigned int mlti_data_size; uint32_t prev_sn, prev_ts; char buffer[RTP_MAX_PACKET_LENGTH + FF_INPUT_BUFFER_PADDING_SIZE]; } rdt_data; 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); for (i=0;i<32;i++) response[i] = tolower(response[i]); /* 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 (rdt_data *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; url_open_buf(&pb, rdt->mlti_data, rdt->mlti_data_size, URL_RDONLY); 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); } rdt->rmctx->pb = pb; if (ff_rm_read_mdpr_codecdata(rdt->rmctx, st, size) < 0) return -1; url_close_buf(pb); return 0; } /** * Actual data handling. */ static int rdt_parse_header(struct RTPDemuxContext *s, const uint8_t *buf, int len, int *seq, uint32_t *timestamp, int *flags) { rdt_data *rdt = s->dynamic_protocol_context; int consumed = 0, sn; if (buf[0] < 0x40 || buf[0] > 0x42) { buf += 9; len -= 9; consumed += 9; } sn = (buf[0]>>1) & 0x1f; *seq = AV_RB16(buf+1); *timestamp = AV_RB32(buf+4); if (!(buf[3] & 1) && (sn != rdt->prev_sn || *timestamp != rdt->prev_ts)) { *flags |= PKT_FLAG_KEY; rdt->prev_sn = sn; rdt->prev_ts = *timestamp; } return consumed + 10; } /**< return 0 on packet, no more left, 1 on packet, 1 on partial packet... */ static int rdt_parse_packet (RTPDemuxContext *s, AVPacket *pkt, uint32_t *timestamp, const uint8_t *buf, int len, int flags) { rdt_data *rdt = s->dynamic_protocol_context; int seq = 1, res; ByteIOContext *pb = rdt->rmctx->pb; RMContext *rm = rdt->rmctx->priv_data; AVStream *st = s->st; if (rm->audio_pkt_cnt == 0) { int pos; url_open_buf (&pb, buf, len, URL_RDONLY); flags = (flags & PKT_FLAG_KEY) ? 2 : 0; rdt->rmctx->pb = pb; res = ff_rm_parse_packet (rdt->rmctx, st, len, pkt, &seq, &flags, timestamp); pos = url_ftell(pb); url_close_buf (pb); if (res < 0) return res; if (rm->audio_pkt_cnt > 0 && st->codec->codec_id == CODEC_ID_AAC) { memcpy (rdt->buffer, buf + pos, len - pos); url_open_buf (&pb, rdt->buffer, len - pos, URL_RDONLY); rdt->rmctx->pb = pb; } } else { ff_rm_retrieve_cache (rdt->rmctx, st, pkt); if (rm->audio_pkt_cnt == 0 && st->codec->codec_id == CODEC_ID_AAC) url_close_buf (pb); } pkt->stream_index = st->index; pkt->pts = *timestamp; return rm->audio_pkt_cnt > 0; } int ff_rdt_parse_packet(RTPDemuxContext *s, AVPacket *pkt, const uint8_t *buf, int len) { int seq, flags = 0; uint32_t timestamp; int rv= 0; if (!buf) { /* 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= rdt_parse_packet(s, pkt, ×tamp, NULL, 0, flags); return rv; } if (len < 12) return -1; rv = rdt_parse_header(s, buf, len, &seq, ×tamp, &flags); if (rv < 0) return rv; buf += rv; len -= rv; s->seq = seq; rv = rdt_parse_packet(s, pkt, ×tamp, buf, len, flags); return rv; } void ff_rdt_subscribe_rule (RTPDemuxContext *s, char *cmd, int size, int stream_nr, int rule_nr) { rdt_data *rdt = s->dynamic_protocol_context; av_strlcatf(cmd, size, "stream=%d;rule=%d,stream=%d;rule=%d", stream_nr, rule_nr, stream_nr, rule_nr + 1); rdt_load_mdpr(rdt, s->st, 0); } 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 (AVStream *stream, void *d, const char *line) { rdt_data *rdt = d; 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); return 0; } static void * rdt_new_extradata (void) { rdt_data *rdt = av_mallocz(sizeof(rdt_data)); av_open_input_stream(&rdt->rmctx, NULL, "", &rdt_demuxer, NULL); rdt->prev_ts = -1; rdt->prev_sn = -1; return rdt; } static void rdt_free_extradata (void *d) { rdt_data *rdt = d; if (rdt->rmctx) av_close_input_stream(rdt->rmctx); av_freep(&rdt->mlti_data); av_free(rdt); } #define RDT_HANDLER(n, s, t) \ static RTPDynamicProtocolHandler ff_rdt_ ## n ## _handler = { \ s, \ t, \ CODEC_ID_NONE, \ rdt_parse_sdp_line, \ rdt_new_extradata, \ rdt_free_extradata \ }; RDT_HANDLER(live_video, "x-pn-multirate-realvideo-live", CODEC_TYPE_VIDEO); RDT_HANDLER(live_audio, "x-pn-multirate-realaudio-live", CODEC_TYPE_AUDIO); RDT_HANDLER(video, "x-pn-realvideo", CODEC_TYPE_VIDEO); RDT_HANDLER(audio, "x-pn-realaudio", CODEC_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); }