Mercurial > libavformat.hg
view flacdec.c @ 4723:a2390c6a35e6 libavformat
Fix index generation in the way that it was supposed to be used. See the
discussion in the ML thread "[PATCH] rmdec.c: merge old/new packet reading
code".
Over time, this code broke somewhat, e.g. seq was never actually written
into (and was thus always 1, therefore the seq condition was always true),
whereas it was supposed to be set to the sequence number of the video slice
in case the video frame is divided over multiple RM packets (slices). The
problem of this is that packets other than those containing the beginning
of a video frame would be indexed as well.
Secondly, flags&2 is supposed to be true for video keyframes and for these
audio packets containing the start of a block. For some codecs (e.g. AAC),
that is every single packet, whereas for others (e.g. cook), that is the
packet containing the first of a series of scrambled packets that are to be
descrambled together. Indexing any of the following would lead to incomplete
and thus useless frames. Problem here is that flags would be reset to 2 to
indicate that the first packet is ready to be returned, and in addition if
no data was left to be returned (which is always true for the first packet),
then we wouldn't actually write the index entry anyway.
All in all, the idea was good and it probably worked at some point, but that
is long ago. This patch should at the very least make it likely for this code
to be executed again at the right times, i.e. the way it was originally
intended to be used.
author | rbultje |
---|---|
date | Sun, 15 Mar 2009 20:14:25 +0000 |
parents | 478a7c56266a |
children | 9a8c5a8c64ea |
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/* * Raw FLAC demuxer * Copyright (c) 2001 Fabrice Bellard * * 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 */ #include "libavcodec/flac.h" #include "avformat.h" #include "raw.h" #include "id3v2.h" #include "oggdec.h" static int flac_read_header(AVFormatContext *s, AVFormatParameters *ap) { uint8_t buf[ID3v2_HEADER_SIZE]; int ret, metadata_last=0, metadata_type, metadata_size, found_streaminfo=0; uint8_t header[4]; uint8_t *buffer=NULL; AVStream *st = av_new_stream(s, 0); if (!st) return AVERROR(ENOMEM); st->codec->codec_type = CODEC_TYPE_AUDIO; st->codec->codec_id = CODEC_ID_FLAC; st->need_parsing = AVSTREAM_PARSE_FULL; /* the parameters will be extracted from the compressed bitstream */ /* skip ID3v2 header if found */ ret = get_buffer(s->pb, buf, ID3v2_HEADER_SIZE); if (ret == ID3v2_HEADER_SIZE && ff_id3v2_match(buf)) { int len = ff_id3v2_tag_len(buf); url_fseek(s->pb, len - ID3v2_HEADER_SIZE, SEEK_CUR); } else { url_fseek(s->pb, 0, SEEK_SET); } /* if fLaC marker is not found, assume there is no header */ if (get_le32(s->pb) != MKTAG('f','L','a','C')) return 0; /* process metadata blocks */ while (!url_feof(s->pb) && !metadata_last) { get_buffer(s->pb, header, 4); ff_flac_parse_block_header(header, &metadata_last, &metadata_type, &metadata_size); switch (metadata_type) { /* allocate and read metadata block for supported types */ case FLAC_METADATA_TYPE_STREAMINFO: case FLAC_METADATA_TYPE_VORBIS_COMMENT: buffer = av_mallocz(metadata_size + FF_INPUT_BUFFER_PADDING_SIZE); if (!buffer) { return AVERROR_NOMEM; } if (get_buffer(s->pb, buffer, metadata_size) != metadata_size) { av_freep(&buffer); return AVERROR_IO; } break; /* skip metadata block for unsupported types */ default: ret = url_fseek(s->pb, metadata_size, SEEK_CUR); if (ret < 0) return ret; } if (metadata_type == FLAC_METADATA_TYPE_STREAMINFO) { FLACStreaminfo si; /* STREAMINFO can only occur once */ if (found_streaminfo) { av_freep(&buffer); return AVERROR_INVALIDDATA; } if (metadata_size != FLAC_STREAMINFO_SIZE) { av_freep(&buffer); return AVERROR_INVALIDDATA; } found_streaminfo = 1; st->codec->extradata = buffer; st->codec->extradata_size = metadata_size; buffer = NULL; /* get codec params from STREAMINFO header */ ff_flac_parse_streaminfo(st->codec, &si, st->codec->extradata); /* set time base and duration */ if (si.samplerate > 0) { av_set_pts_info(st, 64, 1, si.samplerate); if (si.samples > 0) st->duration = si.samples; } } else { /* STREAMINFO must be the first block */ if (!found_streaminfo) { av_freep(&buffer); return AVERROR_INVALIDDATA; } /* process supported blocks other than STREAMINFO */ if (metadata_type == FLAC_METADATA_TYPE_VORBIS_COMMENT) { if (vorbis_comment(s, buffer, metadata_size)) { av_log(s, AV_LOG_WARNING, "error parsing VorbisComment metadata\n"); } } av_freep(&buffer); } } return 0; } static int flac_probe(AVProbeData *p) { uint8_t *bufptr = p->buf; uint8_t *end = p->buf + p->buf_size; if(ff_id3v2_match(bufptr)) bufptr += ff_id3v2_tag_len(bufptr); if(bufptr > end-4 || memcmp(bufptr, "fLaC", 4)) return 0; else return AVPROBE_SCORE_MAX/2; } AVInputFormat flac_demuxer = { "flac", NULL_IF_CONFIG_SMALL("raw FLAC"), 0, flac_probe, flac_read_header, ff_raw_read_partial_packet, .flags= AVFMT_GENERIC_INDEX, .extensions = "flac", .value = CODEC_ID_FLAC, };