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ffmpeg / libavcodec / flac.c @ 2912e87a

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/*
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 * FLAC common code
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 * Copyright (c) 2009 Justin Ruggles
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 *
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 * This file is part of Libav.
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 *
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 * Libav is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * Libav is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * 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 Libav; 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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 */
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#include "libavutil/crc.h"
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#include "flac.h"
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#include "flacdata.h"
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static const int8_t sample_size_table[] = { 0, 8, 12, 0, 16, 20, 24, 0 };
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static int64_t get_utf8(GetBitContext *gb)
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{
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    int64_t val;
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    GET_UTF8(val, get_bits(gb, 8), return -1;)
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    return val;
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}
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int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
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                                FLACFrameInfo *fi, int log_level_offset)
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{
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    int bs_code, sr_code, bps_code;
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    /* frame sync code */
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    if ((get_bits(gb, 15) & 0x7FFF) != 0x7FFC) {
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        av_log(avctx, AV_LOG_ERROR + log_level_offset, "invalid sync code\n");
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        return -1;
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    }
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    /* variable block size stream code */
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    fi->is_var_size = get_bits1(gb);
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    /* block size and sample rate codes */
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    bs_code = get_bits(gb, 4);
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    sr_code = get_bits(gb, 4);
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    /* channels and decorrelation */
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    fi->ch_mode = get_bits(gb, 4);
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    if (fi->ch_mode < FLAC_MAX_CHANNELS) {
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        fi->channels = fi->ch_mode + 1;
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        fi->ch_mode = FLAC_CHMODE_INDEPENDENT;
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    } else if (fi->ch_mode <= FLAC_CHMODE_MID_SIDE) {
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        fi->channels = 2;
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    } else {
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        av_log(avctx, AV_LOG_ERROR + log_level_offset,
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               "invalid channel mode: %d\n", fi->ch_mode);
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        return -1;
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    }
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    /* bits per sample */
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    bps_code = get_bits(gb, 3);
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    if (bps_code == 3 || bps_code == 7) {
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        av_log(avctx, AV_LOG_ERROR + log_level_offset,
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               "invalid sample size code (%d)\n",
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               bps_code);
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        return -1;
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    }
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    fi->bps = sample_size_table[bps_code];
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    /* reserved bit */
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    if (get_bits1(gb)) {
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        av_log(avctx, AV_LOG_ERROR + log_level_offset,
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               "broken stream, invalid padding\n");
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        return -1;
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    }
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    /* sample or frame count */
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    fi->frame_or_sample_num = get_utf8(gb);
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    if (fi->frame_or_sample_num < 0) {
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        av_log(avctx, AV_LOG_ERROR + log_level_offset,
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               "sample/frame number invalid; utf8 fscked\n");
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        return -1;
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    }
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    /* blocksize */
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    if (bs_code == 0) {
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        av_log(avctx, AV_LOG_ERROR + log_level_offset,
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               "reserved blocksize code: 0\n");
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        return -1;
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    } else if (bs_code == 6) {
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        fi->blocksize = get_bits(gb, 8) + 1;
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    } else if (bs_code == 7) {
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        fi->blocksize = get_bits(gb, 16) + 1;
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    } else {
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        fi->blocksize = ff_flac_blocksize_table[bs_code];
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    }
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    /* sample rate */
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    if (sr_code < 12) {
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        fi->samplerate = ff_flac_sample_rate_table[sr_code];
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    } else if (sr_code == 12) {
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        fi->samplerate = get_bits(gb, 8) * 1000;
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    } else if (sr_code == 13) {
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        fi->samplerate = get_bits(gb, 16);
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    } else if (sr_code == 14) {
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        fi->samplerate = get_bits(gb, 16) * 10;
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    } else {
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        av_log(avctx, AV_LOG_ERROR + log_level_offset,
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               "illegal sample rate code %d\n",
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               sr_code);
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        return -1;
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    }
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    /* header CRC-8 check */
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    skip_bits(gb, 8);
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    if (av_crc(av_crc_get_table(AV_CRC_8_ATM), 0, gb->buffer,
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               get_bits_count(gb)/8)) {
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        av_log(avctx, AV_LOG_ERROR + log_level_offset,
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               "header crc mismatch\n");
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        return -1;
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    }
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    return 0;
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}
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int ff_flac_get_max_frame_size(int blocksize, int ch, int bps)
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{
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    /* Technically, there is no limit to FLAC frame size, but an encoder
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       should not write a frame that is larger than if verbatim encoding mode
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       were to be used. */
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    int count;
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    count = 16;                  /* frame header */
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    count += ch * ((7+bps+7)/8); /* subframe headers */
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    if (ch == 2) {
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        /* for stereo, need to account for using decorrelation */
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        count += (( 2*bps+1) * blocksize + 7) / 8;
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    } else {
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        count += ( ch*bps    * blocksize + 7) / 8;
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    }
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    count += 2; /* frame footer */
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    return count;
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}