third_party_ffmpeg/libavcodec/flac.c

/*
 * FLAC common code
 * Copyright (c) 2009 Justin Ruggles
 *
 * This file is part of Libav.
 *
 * Libav 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.
 *
 * Libav 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 Libav; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include "libavutil/crc.h"
#include "flac.h"
#include "flacdata.h"

static const int8_t sample_size_table[] = { 0, 8, 12, 0, 16, 20, 24, 0 };

static int64_t get_utf8(GetBitContext *gb)
{
    int64_t val;
    GET_UTF8(val, get_bits(gb, 8), return -1;)
    return val;
}

int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
                                FLACFrameInfo *fi, int log_level_offset)
{
    int bs_code, sr_code, bps_code;

    /* frame sync code */
    if ((get_bits(gb, 15) & 0x7FFF) != 0x7FFC) {
        av_log(avctx, AV_LOG_ERROR + log_level_offset, "invalid sync code\n");
        return -1;
    }

    /* variable block size stream code */
    fi->is_var_size = get_bits1(gb);

    /* block size and sample rate codes */
    bs_code = get_bits(gb, 4);
    sr_code = get_bits(gb, 4);

    /* channels and decorrelation */
    fi->ch_mode = get_bits(gb, 4);
    if (fi->ch_mode < FLAC_MAX_CHANNELS) {
        fi->channels = fi->ch_mode + 1;
        fi->ch_mode = FLAC_CHMODE_INDEPENDENT;
    } else if (fi->ch_mode <= FLAC_CHMODE_MID_SIDE) {
        fi->channels = 2;
    } else {
        av_log(avctx, AV_LOG_ERROR + log_level_offset,
               "invalid channel mode: %d\n", fi->ch_mode);
        return -1;
    }

    /* bits per sample */
    bps_code = get_bits(gb, 3);
    if (bps_code == 3 || bps_code == 7) {
        av_log(avctx, AV_LOG_ERROR + log_level_offset,
               "invalid sample size code (%d)\n",
               bps_code);
        return -1;
    }
    fi->bps = sample_size_table[bps_code];

    /* reserved bit */
    if (get_bits1(gb)) {
        av_log(avctx, AV_LOG_ERROR + log_level_offset,
               "broken stream, invalid padding\n");
        return -1;
    }

    /* sample or frame count */
    fi->frame_or_sample_num = get_utf8(gb);
    if (fi->frame_or_sample_num < 0) {
        av_log(avctx, AV_LOG_ERROR + log_level_offset,
               "sample/frame number invalid; utf8 fscked\n");
        return -1;
    }

    /* blocksize */
    if (bs_code == 0) {
        av_log(avctx, AV_LOG_ERROR + log_level_offset,
               "reserved blocksize code: 0\n");
        return -1;
    } else if (bs_code == 6) {
        fi->blocksize = get_bits(gb, 8) + 1;
    } else if (bs_code == 7) {
        fi->blocksize = get_bits(gb, 16) + 1;
    } else {
        fi->blocksize = ff_flac_blocksize_table[bs_code];
    }

    /* sample rate */
    if (sr_code < 12) {
        fi->samplerate = ff_flac_sample_rate_table[sr_code];
    } else if (sr_code == 12) {
        fi->samplerate = get_bits(gb, 8) * 1000;
    } else if (sr_code == 13) {
        fi->samplerate = get_bits(gb, 16);
    } else if (sr_code == 14) {
        fi->samplerate = get_bits(gb, 16) * 10;
    } else {
        av_log(avctx, AV_LOG_ERROR + log_level_offset,
               "illegal sample rate code %d\n",
               sr_code);
        return -1;
    }

    /* header CRC-8 check */
    skip_bits(gb, 8);
    if (av_crc(av_crc_get_table(AV_CRC_8_ATM), 0, gb->buffer,
               get_bits_count(gb)/8)) {
        av_log(avctx, AV_LOG_ERROR + log_level_offset,
               "header crc mismatch\n");
        return -1;
    }

    return 0;
}

int ff_flac_get_max_frame_size(int blocksize, int ch, int bps)
{
    /* Technically, there is no limit to FLAC frame size, but an encoder
       should not write a frame that is larger than if verbatim encoding mode
       were to be used. */

    int count;

    count = 16;                  /* frame header */
    count += ch * ((7+bps+7)/8); /* subframe headers */
    if (ch == 2) {
        /* for stereo, need to account for using decorrelation */
        count += (( 2*bps+1) * blocksize + 7) / 8;
    } else {
        count += ( ch*bps    * blocksize + 7) / 8;
    }
    count += 2; /* frame footer */

    return count;
}