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/*
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 * Atrac 1 compatible decoder
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 * Copyright (c) 2009 Maxim Poliakovski
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 * Copyright (c) 2009 Benjamin Larsson
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 *
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 * This file is part of FFmpeg.
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 *
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 * FFmpeg 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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 * FFmpeg 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 FFmpeg; 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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/**
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 * @file libavcodec/atrac1.c
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 * Atrac 1 compatible decoder.
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 * This decoder handles raw ATRAC1 data and probably SDDS data.
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 */
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/* Many thanks to Tim Craig for all the help! */
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#include <math.h>
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#include <stddef.h>
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#include <stdio.h>
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#include "avcodec.h"
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#include "get_bits.h"
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#include "dsputil.h"
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#include "atrac.h"
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#include "atrac1data.h"
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#define AT1_MAX_BFU      52                 ///< max number of block floating units in a sound unit
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#define AT1_SU_SIZE      212                ///< number of bytes in a sound unit
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#define AT1_SU_SAMPLES   512                ///< number of samples in a sound unit
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#define AT1_FRAME_SIZE   AT1_SU_SIZE * 2
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#define AT1_SU_MAX_BITS  AT1_SU_SIZE * 8
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#define AT1_MAX_CHANNELS 2
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#define AT1_QMF_BANDS    3
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#define IDX_LOW_BAND     0
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#define IDX_MID_BAND     1
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#define IDX_HIGH_BAND    2
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/**
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 * Sound unit struct, one unit is used per channel
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 */
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typedef struct {
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    int                 log2_block_count[AT1_QMF_BANDS];    ///< log2 number of blocks in a band
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    int                 num_bfus;                           ///< number of Block Floating Units
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    float*              spectrum[2];
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    DECLARE_ALIGNED(16, float, spec1)[AT1_SU_SAMPLES];     ///< mdct buffer
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    DECLARE_ALIGNED(16, float, spec2)[AT1_SU_SAMPLES];     ///< mdct buffer
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    DECLARE_ALIGNED(16, float, fst_qmf_delay)[46];         ///< delay line for the 1st stacked QMF filter
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    DECLARE_ALIGNED(16, float, snd_qmf_delay)[46];         ///< delay line for the 2nd stacked QMF filter
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    DECLARE_ALIGNED(16, float, last_qmf_delay)[256+23];    ///< delay line for the last stacked QMF filter
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} AT1SUCtx;
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/**
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 * The atrac1 context, holds all needed parameters for decoding
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 */
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typedef struct {
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    AT1SUCtx            SUs[AT1_MAX_CHANNELS];              ///< channel sound unit
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    DECLARE_ALIGNED(16, float, spec)[AT1_SU_SAMPLES];      ///< the mdct spectrum buffer
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    DECLARE_ALIGNED(16, float,  low)[256];
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    DECLARE_ALIGNED(16, float,  mid)[256];
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    DECLARE_ALIGNED(16, float, high)[512];
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    float*              bands[3];
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    DECLARE_ALIGNED(16, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];
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    FFTContext          mdct_ctx[3];
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    int                 channels;
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    DSPContext          dsp;
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} AT1Ctx;
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/** size of the transform in samples in the long mode for each QMF band */
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static const uint16_t samples_per_band[3] = {128, 128, 256};
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static const uint8_t   mdct_long_nbits[3] = {7, 7, 8};
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static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
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                      int rev_spec)
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{
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    FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
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    int transf_size = 1 << nbits;
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    if (rev_spec) {
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        int i;
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        for (i = 0; i < transf_size / 2; i++)
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            FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
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    }
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    ff_imdct_half(mdct_context, out, spec);
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}
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static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
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{
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    int          band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
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    unsigned int start_pos, ref_pos = 0, pos = 0;
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    for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
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        float *prev_buf;
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        int j;
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        band_samples = samples_per_band[band_num];
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        log2_block_count = su->log2_block_count[band_num];
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        /* number of mdct blocks in the current QMF band: 1 - for long mode */
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        /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
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        num_blocks = 1 << log2_block_count;
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        if (num_blocks == 1) {
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            /* mdct block size in samples: 128 (long mode, low & mid bands), */
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            /* 256 (long mode, high band) and 32 (short mode, all bands) */
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            block_size = band_samples >> log2_block_count;
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            /* calc transform size in bits according to the block_size_mode */
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            nbits = mdct_long_nbits[band_num] - log2_block_count;
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            if (nbits != 5 && nbits != 7 && nbits != 8)
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                return -1;
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        } else {
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            block_size = 32;
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            nbits = 5;
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        }
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        start_pos = 0;
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        prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
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        for (j=0; j < num_blocks; j++) {
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            at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num);
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            /* overlap and window */
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            q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
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                                      &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 0, 16);
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            prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
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            start_pos += block_size;
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            pos += block_size;
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        }
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        if (num_blocks == 1)
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            memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
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        ref_pos += band_samples;
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    }
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    /* Swap buffers so the mdct overlap works */
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    FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
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    return 0;
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}
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/**
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 * Parse the block size mode byte
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 */
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static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
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{
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    int log2_block_count_tmp, i;
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    for (i = 0; i < 2; i++) {
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        /* low and mid band */
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        log2_block_count_tmp = get_bits(gb, 2);
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        if (log2_block_count_tmp & 1)
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            return -1;
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        log2_block_cnt[i] = 2 - log2_block_count_tmp;
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    }
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    /* high band */
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    log2_block_count_tmp = get_bits(gb, 2);
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    if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
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        return -1;
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    log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
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    skip_bits(gb, 2);
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    return 0;
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}
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static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
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                              float spec[AT1_SU_SAMPLES])
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{
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    int bits_used, band_num, bfu_num, i;
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    uint8_t idwls[AT1_MAX_BFU];                 ///< the word length indexes for each BFU
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    uint8_t idsfs[AT1_MAX_BFU];                 ///< the scalefactor indexes for each BFU
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    /* parse the info byte (2nd byte) telling how much BFUs were coded */
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    su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
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    /* calc number of consumed bits:
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        num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
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        + info_byte_copy(8bits) + log2_block_count_copy(8bits) */
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    bits_used = su->num_bfus * 10 + 32 +
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                bfu_amount_tab2[get_bits(gb, 2)] +
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                (bfu_amount_tab3[get_bits(gb, 3)] << 1);
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    /* get word length index (idwl) for each BFU */
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    for (i = 0; i < su->num_bfus; i++)
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        idwls[i] = get_bits(gb, 4);
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    /* get scalefactor index (idsf) for each BFU */
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    for (i = 0; i < su->num_bfus; i++)
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        idsfs[i] = get_bits(gb, 6);
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    /* zero idwl/idsf for empty BFUs */
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    for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
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        idwls[i] = idsfs[i] = 0;
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    /* read in the spectral data and reconstruct MDCT spectrum of this channel */
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    for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
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        for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) {
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            int pos;
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            int num_specs = specs_per_bfu[bfu_num];
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            int word_len  = !!idwls[bfu_num] + idwls[bfu_num];
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            float scale_factor = sf_table[idsfs[bfu_num]];
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            bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */
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            /* check for bitstream overflow */
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            if (bits_used > AT1_SU_MAX_BITS)
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                return -1;
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            /* get the position of the 1st spec according to the block size mode */
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            pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
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            if (word_len) {
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                float   max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
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                for (i = 0; i < num_specs; i++) {
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                    /* read in a quantized spec and convert it to
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                     * signed int and then inverse quantization
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                     */
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                    spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
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                }
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            } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
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                memset(&spec[pos], 0, num_specs * sizeof(float));
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            }
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        }
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    }
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    return 0;
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}
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void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
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{
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    float temp[256];
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    float iqmf_temp[512 + 46];
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    /* combine low and middle bands */
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    atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
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    /* delay the signal of the high band by 23 samples */
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    memcpy( su->last_qmf_delay,    &su->last_qmf_delay[256], sizeof(float) *  23);
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    memcpy(&su->last_qmf_delay[23], q->bands[2],             sizeof(float) * 256);
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    /* combine (low + middle) and high bands */
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    atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
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}
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static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
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                               int *data_size, AVPacket *avpkt)
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{
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    const uint8_t *buf = avpkt->data;
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    int buf_size       = avpkt->size;
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    AT1Ctx *q          = avctx->priv_data;
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    int ch, ret, i;
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    GetBitContext gb;
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    float* samples = data;
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    if (buf_size < 212 * q->channels) {
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        av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n");
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        return -1;
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    }
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    for (ch = 0; ch < q->channels; ch++) {
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        AT1SUCtx* su = &q->SUs[ch];
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        init_get_bits(&gb, &buf[212 * ch], 212 * 8);
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        /* parse block_size_mode, 1st byte */
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        ret = at1_parse_bsm(&gb, su->log2_block_count);
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        if (ret < 0)
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            return ret;
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        ret = at1_unpack_dequant(&gb, su, q->spec);
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        if (ret < 0)
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            return ret;
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        ret = at1_imdct_block(su, q);
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        if (ret < 0)
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            return ret;
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        at1_subband_synthesis(q, su, q->out_samples[ch]);
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    }
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    /* round, convert to 16bit and interleave */
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    if (q->channels == 1) {
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        /* mono */
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        q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1 << 15),
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                            32700.0 / (1 << 15), AT1_SU_SAMPLES);
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    } else {
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        /* stereo */
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        for (i = 0; i < AT1_SU_SAMPLES; i++) {
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            samples[i * 2]     = av_clipf(q->out_samples[0][i],
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                                          -32700.0 / (1 << 15),
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                                           32700.0 / (1 << 15));
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            samples[i * 2 + 1] = av_clipf(q->out_samples[1][i],
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                                          -32700.0 / (1 << 15),
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                                           32700.0 / (1 << 15));
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        }
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    }
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    *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
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    return avctx->block_align;
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}
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static av_cold int atrac1_decode_init(AVCodecContext *avctx)
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{
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    AT1Ctx *q = avctx->priv_data;
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    avctx->sample_fmt = SAMPLE_FMT_FLT;
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    q->channels = avctx->channels;
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    /* Init the mdct transforms */
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    ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15));
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    ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15));
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    ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15));
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    ff_init_ff_sine_windows(5);
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    atrac_generate_tables();
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    dsputil_init(&q->dsp, avctx);
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    q->bands[0] = q->low;
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    q->bands[1] = q->mid;
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    q->bands[2] = q->high;
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    /* Prepare the mdct overlap buffers */
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    q->SUs[0].spectrum[0] = q->SUs[0].spec1;
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    q->SUs[0].spectrum[1] = q->SUs[0].spec2;
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    q->SUs[1].spectrum[0] = q->SUs[1].spec1;
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    q->SUs[1].spectrum[1] = q->SUs[1].spec2;
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    return 0;
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}
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static av_cold int atrac1_decode_end(AVCodecContext * avctx) {
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    AT1Ctx *q = avctx->priv_data;
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    ff_mdct_end(&q->mdct_ctx[0]);
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    ff_mdct_end(&q->mdct_ctx[1]);
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    ff_mdct_end(&q->mdct_ctx[2]);
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    return 0;
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}
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AVCodec atrac1_decoder = {
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    .name = "atrac1",
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    .type = CODEC_TYPE_AUDIO,
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    .id = CODEC_ID_ATRAC1,
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    .priv_data_size = sizeof(AT1Ctx),
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    .init = atrac1_decode_init,
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    .close = atrac1_decode_end,
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    .decode = atrac1_decode_frame,
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    .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
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};