PeerStreamer

/*

 * Atrac 1 compatible decoder

 * Copyright (c) 2009 Maxim Poliakovski

 * Copyright (c) 2009 Benjamin Larsson

 *

 * 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 libavcodec/atrac1.c

 * Atrac 1 compatible decoder.

 * This decoder handles raw ATRAC1 data and probably SDDS data.

 */

/* Many thanks to Tim Craig for all the help! */

#include <math.h>

#include <stddef.h>

#include <stdio.h>

#include "avcodec.h"

#include "get_bits.h"

#include "dsputil.h"

#include "fft.h"

#include "atrac.h"

#include "atrac1data.h"

#define AT1_MAX_BFU      52                 ///< max number of block floating units in a sound unit

#define AT1_SU_SIZE      212                ///< number of bytes in a sound unit

#define AT1_SU_SAMPLES   512                ///< number of samples in a sound unit

#define AT1_FRAME_SIZE   AT1_SU_SIZE * 2

#define AT1_SU_MAX_BITS  AT1_SU_SIZE * 8

#define AT1_MAX_CHANNELS 2

#define AT1_QMF_BANDS    3

#define IDX_LOW_BAND     0

#define IDX_MID_BAND     1

#define IDX_HIGH_BAND    2

/**

 * Sound unit struct, one unit is used per channel

 */

typedef struct {

    int                 log2_block_count[AT1_QMF_BANDS];    ///< log2 number of blocks in a band

    int                 num_bfus;                           ///< number of Block Floating Units

    float*              spectrum[2];

    DECLARE_ALIGNED(16, float, spec1)[AT1_SU_SAMPLES];     ///< mdct buffer

    DECLARE_ALIGNED(16, float, spec2)[AT1_SU_SAMPLES];     ///< mdct buffer

    DECLARE_ALIGNED(16, float, fst_qmf_delay)[46];         ///< delay line for the 1st stacked QMF filter

    DECLARE_ALIGNED(16, float, snd_qmf_delay)[46];         ///< delay line for the 2nd stacked QMF filter

    DECLARE_ALIGNED(16, float, last_qmf_delay)[256+23];    ///< delay line for the last stacked QMF filter

} AT1SUCtx;

/**

 * The atrac1 context, holds all needed parameters for decoding

 */

typedef struct {

    AT1SUCtx            SUs[AT1_MAX_CHANNELS];              ///< channel sound unit

    DECLARE_ALIGNED(16, float, spec)[AT1_SU_SAMPLES];      ///< the mdct spectrum buffer

    DECLARE_ALIGNED(16, float,  low)[256];

    DECLARE_ALIGNED(16, float,  mid)[256];

    DECLARE_ALIGNED(16, float, high)[512];

    float*              bands[3];

    DECLARE_ALIGNED(16, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES];

    FFTContext          mdct_ctx[3];

    int                 channels;

    DSPContext          dsp;

} AT1Ctx;

/** size of the transform in samples in the long mode for each QMF band */

static const uint16_t samples_per_band[3] = {128, 128, 256};

static const uint8_t   mdct_long_nbits[3] = {7, 7, 8};

static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,

                      int rev_spec)

{

    FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];

    int transf_size = 1 << nbits;

    if (rev_spec) {

        int i;

        for (i = 0; i < transf_size / 2; i++)

            FFSWAP(float, spec[i], spec[transf_size - 1 - i]);

    }

    ff_imdct_half(mdct_context, out, spec);

}

static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)

{

    int          band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;

    unsigned int start_pos, ref_pos = 0, pos = 0;

    for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {

        float *prev_buf;

        int j;

        band_samples = samples_per_band[band_num];

        log2_block_count = su->log2_block_count[band_num];

        /* number of mdct blocks in the current QMF band: 1 - for long mode */

        /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/

        num_blocks = 1 << log2_block_count;

        if (num_blocks == 1) {

            /* mdct block size in samples: 128 (long mode, low & mid bands), */

            /* 256 (long mode, high band) and 32 (short mode, all bands) */

            block_size = band_samples >> log2_block_count;

            /* calc transform size in bits according to the block_size_mode */

            nbits = mdct_long_nbits[band_num] - log2_block_count;

            if (nbits != 5 && nbits != 7 && nbits != 8)

                return -1;

        } else {

            block_size = 32;

            nbits = 5;

        }

        start_pos = 0;

        prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];

        for (j=0; j < num_blocks; j++) {

            at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num);

            /* overlap and window */

            q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,

                                      &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 0, 16);

            prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];

            start_pos += block_size;

            pos += block_size;

        }

        if (num_blocks == 1)

            memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));

        ref_pos += band_samples;

    }

    /* Swap buffers so the mdct overlap works */

    FFSWAP(float*, su->spectrum[0], su->spectrum[1]);

    return 0;

}

/**

 * Parse the block size mode byte

 */

static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])

{

    int log2_block_count_tmp, i;

    for (i = 0; i < 2; i++) {

        /* low and mid band */

        log2_block_count_tmp = get_bits(gb, 2);

        if (log2_block_count_tmp & 1)

            return -1;

        log2_block_cnt[i] = 2 - log2_block_count_tmp;

    }

    /* high band */

    log2_block_count_tmp = get_bits(gb, 2);

    if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)

        return -1;

    log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;

    skip_bits(gb, 2);

    return 0;

}

static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,

                              float spec[AT1_SU_SAMPLES])

{

    int bits_used, band_num, bfu_num, i;

    uint8_t idwls[AT1_MAX_BFU];                 ///< the word length indexes for each BFU

    uint8_t idsfs[AT1_MAX_BFU];                 ///< the scalefactor indexes for each BFU

    /* parse the info byte (2nd byte) telling how much BFUs were coded */

    su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];

    /* calc number of consumed bits:

        num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)

        + info_byte_copy(8bits) + log2_block_count_copy(8bits) */

    bits_used = su->num_bfus * 10 + 32 +

                bfu_amount_tab2[get_bits(gb, 2)] +

                (bfu_amount_tab3[get_bits(gb, 3)] << 1);

    /* get word length index (idwl) for each BFU */

    for (i = 0; i < su->num_bfus; i++)

        idwls[i] = get_bits(gb, 4);

    /* get scalefactor index (idsf) for each BFU */

    for (i = 0; i < su->num_bfus; i++)

        idsfs[i] = get_bits(gb, 6);

    /* zero idwl/idsf for empty BFUs */

    for (i = su->num_bfus; i < AT1_MAX_BFU; i++)

        idwls[i] = idsfs[i] = 0;

    /* read in the spectral data and reconstruct MDCT spectrum of this channel */

    for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {

        for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) {

            int pos;

            int num_specs = specs_per_bfu[bfu_num];

            int word_len  = !!idwls[bfu_num] + idwls[bfu_num];

            float scale_factor = sf_table[idsfs[bfu_num]];

            bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */

            /* check for bitstream overflow */

            if (bits_used > AT1_SU_MAX_BITS)

                return -1;

            /* get the position of the 1st spec according to the block size mode */

            pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];

            if (word_len) {

                float   max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);

                for (i = 0; i < num_specs; i++) {

                    /* read in a quantized spec and convert it to

                     * signed int and then inverse quantization

                     */

                    spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;

                }

            } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */

                memset(&spec[pos], 0, num_specs * sizeof(float));

            }

        }

    }

    return 0;

}

static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)

{

    float temp[256];

    float iqmf_temp[512 + 46];

    /* combine low and middle bands */

    atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);

    /* delay the signal of the high band by 23 samples */

    memcpy( su->last_qmf_delay,    &su->last_qmf_delay[256], sizeof(float) *  23);

    memcpy(&su->last_qmf_delay[23], q->bands[2],             sizeof(float) * 256);

    /* combine (low + middle) and high bands */

    atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);

}

static int atrac1_decode_frame(AVCodecContext *avctx, void *data,

                               int *data_size, AVPacket *avpkt)

{

    const uint8_t *buf = avpkt->data;

    int buf_size       = avpkt->size;

    AT1Ctx *q          = avctx->priv_data;

    int ch, ret, i;

    GetBitContext gb;

    float* samples = data;

    if (buf_size < 212 * q->channels) {

        av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n");

        return -1;

    }

    for (ch = 0; ch < q->channels; ch++) {

        AT1SUCtx* su = &q->SUs[ch];

        init_get_bits(&gb, &buf[212 * ch], 212 * 8);

        /* parse block_size_mode, 1st byte */

        ret = at1_parse_bsm(&gb, su->log2_block_count);

        if (ret < 0)

            return ret;

        ret = at1_unpack_dequant(&gb, su, q->spec);

        if (ret < 0)

            return ret;

        ret = at1_imdct_block(su, q);

        if (ret < 0)

            return ret;

        at1_subband_synthesis(q, su, q->out_samples[ch]);

    }

    /* round, convert to 16bit and interleave */

    if (q->channels == 1) {

        /* mono */

        q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1 << 15),

                            32700.0 / (1 << 15), AT1_SU_SAMPLES);

    } else {

        /* stereo */

        for (i = 0; i < AT1_SU_SAMPLES; i++) {

            samples[i * 2]     = av_clipf(q->out_samples[0][i],

                                          -32700.0 / (1 << 15),

                                           32700.0 / (1 << 15));

            samples[i * 2 + 1] = av_clipf(q->out_samples[1][i],

                                          -32700.0 / (1 << 15),

                                           32700.0 / (1 << 15));

        }

    }

    *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);

    return avctx->block_align;

}

static av_cold int atrac1_decode_init(AVCodecContext *avctx)

{

    AT1Ctx *q = avctx->priv_data;

    avctx->sample_fmt = SAMPLE_FMT_FLT;

    q->channels = avctx->channels;

    /* Init the mdct transforms */

    ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15));

    ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15));

    ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15));

    ff_init_ff_sine_windows(5);

    atrac_generate_tables();

    dsputil_init(&q->dsp, avctx);

    q->bands[0] = q->low;

    q->bands[1] = q->mid;

    q->bands[2] = q->high;

    /* Prepare the mdct overlap buffers */

    q->SUs[0].spectrum[0] = q->SUs[0].spec1;

    q->SUs[0].spectrum[1] = q->SUs[0].spec2;

    q->SUs[1].spectrum[0] = q->SUs[1].spec1;

    q->SUs[1].spectrum[1] = q->SUs[1].spec2;

    return 0;

}

static av_cold int atrac1_decode_end(AVCodecContext * avctx) {

    AT1Ctx *q = avctx->priv_data;

    ff_mdct_end(&q->mdct_ctx[0]);

    ff_mdct_end(&q->mdct_ctx[1]);

    ff_mdct_end(&q->mdct_ctx[2]);

    return 0;

}

AVCodec atrac1_decoder = {

    .name = "atrac1",

    .type = CODEC_TYPE_AUDIO,

    .id = CODEC_ID_ATRAC1,

    .priv_data_size = sizeof(AT1Ctx),

    .init = atrac1_decode_init,

    .close = atrac1_decode_end,

    .decode = atrac1_decode_frame,

    .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),

};