/*
 * 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
 * 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 "fmtconvert.h"
#include "sinewin.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(32, float, spec1)[AT1_SU_SAMPLES];     ///< mdct buffer
    DECLARE_ALIGNED(32, float, spec2)[AT1_SU_SAMPLES];     ///< mdct buffer
    DECLARE_ALIGNED(32, float, fst_qmf_delay)[46];         ///< delay line for the 1st stacked QMF filter
    DECLARE_ALIGNED(32, float, snd_qmf_delay)[46];         ///< delay line for the 2nd stacked QMF filter
    DECLARE_ALIGNED(32, 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 {
    AVFrame frame;
    AT1SUCtx            SUs[AT1_MAX_CHANNELS];              ///< channel sound unit
    DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES];      ///< the mdct spectrum buffer

    DECLARE_ALIGNED(32, float,  low)[256];
    DECLARE_ALIGNED(32, float,  mid)[256];
    DECLARE_ALIGNED(32, float, high)[512];
    float*              bands[3];
    float              *out_samples[AT1_MAX_CHANNELS];
    FFTContext          mdct_ctx[3];
    int                 channels;
    DSPContext          dsp;
    FmtConvertContext   fmt_conv;
} 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]);
    }
    mdct_context->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 AVERROR_INVALIDDATA;
        } 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, 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 AVERROR_INVALIDDATA;
        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 AVERROR_INVALIDDATA;
    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 = ff_atrac_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 AVERROR_INVALIDDATA;

            /* 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 */
    ff_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 */
    ff_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 *got_frame_ptr, AVPacket *avpkt)
{
    const uint8_t *buf = avpkt->data;
    int buf_size       = avpkt->size;
    AT1Ctx *q          = avctx->priv_data;
    int ch, ret;
    GetBitContext gb;
    float *samples;


    if (buf_size < 212 * q->channels) {
        av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n");
        return AVERROR_INVALIDDATA;
    }

    /* get output buffer */
    q->frame.nb_samples = AT1_SU_SAMPLES;
    if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) {
        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
        return ret;
    }
    samples = (float *)q->frame.data[0];

    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->channels == 1 ? samples : q->out_samples[ch]);
    }

    /* interleave */
    if (q->channels == 2) {
        q->fmt_conv.float_interleave(samples, (const float **)q->out_samples,
                                     AT1_SU_SAMPLES, 2);
    }

    *got_frame_ptr   = 1;
    *(AVFrame *)data = q->frame;

    return avctx->block_align;
}


static av_cold int atrac1_decode_end(AVCodecContext * avctx)
{
    AT1Ctx *q = avctx->priv_data;

    av_freep(&q->out_samples[0]);

    ff_mdct_end(&q->mdct_ctx[0]);
    ff_mdct_end(&q->mdct_ctx[1]);
    ff_mdct_end(&q->mdct_ctx[2]);

    return 0;
}


static av_cold int atrac1_decode_init(AVCodecContext *avctx)
{
    AT1Ctx *q = avctx->priv_data;
    int ret;

    avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

    if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) {
        av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n",
               avctx->channels);
        return AVERROR(EINVAL);
    }
    q->channels = avctx->channels;

    if (avctx->channels == 2) {
        q->out_samples[0] = av_malloc(2 * AT1_SU_SAMPLES * sizeof(*q->out_samples[0]));
        q->out_samples[1] = q->out_samples[0] + AT1_SU_SAMPLES;
        if (!q->out_samples[0]) {
            av_freep(&q->out_samples[0]);
            return AVERROR(ENOMEM);
        }
    }

    /* Init the mdct transforms */
    if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) ||
        (ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) ||
        (ret = ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)))) {
        av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
        atrac1_decode_end(avctx);
        return ret;
    }

    ff_init_ff_sine_windows(5);

    ff_atrac_generate_tables();

    ff_dsputil_init(&q->dsp, avctx);
    ff_fmt_convert_init(&q->fmt_conv, 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;

    avcodec_get_frame_defaults(&q->frame);
    avctx->coded_frame = &q->frame;

    return 0;
}


AVCodec ff_atrac1_decoder = {
    .name           = "atrac1",
    .type           = AVMEDIA_TYPE_AUDIO,
    .id             = CODEC_ID_ATRAC1,
    .priv_data_size = sizeof(AT1Ctx),
    .init           = atrac1_decode_init,
    .close          = atrac1_decode_end,
    .decode         = atrac1_decode_frame,
    .capabilities   = CODEC_CAP_DR1,
    .long_name      = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
};
