/*

 * Direct Stream Transfer (DST) decoder

 * Copyright (c) 2014 Peter Ross <pross@xvid.org>

 *

 * 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

 * Direct Stream Transfer (DST) decoder

 * ISO/IEC 14496-3 Part 3 Subpart 10: Technical description of lossless coding of oversampled audio

 */

#include "config.h"

#include "libavutil/intreadwrite.h"

#include "libavutil/mem.h"

#include "libavutil/mem_internal.h"

#include "libavutil/reverse.h"

#include "codec_internal.h"

#include "decode.h"

#include "get_bits.h"

#include "avcodec.h"

#include "golomb.h"

#include "dsd.h"

#if CONFIG_SWRESAMPLE && FF_API_DSD_PCM

#include "libswresample/swresample.h"

#endif

#define DST_MAX_CHANNELS 6

#define DST_MAX_ELEMENTS (2 * DST_MAX_CHANNELS)

#define DSD_FS44(sample_rate) (sample_rate * 8LL / 44100)

#define DST_SAMPLES_PER_FRAME(sample_rate) (588 * DSD_FS44(sample_rate))

static const int8_t fsets_code_pred_coeff[3][3] = {

    {  -8 },

    { -16,  8 },

    {  -9, -5, 6 },

};

static const int8_t probs_code_pred_coeff[3][3] = {

    {  -8 },

    { -16,  8 },

    { -24, 24, -8 },

};

typedef struct ArithCoder {

    unsigned int a;

    unsigned int c;

} ArithCoder;

typedef struct Table {

    unsigned int elements;

    unsigned int length[DST_MAX_ELEMENTS];

    int coeff[DST_MAX_ELEMENTS][128];

} Table;

typedef struct DSTContext {

    AVClass *class;

    GetBitContext gb;

    ArithCoder ac;

    Table fsets, probs;

    DECLARE_ALIGNED(16, uint8_t, status)[DST_MAX_CHANNELS][16];

    DECLARE_ALIGNED(16, int16_t, filter)[DST_MAX_ELEMENTS][16][256];

#if CONFIG_SWRESAMPLE && FF_API_DSD_PCM

    struct SwrContext *swr;

    uint8_t *scratch;

    unsigned scratch_size;

#endif

} DSTContext;

static av_cold int decode_init(AVCodecContext *avctx)

{

    if (avctx->ch_layout.nb_channels > DST_MAX_CHANNELS) {

        avpriv_request_sample(avctx, "Channel count %d", avctx->ch_layout.nb_channels);

        return AVERROR_PATCHWELCOME;

    }

    // the sample rate is only allowed to be 64,128,256 * 44100 by ISO/IEC 14496-3:2005(E)

    // We are a bit more tolerant here, but this check is needed to bound the size and duration

    if (avctx->sample_rate > 512 * 44100)

        return AVERROR_INVALIDDATA;

    if (DST_SAMPLES_PER_FRAME(avctx->sample_rate) & 7) {

        return AVERROR_PATCHWELCOME;

    }

    avctx->sample_fmt = AV_SAMPLE_FMT_DSD;

#if CONFIG_SWRESAMPLE && FF_API_DSD_PCM

    if (avctx->request_sample_fmt != AV_SAMPLE_FMT_DSD)

        avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

#endif

    return 0;

}

static av_cold int decode_close(AVCodecContext *avctx)

{

#if CONFIG_SWRESAMPLE && FF_API_DSD_PCM

    DSTContext *s = avctx->priv_data;

    swr_free(&s->swr);

    av_freep(&s->scratch);

#endif

    return 0;

}

static int read_map(GetBitContext *gb, Table *t, unsigned int map[DST_MAX_CHANNELS], int channels)

{

    int ch;

    t->elements = 1;

    map[0] = 0;

    if (!get_bits1(gb)) {

        for (ch = 1; ch < channels; ch++) {

            int bits = av_log2(t->elements) + 1;

            map[ch] = get_bits(gb, bits);

            if (map[ch] == t->elements) {

                t->elements++;

                if (t->elements >= DST_MAX_ELEMENTS)

                    return AVERROR_INVALIDDATA;

            } else if (map[ch] > t->elements) {

                return AVERROR_INVALIDDATA;

            }

        }

    } else {

        memset(map, 0, sizeof(*map) * DST_MAX_CHANNELS);

    }

    return 0;

}

static av_always_inline int get_sr_golomb_dst(GetBitContext *gb, unsigned int k)

{

    int v = get_ur_golomb_jpegls(gb, k, get_bits_left(gb), 0);

    if (v && get_bits1(gb))

        v = -v;

    return v;

}

static void read_uncoded_coeff(GetBitContext *gb, int *dst, unsigned int elements,

                               int coeff_bits, int is_signed, int offset)

{

    int i;

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

        dst[i] = (is_signed ? get_sbits(gb, coeff_bits) : get_bits(gb, coeff_bits)) + offset;

    }

}

static int read_table(GetBitContext *gb, Table *t, const int8_t code_pred_coeff[3][3],

                      int length_bits, int coeff_bits, int is_signed, int offset)

{

    unsigned int i, j, k;

    for (i = 0; i < t->elements; i++) {

        t->length[i] = get_bits(gb, length_bits) + 1;

        if (!get_bits1(gb)) {

            read_uncoded_coeff(gb, t->coeff[i], t->length[i], coeff_bits, is_signed, offset);

        } else {

            int method = get_bits(gb, 2), lsb_size;

            if (method == 3)

                return AVERROR_INVALIDDATA;

            read_uncoded_coeff(gb, t->coeff[i], method + 1, coeff_bits, is_signed, offset);

            lsb_size  = get_bits(gb, 3);

            for (j = method + 1; j < t->length[i]; j++) {

                int c, x = 0;

                for (k = 0; k < method + 1; k++)

                    x += code_pred_coeff[method][k] * (unsigned)t->coeff[i][j - k - 1];

                c = get_sr_golomb_dst(gb, lsb_size);

                if (x >= 0)

                    c -= (x + 4) / 8;

                else

                    c += (-x + 3) / 8;

                if (!is_signed) {

                    if (c < offset || c >= offset + (1<<coeff_bits))

                        return AVERROR_INVALIDDATA;

                }

                t->coeff[i][j] = c;

            }

        }

    }

    return 0;

}

static void ac_init(ArithCoder *ac, GetBitContext *gb)

{

    ac->a = 4095;

    ac->c = get_bits(gb, 12);

}

static av_always_inline void ac_get(ArithCoder *ac, GetBitContext *gb, int p, int *e)

{

    unsigned int k = (ac->a >> 8) | ((ac->a >> 7) & 1);

    unsigned int q = k * p;

    unsigned int a_q = ac->a - q;

    *e = ac->c < a_q;

    if (*e) {

        ac->a  = a_q;

    } else {

        ac->a  = q;

        ac->c -= a_q;

    }

    if (ac->a < 2048) {

        int n = 11 - av_log2(ac->a);

        ac->a <<= n;

        ac->c = (ac->c << n) | get_bits(gb, n);

    }

}

static uint8_t prob_dst_x_bit(int c)

{

    return (ff_reverse[c & 127] >> 1) + 1;

}

static int build_filter(int16_t table[DST_MAX_ELEMENTS][16][256], const Table *fsets)

{

    int i, j, k, l;

    for (i = 0; i < fsets->elements; i++) {

        int length = fsets->length[i];

        for (j = 0; j < 16; j++) {

            int total = av_clip(length - j * 8, 0, 8);

            for (k = 0; k < 256; k++) {

                int64_t v = 0;

                for (l = 0; l < total; l++)

                    v += (((k >> l) & 1) * 2 - 1) * fsets->coeff[i][j * 8 + l];

                if ((int16_t)v != v)

                    return AVERROR_INVALIDDATA;

                table[i][j][k] = v;

            }

        }

    }

    return 0;

}

static int decode_frame(AVCodecContext *avctx, AVFrame *frame,

                        int *got_frame_ptr, AVPacket *avpkt)

{

    unsigned samples_per_frame = DST_SAMPLES_PER_FRAME(avctx->sample_rate);

    unsigned map_ch_to_felem[DST_MAX_CHANNELS];

    unsigned map_ch_to_pelem[DST_MAX_CHANNELS];

    unsigned i, ch, same_map, dst_x_bit;

    unsigned half_prob[DST_MAX_CHANNELS];

    const int channels = avctx->ch_layout.nb_channels;

    DSTContext *s = avctx->priv_data;

    GetBitContext *gb = &s->gb;

    ArithCoder *ac = &s->ac;

    uint8_t *dsd;

    int ret;

    if (avpkt->size <= 1)

        return AVERROR_INVALIDDATA;

    frame->nb_samples = samples_per_frame / 8;

    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)

        return ret;

    dsd = frame->data[0];

#if CONFIG_SWRESAMPLE && FF_API_DSD_PCM

    if (avctx->sample_fmt != AV_SAMPLE_FMT_DSD) {

        if (!s->swr && (ret = ff_dsd_to_pcm_init(avctx, &s->swr)) < 0)

            return ret;

        av_fast_malloc(&s->scratch, &s->scratch_size,

                       frame->nb_samples * channels);

        if (!s->scratch)

            return AVERROR(ENOMEM);

        dsd = s->scratch;

    }

#endif

    if ((ret = init_get_bits8(gb, avpkt->data, avpkt->size)) < 0)

        return ret;

    if (!get_bits1(gb)) {

        unsigned total = frame->nb_samples * channels;

        unsigned n = FFMIN(avpkt->size - 1, total);

        skip_bits1(gb);

        if (get_bits(gb, 6))

            return AVERROR_INVALIDDATA;

        // pad short frames with silence

        memcpy(dsd, avpkt->data + 1, n);

        memset(dsd + n, 0x69, total - n);

        goto done;

    }

    /* Segmentation (10.4, 10.5, 10.6) */

    if (!get_bits1(gb)) {

        avpriv_request_sample(avctx, "Not Same Segmentation");

        return AVERROR_PATCHWELCOME;

    }

    if (!get_bits1(gb)) {

        avpriv_request_sample(avctx, "Not Same Segmentation For All Channels");

        return AVERROR_PATCHWELCOME;

    }

    if (!get_bits1(gb)) {

        avpriv_request_sample(avctx, "Not End Of Channel Segmentation");

        return AVERROR_PATCHWELCOME;

    }

    /* Mapping (10.7, 10.8, 10.9) */

    same_map = get_bits1(gb);

    if ((ret = read_map(gb, &s->fsets, map_ch_to_felem, channels)) < 0)

        return ret;

    if (same_map) {

        s->probs.elements = s->fsets.elements;

        memcpy(map_ch_to_pelem, map_ch_to_felem, sizeof(map_ch_to_felem));

    } else {

        avpriv_request_sample(avctx, "Not Same Mapping");

        if ((ret = read_map(gb, &s->probs, map_ch_to_pelem, channels)) < 0)

            return ret;

    }

    /* Half Probability (10.10) */

    for (ch = 0; ch < channels; ch++)

        half_prob[ch] = get_bits1(gb);

    /* Filter Coef Sets (10.12) */

    ret = read_table(gb, &s->fsets, fsets_code_pred_coeff, 7, 9, 1, 0);

    if (ret < 0)

        return ret;

    /* Probability Tables (10.13) */

    ret = read_table(gb, &s->probs, probs_code_pred_coeff, 6, 7, 0, 1);

    if (ret < 0)

        return ret;

    /* Arithmetic Coded Data (10.11) */

    if (get_bits1(gb))

        return AVERROR_INVALIDDATA;

    ac_init(ac, gb);

    ret = build_filter(s->filter, &s->fsets);

    if (ret < 0)

        return ret;

    memset(s->status, 0xAA, sizeof(s->status));

    memset(dsd, 0, frame->nb_samples * channels);

    ac_get(ac, gb, prob_dst_x_bit(s->fsets.coeff[0][0]), &dst_x_bit);

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

        for (ch = 0; ch < channels; ch++) {

            const unsigned felem = map_ch_to_felem[ch];

            int16_t (*filter)[256] = s->filter[felem];

            uint8_t *status = s->status[ch];

            int prob, residual, v;

#define F(x) filter[(x)][status[(x)]]

            const int16_t predict = F( 0) + F( 1) + F( 2) + F( 3) +

                                    F( 4) + F( 5) + F( 6) + F( 7) +

                                    F( 8) + F( 9) + F(10) + F(11) +

                                    F(12) + F(13) + F(14) + F(15);

#undef F

            if (!half_prob[ch] || i >= s->fsets.length[felem]) {

                unsigned pelem = map_ch_to_pelem[ch];

                unsigned index = FFABS(predict) >> 3;

                prob = s->probs.coeff[pelem][FFMIN(index, s->probs.length[pelem] - 1)];

            } else {

                prob = 128;

            }

            ac_get(ac, gb, prob, &residual);

            v = ((predict >> 15) ^ residual) & 1;

            dsd[(i >> 3) * channels + ch] |= v << (7 - (i & 0x7 ));

            AV_WL64A(status + 8, (AV_RL64A(status + 8) << 1) | ((AV_RL64A(status) >> 63) & 1));

            AV_WL64A(status, (AV_RL64A(status) << 1) | v);

        }

    }

done:

#if CONFIG_SWRESAMPLE && FF_API_DSD_PCM

    if (s->swr) {

        ret = swr_convert(s->swr, &frame->data[0], frame->nb_samples,

                          (const uint8_t *const []){ s->scratch },

                          frame->nb_samples);

        if (ret != frame->nb_samples)

            return ret < 0 ? ret : AVERROR_BUG;

    }

#endif

    *got_frame_ptr = 1;

    return avpkt->size;

}

const FFCodec ff_dst_decoder = {

    .p.name         = "dst",

    CODEC_LONG_NAME("DST (Digital Stream Transfer)"),

    .p.type         = AVMEDIA_TYPE_AUDIO,

    .p.id           = AV_CODEC_ID_DST,

    .priv_data_size = sizeof(DSTContext),

    .init           = decode_init,

    .close          = decode_close,

    FF_CODEC_DECODE_CB(decode_frame),

    .p.capabilities = AV_CODEC_CAP_DR1,

};