/*

 * Musepack SV8 decoder

 * Copyright (c) 2007 Konstantin Shishkov

 *

 * 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

 * MPEG Audio Layer 1/2 -like codec with frames of 1152 samples

 * divided into 32 subbands.

 */

#include "libavutil/channel_layout.h"

#include "libavutil/lfg.h"

#include "libavutil/thread.h"

#include "avcodec.h"

#include "codec_internal.h"

#include "decode.h"

#include "get_bits.h"

#include "mpegaudiodsp.h"

#include "mpc.h"

#include "mpc8data.h"

#include "mpc8huff.h"

static VLC band_vlc, scfi_vlc[2], dscf_vlc[2], res_vlc[2];

static VLC q1_vlc, q2_vlc[2], q3_vlc[2], quant_vlc[4][2], q9up_vlc;

static inline int mpc8_dec_base(GetBitContext *gb, int k, int n)

{

    int len = mpc8_cnk_len[k-1][n-1] - 1;

    int code = len ? get_bits_long(gb, len) : 0;

    if (code >= mpc8_cnk_lost[k-1][n-1])

        code = ((code << 1) | get_bits1(gb)) - mpc8_cnk_lost[k-1][n-1];

    return code;

}

static inline int mpc8_dec_enum(GetBitContext *gb, int k, int n)

{

    int bits = 0;

    const uint32_t * C = mpc8_cnk[k-1];

    int code = mpc8_dec_base(gb, k, n);

    do {

        n--;

        if (code >= C[n]) {

            bits |= 1U << n;

            code -= C[n];

            C -= 32;

            k--;

        }

    } while(k > 0);

    return bits;

}

static inline int mpc8_get_mod_golomb(GetBitContext *gb, int m)

{

    if(mpc8_cnk_len[0][m] < 1) return 0;

    return mpc8_dec_base(gb, 1, m+1);

}

static int mpc8_get_mask(GetBitContext *gb, int size, int t)

{

    int mask = 0;

    if(t && t != size)

         mask = mpc8_dec_enum(gb, FFMIN(t, size - t), size);

    if((t << 1) > size) mask = ~mask;

    return mask;

}

static av_cold void build_vlc(VLC *vlc, unsigned *buf_offset,

                              const uint8_t codes_counts[16],

                              const uint8_t **syms, int offset)

{

    static VLCElem vlc_buf[9296];

    uint8_t len[MPC8_MAX_VLC_SIZE];

    unsigned num = 0;

    vlc->table           = &vlc_buf[*buf_offset];

    vlc->table_allocated = FF_ARRAY_ELEMS(vlc_buf) - *buf_offset;

    for (int i = 16; i > 0; i--)

        for (unsigned tmp = num + codes_counts[i - 1]; num < tmp; num++)

            len[num] = i;

    ff_vlc_init_from_lengths(vlc, FFMIN(len[0], 9), num, len, 1,

                             *syms, 1, 1, offset, VLC_INIT_STATIC_OVERLONG, NULL);

    *buf_offset += vlc->table_size;

    *syms       += num;

}

static av_cold void mpc8_init_static(void)

{

    const uint8_t *q_syms   = mpc8_q_syms,  *bands_syms = mpc8_bands_syms;

    const uint8_t *res_syms = mpc8_res_syms, *scfi_syms = mpc8_scfi_syms;

    const uint8_t *dscf_syms = mpc8_dscf_syms;

    unsigned offset = 0;

    build_vlc(&band_vlc, &offset, mpc8_bands_len_counts, &bands_syms, 0);

    build_vlc(&q1_vlc,   &offset, mpc8_q1_len_counts,   &q_syms, 0);

    build_vlc(&q9up_vlc, &offset, mpc8_q9up_len_counts, &q_syms, 0);

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

        build_vlc(&scfi_vlc[i], &offset, mpc8_scfi_len_counts[i], &scfi_syms, 0);

        build_vlc(&dscf_vlc[i], &offset, mpc8_dscf_len_counts[i], &dscf_syms, 0);

        build_vlc(&res_vlc[i],  &offset, mpc8_res_len_counts[i],  &res_syms,  0);

        build_vlc(&q2_vlc[i], &offset, mpc8_q2_len_counts[i], &q_syms, 0);

        build_vlc(&q3_vlc[i], &offset, mpc8_q34_len_counts[i],

                  &q_syms, -48 - 16 * i);

        for (int j = 0; j < 4; j++)

            build_vlc(&quant_vlc[j][i], &offset, mpc8_q5_8_len_counts[i][j],

                      &q_syms, -((8 << j) - 1));

    }

    ff_mpa_synth_init_fixed();

}

static av_cold int mpc8_decode_init(AVCodecContext * avctx)

{

    static AVOnce init_static_once = AV_ONCE_INIT;

    MPCContext *c = avctx->priv_data;

    GetBitContext gb;

    int channels;

    if(avctx->extradata_size < 2){

        av_log(avctx, AV_LOG_ERROR, "Too small extradata size (%i)!\n", avctx->extradata_size);

        return -1;

    }

    memset(c->oldDSCF, 0, sizeof(c->oldDSCF));

    av_lfg_init(&c->rnd, 0xDEADBEEF);

    ff_mpadsp_init(&c->mpadsp);

    init_get_bits(&gb, avctx->extradata, 16);

    uint8_t sample_rate_idx = get_bits(&gb, 3);

    static const int sample_rates[] = { 44100, 48000, 37800, 32000 };

    if (sample_rate_idx >= FF_ARRAY_ELEMS(sample_rates)) {

        av_log(avctx, AV_LOG_ERROR, "invalid sample rate index (%u)\n", sample_rate_idx);

        return AVERROR_INVALIDDATA;

    }

    avctx->sample_rate = sample_rates[sample_rate_idx];

    c->maxbands = get_bits(&gb, 5) + 1;

    if (c->maxbands >= BANDS) {

        av_log(avctx,AV_LOG_ERROR, "maxbands %d too high\n", c->maxbands);

        return AVERROR_INVALIDDATA;

    }

    channels = get_bits(&gb, 4) + 1;

    if (channels > 2) {

        avpriv_request_sample(avctx, "Multichannel MPC SV8");

        return AVERROR_PATCHWELCOME;

    }

    c->MSS = get_bits1(&gb);

    c->frames = 1 << (get_bits(&gb, 3) * 2);

    avctx->sample_fmt = AV_SAMPLE_FMT_S16P;

    av_channel_layout_uninit(&avctx->ch_layout);

    av_channel_layout_default(&avctx->ch_layout, channels);

    ff_thread_once(&init_static_once, mpc8_init_static);

    return 0;

}

static int mpc8_decode_frame(AVCodecContext *avctx, AVFrame *frame,

                             int *got_frame_ptr, AVPacket *avpkt)

{

    const uint8_t *buf = avpkt->data;

    int buf_size = avpkt->size;

    MPCContext *c = avctx->priv_data;

    GetBitContext gb2, *gb = &gb2;

    int i, j, k, ch, cnt, res, t;

    Band *bands = c->bands;

    int off;

    int maxband, keyframe;

    int last[2];

    keyframe = c->cur_frame == 0;

    if(keyframe){

        memset(c->Q, 0, sizeof(c->Q));

        c->last_bits_used = 0;

    }

    if ((res = init_get_bits8(gb, buf, buf_size)) < 0)

        return res;

    skip_bits(gb, c->last_bits_used & 7);

    if(keyframe)

        maxband = mpc8_get_mod_golomb(gb, c->maxbands + 1);

    else{

        maxband = c->last_max_band + get_vlc2(gb, band_vlc.table, MPC8_BANDS_BITS, 2);

        if(maxband > 32) maxband -= 33;

    }

    if (get_bits_left(gb) < 0) {

        *got_frame_ptr = 0;

        return buf_size;

    }

    if(maxband > c->maxbands + 1) {

        av_log(avctx, AV_LOG_ERROR, "maxband %d too large\n",maxband);

        return AVERROR_INVALIDDATA;

    }

    c->last_max_band = maxband;

    /* read subband indexes */

    if(maxband){

        last[0] = last[1] = 0;

        for(i = maxband - 1; i >= 0; i--){

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

                last[ch] = get_vlc2(gb, res_vlc[last[ch] > 2].table, MPC8_RES_BITS, 2) + last[ch];

                if(last[ch] > 15) last[ch] -= 17;

                bands[i].res[ch] = last[ch];

            }

        }

        if(c->MSS){

            int mask;

            cnt = 0;

            for(i = 0; i < maxband; i++)

                if(bands[i].res[0] || bands[i].res[1])

                    cnt++;

            t = mpc8_get_mod_golomb(gb, cnt);

            mask = mpc8_get_mask(gb, cnt, t);

            for(i = maxband - 1; i >= 0; i--)

                if(bands[i].res[0] || bands[i].res[1]){

                    bands[i].msf = mask & 1;

                    mask >>= 1;

                }

        }

    }

    for(i = maxband; i < c->maxbands; i++)

        bands[i].res[0] = bands[i].res[1] = 0;

    if(keyframe){

        for(i = 0; i < 32; i++)

            c->oldDSCF[0][i] = c->oldDSCF[1][i] = 1;

    }

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

        if(bands[i].res[0] || bands[i].res[1]){

            cnt = !!bands[i].res[0] + !!bands[i].res[1] - 1;

            if(cnt >= 0){

                t = get_vlc2(gb, scfi_vlc[cnt].table, scfi_vlc[cnt].bits, 1);

                if(bands[i].res[0]) bands[i].scfi[0] = t >> (2 * cnt);

                if(bands[i].res[1]) bands[i].scfi[1] = t & 3;

            }

        }

    }

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

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

            if(!bands[i].res[ch]) continue;

            if(c->oldDSCF[ch][i]){

                bands[i].scf_idx[ch][0] = get_bits(gb, 7) - 6;

                c->oldDSCF[ch][i] = 0;

            }else{

                t = get_vlc2(gb, dscf_vlc[1].table, MPC8_DSCF1_BITS, 2);

                if(t == 64)

                    t += get_bits(gb, 6);

                bands[i].scf_idx[ch][0] = ((bands[i].scf_idx[ch][2] + t - 25) & 0x7F) - 6;

            }

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

                if((bands[i].scfi[ch] << j) & 2)

                    bands[i].scf_idx[ch][j + 1] = bands[i].scf_idx[ch][j];

                else{

                    t = get_vlc2(gb, dscf_vlc[0].table, MPC8_DSCF0_BITS, 2);

                    if(t == 31)

                        t = 64 + get_bits(gb, 6);

                    bands[i].scf_idx[ch][j + 1] = ((bands[i].scf_idx[ch][j] + t - 25) & 0x7F) - 6;

                }

            }

        }

    }

    for(i = 0, off = 0; i < maxband; i++, off += SAMPLES_PER_BAND){

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

            res = bands[i].res[ch];

            switch(res){

            case -1:

                for(j = 0; j < SAMPLES_PER_BAND; j++)

                    c->Q[ch][off + j] = (av_lfg_get(&c->rnd) & 0x3FC) - 510;

                break;

            case 0:

                break;

            case 1:

                for(j = 0; j < SAMPLES_PER_BAND; j += SAMPLES_PER_BAND / 2){

                    cnt = get_vlc2(gb, q1_vlc.table, MPC8_Q1_BITS, 2);

                    t = mpc8_get_mask(gb, 18, cnt);

                    for(k = 0; k < SAMPLES_PER_BAND / 2; k++)

                        c->Q[ch][off + j + k] = t & (1 << (SAMPLES_PER_BAND / 2 - k - 1))

                                                ? (get_bits1(gb) << 1) - 1 : 0;

                }

                break;

            case 2:

                cnt = 6;//2*mpc8_thres[res]

                for(j = 0; j < SAMPLES_PER_BAND; j += 3){

                    t = get_vlc2(gb, q2_vlc[cnt > 3].table, MPC8_Q2_BITS, 2);

                    c->Q[ch][off + j + 0] = mpc8_idx50[t];

                    c->Q[ch][off + j + 1] = mpc8_idx51[t];

                    c->Q[ch][off + j + 2] = mpc8_idx52[t];

                    cnt = (cnt >> 1) + mpc8_huffq2[t];

                }

                break;

            case 3:

            case 4:

                for(j = 0; j < SAMPLES_PER_BAND; j += 2){

                    t = get_vlc2(gb, q3_vlc[res - 3].table, MPC8_Q3_BITS, 2);

                    c->Q[ch][off + j + 1] = t >> 4;

                    c->Q[ch][off + j + 0] = sign_extend(t, 4);

                }

                break;

            case 5:

            case 6:

            case 7:

            case 8:

                cnt = 2 * mpc8_thres[res];

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

                    const VLC *vlc = &quant_vlc[res - 5][cnt > mpc8_thres[res]];

                    c->Q[ch][off + j] = get_vlc2(gb, vlc->table, vlc->bits, 2);

                    cnt = (cnt >> 1) + FFABS(c->Q[ch][off + j]);

                }

                break;

            default:

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

                    c->Q[ch][off + j] = get_vlc2(gb, q9up_vlc.table, MPC8_Q9UP_BITS, 2);

                    if(res != 9){

                        c->Q[ch][off + j] <<= res - 9;

                        c->Q[ch][off + j] |= get_bits(gb, res - 9);

                    }

                    c->Q[ch][off + j] -= (1 << (res - 2)) - 1;

                }

            }

        }

    }

    frame->nb_samples = MPC_FRAME_SIZE;

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

        return res;

    ff_mpc_dequantize_and_synth(c, maxband - 1,

                                (int16_t **)frame->extended_data,

                                avctx->ch_layout.nb_channels);

    c->cur_frame++;

    c->last_bits_used = get_bits_count(gb);

    if(c->cur_frame >= c->frames)

        c->cur_frame = 0;

    if (get_bits_left(gb) < 0) {

        av_log(avctx, AV_LOG_ERROR, "Overread %d\n", -get_bits_left(gb));

        c->last_bits_used = buf_size << 3;

    } else if (c->cur_frame == 0 && get_bits_left(gb) < 8) {// we have only padding left

        c->last_bits_used = buf_size << 3;

    }

    *got_frame_ptr = 1;

    return c->cur_frame ? c->last_bits_used >> 3 : buf_size;

}

static av_cold void mpc8_decode_flush(AVCodecContext *avctx)

{

    MPCContext *c = avctx->priv_data;

    c->cur_frame = 0;

}

const FFCodec ff_mpc8_decoder = {

    .p.name         = "mpc8",

    CODEC_LONG_NAME("Musepack SV8"),

    .p.type         = AVMEDIA_TYPE_AUDIO,

    .p.id           = AV_CODEC_ID_MUSEPACK8,

    .priv_data_size = sizeof(MPCContext),

    .init           = mpc8_decode_init,

    FF_CODEC_DECODE_CB(mpc8_decode_frame),

    .flush          = mpc8_decode_flush,

    .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,

    CODEC_SAMPLEFMTS(AV_SAMPLE_FMT_S16P),

};