/*
  * 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 "libavutil/avassert.h"
 #include "libavutil/intreadwrite.h"
 #include "internal.h"
 #include "get_bits.h"
 #include "avcodec.h"
 #include "golomb.h"
 #include "mathops.h"
 #include "dsd.h"
 
 #define DST_MAX_CHANNELS 6
 #define DST_MAX_ELEMENTS (2 * DST_MAX_CHANNELS)
 
 #define DSD_FS44(sample_rate) (sample_rate * 8 / 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(64, uint8_t, status)[DST_MAX_CHANNELS][16];
     DECLARE_ALIGNED(16, int16_t, filter)[DST_MAX_ELEMENTS][16][256];
     DSDContext dsdctx[DST_MAX_CHANNELS];
 } DSTContext;
 
 static av_cold int decode_init(AVCodecContext *avctx)
 {
     DSTContext *s = avctx->priv_data;
     int i;
 
     if (avctx->channels > DST_MAX_CHANNELS) {
         avpriv_request_sample(avctx, "Channel count %d", avctx->channels);
         return AVERROR_PATCHWELCOME;
     }
 
     avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
 
     for (i = 0; i < avctx->channels; i++)
         memset(s->dsdctx[i].buf, 0x69, sizeof(s->dsdctx[i].buf));
 
     ff_init_dsd_data();
 
     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(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] * 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;
                 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 void 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++) {
                 int v = 0;
 
                 for (l = 0; l < total; l++)
                     v += (((k >> l) & 1) * 2 - 1) * fsets->coeff[i][j * 8 + l];
                 table[i][j][k] = v;
             }
         }
     }
 }
 
 static int decode_frame(AVCodecContext *avctx, void *data,
                         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->channels;
     DSTContext *s = avctx->priv_data;
     GetBitContext *gb = &s->gb;
     ArithCoder *ac = &s->ac;
     AVFrame *frame = data;
     uint8_t *dsd;
     float *pcm;
     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];
     pcm = (float *)frame->data[0];
 
     if ((ret = init_get_bits8(gb, avpkt->data, avpkt->size)) < 0)
         return ret;
 
     if (!get_bits1(gb)) {
         skip_bits1(gb);
         if (get_bits(gb, 6))
             return AVERROR_INVALIDDATA;
         memcpy(frame->data[0], avpkt->data + 1, FFMIN(avpkt->size - 1, frame->nb_samples * avctx->channels));
         goto dsd;
     }
 
     /* 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, avctx->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, avctx->channels)) < 0)
             return ret;
     }
 
     /* Half Probability (10.10) */
 
     for (ch = 0; ch < avctx->channels; ch++)
         half_prob[ch] = get_bits1(gb);
 
     /* Filter Coef Sets (10.12) */
 
     read_table(gb, &s->fsets, fsets_code_pred_coeff, 7, 9, 1, 0);
 
     /* Probability Tables (10.13) */
 
     read_table(gb, &s->probs, probs_code_pred_coeff, 6, 7, 0, 1);
 
     /* Arithmetic Coded Data (10.11) */
 
     if (get_bits1(gb))
         return AVERROR_INVALIDDATA;
     ac_init(ac, gb);
 
     build_filter(s->filter, &s->fsets);
 
     memset(s->status, 0xAA, sizeof(s->status));
     memset(dsd, 0, frame->nb_samples * 4 * avctx->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) << 2] |= v << (7 - (i & 0x7 ));
 
             AV_WN64A(status + 8, (AV_RN64A(status + 8) << 1) | ((AV_RN64A(status) >> 63) & 1));
             AV_WN64A(status, (AV_RN64A(status) << 1) | v);
         }
     }
 
 dsd:
     for (i = 0; i < avctx->channels; i++) {
         ff_dsd2pcm_translate(&s->dsdctx[i], frame->nb_samples, 0,
                              frame->data[0] + i * 4,
                              avctx->channels * 4, pcm + i, avctx->channels);
     }
 
     *got_frame_ptr = 1;
 
     return avpkt->size;
 }
 
 AVCodec ff_dst_decoder = {
     .name           = "dst",
     .long_name      = NULL_IF_CONFIG_SMALL("DST (Digital Stream Transfer)"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_DST,
     .priv_data_size = sizeof(DSTContext),
     .init           = decode_init,
     .decode         = decode_frame,
     .capabilities   = AV_CODEC_CAP_DR1,
     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
                                                       AV_SAMPLE_FMT_NONE },
 };