/*
  * RealAudio 2.0 (28.8K)
  * Copyright (c) 2003 the ffmpeg project
  *

  * 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

  */
 
 #include "avcodec.h"

 #define ALT_BITSTREAM_READER_LE

 #include "get_bits.h"

 #include "ra288.h"

 #include "lpc.h"

 #include "celp_math.h"

 #include "celp_filters.h"

 #define MAX_BACKWARD_FILTER_ORDER  36
 #define MAX_BACKWARD_FILTER_LEN    40
 #define MAX_BACKWARD_FILTER_NONREC 35
 

 typedef struct {

     float sp_lpc[36];      ///< LPC coefficients for speech data (spec: A)

     float gain_lpc[10];    ///< LPC coefficients for gain        (spec: GB)

     /** speech data history                                      (spec: SB).
      *  Its first 70 coefficients are updated only at backward filtering.
      */
     float sp_hist[111];

     /// speech part of the gain autocorrelation                  (spec: REXP)

     float sp_rec[37];
 

     /** log-gain history                                         (spec: SBLG).
      *  Its first 28 coefficients are updated only at backward filtering.
      */
     float gain_hist[38];

     /// recursive part of the gain autocorrelation               (spec: REXPLG)

     float gain_rec[11];

 } RA288Context;

 static av_cold int ra288_decode_init(AVCodecContext *avctx)
 {

     avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

     return 0;
 }
 

 static void apply_window(float *tgt, const float *m1, const float *m2, int n)

 {
     while (n--)

         *tgt++ = *m1++ * *m2++;

 }
 

 static void convolve(float *tgt, const float *src, int len, int n)
 {
     for (; n >= 0; n--)

         tgt[n] = ff_dot_productf(src, src - n, len);

 
 }
 

 static void decode(RA288Context *ractx, float gain, int cb_coef)

 {

     int i;

     double sumsum;
     float sum, buffer[5];

     float *block = ractx->sp_hist + 70 + 36; // current block
     float *gain_block = ractx->gain_hist + 28;

     memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));

     /* block 46 of G.728 spec */

     sum = 32.;
     for (i=0; i < 10; i++)

         sum -= gain_block[9-i] * ractx->gain_lpc[i];

     /* block 47 of G.728 spec */

     sum = av_clipf(sum, 0, 60);

     /* block 48 of G.728 spec */

     /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */

     sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));

     for (i=0; i < 5; i++)

         buffer[i] = codetable[cb_coef][i] * sumsum;

     sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.);

 
     sum = FFMAX(sum, 1);

 
     /* shift and store */

     memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));

     gain_block[9] = 10 * log10(sum) - 32;

     ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);

 }
 

 /**

  * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.

  *

  * @param order   filter order
  * @param n       input length
  * @param non_rec number of non-recursive samples
  * @param out     filter output

  * @param hist    pointer to the input history of the filter

  * @param out     pointer to the non-recursive part of the output

  * @param out2    pointer to the recursive part of the output
  * @param window  pointer to the windowing function table
  */

 static void do_hybrid_window(int order, int n, int non_rec, float *out,
                              float *hist, float *out2, const float *window)

 {

     int i;

     float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
     float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
     float work[MAX_BACKWARD_FILTER_ORDER + MAX_BACKWARD_FILTER_LEN + MAX_BACKWARD_FILTER_NONREC];

     apply_window(work, window, hist, order + n + non_rec);

     convolve(buffer1, work + order    , n      , order);
     convolve(buffer2, work + order + n, non_rec, order);

     for (i=0; i <= order; i++) {
         out2[i] = out2[i] * 0.5625 + buffer1[i];
         out [i] = out2[i]          + buffer2[i];

     }

     /* Multiply by the white noise correcting factor (WNCF). */

     *out *= 257./256.;

 }
 

 /**

  * Backward synthesis filter, find the LPC coefficients from past speech data.

  */

 static void backward_filter(float *hist, float *rec, const float *window,
                             float *lpc, const float *tab,
                             int order, int n, int non_rec, int move_size)

 {

     float temp[MAX_BACKWARD_FILTER_ORDER+1];

     do_hybrid_window(order, n, non_rec, temp, hist, rec, window);

     if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
         apply_window(lpc, lpc, tab, order);

     memmove(hist, hist + n, move_size*sizeof(*hist));

 }
 

 static int ra288_decode_frame(AVCodecContext * avctx, void *data,

                               int *data_size, AVPacket *avpkt)

 {

     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;

     float *out = data;

     int i, j;

     RA288Context *ractx = avctx->priv_data;

     GetBitContext gb;

     if (buf_size < avctx->block_align) {
         av_log(avctx, AV_LOG_ERROR,
                "Error! Input buffer is too small [%d<%d]\n",
                buf_size, avctx->block_align);
         return 0;
     }
 

     if (*data_size < 32*5*4)

         return -1;
 

     init_get_bits(&gb, buf, avctx->block_align * 8);

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

         float gain = amptable[get_bits(&gb, 3)];

         int cb_coef = get_bits(&gb, 6 + (i&1));

         decode(ractx, gain, cb_coef);

         for (j=0; j < 5; j++)

             *(out++) = ractx->sp_hist[70 + 36 + j];

         if ((i & 7) == 3) {
             backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window,
                             ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
 
             backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window,
                             ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
         }

     }
 

     *data_size = (char *)out - (char *)data;

     return avctx->block_align;

 }
 

 AVCodec ff_ra_288_decoder =

 {
     "real_288",

     AVMEDIA_TYPE_AUDIO,

     CODEC_ID_RA_288,

     sizeof(RA288Context),

     ra288_decode_init,

     NULL,
     NULL,
     ra288_decode_frame,

     .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),

 };