/*
  * Real Audio 1.0 (14.4K)

  * 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

  */
 

 #ifndef AVCODEC_RA144_H
 #define AVCODEC_RA144_H

 #include <stdint.h>

 #include "lpc.h"

 #include "audio_frame_queue.h"

 #include "audiodsp.h"

 #define NBLOCKS         4       ///< number of subblocks within a block
 #define BLOCKSIZE       40      ///< subblock size in 16-bit words
 #define BUFFERSIZE      146     ///< the size of the adaptive codebook

 #define FIXED_CB_SIZE   128     ///< size of fixed codebooks

 #define FRAME_SIZE      20      ///< size of encoded frame

 #define LPC_ORDER       10      ///< order of LPC filter

 typedef struct RA144Context {

     AVCodecContext *avctx;

     AudioDSPContext adsp;

     LPCContext lpc_ctx;

     AudioFrameQueue afq;

     int last_frame;

 
     unsigned int     old_energy;        ///< previous frame energy
 
     unsigned int     lpc_tables[2][10];
 
     /** LPC coefficients: lpc_coef[0] is the coefficients of the current frame
      *  and lpc_coef[1] of the previous one. */
     unsigned int    *lpc_coef[2];
 
     unsigned int     lpc_refl_rms[2];
 

     int16_t curr_block[NBLOCKS * BLOCKSIZE];
 

     /** The current subblock padded by the last 10 values of the previous one. */
     int16_t curr_sblock[50];
 
     /** Adaptive codebook, its size is two units bigger to avoid a
      *  buffer overflow. */

     int16_t adapt_cb[146+2];

 
     DECLARE_ALIGNED(16, int16_t, buffer_a)[FFALIGN(BLOCKSIZE,16)];

 } RA144Context;
 

 void ff_copy_and_dup(int16_t *target, const int16_t *source, int offset);
 int ff_eval_refl(int *refl, const int16_t *coefs, AVCodecContext *avctx);
 void ff_eval_coefs(int *coefs, const int *refl);
 void ff_int_to_int16(int16_t *out, const int *inp);
 int ff_t_sqrt(unsigned int x);
 unsigned int ff_rms(const int *data);
 int ff_interp(RA144Context *ractx, int16_t *out, int a, int copyold,
               int energy);
 unsigned int ff_rescale_rms(unsigned int rms, unsigned int energy);

 int ff_irms(AudioDSPContext *adsp, const int16_t *data/*align 16*/);

 void ff_subblock_synthesis(RA144Context *ractx, const int16_t *lpc_coefs,

                            int cba_idx, int cb1_idx, int cb2_idx,
                            int gval, int gain);

 extern const int16_t ff_gain_val_tab[256][3];
 extern const uint8_t ff_gain_exp_tab[256];
 extern const int8_t ff_cb1_vects[128][40];
 extern const int8_t ff_cb2_vects[128][40];
 extern const uint16_t ff_cb1_base[128];
 extern const uint16_t ff_cb2_base[128];
 extern const int16_t ff_energy_tab[32];
 extern const int16_t * const ff_lpc_refl_cb[10];

 #endif /* AVCODEC_RA144_H */