ffmpeg/libavcodec/ra144.h

   1 /*
   2  * Real Audio 1.0 (14.4K)
   3  * Copyright (c) 2003 The FFmpeg Project
   4  *
   5  * This file is part of FFmpeg.
   6  *
   7  * FFmpeg is free software; you can redistribute it and/or
   8  * modify it under the terms of the GNU Lesser General Public
   9  * License as published by the Free Software Foundation; either
  10  * version 2.1 of the License, or (at your option) any later version.
  11  *
  12  * FFmpeg is distributed in the hope that it will be useful,
  13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15  * Lesser General Public License for more details.
  16  *
  17  * You should have received a copy of the GNU Lesser General Public
  18  * License along with FFmpeg; if not, write to the Free Software
  19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20  */
  21
  22 #ifndef AVCODEC_RA144_H
  23 #define AVCODEC_RA144_H
  24
  25 #include <stdint.h>
  26 #include "lpc.h"
  27 #include "audio_frame_queue.h"
  28 #include "audiodsp.h"
  29
  30 #define NBLOCKS         4       ///< number of subblocks within a block
  31 #define BLOCKSIZE       40      ///< subblock size in 16-bit words
  32 #define BUFFERSIZE      146     ///< the size of the adaptive codebook
  33 #define FIXED_CB_SIZE   128     ///< size of fixed codebooks
  34 #define FRAME_SIZE      20      ///< size of encoded frame
  35 #define LPC_ORDER       10      ///< order of LPC filter
  36
  37 typedef struct RA144Context {
  38     AVCodecContext *avctx;
  39     AudioDSPContext adsp;
  40     LPCContext lpc_ctx;
  41     AudioFrameQueue afq;
  42     int last_frame;
  43
  44     unsigned int     old_energy;        ///< previous frame energy
  45
  46     unsigned int     lpc_tables[2][10];
  47
  48     /** LPC coefficients: lpc_coef[0] is the coefficients of the current frame
  49      *  and lpc_coef[1] of the previous one. */
  50     unsigned int    *lpc_coef[2];
  51
  52     unsigned int     lpc_refl_rms[2];
  53
  54     int16_t curr_block[NBLOCKS * BLOCKSIZE];
  55
  56     /** The current subblock padded by the last 10 values of the previous one. */
  57     int16_t curr_sblock[50];
  58
  59     /** Adaptive codebook, its size is two units bigger to avoid a
  60      *  buffer overflow. */
  61     int16_t adapt_cb[146+2];
  62
  63     DECLARE_ALIGNED(16, int16_t, buffer_a)[FFALIGN(BLOCKSIZE,16)];
  64 } RA144Context;
  65
  66 void ff_copy_and_dup(int16_t *target, const int16_t *source, int offset);
  67 int ff_eval_refl(int *refl, const int16_t *coefs, AVCodecContext *avctx);
  68 void ff_eval_coefs(int *coefs, const int *refl);
  69 void ff_int_to_int16(int16_t *out, const int *inp);
  70 int ff_t_sqrt(unsigned int x);
  71 unsigned int ff_rms(const int *data);
  72 int ff_interp(RA144Context *ractx, int16_t *out, int a, int copyold,
  73               int energy);
  74 unsigned int ff_rescale_rms(unsigned int rms, unsigned int energy);
  75 int ff_irms(AudioDSPContext *adsp, const int16_t *data/*align 16*/);
  76 void ff_subblock_synthesis(RA144Context *ractx, const int16_t *lpc_coefs,
  77                            int cba_idx, int cb1_idx, int cb2_idx,
  78                            int gval, int gain);
  79
  80 extern const int16_t ff_gain_val_tab[256][3];
  81 extern const uint8_t ff_gain_exp_tab[256];
  82 extern const int8_t ff_cb1_vects[128][40];
  83 extern const int8_t ff_cb2_vects[128][40];
  84 extern const uint16_t ff_cb1_base[128];
  85 extern const uint16_t ff_cb2_base[128];
  86 extern const int16_t ff_energy_tab[32];
  87 extern const int16_t * const ff_lpc_refl_cb[10];
  88
  89 #endif /* AVCODEC_RA144_H */