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* 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
* 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 "libavutil/mem_internal.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 */