xref: /third_party/ffmpeg/libavcodec/ra144.h (revision cabdff1a) 
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
27#include "libavutil/mem_internal.h"
28
29#include "lpc.h"
30#include "audio_frame_queue.h"
31#include "audiodsp.h"
32
33#define NBLOCKS         4       ///< number of subblocks within a block
34#define BLOCKSIZE       40      ///< subblock size in 16-bit words
35#define BUFFERSIZE      146     ///< the size of the adaptive codebook
36#define FIXED_CB_SIZE   128     ///< size of fixed codebooks
37#define FRAME_SIZE      20      ///< size of encoded frame
38#define LPC_ORDER       10      ///< order of LPC filter
39
40typedef struct RA144Context {
41    AVCodecContext *avctx;
42    AudioDSPContext adsp;
43    LPCContext lpc_ctx;
44    AudioFrameQueue afq;
45    int last_frame;
46
47    unsigned int     old_energy;        ///< previous frame energy
48
49    unsigned int     lpc_tables[2][10];
50
51    /** LPC coefficients: lpc_coef[0] is the coefficients of the current frame
52     *  and lpc_coef[1] of the previous one. */
53    unsigned int    *lpc_coef[2];
54
55    unsigned int     lpc_refl_rms[2];
56
57    int16_t curr_block[NBLOCKS * BLOCKSIZE];
58
59    /** The current subblock padded by the last 10 values of the previous one. */
60    int16_t curr_sblock[50];
61
62    /** Adaptive codebook, its size is two units bigger to avoid a
63     *  buffer overflow. */
64    int16_t adapt_cb[146+2];
65
66    DECLARE_ALIGNED(16, int16_t, buffer_a)[FFALIGN(BLOCKSIZE,16)];
67} RA144Context;
68
69void ff_copy_and_dup(int16_t *target, const int16_t *source, int offset);
70int ff_eval_refl(int *refl, const int16_t *coefs, AVCodecContext *avctx);
71void ff_eval_coefs(int *coefs, const int *refl);
72void ff_int_to_int16(int16_t *out, const int *inp);
73int ff_t_sqrt(unsigned int x);
74unsigned int ff_rms(const int *data);
75int ff_interp(RA144Context *ractx, int16_t *out, int a, int copyold,
76              int energy);
77unsigned int ff_rescale_rms(unsigned int rms, unsigned int energy);
78int ff_irms(AudioDSPContext *adsp, const int16_t *data/*align 16*/);
79void ff_subblock_synthesis(RA144Context *ractx, const int16_t *lpc_coefs,
80                           int cba_idx, int cb1_idx, int cb2_idx,
81                           int gval, int gain);
82
83extern const int16_t ff_gain_val_tab[256][3];
84extern const uint8_t ff_gain_exp_tab[256];
85extern const int8_t ff_cb1_vects[128][40];
86extern const int8_t ff_cb2_vects[128][40];
87extern const uint16_t ff_cb1_base[128];
88extern const uint16_t ff_cb2_base[128];
89extern const int16_t ff_energy_tab[32];
90extern const int16_t * const ff_lpc_refl_cb[10];
91
92#endif /* AVCODEC_RA144_H */
93