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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 */ |