Imported Debian version 2.5.3~trusty1
[deb_ffmpeg.git] / 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 */