repositories / deb_ffmpeg.git / blame
| Commit | Line | Data |
|---|---|---|
| 2ba45a60 DM |
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 */ |