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 40 typedef 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 69 void ff_copy_and_dup(int16_t *target, const int16_t *source, int offset); 70 int ff_eval_refl(int *refl, const int16_t *coefs, AVCodecContext *avctx); 71 void ff_eval_coefs(int *coefs, const int *refl); 72 void ff_int_to_int16(int16_t *out, const int *inp); 73 int ff_t_sqrt(unsigned int x); 74 unsigned int ff_rms(const int *data); 75 int ff_interp(RA144Context *ractx, int16_t *out, int a, int copyold, 76 int energy); 77 unsigned int ff_rescale_rms(unsigned int rms, unsigned int energy); 78 int ff_irms(AudioDSPContext *adsp, const int16_t *data/*align 16*/); 79 void 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 83 extern const int16_t ff_gain_val_tab[256][3]; 84 extern const uint8_t ff_gain_exp_tab[256]; 85 extern const int8_t ff_cb1_vects[128][40]; 86 extern const int8_t ff_cb2_vects[128][40]; 87 extern const uint16_t ff_cb1_base[128]; 88 extern const uint16_t ff_cb2_base[128]; 89 extern const int16_t ff_energy_tab[32]; 90 extern const int16_t * const ff_lpc_refl_cb[10]; 91 92 #endif /* AVCODEC_RA144_H */ 93