| /* |
| * Real Audio 1.0 (14.4K) |
| * Copyright (c) 2003 The FFmpeg project |
| * |
| * This file is part of FFmpeg. |
| * |
| * FFmpeg is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Lesser General Public |
| * License as published by the Free Software Foundation; either |
| * version 2.1 of the License, or (at your option) any later version. |
| * |
| * FFmpeg is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| * Lesser General Public License for more details. |
| * |
| * You should have received a copy of the GNU Lesser General Public |
| * License along with FFmpeg; if not, write to the Free Software |
| * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| */ |
| |
| #ifndef AVCODEC_RA144_H |
| #define AVCODEC_RA144_H |
| |
| #include <stdint.h> |
| #include "lpc.h" |
| #include "audio_frame_queue.h" |
| #include "audiodsp.h" |
| |
| #define NBLOCKS 4 ///< number of subblocks within a block |
| #define BLOCKSIZE 40 ///< subblock size in 16-bit words |
| #define BUFFERSIZE 146 ///< the size of the adaptive codebook |
| #define FIXED_CB_SIZE 128 ///< size of fixed codebooks |
| #define FRAME_SIZE 20 ///< size of encoded frame |
| #define LPC_ORDER 10 ///< order of LPC filter |
| |
| typedef struct RA144Context { |
| AVCodecContext *avctx; |
| AudioDSPContext adsp; |
| LPCContext lpc_ctx; |
| AudioFrameQueue afq; |
| int last_frame; |
| |
| unsigned int old_energy; ///< previous frame energy |
| |
| unsigned int lpc_tables[2][10]; |
| |
| /** LPC coefficients: lpc_coef[0] is the coefficients of the current frame |
| * and lpc_coef[1] of the previous one. */ |
| unsigned int *lpc_coef[2]; |
| |
| unsigned int lpc_refl_rms[2]; |
| |
| int16_t curr_block[NBLOCKS * BLOCKSIZE]; |
| |
| /** The current subblock padded by the last 10 values of the previous one. */ |
| int16_t curr_sblock[50]; |
| |
| /** Adaptive codebook, its size is two units bigger to avoid a |
| * buffer overflow. */ |
| int16_t adapt_cb[146+2]; |
| |
| DECLARE_ALIGNED(16, int16_t, buffer_a)[FFALIGN(BLOCKSIZE,16)]; |
| } RA144Context; |
| |
| void ff_copy_and_dup(int16_t *target, const int16_t *source, int offset); |
| int ff_eval_refl(int *refl, const int16_t *coefs, AVCodecContext *avctx); |
| void ff_eval_coefs(int *coefs, const int *refl); |
| void ff_int_to_int16(int16_t *out, const int *inp); |
| int ff_t_sqrt(unsigned int x); |
| unsigned int ff_rms(const int *data); |
| int ff_interp(RA144Context *ractx, int16_t *out, int a, int copyold, |
| int energy); |
| unsigned int ff_rescale_rms(unsigned int rms, unsigned int energy); |
| int ff_irms(AudioDSPContext *adsp, const int16_t *data/*align 16*/); |
| void ff_subblock_synthesis(RA144Context *ractx, const int16_t *lpc_coefs, |
| int cba_idx, int cb1_idx, int cb2_idx, |
| int gval, int gain); |
| |
| extern const int16_t ff_gain_val_tab[256][3]; |
| extern const uint8_t ff_gain_exp_tab[256]; |
| extern const int8_t ff_cb1_vects[128][40]; |
| extern const int8_t ff_cb2_vects[128][40]; |
| extern const uint16_t ff_cb1_base[128]; |
| extern const uint16_t ff_cb2_base[128]; |
| extern const int16_t ff_energy_tab[32]; |
| extern const int16_t * const ff_lpc_refl_cb[10]; |
| |
| #endif /* AVCODEC_RA144_H */ |