/* * 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 "libavutil/mem_internal.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 */