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99497b4683
* commit '9a9e2f1c8aa4539a261625145e5c1f46a8106ac2': dsputil: Split audio operations off into a separate context Conflicts: configure libavcodec/takdec.c libavcodec/x86/Makefile libavcodec/x86/dsputil.asm libavcodec/x86/dsputil_init.c libavcodec/x86/dsputil_mmx.c libavcodec/x86/dsputil_x86.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
90 lines
3.2 KiB
C
90 lines
3.2 KiB
C
/*
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* Real Audio 1.0 (14.4K)
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* Copyright (c) 2003 the ffmpeg project
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#ifndef AVCODEC_RA144_H
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#define AVCODEC_RA144_H
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#include <stdint.h>
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#include "lpc.h"
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#include "audio_frame_queue.h"
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#include "audiodsp.h"
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#define NBLOCKS 4 ///< number of subblocks within a block
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#define BLOCKSIZE 40 ///< subblock size in 16-bit words
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#define BUFFERSIZE 146 ///< the size of the adaptive codebook
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#define FIXED_CB_SIZE 128 ///< size of fixed codebooks
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#define FRAME_SIZE 20 ///< size of encoded frame
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#define LPC_ORDER 10 ///< order of LPC filter
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typedef struct RA144Context {
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AVCodecContext *avctx;
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AudioDSPContext adsp;
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LPCContext lpc_ctx;
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AudioFrameQueue afq;
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int last_frame;
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unsigned int old_energy; ///< previous frame energy
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unsigned int lpc_tables[2][10];
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/** LPC coefficients: lpc_coef[0] is the coefficients of the current frame
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* and lpc_coef[1] of the previous one. */
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unsigned int *lpc_coef[2];
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unsigned int lpc_refl_rms[2];
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int16_t curr_block[NBLOCKS * BLOCKSIZE];
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/** The current subblock padded by the last 10 values of the previous one. */
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int16_t curr_sblock[50];
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/** Adaptive codebook, its size is two units bigger to avoid a
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* buffer overflow. */
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int16_t adapt_cb[146+2];
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DECLARE_ALIGNED(16, int16_t, buffer_a)[FFALIGN(BLOCKSIZE,16)];
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} RA144Context;
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void ff_copy_and_dup(int16_t *target, const int16_t *source, int offset);
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int ff_eval_refl(int *refl, const int16_t *coefs, AVCodecContext *avctx);
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void ff_eval_coefs(int *coefs, const int *refl);
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void ff_int_to_int16(int16_t *out, const int *inp);
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int ff_t_sqrt(unsigned int x);
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unsigned int ff_rms(const int *data);
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int ff_interp(RA144Context *ractx, int16_t *out, int a, int copyold,
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int energy);
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unsigned int ff_rescale_rms(unsigned int rms, unsigned int energy);
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int ff_irms(AudioDSPContext *adsp, const int16_t *data/*align 16*/);
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void ff_subblock_synthesis(RA144Context *ractx, const int16_t *lpc_coefs,
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int cba_idx, int cb1_idx, int cb2_idx,
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int gval, int gain);
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extern const int16_t ff_gain_val_tab[256][3];
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extern const uint8_t ff_gain_exp_tab[256];
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extern const int8_t ff_cb1_vects[128][40];
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extern const int8_t ff_cb2_vects[128][40];
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extern const uint16_t ff_cb1_base[128];
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extern const uint16_t ff_cb2_base[128];
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extern const int16_t ff_energy_tab[32];
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extern const int16_t * const ff_lpc_refl_cb[10];
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#endif /* AVCODEC_RA144_H */
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