ffmpeg / libavcodec / ra144.c @ 11bb2eb0
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/*


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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 021101301 USA

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*/

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#include "avcodec.h" 
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#include "bitstream.h" 
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#include "ra144.h" 
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#define NBLOCKS 4 /* number of segments within a block */ 
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#define BLOCKSIZE 40 /* (quarter) block size in 16bit words (80 bytes) */ 
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#define HALFBLOCK 20 /* BLOCKSIZE/2 */ 
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#define BUFFERSIZE 146 /* for do_output */ 
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/* internal globals */

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typedef struct { 
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unsigned int old_energy; ///< previous frame energy 
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/* the swapped buffers */

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unsigned int lpc_tables[4][10]; 
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unsigned int *lpc_refl; ///< LPC reflection coefficients 
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unsigned int *lpc_coef; ///< LPC coefficients 
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unsigned int *lpc_refl_old; ///< previous frame LPC reflection coefs 
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unsigned int *lpc_coef_old; ///< previous frame LPC coefficients 
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unsigned int buffer[5]; 
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uint16_t adapt_cb[148]; ///< adaptive codebook 
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} RA144Context; 
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static int ra144_decode_init(AVCodecContext * avctx) 
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{ 
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RA144Context *ractx = avctx>priv_data; 
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ractx>lpc_refl = ractx>lpc_tables[0];

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ractx>lpc_coef = ractx>lpc_tables[1];

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ractx>lpc_refl_old = ractx>lpc_tables[2];

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ractx>lpc_coef_old = ractx>lpc_tables[3];

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return 0; 
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} 
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/**

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* Evaluate sqrt(x << 24). x must fit in 20 bits. This value is evaluated in an

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* odd way to make the output identical to the binary decoder.

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*/

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static int t_sqrt(unsigned int x) 
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{ 
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int s = 0; 
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while (x > 0xfff) { 
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s++; 
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x = x >> 2;

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} 
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return (ff_sqrt(x << 20) << s) << 2; 
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} 
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/**

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* Evaluate the LPC filter coefficients from the reflection coefficients.

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* Does the inverse of the eval_refl() function.

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*/

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static void eval_coefs(const int *refl, int *coefs) 
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{ 
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int buffer[10]; 
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int *b1 = buffer;

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int *b2 = coefs;

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int x, y;

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for (x=0; x < 10; x++) { 
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b1[x] = refl[x] << 4;

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for (y=0; y < x; y++) 
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b1[y] = ((refl[x] * b2[xy1]) >> 12) + b2[y]; 
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FFSWAP(int *, b1, b2);

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} 
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for (x=0; x < 10; x++) 
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coefs[x] >>= 4;

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} 
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/* rotate block */

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static void rotate_block(const int16_t *source, int16_t *target, int offset) 
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{ 
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int i=0, k=0; 
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source += BUFFERSIZE  offset; 
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while (i<BLOCKSIZE) {

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target[i++] = source[k++]; 
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if (k == offset)

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k = 0;

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} 
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} 
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/* inverse root mean square */

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static int irms(const int16_t *data, int factor) 
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{ 
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unsigned int i, sum = 0; 
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for (i=0; i < BLOCKSIZE; i++) 
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sum += data[i] * data[i]; 
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if (sum == 0) 
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return 0; /* OOPS  division by zero */ 
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return (0x20000000 / (t_sqrt(sum) >> 8)) * factor; 
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} 
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/* multiply/add wavetable */

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static void add_wav(int n, int skip_first, int *m, const int16_t *s1, 
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const int8_t *s2, const int8_t *s3, int16_t *dest) 
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{ 
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int i;

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int v[3]; 
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v[0] = 0; 
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for (i=!skip_first; i<3; i++) 
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v[i] = (wavtable1[n][i] * m[i]) >> (wavtable2[n][i] + 1);

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for (i=0; i < BLOCKSIZE; i++) 
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dest[i] = ((*(s1++))*v[0] + (*(s2++))*v[1] + (*(s3++))*v[2]) >> 12; 
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} 
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static void final(const int16_t *i1, const int16_t *i2, 
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void *out, int *statbuf, int len) 
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{ 
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int x, i;

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uint16_t work[50];

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int16_t *ptr = work; 
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memcpy(work, statbuf,20);

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memcpy(work + 10, i2, len * 2); 
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for (i=0; i<len; i++) { 
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int sum = 0; 
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int new_val;

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for(x=0; x<10; x++) 
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sum += i1[9x] * ptr[x];

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sum >>= 12;

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new_val = ptr[10]  sum;

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if (new_val < 32768  new_val > 32767) { 
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memset(out, 0, len * 2); 
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memset(statbuf, 0, 20); 
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return;

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} 
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ptr[10] = new_val;

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ptr++; 
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} 
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memcpy(out, work+10, len * 2); 
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memcpy(statbuf, work + 40, 20); 
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} 
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static unsigned int rescale_rms(int rms, int energy) 
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{ 
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return (rms * energy) >> 10; 
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} 
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static unsigned int rms(const int *data) 
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{ 
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int x;

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unsigned int res = 0x10000; 
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int b = 0; 
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for (x=0; x<10; x++) { 
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res = (((0x1000000  (*data) * (*data)) >> 12) * res) >> 12; 
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if (res == 0) 
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return 0; 
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while (res <= 0x3fff) { 
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b++; 
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res <<= 2;

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} 
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data++; 
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} 
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if (res > 0) 
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res = t_sqrt(res); 
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res >>= (b + 10);

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return res;

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} 
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/* do quarterblock output */

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static void do_output_subblock(RA144Context *ractx, 
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const uint16_t *lpc_coefs, unsigned int gval, 
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int16_t *output_buffer, GetBitContext *gb) 
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{ 
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uint16_t buffer_a[40];

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uint16_t *block; 
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int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none 
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int gain = get_bits(gb, 8); 
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int cb1_idx = get_bits(gb, 7); 
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int cb2_idx = get_bits(gb, 7); 
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int m[3]; 
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if (cba_idx) {

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cba_idx += HALFBLOCK  1;

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rotate_block(ractx>adapt_cb, buffer_a, cba_idx); 
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m[0] = irms(buffer_a, gval) >> 12; 
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} else {

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m[0] = 0; 
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} 
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m[1] = ((ftable1[cb1_idx] >> 4) * gval) >> 8; 
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m[2] = ((ftable2[cb2_idx] >> 4) * gval) >> 8; 
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memmove(ractx>adapt_cb, ractx>adapt_cb + BLOCKSIZE, 
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(BUFFERSIZE  BLOCKSIZE) * 2);

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block = ractx>adapt_cb + BUFFERSIZE  BLOCKSIZE; 
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add_wav(gain, cba_idx, m, buffer_a, etable1[cb1_idx], etable2[cb2_idx], 
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block); 
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final(lpc_coefs, block, output_buffer, ractx>buffer, BLOCKSIZE); 
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} 
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static void int_to_int16(int16_t *decsp, const int *inp) 
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{ 
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int i;

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for (i=0; i<30; i++) 
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*(decsp++) = *(inp++); 
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} 
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/**

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* Evaluate the reflection coefficients from the filter coefficients.

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* Does the inverse of the eval_coefs() function.

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*

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* @return 1 if one of the reflection coefficients is of magnitude greater than

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* 4095, 0 if not.

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*/

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static int eval_refl(const int16_t *coefs, int *refl, RA144Context *ractx) 
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{ 
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int retval = 0; 
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int b, c, i;

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unsigned int u; 
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int buffer1[10]; 
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int buffer2[10]; 
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int *bp1 = buffer1;

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int *bp2 = buffer2;

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for (i=0; i < 10; i++) 
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buffer2[i] = coefs[i]; 
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u = refl[9] = bp2[9]; 
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if (u + 0x1000 > 0x1fff) { 
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av_log(ractx, AV_LOG_ERROR, "Overflow. Broken sample?\n");

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return 0; 
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} 
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for (c=8; c >= 0; c) { 
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if (u == 0x1000) 
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u++; 
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if (u == 0xfffff000) 
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u; 
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b = 0x1000((u * u) >> 12); 
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if (b == 0) 
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b++; 
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for (u=0; u<=c; u++) 
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bp1[u] = ((bp2[u]  ((refl[c+1] * bp2[cu]) >> 12)) * (0x1000000 / b)) >> 12; 
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refl[c] = u = bp1[c]; 
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if ((u + 0x1000) > 0x1fff) 
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retval = 1;

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FFSWAP(int *, bp1, bp2);

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} 
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return retval;

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} 
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static int interp(RA144Context *ractx, int16_t *decsp, int block_num, 
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int copynew, int energy) 
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{ 
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int work[10]; 
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int a = block_num + 1; 
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int b = NBLOCKS  a;

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int x;

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// Interpolate block coefficients from the this frame forth block and

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// last frame forth block

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for (x=0; x<30; x++) 
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decsp[x] = (a * ractx>lpc_coef[x] + b * ractx>lpc_coef_old[x])>> 2;

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if (eval_refl(decsp, work, ractx)) {

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// The interpolated coefficients are unstable, copy either new or old

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// coefficients

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if (copynew) {

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int_to_int16(decsp, ractx>lpc_coef); 
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return rescale_rms(rms(ractx>lpc_refl), energy);

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} else {

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int_to_int16(decsp, ractx>lpc_coef_old); 
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return rescale_rms(rms(ractx>lpc_refl_old), energy);

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} 
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} else {

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return rescale_rms(rms(work), energy);

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} 
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} 
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/* Uncompress one block (20 bytes > 160*2 bytes) */

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static int ra144_decode_frame(AVCodecContext * avctx, 
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void *vdata, int *data_size, 
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const uint8_t * buf, int buf_size) 
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{ 
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static const uint8_t sizes[10] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2}; 
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unsigned int refl_rms[4]; // RMS of the reflection coefficients 
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uint16_t block_coefs[4][30]; // LPC coefficients of each subblock 
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int i, c;

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int16_t *data = vdata; 
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unsigned int energy; 
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RA144Context *ractx = avctx>priv_data; 
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GetBitContext gb; 
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if(buf_size < 20) { 
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av_log(avctx, AV_LOG_ERROR, 
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"Frame too small (%d bytes). Truncated file?\n", buf_size);

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return buf_size;

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} 
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init_get_bits(&gb, buf, 20 * 8); 
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for (i=0; i<10; i++) 
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// "<< 1"? Doesn't this make one value out of two of the table useless?

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ractx>lpc_refl[i] = decodetable[i][get_bits(&gb, sizes[i]) << 1];

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eval_coefs(ractx>lpc_refl, ractx>lpc_coef); 
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energy = decodeval[get_bits(&gb, 5) << 1]; // Useless table entries? 
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refl_rms[0] = interp(ractx, block_coefs[0], 0, 0, ractx>old_energy); 
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refl_rms[1] = interp(ractx, block_coefs[1], 1, energy > ractx>old_energy, 
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t_sqrt(energy*ractx>old_energy) >> 12);

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refl_rms[2] = interp(ractx, block_coefs[2], 2, 1, energy); 
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refl_rms[3] = rescale_rms(rms(ractx>lpc_refl), energy);

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int_to_int16(block_coefs[3], ractx>lpc_coef);

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/* do output */

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for (c=0; c<4; c++) { 
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do_output_subblock(ractx, block_coefs[c], refl_rms[c], data, &gb); 
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for (i=0; i<BLOCKSIZE; i++) { 
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*data = av_clip_int16(*data << 2);

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data++; 
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} 
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} 
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ractx>old_energy = energy; 
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FFSWAP(unsigned int *, ractx>lpc_refl_old, ractx>lpc_refl); 
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FFSWAP(unsigned int *, ractx>lpc_coef_old, ractx>lpc_coef); 
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*data_size = 2*160; 
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return 20; 
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} 
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AVCodec ra_144_decoder = 
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{ 
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"real_144",

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CODEC_TYPE_AUDIO, 
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CODEC_ID_RA_144, 
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sizeof(RA144Context),

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ra144_decode_init, 
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NULL,

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NULL,

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ra144_decode_frame, 
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.long_name = "RealAudio 1.0 (14.4K)",

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}; 