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1/*
2 * COOK compatible decoder, fixed point implementation.
3 * Copyright (c) 2007 Ian Braithwaite
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
23/**
24 * @file cook_fixpoint.h
25 *
26 * Cook AKA RealAudio G2 fixed point functions.
27 *
28 * Fixed point values are represented as 32 bit signed integers,
29 * which can be added and subtracted directly in C (without checks for
30 * overflow/saturation.
31 * Two multiplication routines are provided:
32 * 1) Multiplication by powers of two (2^-31 .. 2^31), implemented
33 * with C's bit shift operations.
34 * 2) Multiplication by 16 bit fractions (0 <= x < 1), implemented
35 * in C using two 32 bit integer multiplications.
36 */
37
38#ifdef ROCKBOX
39/* get definitions of MULT31, MULT31_SHIFT16, vect_add, from codelib */
40#include "codeclib_misc.h"
41#include "codeclib.h"
42#endif
43
44/* cplscales was moved from cookdata_fixpoint.h since only *
45 * cook_fixpoint.h should see/use it. */
46static const FIXPU* cplscales[5] = {
47 cplscale2, cplscale3, cplscale4, cplscale5, cplscale6
48};
49
50/**
51 * Fixed point multiply by power of two.
52 *
53 * @param x fix point value
54 * @param i integer power-of-two, -31..+31
55 */
56static inline FIXP fixp_pow2(FIXP x, int i)
57{
58 if (i < 0)
59 return (x >> -i);
60 else
61 return x << i; /* no check for overflow */
62}
63
64/**
65 * Fixed point multiply by fraction.
66 *
67 * @param a fix point value
68 * @param b fix point fraction, 0 <= b < 1
69 */
70#ifdef ROCKBOX
71#define fixp_mult_su(x,y) (MULT31_SHIFT16(x,y))
72#else
73static inline FIXP fixp_mult_su(FIXP a, FIXPU b)
74{
75 int32_t hb = (a >> 16) * b;
76 uint32_t lb = (a & 0xffff) * b;
77
78 return hb + (lb >> 16) + ((lb & 0x8000) >> 15);
79}
80#endif
81
82/* Faster version of the above using 32x32=64 bit multiply */
83#ifdef ROCKBOX
84#define fixmul31(x,y) (MULT31(x,y))
85#else
86static inline int32_t fixmul31(int32_t x, int32_t y)
87{
88 int64_t temp;
89
90 temp = x;
91 temp *= y;
92
93 temp >>= 31; //16+31-16 = 31 bits
94
95 return (int32_t)temp;
96}
97#endif
98
99/**
100 * Clips a signed integer value into the amin-amax range.
101 * @param a value to clip
102 * @param amin minimum value of the clip range
103 * @param amax maximum value of the clip range
104 * @return clipped value
105 */
106static inline int av_clip(int a, int amin, int amax)
107{
108 if (a < amin) return amin;
109 else if (a > amax) return amax;
110 else return a;
111}
112
113/**
114 * The real requantization of the mltcoefs
115 *
116 * @param q pointer to the COOKContext
117 * @param index index
118 * @param quant_index quantisation index for this band
119 * @param subband_coef_index array of indexes to quant_centroid_tab
120 * @param subband_coef_sign use random noise instead of predetermined value
121 * @param mlt_ptr pointer to the mlt coefficients
122 */
123
124static void scalar_dequant_math(COOKContext *q, int index,
125 int quant_index, int* subband_coef_index,
126 int* subband_coef_sign, REAL_T *mlt_p)
127 ICODE_ATTR_COOK_DECODE;
128static void scalar_dequant_math(COOKContext *q, int index,
129 int quant_index, int* subband_coef_index,
130 int* subband_coef_sign, REAL_T *mlt_p)
131{
132 /* Num. half bits to right shift */
133 const int s = 33 - quant_index + av_log2(q->samples_per_channel);
134 const FIXP *table = quant_tables[s & 1][index];
135 FIXP f;
136 int i;
137
138
139 if(s >= 64)
140 memset(mlt_p, 0, sizeof(REAL_T)*SUBBAND_SIZE);
141 else
142 {
143 for(i=0 ; i<SUBBAND_SIZE ; i++) {
144 f = (table[subband_coef_index[i]]) >> (s >> 1);
145 /* noise coding if subband_coef_index[i] == 0 */
146 if (((subband_coef_index[i] == 0) && cook_random(q)) ||
147 ((subband_coef_index[i] != 0) && subband_coef_sign[i]))
148 f = -f;
149
150 *mlt_p++ = f;
151 }
152 }
153}
154
155/**
156 * The modulated lapped transform, this takes transform coefficients
157 * and transforms them into timedomain samples.
158 * A window step is also included.
159 *
160 * @param q pointer to the COOKContext
161 * @param inbuffer pointer to the mltcoefficients
162 * @param outbuffer pointer to the timedomain buffer
163 * @param mlt_tmp pointer to temporary storage space
164 */
165#include "../lib/mdct_lookup.h"
166
167void imlt_math(COOKContext *q, FIXP *in) ICODE_ATTR;
168void imlt_math(COOKContext *q, FIXP *in)
169{
170 const int n = q->samples_per_channel;
171 const int step = 2 << (10 - av_log2(n));
172 REAL_T *mdct_out = q->mono_mdct_output;
173 REAL_T tmp;
174 int i = 0, j = 0;
175
176 ff_imdct_calc(q->mdct_nbits, q->mono_mdct_output, in);
177
178 do {
179 tmp = mdct_out[i];
180 mdct_out[i ] = fixmul31(-mdct_out[n+i], (sincos_lookup0[j ]));
181 mdct_out[n+i] = fixmul31(tmp , (sincos_lookup0[j+1]));
182
183 j += step;
184 } while (++i < n/2);
185
186 do {
187 j -= step;
188
189 tmp = mdct_out[i];
190 mdct_out[i ] = fixmul31(-mdct_out[n+i], (sincos_lookup0[j+1]));
191 mdct_out[n+i] = fixmul31(tmp , (sincos_lookup0[j ]));
192 } while (++i < n);
193}
194
195/**
196 * Perform buffer overlapping.
197 *
198 * @param q pointer to the COOKContext
199 * @param gain gain correction to apply first to output buffer
200 * @param buffer data to overlap
201 */
202void overlap_math(COOKContext *q, int gain, FIXP buffer[]) ICODE_ATTR;
203void overlap_math(COOKContext *q, int gain, FIXP buffer[])
204{
205 int i;
206#ifdef ROCKBOX
207 if(LIKELY(gain == 0))
208 {
209 vect_add(q->mono_mdct_output, buffer, q->samples_per_channel);
210
211 } else if (gain > 0){
212 for(i=0 ; i<q->samples_per_channel ; i++) {
213 q->mono_mdct_output[i] = (q->mono_mdct_output[i]<< gain) + buffer[i]; }
214
215 } else {
216 for(i=0 ; i<q->samples_per_channel ; i++) {
217 q->mono_mdct_output[i] = (q->mono_mdct_output[i]>>-gain) + buffer[i];
218 }
219 }
220#else
221 for(i=0 ; i<q->samples_per_channel ; i++) {
222 q->mono_mdct_output[i] =
223 fixp_pow2(q->mono_mdct_output[i], gain) + buffer[i];
224 }
225#endif
226}
227
228
229/**
230 * the actual requantization of the timedomain samples
231 *
232 * @param q pointer to the COOKContext
233 * @param buffer pointer to the timedomain buffer
234 * @param gain_index index for the block multiplier
235 * @param gain_index_next index for the next block multiplier
236 */
237static inline void
238interpolate_math(COOKContext *q, register FIXP* buffer,
239 int gain_index, int gain_index_next)
240{
241 int i;
242 int gain_size_factor = q->samples_per_channel / 8;
243
244 if(gain_index == gain_index_next){ //static gain
245 for(i = 0; i < gain_size_factor; i++) {
246 buffer[i] = fixp_pow2(buffer[i], gain_index);
247 }
248 } else { //smooth gain
249 int step = (gain_index_next - gain_index)
250 << (7 - av_log2(gain_size_factor));
251 int x = 0;
252 register FIXP* bufferend = buffer+gain_size_factor;
253 while(buffer < bufferend )
254 {
255 *buffer = fixp_pow2(
256 fixp_mult_su(*buffer, pow128_tab[x]),
257 gain_index+1);
258 buffer++;
259
260 x += step;
261 gain_index += ( (x + 128) >> 7 ) - 1;
262 x = ( (x + 128) & 127 );
263 }
264 }
265}
266
267
268/**
269 * Decoupling calculation for joint stereo coefficients.
270 *
271 * @param x mono coefficient
272 * @param table number of decoupling table
273 * @param i table index
274 */
275static inline FIXP cplscale_math(FIXP x, int table, int i)
276{
277 return fixp_mult_su(x, cplscales[table-2][i]);
278}