2 * COOK compatible decoder, fixed point implementation.
3 * Copyright (c) 2007 Ian Braithwaite
5 * This file is part of FFmpeg.
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.
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.
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
26 * Cook AKA RealAudio G2 fixed point functions.
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.
38 /* The following table is taken from libavutil/mathematics.c */
39 const uint8_t ff_log2_tab
[256]={
40 0,0,1,1,2,2,2,2,3,3,3,3,3,3,3,3,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,
41 5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,
42 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
43 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
44 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
45 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
46 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
47 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7
50 /* cplscales was moved from cookdata_fixpoint.h since only *
51 * cook_fixpoint.h should see/use it. */
52 static const FIXPU
* cplscales
[5] = {
53 cplscale2
, cplscale3
, cplscale4
, cplscale5
, cplscale6
57 * Fixed point multiply by power of two.
59 * @param x fix point value
60 * @param i integer power-of-two, -31..+31
62 static inline FIXP
fixp_pow2(FIXP x
, int i
)
65 return (x
>> -i
) + ((x
>> (-i
-1)) & 1);
67 return x
<< i
; /* no check for overflow */
71 * Fixed point multiply by fraction.
73 * @param a fix point value
74 * @param b fix point fraction, 0 <= b < 1
76 static inline FIXP
fixp_mult_su(FIXP a
, FIXPU b
)
78 int32_t hb
= (a
>> 16) * b
;
79 uint32_t lb
= (a
& 0xffff) * b
;
81 return hb
+ (lb
>> 16) + ((lb
& 0x8000) >> 15);
84 /* math functions taken from libavutil/common.h */
86 static inline int av_log2(unsigned int v
)
103 * Clips a signed integer value into the amin-amax range.
104 * @param a value to clip
105 * @param amin minimum value of the clip range
106 * @param amax maximum value of the clip range
107 * @return clipped value
109 static inline int av_clip(int a
, int amin
, int amax
)
111 if (a
< amin
) return amin
;
112 else if (a
> amax
) return amax
;
117 * The real requantization of the mltcoefs
119 * @param q pointer to the COOKContext
121 * @param quant_index quantisation index for this band
122 * @param subband_coef_index array of indexes to quant_centroid_tab
123 * @param subband_coef_sign use random noise instead of predetermined value
124 * @param mlt_ptr pointer to the mlt coefficients
126 static void scalar_dequant_math(COOKContext
*q
, int index
,
127 int quant_index
, int* subband_coef_index
,
128 int* subband_coef_sign
, REAL_T
*mlt_p
)
130 /* Num. half bits to right shift */
131 const int s
= 33 - quant_index
+ av_log2(q
->samples_per_channel
);
132 const FIXP
*table
= quant_tables
[s
& 1][index
];
136 for(i
=0 ; i
<SUBBAND_SIZE
; i
++) {
137 f
= table
[subband_coef_index
[i
]];
138 /* noise coding if subband_coef_index[i] == 0 */
139 if (((subband_coef_index
[i
] == 0) && cook_random(q
)) ||
140 ((subband_coef_index
[i
] != 0) && subband_coef_sign
[i
]))
143 mlt_p
[i
] = (s
>= 64) ? 0 : fixp_pow2(f
, -(s
/2));
149 * The modulated lapped transform, this takes transform coefficients
150 * and transforms them into timedomain samples.
151 * A window step is also included.
153 * @param q pointer to the COOKContext
154 * @param inbuffer pointer to the mltcoefficients
155 * @param outbuffer pointer to the timedomain buffer
156 * @param mlt_tmp pointer to temporary storage space
158 #include "cook_fixp_mdct.h"
160 static inline void imlt_math(COOKContext
*q
, FIXP
*in
)
162 const int n
= q
->samples_per_channel
;
163 const int step
= 4 << (10 - av_log2(n
));
164 int i
= 0, j
= step
>>1;
166 cook_mdct_backward(2 * n
, in
, q
->mono_mdct_output
);
169 FIXP tmp
= q
->mono_mdct_output
[i
];
171 q
->mono_mdct_output
[i
] =
172 fixp_mult_su(-q
->mono_mdct_output
[n
+ i
], sincos_lookup
[j
]);
173 q
->mono_mdct_output
[n
+ i
] = fixp_mult_su(tmp
, sincos_lookup
[j
+1]);
177 FIXP tmp
= q
->mono_mdct_output
[i
];
180 q
->mono_mdct_output
[i
] =
181 fixp_mult_su(-q
->mono_mdct_output
[n
+ i
], sincos_lookup
[j
+1]);
182 q
->mono_mdct_output
[n
+ i
] = fixp_mult_su(tmp
, sincos_lookup
[j
]);
186 #include <codecs/lib/codeclib.h>
188 static inline void imlt_math(COOKContext
*q
, FIXP
*in
)
190 const int n
= q
->samples_per_channel
;
191 const int step
= 4 << (10 - av_log2(n
));
192 int i
= 0, j
= step
>>1;
194 mdct_backward(2 * n
, in
, q
->mono_mdct_output
);
197 FIXP tmp
= q
->mono_mdct_output
[i
];
199 q
->mono_mdct_output
[i
] =
200 fixp_mult_su(-q
->mono_mdct_output
[n
+ i
], sincos_lookup
[j
]);
201 q
->mono_mdct_output
[n
+ i
] = fixp_mult_su(tmp
, sincos_lookup
[j
+1]);
205 FIXP tmp
= q
->mono_mdct_output
[i
];
208 q
->mono_mdct_output
[i
] =
209 fixp_mult_su(-q
->mono_mdct_output
[n
+ i
], sincos_lookup
[j
+1]);
210 q
->mono_mdct_output
[n
+ i
] = fixp_mult_su(tmp
, sincos_lookup
[j
]);
216 * Perform buffer overlapping.
218 * @param q pointer to the COOKContext
219 * @param gain gain correction to apply first to output buffer
220 * @param buffer data to overlap
222 static inline void overlap_math(COOKContext
*q
, int gain
, FIXP buffer
[])
225 for(i
=0 ; i
<q
->samples_per_channel
; i
++) {
226 q
->mono_mdct_output
[i
] =
227 fixp_pow2(q
->mono_mdct_output
[i
], gain
) + buffer
[i
];
233 * the actual requantization of the timedomain samples
235 * @param q pointer to the COOKContext
236 * @param buffer pointer to the timedomain buffer
237 * @param gain_index index for the block multiplier
238 * @param gain_index_next index for the next block multiplier
241 interpolate_math(COOKContext
*q
, FIXP
* buffer
,
242 int gain_index
, int gain_index_next
)
245 int gain_size_factor
= q
->samples_per_channel
/ 8;
247 if(gain_index
== gain_index_next
){ //static gain
248 for(i
= 0; i
< gain_size_factor
; i
++) {
249 buffer
[i
] = fixp_pow2(buffer
[i
], gain_index
);
251 } else { //smooth gain
252 int step
= (gain_index_next
- gain_index
)
253 << (7 - av_log2(gain_size_factor
));
256 for(i
= 0; i
< gain_size_factor
; i
++) {
257 buffer
[i
] = fixp_mult_su(buffer
[i
], pow128_tab
[x
]);
258 buffer
[i
] = fixp_pow2(buffer
[i
], gain_index
+1);
261 gain_index
+= (x
+ 128) / 128 - 1;
269 * Decoupling calculation for joint stereo coefficients.
271 * @param x mono coefficient
272 * @param table number of decoupling table
273 * @param i table index
275 static inline FIXP
cplscale_math(FIXP x
, int table
, int i
)
277 return fixp_mult_su(x
, cplscales
[table
-2][i
]);
282 * Final converion from floating point values to
283 * signed, 16 bit sound samples. Round and clip.
285 * @param q pointer to the COOKContext
286 * @param out pointer to the output buffer
287 * @param chan 0: left or single channel, 1: right channel
289 static inline void output_math(COOKContext
*q
, int16_t *out
, int chan
)
293 for (j
= 0; j
< q
->samples_per_channel
; j
++) {
294 out
[chan
+ q
->nb_channels
* j
] = fixp_pow2(q
->mono_mdct_output
[j
], -11);