3 * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
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
23 * @file libavcodec/resample2.c
25 * @author Michael Niedermayer <michaelni@gmx.at>
31 #ifndef CONFIG_RESAMPLE_HP
32 #define FILTER_SHIFT 15
35 #define FELEM2 int32_t
36 #define FELEML int64_t
37 #define FELEM_MAX INT16_MAX
38 #define FELEM_MIN INT16_MIN
40 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
41 #define FILTER_SHIFT 30
44 #define FELEM2 int64_t
45 #define FELEML int64_t
46 #define FELEM_MAX INT32_MAX
47 #define FELEM_MIN INT32_MIN
48 #define WINDOW_TYPE 12
50 #define FILTER_SHIFT 0
55 #define WINDOW_TYPE 24
59 typedef struct AVResampleContext
{
67 int compensation_distance
;
74 * 0th order modified bessel function of the first kind.
76 static double bessel(double x
){
90 * builds a polyphase filterbank.
91 * @param factor resampling factor
92 * @param scale wanted sum of coefficients for each filter
93 * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
95 void av_build_filter(FELEM
*filter
, double factor
, int tap_count
, int phase_count
, int scale
, int type
){
97 double x
, y
, w
, tab
[tap_count
];
98 const int center
= (tap_count
-1)/2;
100 /* if upsampling, only need to interpolate, no filter */
104 for(ph
=0;ph
<phase_count
;ph
++) {
106 for(i
=0;i
<tap_count
;i
++) {
107 x
= M_PI
* ((double)(i
- center
) - (double)ph
/ phase_count
) * factor
;
112 const float d
= -0.5; //first order derivative = -0.5
113 x
= fabs(((double)(i
- center
) - (double)ph
/ phase_count
) * factor
);
114 if(x
<1.0) y
= 1 - 3*x
*x
+ 2*x
*x
*x
+ d
*( -x
*x
+ x
*x
*x
);
115 else y
= d
*(-4 + 8*x
- 5*x
*x
+ x
*x
*x
);
118 w
= 2.0*x
/ (factor
*tap_count
) + M_PI
;
119 y
*= 0.3635819 - 0.4891775 * cos(w
) + 0.1365995 * cos(2*w
) - 0.0106411 * cos(3*w
);
122 w
= 2.0*x
/ (factor
*tap_count
*M_PI
);
123 y
*= bessel(type
*sqrt(FFMAX(1-w
*w
, 0)));
131 /* normalize so that an uniform color remains the same */
132 for(i
=0;i
<tap_count
;i
++) {
133 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
134 filter
[ph
* tap_count
+ i
] = tab
[i
] / norm
;
136 filter
[ph
* tap_count
+ i
] = av_clip(lrintf(tab
[i
] * scale
/ norm
), FELEM_MIN
, FELEM_MAX
);
144 double sine
[LEN
+ tap_count
];
145 double filtered
[LEN
];
146 double maxff
=-2, minff
=2, maxsf
=-2, minsf
=2;
147 for(i
=0; i
<LEN
; i
++){
148 double ss
=0, sf
=0, ff
=0;
149 for(j
=0; j
<LEN
+tap_count
; j
++)
150 sine
[j
]= cos(i
*j
*M_PI
/LEN
);
151 for(j
=0; j
<LEN
; j
++){
154 for(k
=0; k
<tap_count
; k
++)
155 sum
+= filter
[ph
* tap_count
+ k
] * sine
[k
+j
];
156 filtered
[j
]= sum
/ (1<<FILTER_SHIFT
);
157 ss
+= sine
[j
+ center
] * sine
[j
+ center
];
158 ff
+= filtered
[j
] * filtered
[j
];
159 sf
+= sine
[j
+ center
] * filtered
[j
];
164 maxff
= FFMAX(maxff
, ff
);
165 minff
= FFMIN(minff
, ff
);
166 maxsf
= FFMAX(maxsf
, sf
);
167 minsf
= FFMIN(minsf
, sf
);
169 av_log(NULL
, AV_LOG_ERROR
, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i
, ss
, maxff
, minff
, maxsf
, minsf
);
178 AVResampleContext
*av_resample_init(int out_rate
, int in_rate
, int filter_size
, int phase_shift
, int linear
, double cutoff
){
179 AVResampleContext
*c
= av_mallocz(sizeof(AVResampleContext
));
180 double factor
= FFMIN(out_rate
* cutoff
/ in_rate
, 1.0);
181 int phase_count
= 1<<phase_shift
;
183 c
->phase_shift
= phase_shift
;
184 c
->phase_mask
= phase_count
-1;
187 c
->filter_length
= FFMAX((int)ceil(filter_size
/factor
), 1);
188 c
->filter_bank
= av_mallocz(c
->filter_length
*(phase_count
+1)*sizeof(FELEM
));
189 av_build_filter(c
->filter_bank
, factor
, c
->filter_length
, phase_count
, 1<<FILTER_SHIFT
, WINDOW_TYPE
);
190 memcpy(&c
->filter_bank
[c
->filter_length
*phase_count
+1], c
->filter_bank
, (c
->filter_length
-1)*sizeof(FELEM
));
191 c
->filter_bank
[c
->filter_length
*phase_count
]= c
->filter_bank
[c
->filter_length
- 1];
193 c
->src_incr
= out_rate
;
194 c
->ideal_dst_incr
= c
->dst_incr
= in_rate
* phase_count
;
195 c
->index
= -phase_count
*((c
->filter_length
-1)/2);
200 void av_resample_close(AVResampleContext
*c
){
201 av_freep(&c
->filter_bank
);
205 void av_resample_compensate(AVResampleContext
*c
, int sample_delta
, int compensation_distance
){
206 // sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
207 c
->compensation_distance
= compensation_distance
;
208 c
->dst_incr
= c
->ideal_dst_incr
- c
->ideal_dst_incr
* (int64_t)sample_delta
/ compensation_distance
;
211 int av_resample(AVResampleContext
*c
, short *dst
, short *src
, int *consumed
, int src_size
, int dst_size
, int update_ctx
){
215 int dst_incr_frac
= c
->dst_incr
% c
->src_incr
;
216 int dst_incr
= c
->dst_incr
/ c
->src_incr
;
217 int compensation_distance
= c
->compensation_distance
;
219 if(compensation_distance
== 0 && c
->filter_length
== 1 && c
->phase_shift
==0){
220 int64_t index2
= ((int64_t)index
)<<32;
221 int64_t incr
= (1LL<<32) * c
->dst_incr
/ c
->src_incr
;
222 dst_size
= FFMIN(dst_size
, (src_size
-1-index
) * (int64_t)c
->src_incr
/ c
->dst_incr
);
224 for(dst_index
=0; dst_index
< dst_size
; dst_index
++){
225 dst
[dst_index
] = src
[index2
>>32];
228 frac
+= dst_index
* dst_incr_frac
;
229 index
+= dst_index
* dst_incr
;
230 index
+= frac
/ c
->src_incr
;
233 for(dst_index
=0; dst_index
< dst_size
; dst_index
++){
234 FELEM
*filter
= c
->filter_bank
+ c
->filter_length
*(index
& c
->phase_mask
);
235 int sample_index
= index
>> c
->phase_shift
;
238 if(sample_index
< 0){
239 for(i
=0; i
<c
->filter_length
; i
++)
240 val
+= src
[FFABS(sample_index
+ i
) % src_size
] * filter
[i
];
241 }else if(sample_index
+ c
->filter_length
> src_size
){
245 for(i
=0; i
<c
->filter_length
; i
++){
246 val
+= src
[sample_index
+ i
] * (FELEM2
)filter
[i
];
247 v2
+= src
[sample_index
+ i
] * (FELEM2
)filter
[i
+ c
->filter_length
];
249 val
+=(v2
-val
)*(FELEML
)frac
/ c
->src_incr
;
251 for(i
=0; i
<c
->filter_length
; i
++){
252 val
+= src
[sample_index
+ i
] * (FELEM2
)filter
[i
];
256 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
257 dst
[dst_index
] = av_clip_int16(lrintf(val
));
259 val
= (val
+ (1<<(FILTER_SHIFT
-1)))>>FILTER_SHIFT
;
260 dst
[dst_index
] = (unsigned)(val
+ 32768) > 65535 ? (val
>>31) ^ 32767 : val
;
263 frac
+= dst_incr_frac
;
265 if(frac
>= c
->src_incr
){
270 if(dst_index
+ 1 == compensation_distance
){
271 compensation_distance
= 0;
272 dst_incr_frac
= c
->ideal_dst_incr
% c
->src_incr
;
273 dst_incr
= c
->ideal_dst_incr
/ c
->src_incr
;
277 *consumed
= FFMAX(index
, 0) >> c
->phase_shift
;
278 if(index
>=0) index
&= c
->phase_mask
;
280 if(compensation_distance
){
281 compensation_distance
-= dst_index
;
282 assert(compensation_distance
> 0);
287 c
->dst_incr
= dst_incr_frac
+ c
->src_incr
*dst_incr
;
288 c
->compensation_distance
= compensation_distance
;
291 if(update_ctx
&& !c
->compensation_distance
){
293 av_resample_compensate(c
, rand() % (8000*2) - 8000, 8000*2);
294 av_log(NULL
, AV_LOG_DEBUG
, "%d %d %d\n", c
->dst_incr
, c
->ideal_dst_incr
, c
->compensation_distance
);