1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
|
/*=============================================================================
//
// This software has been released under the terms of the GNU Public
// license. See http://www.gnu.org/copyleft/gpl.html for details.
//
// Copyright 2002 Anders Johansson ajh@atri.curtin.edu.au
//
//=============================================================================
*/
/* This audio filter changes the sample rate. */
#define PLUGIN
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <inttypes.h>
#include "../config.h"
#include "../mp_msg.h"
#include "../libao2/afmt.h"
#include "af.h"
#include "dsp.h"
/* Below definition selects the length of each poly phase component.
Valid definitions are L8 and L16, where the number denotes the
length of the filter. This definition affects the computational
complexity (see play()), the performance (see filter.h) and the
memory usage. The filterlenght is choosen to 8 if the machine is
slow and to 16 if the machine is fast and has MMX.
*/
#if !defined(HAVE_SSE) && !defined(HAVE_3DNOW) // This machine is slow
#define L 8 // Filter length
// Unrolled loop to speed up execution
#define FIR(x,w,y) \
(y[0]) = ( w[0]*x[0]+w[1]*x[1]+w[2]*x[2]+w[3]*x[3] \
+ w[4]*x[4]+w[5]*x[5]+w[6]*x[6]+w[7]*x[7] ) >> 16
#else /* Fast machine */
#define L 16
// Unrolled loop to speed up execution
#define FIR(x,w,y) \
y[0] = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] \
+ w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] \
+ w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \
+ w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] ) >> 16
#endif /* Fast machine */
// Macro to add data to circular que
#define ADDQUE(xi,xq,in)\
xq[xi]=xq[xi+L]=(*in);\
xi=(--xi)&(L-1);
// local data
typedef struct af_resample_s
{
int16_t* w; // Current filter weights
int16_t** xq; // Circular buffers
uint32_t xi; // Index for circular buffers
uint32_t wi; // Index for w
uint32_t i; // Number of new samples to put in x queue
uint32_t dn; // Down sampling factor
uint32_t up; // Up sampling factor
int sloppy; // Enable sloppy resampling to reduce memory usage
int fast; // Enable linear interpolation instead of filtering
} af_resample_t;
// Euclids algorithm for calculating Greatest Common Divisor GCD(a,b)
static inline int gcd(register int a, register int b)
{
register int r = min(a,b);
a=max(a,b);
b=r;
r=a%b;
while(r!=0){
a=b;
b=r;
r=a%b;
}
return b;
}
static int upsample(af_data_t* c,af_data_t* l, af_resample_t* s)
{
uint32_t ci = l->nch; // Index for channels
uint32_t len = 0; // Number of input samples
uint32_t nch = l->nch; // Number of channels
uint32_t inc = s->up/s->dn;
uint32_t level = s->up%s->dn;
uint32_t up = s->up;
uint32_t dn = s->dn;
register int16_t* w = s->w;
register uint32_t wi = 0;
register uint32_t xi = 0;
// Index current channel
while(ci--){
// Temporary pointers
register int16_t* x = s->xq[ci];
register int16_t* in = ((int16_t*)c->audio)+ci;
register int16_t* out = ((int16_t*)l->audio)+ci;
int16_t* end = in+c->len/2; // Block loop end
wi = s->wi; xi = s->xi;
while(in < end){
register uint32_t i = inc;
if(wi<level) i++;
ADDQUE(xi,x,in);
in+=nch;
while(i--){
// Run the FIR filter
FIR((&x[xi]),(&w[wi*L]),out);
len++; out+=nch;
// Update wi to point at the correct polyphase component
wi=(wi+dn)%up;
}
}
}
// Save values that needs to be kept for next time
s->wi = wi;
s->xi = xi;
return len;
}
static int downsample(af_data_t* c,af_data_t* l, af_resample_t* s)
{
uint32_t ci = l->nch; // Index for channels
uint32_t len = 0; // Number of output samples
uint32_t nch = l->nch; // Number of channels
uint32_t inc = s->dn/s->up;
uint32_t level = s->dn%s->up;
uint32_t up = s->up;
uint32_t dn = s->dn;
register int32_t i = 0;
register uint32_t wi = 0;
register uint32_t xi = 0;
// Index current channel
while(ci--){
// Temporary pointers
register int16_t* x = s->xq[ci];
register int16_t* in = ((int16_t*)c->audio)+ci;
register int16_t* out = ((int16_t*)l->audio)+ci;
register int16_t* end = in+c->len/2; // Block loop end
i = s->i; wi = s->wi; xi = s->xi;
while(in < end){
ADDQUE(xi,x,in);
in+=nch;
if((--i)<=0){
// Run the FIR filter
FIR((&x[xi]),(&s->w[wi*L]),out);
len++; out+=nch;
// Update wi to point at the correct polyphase component
wi=(wi+dn)%up;
// Insert i number of new samples in queue
i = inc;
if(wi<level) i++;
}
}
}
// Save values that needs to be kept for next time
s->wi = wi;
s->xi = xi;
s->i = i;
return len;
}
// Initialization and runtime control
static int control(struct af_instance_s* af, int cmd, void* arg)
{
switch(cmd){
case AF_CONTROL_REINIT:{
af_resample_t* s = (af_resample_t*)af->setup;
af_data_t* n = (af_data_t*)arg; // New configureation
int i,d = 0;
int rv = AF_OK;
// Make sure this filter isn't redundant
if(af->data->rate == n->rate)
return AF_DETACH;
// Create space for circular bufers (if nesessary)
if(af->data->nch != n->nch){
// First free the old ones
if(s->xq){
for(i=1;i<af->data->nch;i++)
if(s->xq[i])
free(s->xq[i]);
free(s->xq);
}
// ... then create new
s->xq = malloc(n->nch*sizeof(int16_t*));
for(i=0;i<n->nch;i++)
s->xq[i] = malloc(2*L*sizeof(int16_t));
s->xi = 0;
}
// Set parameters
af->data->nch = n->nch;
af->data->format = AFMT_S16_NE;
af->data->bps = 2;
if(af->data->format != n->format || af->data->bps != n->bps)
rv = AF_FALSE;
n->format = AFMT_S16_NE;
n->bps = 2;
// Calculate up and down sampling factors
d=gcd(af->data->rate,n->rate);
// If sloppy resampling is enabled limit the upsampling factor
if(s->sloppy && (af->data->rate/d > 5000)){
int up=af->data->rate/2;
int dn=n->rate/2;
int m=2;
while(af->data->rate/(d*m) > 5000){
d=gcd(up,dn);
up/=2; dn/=2; m*=2;
}
d*=m;
}
// Check if the the design needs to be redone
if(s->up != af->data->rate/d || s->dn != n->rate/d){
float* w;
float* wt;
float fc;
int j;
s->up = af->data->rate/d;
s->dn = n->rate/d;
// Calculate cuttof frequency for filter
fc = 1/(float)(max(s->up,s->dn));
// Allocate space for polyphase filter bank and protptype filter
w = malloc(sizeof(float) * s->up *L);
if(NULL != s->w)
free(s->w);
s->w = malloc(L*s->up*sizeof(int16_t));
// Design prototype filter type using Kaiser window with beta = 10
if(NULL == w || NULL == s->w ||
-1 == design_fir(s->up*L, w, &fc, LP|KAISER , 10.0)){
mp_msg(MSGT_AFILTER,MSGL_ERR,"[resample] Unable to design prototype filter.\n");
return AF_ERROR;
}
// Copy data from prototype to polyphase filter
wt=w;
for(j=0;j<L;j++){//Columns
for(i=0;i<s->up;i++){//Rows
float t=(float)s->up*32767.0*(*wt);
s->w[i*L+j] = (int16_t)((t>=0.0)?(t+0.5):(t-0.5));
wt++;
}
}
free(w);
mp_msg(MSGT_AFILTER,MSGL_V,"[resample] New filter designed up: %i down: %i\n", s->up, s->dn);
}
// Set multiplier and delay
af->delay = (double)(1000*L/2)/((double)n->rate);
af->mul.n = s->up;
af->mul.d = s->dn;
return rv;
}
case AF_CONTROL_COMMAND_LINE:{
af_resample_t* s = (af_resample_t*)af->setup;
int rate=0;
sscanf((char*)arg,"%i:%i:%i",&rate,&(s->sloppy), &(s->fast));
return af->control(af,AF_CONTROL_RESAMPLE,&rate);
}
case AF_CONTROL_RESAMPLE:
// Reinit must be called after this function has been called
// Sanity check
if(((int*)arg)[0] < 8000 || ((int*)arg)[0] > 192000){
mp_msg(MSGT_AFILTER,MSGL_ERR,"[resample] The output sample frequency must be between 8kHz and 192kHz. Current value is %i \n",((int*)arg)[0]);
return AF_ERROR;
}
af->data->rate=((int*)arg)[0];
mp_msg(MSGT_AFILTER,MSGL_V,"[resample] Changing sample rate to %iHz\n",af->data->rate);
return AF_OK;
}
return AF_UNKNOWN;
}
// Deallocate memory
static void uninit(struct af_instance_s* af)
{
if(af->data)
free(af->data);
}
// Filter data through filter
static af_data_t* play(struct af_instance_s* af, af_data_t* data)
{
int len = 0; // Length of output data
af_data_t* c = data; // Current working data
af_data_t* l = af->data; // Local data
af_resample_t* s = (af_resample_t*)af->setup;
if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
return NULL;
// Run resampling
if(s->up>s->dn)
len = upsample(c,l,s);
else
len = downsample(c,l,s);
// Set output data
c->audio = l->audio;
c->len = len*2;
c->rate = l->rate;
return c;
}
// Allocate memory and set function pointers
static int open(af_instance_t* af){
af->control=control;
af->uninit=uninit;
af->play=play;
af->mul.n=1;
af->mul.d=1;
af->data=calloc(1,sizeof(af_data_t));
af->setup=calloc(1,sizeof(af_resample_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
return AF_OK;
}
// Description of this plugin
af_info_t af_info_resample = {
"Sample frequency conversion",
"resample",
"Anders",
"",
AF_FLAGS_REENTRANT,
open
};
|