xref: /illumos-gate/usr/src/cmd/audio/utilities/Resample.cc (revision 014ec826f15c2551c9f84a5fe6f816ffb5bee7da)
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21  */
22 /*
23  * Copyright (c) 1992-2001 by Sun Microsystems, Inc.
24  * All rights reserved.
25  */
26 
27 /*
28  * This is a impementation of sampling rate conversion for low-passed signals.
29  *
30  * To convert the input signal of sampling rate of fi to another rate of fo,
31  * the least common multiple of fi and fo is derived first:
32  *	fm = L * fi = M * fo.
33  * Then the input signal is up-sampled to fm by inserting (L -1) zero valued
34  * samples after each input sample, low-pass filtered, and down-smapled by
35  * saving one sample for every M samples. The implementation takes advantages
36  * of the (L - 1) zero values in the filter input and the (M - 1) output
37  * samples to be disregarded.
38  *
39  * If L = 1 or M = 1, a simpler decimation or interpolation is used.
40  */
41 #include <memory.h>
42 #include <math.h>
43 
44 #include <Resample.h>
45 
46 extern "C" {
47 	char *bcopy(char *, char *, int);
48 	char *memmove(char *, char *, int);
49 }
50 
51 #define	BCOPY(src, dest, num) memmove(dest, src, num)
52 
53 // convolve(), short2double() and double2short() are defined in Fir.cc.
54 extern double convolve(double *, double *, int);
55 extern void short2double(double *, short *, int);
56 extern short double2short(double);
57 
58 // generate truncated ideal LPF
59 static void sinc_coef(int	fold,	// sample rate change
60 	int	order,			// LP FIR filter order
61 	double  *coef)			// filter coefficients
62 {
63 	int	i;
64 	float   alpha;
65 	double  bandwidth = M_PI / fold; // digital bandwidth of pass band
66 
67 	int half = order >> 1;
68 	if (order & 1) {		// order is odd, center = half + 0.5
69 		float center = half + 0.5;
70 		for (i = 0; i <= half; i++) {
71 			alpha = center - i;
72 			coef[i] = sin(bandwidth * alpha) / (M_PI * alpha);
73 		}
74 	} else {	// order is even, center = half
75 		for (i = 0; i < half; i++) {
76 			alpha = half - i;
77 			coef[i] = sin(bandwidth * alpha) / (M_PI * alpha);
78 		}
79 		coef[i++] = bandwidth / M_PI;
80 	}
81 	for (; i <= order; i++)		// symmetric FIR
82 		coef[i] = coef[order - i];
83 }
84 
85 /*
86  * poly_conv() = coef[0] * data[length - 1] +
87  *		 coef[inc_coef] * data[length - 2] +
88  *		 ...
89  *		 coef[(length - 1) * inc_coef] * data[0] +
90  *
91  * convolution of coef[order + 1] and data[length] up-sampled by a factor
92  * of inc_coef. The terms of coef[1, 2, ... inc_coef - 1] multiplying 0
93  * are ignored.
94  */
95 // polyphase filter convolution
96 double
97 poly_conv(double	*coef,		// filter coef array
98 	    int		order,		// filter order
99 	    int		inc_coef,	// 1-to-L up-sample for data[length]
100 	    double	*data,		// data array
101 	    int		length)		// data array length
102 {
103 	if ((order < 0) || (inc_coef < 1) || (length < 1))
104 		return (0.0);
105 	else {
106 		double sum = 0.0;
107 		double *coef_end = coef + order;
108 		double *data_end = data + length;
109 		while ((coef <= coef_end) && (data < data_end)) {
110 			sum += *coef * *--data_end;
111 			coef += inc_coef;
112 		}
113 		return (sum);
114 	}
115 }
116 
117 int
118 gcf(int a, int b)		// greatest common factor between a and b
119 {
120 	int remainder = a % b;
121 	return (remainder == 0)? b : gcf(b, remainder);
122 }
123 
124 void ResampleFilter::
125 updateState(
126 	double *in,
127 	int size)
128 {
129 	if (up <= 1)
130 		Fir::updateState(in, size);
131 	else if (size >= num_state)
132 		memcpy(state, in + size - num_state,
133 		    num_state * sizeof (double));
134 	else {
135 		int old = num_state - size;
136 		BCOPY((char *)(state + size), (char *)state,
137 		    old * sizeof (double));
138 		memcpy(state + old, in, size * sizeof (double));
139 	}
140 }
141 
142 ResampleFilter::			// constructor
143 ResampleFilter(
144 	int rate_in,			// sampling rate of input signal
145 	int rate_out)			// sampling rate of output signal
146 {
147 	// filter sampling rate = common multiple of rate_in and rate_out
148 	int commonfactor = gcf(rate_in, rate_out);
149 	up = rate_out / commonfactor;
150 	down = rate_in / commonfactor;
151 
152 	int fold = (up > down)? up : down;	// take the bigger rate change
153 	order = (fold << 4) - 2;		// filter order = fold * 16 - 2
154 	coef = new double[order + 1];
155 	sinc_coef(fold, order, coef);		// required bandwidth = PI/fold
156 
157 	if (up > 1) {				// need (order/up) states
158 		num_state = (order + up  - 1) / up;
159 		state = new double[num_state];
160 		for (int i = 0; i < num_state; i++)	// reset states
161 			state[i] = 0.0;
162 	} else {
163 		num_state = order;
164 		state = new double[order];	// need order states
165 		resetState();
166 	}
167 	delay = (order + 1) >> 1;	// assuming symmetric FIR
168 	down_offset = 0;
169 	up_offset = 0;
170 }
171 
172 // down-to-1 decimation
173 int ResampleFilter::
174 decimate_noadjust(short	*in,
175 		int	size,
176 		short	*out)
177 {
178 	int	i;
179 
180 	if (size <= 0)
181 		return (0);
182 	else if (down <= 1)		// normal filter
183 		return (Fir::filter_noadjust(in, size, out));
184 	else if (size <= down_offset) {	// just update states
185 		update_short(in, size);
186 		down_offset -= size;
187 		return (0);
188 	}
189 
190 	double *in_buf = new double[size];
191 	short2double(in_buf, in, size);
192 
193 	// filter and decimate and output
194 	short   *out_ptr = out;
195 	int init_size = (size <= order)? size : order;
196 	for (i = down_offset; i < init_size; i += down)
197 		*out_ptr++ = double2short(convolve(coef, in_buf, i + 1) +
198 		    convolve(coef + i + 1, state + i, order - i));
199 	for (; i < size; i += down)
200 		*out_ptr++ = double2short(convolve(coef, in_buf + i - order,
201 		    order + 1));
202 	down_offset = i - size;
203 
204 	updateState(in_buf, size);
205 	delete in_buf;
206 	return (out_ptr - out);
207 }
208 
209 // decimate with group delay adjusted to 0
210 int ResampleFilter::
211 decimate(short	*in,
212 	int	size,
213 	short	*out)
214 {
215 	if (delay <= 0)
216 		return (decimate_noadjust(in, size, out));
217 	else if (size <= delay) {
218 		update_short(in, size);
219 		delay -= size;
220 		return (0);
221 	} else {
222 		update_short(in, delay);
223 		in += delay;
224 		size -= delay;
225 		delay = 0;
226 		return (decimate_noadjust(in, size, out));
227 	}
228 }
229 
230 // flush decimate filter
231 int ResampleFilter::
232 decimate_flush(short	*out)
233 {
234 	int num_in = Fir::getFlushSize();
235 	short *in = new short[num_in];
236 	memset(in, 0, num_in * sizeof (short));
237 	int num_out = decimate_noadjust(in, num_in, out);
238 	delay += num_in;
239 	delete in;
240 	return (num_out);
241 }
242 
243 // 1-to-up interpolation
244 int ResampleFilter::
245 interpolate_noadjust(short	*in,
246 		    int		size,
247 		    short	*out)
248 {
249 	int	i, j;
250 
251 	if (size <= 0)
252 		return (0);
253 	else if (up <= 1)			// regular filter
254 		return (Fir::filter_noadjust(in, size, out));
255 
256 	double *in_buf = new double[size];
257 	short2double(in_buf, in, size);
258 	short *out_ptr = out;
259 	// befor the 1st input sample, generate  "-up_offset" output samples
260 	int coef_offset = up + up_offset;
261 	for (j = up_offset; j < 0; j++) {
262 		*out_ptr++ = double2short(up * poly_conv(coef + coef_offset,
263 		    order - coef_offset, up, state, num_state));
264 		coef_offset++;
265 	}
266 	// for each of the rest input samples, generate up output samples
267 	for (i = 1; i < size; i++) {
268 		for (j = 0; j < up; j++) {
269 			*out_ptr++ = double2short(up * (poly_conv(coef + j,
270 			    order - j, up, in_buf, i) + poly_conv(
271 			    coef + coef_offset, order - coef_offset, up, state,
272 			    num_state)));
273 			coef_offset++;
274 		}
275 	}
276 
277 	// for the last input samples, generate "up_offset + up" output samples
278 	for (j = 0; j < (up_offset + up); j++) {
279 		*out_ptr++ = double2short(up * (poly_conv(coef + j,
280 		    order - j, up, in_buf, size) + poly_conv(
281 		    coef + coef_offset, order - coef_offset, up, state,
282 		    num_state)));
283 		coef_offset++;
284 	}
285 	updateState(in_buf, size);
286 	delete in_buf;
287 	return (out_ptr - out);
288 }
289 
290 // flush interpolate filter
291 int ResampleFilter::
292 interpolate_flush(short	*out)
293 {
294 	int num = (Fir::getFlushSize() + up - 1) / up;
295 
296 	short *in = new short[num];
297 	memset(in, 0, num * sizeof (short));
298 	int out_num = interpolate_noadjust(in, num, out);
299 	delay += num * up;
300 	delete in;
301 	return (out_num);
302 }
303 
304 // interpolate with delay adjusted
305 int ResampleFilter::
306 interpolate(short *in,
307 	    int size,
308 	    short *out)
309 {
310 	if (size <= 0)
311 		return (interpolate_flush(out));
312 	else if (delay <= 0)
313 		return (interpolate_noadjust(in, size, out));
314 	else {
315 		int delay_in = (delay + up - 1) / up;
316 		if (size < delay_in)
317 			delay_in = size;
318 		double *in_buf = new double[delay_in];
319 		short2double(in_buf, in, delay_in);
320 		updateState(in_buf, delay_in);
321 		delete in_buf;
322 		delay -= delay_in * up;
323 		up_offset = delay;
324 		return (interpolate_noadjust(in + delay_in, size -
325 		    delay_in, out));
326 	}
327 }
328 
329 int ResampleFilter::
330 filter_noadjust(short	*in,		// non-integer sampling rate conversion
331 	int	size,
332 	short	*out)
333 {
334 	int	i, j;
335 
336 	if (size <= 0)
337 		return (0);
338 	else if (up <= 1)
339 		return (decimate_noadjust(in, size, out));
340 	else if (down <= 1)
341 		return (interpolate_noadjust(in, size, out));
342 
343 	double *in_buf = new double[size];
344 	short2double(in_buf, in, size);
345 	short *init_out = out;
346 	int coef_offset = up_offset + down_offset + up;
347 
348 	/*
349 	 * before the 1st input sample,
350 	 * start from "up_offset + down_offset"th up-sampled sample
351 	 */
352 	for (j = up_offset + down_offset; j < 0; j += down) {
353 		*out++ = double2short(up * poly_conv(coef + coef_offset,
354 		    order - coef_offset, up, state, num_state));
355 		coef_offset += down;
356 	}
357 
358 	// process the input samples until the last one
359 	for (i = 1; i < size; i++) {
360 		for (; j < up; j += down) {
361 			*out++ = double2short(up * (poly_conv(coef + j,
362 			    order - j, up, in_buf, i) + poly_conv(
363 			    coef + coef_offset, order - coef_offset, up, state,
364 			    num_state)));
365 			coef_offset += down;
366 		}
367 		j -= up;
368 	}
369 
370 	// for the last input sample, end at the "up + up_offset"th
371 	for (; j < (up + up_offset); j += down) {
372 		*out++ = double2short(up * (poly_conv(coef + j, order - j, up,
373 		    in_buf, size) + poly_conv(coef + coef_offset,
374 		    order - coef_offset, up, state, num_state)));
375 		coef_offset += down;
376 	}
377 	down_offset = j - (up + up_offset);
378 
379 	updateState(in_buf, size);
380 	delete in_buf;
381 	return (out - init_out);
382 }
383 
384 int ResampleFilter::
385 getFlushSize(void)
386 {
387 	int num_in = (Fir::getFlushSize() + up - 1) / up;
388 	return ((num_in * up + down - 1 - down_offset) / down);
389 }
390 
391 int ResampleFilter::
392 flush(short	*out)		// flush resampling filter
393 
394 {
395 	if (down <= 1)
396 		return (interpolate_flush(out));
397 	else if (up <= 1)
398 		return (decimate_flush(out));
399 
400 	int num = (Fir::getFlushSize() + up - 1) / up;
401 
402 	short *in = new short[num];
403 	memset(in, 0, num * sizeof (short));
404 	int out_num = filter_noadjust(in, num, out);
405 	delete in;
406 	delay += num * up;
407 	return (out_num);
408 }
409 
410 /*
411  * sampling rate conversion with filter delay adjusted
412  */
413 int ResampleFilter::
414 filter(
415 	short	*in,
416 	int	size,
417 	short	*out)
418 {
419 	if (size <= 0)
420 		return (flush(out));
421 	else if (up <= 1)
422 		return (decimate(in, size, out));
423 	else if (down <= 1)
424 		return (interpolate(in, size, out));
425 	else if (delay <= 0)
426 		return (filter_noadjust(in, size, out));
427 	else {
428 		int delay_in = (delay + up - 1) / up;
429 		if (size < delay_in)
430 			delay_in = size;
431 		double *in_buf = new double[delay_in];
432 		short2double(in_buf, in, delay_in);
433 		updateState(in_buf, delay_in);
434 		delete in_buf;
435 		delay -= up * delay_in;
436 		if (delay <= 0) {
437 			up_offset = delay;
438 			down_offset = 0;
439 		}
440 		return (filter_noadjust(in + delay_in, size - delay_in, out));
441 	}
442 }
443