summaryrefslogtreecommitdiffstats
path: root/drivers/isdn/mISDN/dsp_audio.c
blob: bbef98e7a16efb438ef38856d8e043397f674804 (plain)
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
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
/*
 * Audio support data for mISDN_dsp.
 *
 * Copyright 2002/2003 by Andreas Eversberg (jolly@eversberg.eu)
 * Rewritten by Peter
 *
 * This software may be used and distributed according to the terms
 * of the GNU General Public License, incorporated herein by reference.
 *
 */

#include <linux/delay.h>
#include <linux/mISDNif.h>
#include <linux/mISDNdsp.h>
#include <linux/export.h>
#include <linux/bitrev.h>
#include "core.h"
#include "dsp.h"

/* ulaw[unsigned char] -> signed 16-bit */
s32 dsp_audio_ulaw_to_s32[256];
/* alaw[unsigned char] -> signed 16-bit */
s32 dsp_audio_alaw_to_s32[256];

s32 *dsp_audio_law_to_s32;
EXPORT_SYMBOL(dsp_audio_law_to_s32);

/* signed 16-bit -> law */
u8 dsp_audio_s16_to_law[65536];
EXPORT_SYMBOL(dsp_audio_s16_to_law);

/* alaw -> ulaw */
u8 dsp_audio_alaw_to_ulaw[256];
/* ulaw -> alaw */
static u8 dsp_audio_ulaw_to_alaw[256];
u8 dsp_silence;


/*****************************************************
 * generate table for conversion of s16 to alaw/ulaw *
 *****************************************************/

#define AMI_MASK 0x55

static inline unsigned char linear2alaw(short int linear)
{
	int mask;
	int seg;
	int pcm_val;
	static int seg_end[8] = {
		0xFF, 0x1FF, 0x3FF, 0x7FF, 0xFFF, 0x1FFF, 0x3FFF, 0x7FFF
	};

	pcm_val = linear;
	if (pcm_val >= 0) {
		/* Sign (7th) bit = 1 */
		mask = AMI_MASK | 0x80;
	} else {
		/* Sign bit = 0 */
		mask = AMI_MASK;
		pcm_val = -pcm_val;
	}

	/* Convert the scaled magnitude to segment number. */
	for (seg = 0; seg < 8; seg++) {
		if (pcm_val <= seg_end[seg])
			break;
	}
	/* Combine the sign, segment, and quantization bits. */
	return  ((seg << 4) |
		 ((pcm_val >> ((seg)  ?  (seg + 3)  :  4)) & 0x0F)) ^ mask;
}


static inline short int alaw2linear(unsigned char alaw)
{
	int i;
	int seg;

	alaw ^= AMI_MASK;
	i = ((alaw & 0x0F) << 4) + 8 /* rounding error */;
	seg = (((int) alaw & 0x70) >> 4);
	if (seg)
		i = (i + 0x100) << (seg - 1);
	return (short int) ((alaw & 0x80)  ?  i  :  -i);
}

static inline short int ulaw2linear(unsigned char ulaw)
{
	short mu, e, f, y;
	static short etab[] = {0, 132, 396, 924, 1980, 4092, 8316, 16764};

	mu = 255 - ulaw;
	e = (mu & 0x70) / 16;
	f = mu & 0x0f;
	y = f * (1 << (e + 3));
	y += etab[e];
	if (mu & 0x80)
		y = -y;
	return y;
}

#define BIAS 0x84   /*!< define the add-in bias for 16 bit samples */

static unsigned char linear2ulaw(short sample)
{
	static int exp_lut[256] = {
		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,
		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,
		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,
		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,
		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,
		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,
		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,
		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};
	int sign, exponent, mantissa;
	unsigned char ulawbyte;

	/* Get the sample into sign-magnitude. */
	sign = (sample >> 8) & 0x80;	  /* set aside the sign */
	if (sign != 0)
		sample = -sample;	      /* get magnitude */

	/* Convert from 16 bit linear to ulaw. */
	sample = sample + BIAS;
	exponent = exp_lut[(sample >> 7) & 0xFF];
	mantissa = (sample >> (exponent + 3)) & 0x0F;
	ulawbyte = ~(sign | (exponent << 4) | mantissa);

	return ulawbyte;
}

void dsp_audio_generate_law_tables(void)
{
	int i;
	for (i = 0; i < 256; i++)
		dsp_audio_alaw_to_s32[i] = alaw2linear(bitrev8((u8)i));

	for (i = 0; i < 256; i++)
		dsp_audio_ulaw_to_s32[i] = ulaw2linear(bitrev8((u8)i));

	for (i = 0; i < 256; i++) {
		dsp_audio_alaw_to_ulaw[i] =
			linear2ulaw(dsp_audio_alaw_to_s32[i]);
		dsp_audio_ulaw_to_alaw[i] =
			linear2alaw(dsp_audio_ulaw_to_s32[i]);
	}
}

void
dsp_audio_generate_s2law_table(void)
{
	int i;

	if (dsp_options & DSP_OPT_ULAW) {
		/* generating ulaw-table */
		for (i = -32768; i < 32768; i++) {
			dsp_audio_s16_to_law[i & 0xffff] =
				bitrev8(linear2ulaw(i));
		}
	} else {
		/* generating alaw-table */
		for (i = -32768; i < 32768; i++) {
			dsp_audio_s16_to_law[i & 0xffff] =
				bitrev8(linear2alaw(i));
		}
	}
}


/*
 * the seven bit sample is the number of every second alaw-sample ordered by
 * aplitude. 0x00 is negative, 0x7f is positive amplitude.
 */
u8 dsp_audio_seven2law[128];
u8 dsp_audio_law2seven[256];

/********************************************************************
 * generate table for conversion law from/to 7-bit alaw-like sample *
 ********************************************************************/

void
dsp_audio_generate_seven(void)
{
	int i, j, k;
	u8 spl;
	u8 sorted_alaw[256];

	/* generate alaw table, sorted by the linear value */
	for (i = 0; i < 256; i++) {
		j = 0;
		for (k = 0; k < 256; k++) {
			if (dsp_audio_alaw_to_s32[k]
			    < dsp_audio_alaw_to_s32[i])
				j++;
		}
		sorted_alaw[j] = i;
	}

	/* generate tabels */
	for (i = 0; i < 256; i++) {
		/* spl is the source: the law-sample (converted to alaw) */
		spl = i;
		if (dsp_options & DSP_OPT_ULAW)
			spl = dsp_audio_ulaw_to_alaw[i];
		/* find the 7-bit-sample */
		for (j = 0; j < 256; j++) {
			if (sorted_alaw[j] == spl)
				break;
		}
		/* write 7-bit audio value */
		dsp_audio_law2seven[i] = j >> 1;
	}
	for (i = 0; i < 128; i++) {
		spl = sorted_alaw[i << 1];
		if (dsp_options & DSP_OPT_ULAW)
			spl = dsp_audio_alaw_to_ulaw[spl];
		dsp_audio_seven2law[i] = spl;
	}
}


/* mix 2*law -> law */
u8 dsp_audio_mix_law[65536];

/******************************************************
 * generate mix table to mix two law samples into one *
 ******************************************************/

void
dsp_audio_generate_mix_table(void)
{
	int i, j;
	s32 sample;

	i = 0;
	while (i < 256) {
		j = 0;
		while (j < 256) {
			sample = dsp_audio_law_to_s32[i];
			sample += dsp_audio_law_to_s32[j];
			if (sample > 32767)
				sample = 32767;
			if (sample < -32768)
				sample = -32768;
			dsp_audio_mix_law[(i << 8) | j] =
				dsp_audio_s16_to_law[sample & 0xffff];
			j++;
		}
		i++;
	}
}


/*************************************
 * generate different volume changes *
 *************************************/

static u8 dsp_audio_reduce8[256];
static u8 dsp_audio_reduce7[256];
static u8 dsp_audio_reduce6[256];
static u8 dsp_audio_reduce5[256];
static u8 dsp_audio_reduce4[256];
static u8 dsp_audio_reduce3[256];
static u8 dsp_audio_reduce2[256];
static u8 dsp_audio_reduce1[256];
static u8 dsp_audio_increase1[256];
static u8 dsp_audio_increase2[256];
static u8 dsp_audio_increase3[256];
static u8 dsp_audio_increase4[256];
static u8 dsp_audio_increase5[256];
static u8 dsp_audio_increase6[256];
static u8 dsp_audio_increase7[256];
static u8 dsp_audio_increase8[256];

static u8 *dsp_audio_volume_change[16] = {
	dsp_audio_reduce8,
	dsp_audio_reduce7,
	dsp_audio_reduce6,
	dsp_audio_reduce5,
	dsp_audio_reduce4,
	dsp_audio_reduce3,
	dsp_audio_reduce2,
	dsp_audio_reduce1,
	dsp_audio_increase1,
	dsp_audio_increase2,
	dsp_audio_increase3,
	dsp_audio_increase4,
	dsp_audio_increase5,
	dsp_audio_increase6,
	dsp_audio_increase7,
	dsp_audio_increase8,
};

void
dsp_audio_generate_volume_changes(void)
{
	register s32 sample;
	int i;
	int num[]   = { 110, 125, 150, 175, 200, 300, 400, 500 };
	int denum[] = { 100, 100, 100, 100, 100, 100, 100, 100 };

	i = 0;
	while (i < 256) {
		dsp_audio_reduce8[i] = dsp_audio_s16_to_law[
			(dsp_audio_law_to_s32[i] * denum[7] / num[7]) & 0xffff];
		dsp_audio_reduce7[i] = dsp_audio_s16_to_law[
			(dsp_audio_law_to_s32[i] * denum[6] / num[6]) & 0xffff];
		dsp_audio_reduce6[i] = dsp_audio_s16_to_law[
			(dsp_audio_law_to_s32[i] * denum[5] / num[5]) & 0xffff];
		dsp_audio_reduce5[i] = dsp_audio_s16_to_law[
			(dsp_audio_law_to_s32[i] * denum[4] / num[4]) & 0xffff];
		dsp_audio_reduce4[i] = dsp_audio_s16_to_law[
			(dsp_audio_law_to_s32[i] * denum[3] / num[3]) & 0xffff];
		dsp_audio_reduce3[i] = dsp_audio_s16_to_law[
			(dsp_audio_law_to_s32[i] * denum[2] / num[2]) & 0xffff];
		dsp_audio_reduce2[i] = dsp_audio_s16_to_law[
			(dsp_audio_law_to_s32[i] * denum[1] / num[1]) & 0xffff];
		dsp_audio_reduce1[i] = dsp_audio_s16_to_law[
			(dsp_audio_law_to_s32[i] * denum[0] / num[0]) & 0xffff];
		sample = dsp_audio_law_to_s32[i] * num[0] / denum[0];
		if (sample < -32768)
			sample = -32768;
		else if (sample > 32767)
			sample = 32767;
		dsp_audio_increase1[i] = dsp_audio_s16_to_law[sample & 0xffff];
		sample = dsp_audio_law_to_s32[i] * num[1] / denum[1];
		if (sample < -32768)
			sample = -32768;
		else if (sample > 32767)
			sample = 32767;
		dsp_audio_increase2[i] = dsp_audio_s16_to_law[sample & 0xffff];
		sample = dsp_audio_law_to_s32[i] * num[2] / denum[2];
		if (sample < -32768)
			sample = -32768;
		else if (sample > 32767)
			sample = 32767;
		dsp_audio_increase3[i] = dsp_audio_s16_to_law[sample & 0xffff];
		sample = dsp_audio_law_to_s32[i] * num[3] / denum[3];
		if (sample < -32768)
			sample = -32768;
		else if (sample > 32767)
			sample = 32767;
		dsp_audio_increase4[i] = dsp_audio_s16_to_law[sample & 0xffff];
		sample = dsp_audio_law_to_s32[i] * num[4] / denum[4];
		if (sample < -32768)
			sample = -32768;
		else if (sample > 32767)
			sample = 32767;
		dsp_audio_increase5[i] = dsp_audio_s16_to_law[sample & 0xffff];
		sample = dsp_audio_law_to_s32[i] * num[5] / denum[5];
		if (sample < -32768)
			sample = -32768;
		else if (sample > 32767)
			sample = 32767;
		dsp_audio_increase6[i] = dsp_audio_s16_to_law[sample & 0xffff];
		sample = dsp_audio_law_to_s32[i] * num[6] / denum[6];
		if (sample < -32768)
			sample = -32768;
		else if (sample > 32767)
			sample = 32767;
		dsp_audio_increase7[i] = dsp_audio_s16_to_law[sample & 0xffff];
		sample = dsp_audio_law_to_s32[i] * num[7] / denum[7];
		if (sample < -32768)
			sample = -32768;
		else if (sample > 32767)
			sample = 32767;
		dsp_audio_increase8[i] = dsp_audio_s16_to_law[sample & 0xffff];

		i++;
	}
}


/**************************************
 * change the volume of the given skb *
 **************************************/

/* this is a helper function for changing volume of skb. the range may be
 * -8 to 8, which is a shift to the power of 2. 0 == no volume, 3 == volume*8
 */
void
dsp_change_volume(struct sk_buff *skb, int volume)
{
	u8 *volume_change;
	int i, ii;
	u8 *p;
	int shift;

	if (volume == 0)
		return;

	/* get correct conversion table */
	if (volume < 0) {
		shift = volume + 8;
		if (shift < 0)
			shift = 0;
	} else {
		shift = volume + 7;
		if (shift > 15)
			shift = 15;
	}
	volume_change = dsp_audio_volume_change[shift];
	i = 0;
	ii = skb->len;
	p = skb->data;
	/* change volume */
	while (i < ii) {
		*p = volume_change[*p];
		p++;
		i++;
	}
}
OpenPOWER on IntegriCloud