aboutsummaryrefslogtreecommitdiffstats
path: root/src/pcm_utils.c
blob: a68542f4a830e185958682d1f17dd7e301d537c5 (plain) (blame)
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
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
/* the Music Player Daemon (MPD)
 * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
 * This project's homepage is: http://www.musicpd.org
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * This program is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 * You should have received a copy of the GNU General Public License
 * along with this program; if not, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */

#include "pcm_utils.h"
#include "pcm_channels.h"
#include "log.h"
#include "utils.h"
#include "conf.h"
#include "audio_format.h"

#include <assert.h>
#include <string.h>
#include <math.h>

static inline int
pcm_dither(void)
{
	return (rand() & 511) - (rand() & 511);
}

/**
 * Check if the value is within the range of the provided bit size,
 * and caps it if necessary.
 */
static int32_t
pcm_range(int32_t sample, unsigned bits)
{
	if (mpd_unlikely(sample < (-1 << (bits - 1))))
		return -1 << (bits - 1);
	if (mpd_unlikely(sample >= (1 << (bits - 1))))
		return (1 << (bits - 1)) - 1;
	return sample;
}

static void
pcm_volume_change_8(int8_t *buffer, unsigned num_samples, int volume)
{
	while (num_samples > 0) {
		int32_t sample = *buffer;

		sample = (sample * volume + pcm_dither() + PCM_VOLUME_1 / 2)
			/ PCM_VOLUME_1;

		*buffer++ = pcm_range(sample, 8);
		--num_samples;
	}
}

static void
pcm_volume_change_16(int16_t *buffer, unsigned num_samples, int volume)
{
	while (num_samples > 0) {
		int32_t sample = *buffer;

		sample = (sample * volume + pcm_dither() + PCM_VOLUME_1 / 2)
			/ PCM_VOLUME_1;

		*buffer++ = pcm_range(sample, 16);
		--num_samples;
	}
}

static void
pcm_volume_change_24(int32_t *buffer, unsigned num_samples, int volume)
{
	while (num_samples > 0) {
		int64_t sample = *buffer;

		sample = (sample * volume + pcm_dither() + PCM_VOLUME_1 / 2)
			/ PCM_VOLUME_1;

		*buffer++ = pcm_range(sample, 24);
		--num_samples;
	}
}

void pcm_volume(char *buffer, int bufferSize,
		const struct audio_format *format,
		int volume)
{
	if (volume >= PCM_VOLUME_1)
		return;

	if (volume <= 0) {
		memset(buffer, 0, bufferSize);
		return;
	}

	switch (format->bits) {
	case 8:
		pcm_volume_change_8((int8_t *)buffer, bufferSize, volume);
		break;

	case 16:
		pcm_volume_change_16((int16_t *)buffer, bufferSize / 2,
				     volume);
		break;

	case 24:
		pcm_volume_change_24((int32_t*)buffer, bufferSize / 4,
				     volume);
		break;

	default:
		FATAL("%u bits not supported by pcm_volume!\n",
		      format->bits);
	}
}

static void
pcm_add_8(int8_t *buffer1, const int8_t *buffer2,
	  unsigned num_samples, int volume1, int volume2)
{
	while (num_samples > 0) {
		int32_t sample1 = *buffer1;
		int32_t sample2 = *buffer2++;

		sample1 = ((sample1 * volume1 + sample2 * volume2) +
			   pcm_dither() + PCM_VOLUME_1 / 2) / PCM_VOLUME_1;

		*buffer1++ = pcm_range(sample1, 8);
		--num_samples;
	}
}

static void
pcm_add_16(int16_t *buffer1, const int16_t *buffer2,
	   unsigned num_samples, int volume1, int volume2)
{
	while (num_samples > 0) {
		int32_t sample1 = *buffer1;
		int32_t sample2 = *buffer2++;

		sample1 = ((sample1 * volume1 + sample2 * volume2) +
			   pcm_dither() + PCM_VOLUME_1 / 2) / PCM_VOLUME_1;

		*buffer1++ = pcm_range(sample1, 16);
		--num_samples;
	}
}

static void
pcm_add_24(int32_t *buffer1, const int32_t *buffer2,
	   unsigned num_samples, unsigned volume1, unsigned volume2)
{
	while (num_samples > 0) {
		int64_t sample1 = *buffer1;
		int64_t sample2 = *buffer2++;

		sample1 = ((sample1 * volume1 + sample2 * volume2) +
			   pcm_dither() + PCM_VOLUME_1 / 2) / PCM_VOLUME_1;

		*buffer1++ = pcm_range(sample1, 24);
		--num_samples;
	}
}

static void pcm_add(char *buffer1, const char *buffer2, size_t size,
                    int vol1, int vol2,
                    const struct audio_format *format)
{
	switch (format->bits) {
	case 8:
		pcm_add_8((int8_t *)buffer1, (const int8_t *)buffer2,
			  size, vol1, vol2);
		break;

	case 16:
		pcm_add_16((int16_t *)buffer1, (const int16_t *)buffer2,
			   size / 2, vol1, vol2);
		break;

	case 24:
		pcm_add_24((int32_t*)buffer1,
			   (const int32_t*)buffer2,
			   size / 4, vol1, vol2);
		break;

	default:
		FATAL("%u bits not supported by pcm_add!\n", format->bits);
	}
}

void pcm_mix(char *buffer1, const char *buffer2, size_t size,
             const struct audio_format *format, float portion1)
{
	int vol1;
	float s = sin(M_PI_2 * portion1);
	s *= s;

	vol1 = s * PCM_VOLUME_1 + 0.5;
	vol1 = vol1 > PCM_VOLUME_1 ? PCM_VOLUME_1 : (vol1 < 0 ? 0 : vol1);

	pcm_add(buffer1, buffer2, size, vol1, PCM_VOLUME_1 - vol1, format);
}

void pcm_convert_init(struct pcm_convert_state *state)
{
	memset(state, 0, sizeof(*state));

	pcm_resample_init(&state->resample);
	pcm_dither_24_init(&state->dither);
}

static void
pcm_convert_8_to_16(int16_t *out, const int8_t *in,
		    unsigned num_samples)
{
	while (num_samples > 0) {
		*out++ = *in++ << 8;
		--num_samples;
	}
}

static void
pcm_convert_24_to_16(struct pcm_dither_24 *dither,
		     int16_t *out, const int32_t *in,
		     unsigned num_samples)
{
	pcm_dither_24_to_16(dither, out, in, num_samples);
}

static const int16_t *
pcm_convert_to_16(struct pcm_convert_state *convert,
		  uint8_t bits, const void *src,
		  size_t src_size, size_t *dest_size_r)
{
	static int16_t *buf;
	static size_t len;
	unsigned num_samples;

	switch (bits) {
	case 8:
		num_samples = src_size;
		*dest_size_r = src_size << 1;
		if (*dest_size_r > len) {
			len = *dest_size_r;
			buf = xrealloc(buf, len);
		}

		pcm_convert_8_to_16((int16_t *)buf,
				    (const int8_t *)src,
				    num_samples);
		return buf;

	case 16:
		*dest_size_r = src_size;
		return src;

	case 24:
		num_samples = src_size / 4;
		*dest_size_r = num_samples * 2;
		if (*dest_size_r > len) {
			len = *dest_size_r;
			buf = xrealloc(buf, len);
		}

		pcm_convert_24_to_16(&convert->dither,
				     (int16_t *)buf,
				     (const int32_t *)src,
				     num_samples);
		return buf;
	}

	ERROR("only 8 or 16 bits are supported for conversion!\n");
	return NULL;
}

static void
pcm_convert_8_to_24(int32_t *out, const int8_t *in,
		    unsigned num_samples)
{
	while (num_samples > 0) {
		*out++ = *in++ << 16;
		--num_samples;
	}
}

static void
pcm_convert_16_to_24(int32_t *out, const int16_t *in,
		     unsigned num_samples)
{
	while (num_samples > 0) {
		*out++ = *in++ << 8;
		--num_samples;
	}
}

static const int32_t *
pcm_convert_to_24(uint8_t bits, const void *src,
		  size_t src_size, size_t *dest_size_r)
{
	static int32_t *buf;
	static size_t len;
	unsigned num_samples;

	switch (bits) {
	case 8:
		num_samples = src_size;
		*dest_size_r = src_size * 4;
		if (*dest_size_r > len) {
			len = *dest_size_r;
			buf = xrealloc(buf, len);
		}

		pcm_convert_8_to_24(buf, (const int8_t *)src,
				    num_samples);
		return buf;

	case 16:
		num_samples = src_size / 2;
		*dest_size_r = num_samples * 4;
		if (*dest_size_r > len) {
			len = *dest_size_r;
			buf = xrealloc(buf, len);
		}

		pcm_convert_16_to_24(buf, (const int16_t *)src,
				     num_samples);
		return buf;

	case 24:
		*dest_size_r = src_size;
		return src;
	}

	ERROR("only 8 or 24 bits are supported for conversion!\n");
	return NULL;
}

static size_t
pcm_convert_16(const struct audio_format *src_format,
	       const void *src_buffer, size_t src_size,
	       const struct audio_format *dest_format,
	       int16_t *dest_buffer,
	       struct pcm_convert_state *state)
{
	const int16_t *buf;
	size_t len;
	size_t dest_size = pcm_convert_size(src_format, src_size, dest_format);

	assert(dest_format->bits == 16);

	buf = pcm_convert_to_16(state, src_format->bits,
				src_buffer, src_size, &len);
	if (!buf)
		exit(EXIT_FAILURE);

	if (src_format->channels != dest_format->channels) {
		buf = pcm_convert_channels_16(dest_format->channels,
					      src_format->channels,
					      buf, len, &len);
		if (!buf)
			exit(EXIT_FAILURE);
	}

	if (src_format->sample_rate == dest_format->sample_rate) {
		assert(dest_size >= len);
		memcpy(dest_buffer, buf, len);
	} else {
		len = pcm_resample_16(dest_format->channels,
				      src_format->sample_rate, buf, len,
				      dest_format->sample_rate,
				      dest_buffer, dest_size,
				      &state->resample);
		if (len == 0)
			exit(EXIT_FAILURE);
	}

	return len;
}

static size_t
pcm_convert_24(const struct audio_format *src_format,
	       const void *src_buffer, size_t src_size,
	       const struct audio_format *dest_format,
	       int32_t *dest_buffer,
	       struct pcm_convert_state *state)
{
	const int32_t *buf;
	size_t len;
	size_t dest_size = pcm_convert_size(src_format, src_size, dest_format);

	assert(dest_format->bits == 24);

	buf = pcm_convert_to_24(src_format->bits,
				src_buffer, src_size, &len);
	if (!buf)
		exit(EXIT_FAILURE);

	if (src_format->channels != dest_format->channels) {
		buf = pcm_convert_channels_24(dest_format->channels,
					      src_format->channels,
					      buf, len, &len);
		if (!buf)
			exit(EXIT_FAILURE);
	}

	if (src_format->sample_rate == dest_format->sample_rate) {
		assert(dest_size >= len);
		memcpy(dest_buffer, buf, len);
	} else {
		len = pcm_resample_24(dest_format->channels,
				      src_format->sample_rate, buf, len,
				      dest_format->sample_rate,
				      (int32_t*)dest_buffer, dest_size,
				      &state->resample);
		if (len == 0)
			exit(EXIT_FAILURE);
	}

	return len;
}

/* outFormat bits must be 16 and channels must be 1 or 2! */
size_t pcm_convert(const struct audio_format *inFormat,
		   const char *src, size_t src_size,
		   const struct audio_format *outFormat,
		   char *outBuffer,
		   struct pcm_convert_state *convState)
{
	switch (outFormat->bits) {
	case 16:
		return pcm_convert_16(inFormat, src, src_size,
				      outFormat, (int16_t*)outBuffer,
				      convState);
	case 24:
		return pcm_convert_24(inFormat, src, src_size,
				      outFormat, (int32_t*)outBuffer,
				      convState);

	default:
		FATAL("cannot convert to %u bit\n", outFormat->bits);
	}
}

size_t pcm_convert_size(const struct audio_format *inFormat, size_t src_size,
			const struct audio_format *outFormat)
{
	const double ratio = (double)outFormat->sample_rate /
	                     (double)inFormat->sample_rate;
	size_t dest_size = src_size;

	/* no partial frames allowed */
	assert((src_size % audio_format_frame_size(inFormat)) == 0);

	dest_size /= audio_format_frame_size(inFormat);
	dest_size = floor(0.5 + (double)dest_size * ratio);
	dest_size *= audio_format_frame_size(outFormat);

	return dest_size;
}