/[rdesktop]/sourceforge.net/trunk/rdesktop/rdpsnd_dsp.c
This is repository of my old source code which isn't updated any more. Go to git.rot13.org for current projects!
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Contents of /sourceforge.net/trunk/rdesktop/rdpsnd_dsp.c

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Revision 1292 - (show annotations)
Mon Oct 2 08:01:24 2006 UTC (17 years, 7 months ago) by stargo
File MIME type: text/plain
File size: 10388 byte(s)
add (disabled) real linear resampler. quality is currently not as good
as previous simpler implementation, but will be improved

1 /*
2 rdesktop: A Remote Desktop Protocol client.
3 Sound DSP routines
4 Copyright (C) Michael Gernoth 2006
5
6 This program is free software; you can redistribute it and/or modify
7 it under the terms of the GNU General Public License as published by
8 the Free Software Foundation; either version 2 of the License, or
9 (at your option) any later version.
10
11 This program is distributed in the hope that it will be useful,
12 but WITHOUT ANY WARRANTY; without even the implied warranty of
13 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
14 GNU General Public License for more details.
15
16 You should have received a copy of the GNU General Public License
17 along with this program; if not, write to the Free Software
18 Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
19 */
20
21 #include <strings.h>
22
23 #include "rdesktop.h"
24 #include "rdpsnd.h"
25 #include "rdpsnd_dsp.h"
26
27 #ifdef HAVE_LIBSAMPLERATE
28 #include <samplerate.h>
29
30 #define SRC_CONVERTER SRC_SINC_MEDIUM_QUALITY
31 #endif
32
33 #define MAX_VOLUME 65535
34
35 static uint16 softvol_left = MAX_VOLUME;
36 static uint16 softvol_right = MAX_VOLUME;
37 static uint32 resample_to_srate = 44100;
38 static uint16 resample_to_bitspersample = 16;
39 static uint16 resample_to_channels = 2;
40 #ifdef HAVE_LIBSAMPLERATE
41 static SRC_STATE *src_converter = NULL;
42 #endif
43
44 void
45 rdpsnd_dsp_softvol_set(uint16 left, uint16 right)
46 {
47 softvol_left = left;
48 softvol_right = right;
49 DEBUG(("rdpsnd_dsp_softvol_set: left: %u, right: %u\n", left, right));
50 }
51
52 void
53 rdpsnd_dsp_softvol(unsigned char *buffer, unsigned int size, WAVEFORMATEX * format)
54 {
55 unsigned int factor_left, factor_right;
56 unsigned char *posin = buffer;
57 unsigned char *posout = buffer;
58
59 if ((softvol_left == MAX_VOLUME) && (softvol_right == MAX_VOLUME))
60 return;
61
62 factor_left = (softvol_left * 256) / MAX_VOLUME;
63 factor_right = (softvol_right * 256) / MAX_VOLUME;
64
65 if (format->nChannels == 1)
66 {
67 factor_left = factor_right = (factor_left + factor_right) / 2;
68 }
69
70 if (format->wBitsPerSample == 8)
71 {
72 sint8 val;
73
74 while (posout < buffer + size)
75 {
76 /* Left */
77 val = *posin++;
78 val = (val * factor_left) >> 8;
79 *posout++ = val;
80
81 /* Right */
82 val = *posin++;
83 val = (val * factor_right) >> 8;
84 *posout++ = val;
85 }
86 }
87 else
88 {
89 sint16 val;
90
91 while (posout < buffer + size)
92 {
93 /* Left */
94 val = *posin++;
95 val |= *posin++ << 8;
96 val = (val * factor_left) >> 8;
97 *posout++ = val & 0xff;
98 *posout++ = val >> 8;
99
100 /* Right */
101 val = *posin++;
102 val |= *posin++ << 8;
103 val = (val * factor_right) >> 8;
104 *posout++ = val & 0xff;
105 *posout++ = val >> 8;
106 }
107 }
108
109 DEBUG(("using softvol with factors left: %d, right: %d (%d/%d)\n", factor_left,
110 factor_right, format->wBitsPerSample, format->nChannels));
111 }
112
113 void
114 rdpsnd_dsp_swapbytes(unsigned char *buffer, unsigned int size, WAVEFORMATEX * format)
115 {
116 int i;
117 uint8 swap;
118
119 if (format->wBitsPerSample == 8)
120 return;
121
122 for (i = 0; i < size; i += 2)
123 {
124 swap = *(buffer + i);
125 *(buffer + i) = *(buffer + i + 1);
126 *(buffer + i + 1) = swap;
127 }
128 }
129
130 BOOL
131 rdpsnd_dsp_resample_set(uint32 device_srate, uint16 device_bitspersample, uint16 device_channels)
132 {
133 #ifdef HAVE_LIBSAMPLERATE
134 int err;
135 #endif
136
137 if (device_bitspersample != 16 && device_bitspersample != 8)
138 return False;
139
140 if (device_channels != 1 && device_channels != 2)
141 return False;
142
143 resample_to_srate = device_srate;
144 resample_to_bitspersample = device_bitspersample;
145 resample_to_channels = device_channels;
146
147 #ifdef HAVE_LIBSAMPLERATE
148 if (src_converter != NULL)
149 src_converter = src_delete(src_converter);
150
151 if ((src_converter = src_new(SRC_CONVERTER, device_channels, &err)) == NULL)
152 {
153 warning("src_new failed: %d!\n", err);
154 return False;
155 }
156 #endif
157
158 return True;
159 }
160
161 BOOL
162 rdpsnd_dsp_resample_supported(WAVEFORMATEX * format)
163 {
164 if (format->wFormatTag != WAVE_FORMAT_PCM)
165 return False;
166 if ((format->nChannels != 1) && (format->nChannels != 2))
167 return False;
168 if ((format->wBitsPerSample != 8) && (format->wBitsPerSample != 16))
169 return False;
170
171 return True;
172 }
173
174 uint32
175 rdpsnd_dsp_resample(unsigned char **out, unsigned char *in, unsigned int size,
176 WAVEFORMATEX * format, BOOL stream_be)
177 {
178 #ifdef HAVE_LIBSAMPLERATE
179 SRC_DATA resample_data;
180 float *infloat, *outfloat;
181 int err;
182 #else
183 int ratio1k = (resample_to_srate * 1000) / format->nSamplesPerSec;
184 #endif
185 int innum, outnum;
186 unsigned char *tmpdata = NULL, *tmp = NULL;
187 int samplewidth = format->wBitsPerSample / 8;
188 int outsize = 0;
189 int i;
190
191 if ((resample_to_bitspersample == format->wBitsPerSample) &&
192 (resample_to_channels == format->nChannels) &&
193 (resample_to_srate == format->nSamplesPerSec))
194 return 0;
195
196 #ifdef B_ENDIAN
197 if (!stream_be)
198 rdpsnd_dsp_swapbytes(in, size, format);
199 #endif
200
201 if (resample_to_channels != format->nChannels)
202 {
203 int newsize = (size / format->nChannels) * resample_to_channels;
204 tmpdata = xmalloc(newsize);
205
206 for (i = 0; i < newsize / samplewidth; i++)
207 {
208 if (format->nChannels > resample_to_channels)
209 memcpy(tmpdata + (i * samplewidth),
210 in +
211 (((i * format->nChannels) / resample_to_channels) *
212 samplewidth), samplewidth);
213 else
214 memcpy(tmpdata + (i * samplewidth),
215 in +
216 (((i / resample_to_channels) * format->nChannels +
217 (i % format->nChannels)) * samplewidth), samplewidth);
218
219 }
220
221 in = tmpdata;
222 size = newsize;
223 }
224
225
226 /* Expand 8bit input-samples to 16bit */
227 #ifndef HAVE_LIBSAMPLERATE /* libsamplerate needs 16bit samples */
228 if (format->wBitsPerSample != resample_to_bitspersample)
229 #endif
230 {
231 /* source: 8 bit, dest: 16bit */
232 if (format->wBitsPerSample == 8)
233 {
234 tmp = tmpdata;
235 tmpdata = xmalloc(size * 2);
236 for (i = 0; i < size; i++)
237 {
238 tmpdata[i * 2] = in[i];
239 tmpdata[(i * 2) + 1] = 0x00;
240 }
241 in = tmpdata;
242 samplewidth = 16 / 2;
243 size *= 2;
244
245 if (tmp != NULL)
246 xfree(tmp);
247 }
248 }
249
250 innum = size / samplewidth;
251
252 /* Do the resampling */
253 #ifdef HAVE_LIBSAMPLERATE
254 if (src_converter == NULL)
255 {
256 warning("no samplerate converter available!!\n");
257 return 0;
258 }
259
260 outnum = ((float) innum * ((float) resample_to_srate / (float) format->nSamplesPerSec)) + 1;
261
262 infloat = xmalloc(sizeof(float) * innum);
263 outfloat = xmalloc(sizeof(float) * outnum);
264
265 src_short_to_float_array((short *) in, infloat, innum);
266
267 bzero(&resample_data, sizeof(resample_data));
268 resample_data.data_in = infloat;
269 resample_data.data_out = outfloat;
270 resample_data.input_frames = innum / resample_to_channels;
271 resample_data.output_frames = outnum / resample_to_channels;
272 resample_data.src_ratio = (double) resample_to_srate / (double) format->nSamplesPerSec;
273 resample_data.end_of_input = 0;
274
275 if ((err = src_process(src_converter, &resample_data)) != 0)
276 error("src_process: %s", src_strerror(err));
277
278 xfree(infloat);
279
280 outsize = resample_data.output_frames_gen * resample_to_channels * samplewidth;
281 *out = xmalloc(outsize);
282 src_float_to_short_array(outfloat, (short *) *out,
283 resample_data.output_frames_gen * resample_to_channels);
284 xfree(outfloat);
285
286 #else
287 /* Michaels simple linear resampler */
288 if (resample_to_srate < format->nSamplesPerSec)
289 {
290 warning("downsampling currently not supported!\n");
291 return 0;
292 }
293
294 outnum = (innum * ratio1k) / 1000;
295
296 outsize = outnum * samplewidth;
297 *out = xmalloc(outsize);
298 bzero(*out, outsize);
299
300 for (i = 0; i < outsize / (resample_to_channels * samplewidth); i++)
301 {
302 int source = (i * 1000) / ratio1k;
303 #if 0 /* Partial for linear resampler */
304 int part = (i * 100000) / ratio1k - source * 100;
305 #endif
306 int j;
307
308 if (source * resample_to_channels + samplewidth > size)
309 break;
310
311 #if 0 /* Linear resampling, TODO: soundquality fixes */
312 if (samplewidth == 1)
313 {
314 sint8 cval1, cval2;
315 for (j = 0; j < resample_to_channels; j++)
316 {
317 memcpy(&cval1,
318 in + (source * resample_to_channels * samplewidth) +
319 (samplewidth * j), samplewidth);
320 memcpy(&cval2,
321 in + ((source + 1) * resample_to_channels * samplewidth) +
322 (samplewidth * j), samplewidth);
323
324 cval1 += (cval2 * part) / 100;
325
326 memcpy(*out + (i * resample_to_channels * samplewidth) +
327 (samplewidth * j), &cval1, samplewidth);
328 }
329 }
330 else
331 {
332 sint16 sval1, sval2;
333 for (j = 0; j < resample_to_channels; j++)
334 {
335 memcpy(&sval1,
336 in + (source * resample_to_channels * samplewidth) +
337 (samplewidth * j), samplewidth);
338 memcpy(&sval2,
339 in + ((source + 1) * resample_to_channels * samplewidth) +
340 (samplewidth * j), samplewidth);
341
342 sval1 += (sval2 * part) / 100;
343
344 memcpy(*out + (i * resample_to_channels * samplewidth) +
345 (samplewidth * j), &sval1, samplewidth);
346 }
347 }
348 #else /* Nearest neighbor search */
349 for (j = 0; j < resample_to_channels; j++)
350 {
351 memcpy(*out + (i * resample_to_channels * samplewidth) + (samplewidth * j),
352 in + (source * resample_to_channels * samplewidth) +
353 (samplewidth * j), samplewidth);
354 }
355 #endif
356 }
357 outsize = i * resample_to_channels * samplewidth;
358 #endif
359
360 if (tmpdata != NULL)
361 xfree(tmpdata);
362
363 /* Shrink 16bit output-samples to 8bit */
364 #ifndef HAVE_LIBSAMPLERATE /* libsamplerate produces 16bit samples */
365 if (format->wBitsPerSample != resample_to_bitspersample)
366 #endif
367 {
368 /* source: 16 bit, dest: 8 bit */
369 if (resample_to_bitspersample == 8)
370 {
371 for (i = 0; i < outsize; i++)
372 {
373 *out[i] = *out[i * 2];
374 }
375 outsize /= 2;
376 }
377 }
378
379 #ifdef B_ENDIAN
380 if (!stream_be)
381 rdpsnd_dsp_swapbytes(*out, outsize, format);
382 #endif
383 return outsize;
384 }
385
386 STREAM
387 rdpsnd_dsp_process(STREAM s, struct audio_driver * current_driver, WAVEFORMATEX * format)
388 {
389 static struct stream out;
390 BOOL stream_be = False;
391
392 /* softvol and byteswap do not change the amount of data they
393 return, so they can operate on the input-stream */
394 if (current_driver->wave_out_volume == rdpsnd_dsp_softvol_set)
395 rdpsnd_dsp_softvol(s->data, s->size, format);
396
397 #ifdef B_ENDIAN
398 if (current_driver->need_byteswap_on_be)
399 {
400 rdpsnd_dsp_swapbytes(s->data, s->size, format);
401 stream_be = True;
402 }
403 #endif
404
405 out.data = NULL;
406
407 if (current_driver->need_resampling)
408 out.size = rdpsnd_dsp_resample(&out.data, s->data, s->size, format, stream_be);
409
410 if (out.data == NULL)
411 {
412 out.data = xmalloc(s->size);
413 memcpy(out.data, s->data, s->size);
414 out.size = s->size;
415 }
416
417 out.p = out.data;
418 out.end = out.p + out.size;
419
420 return &out;
421 }

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