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source: vbox/trunk/src/VBox/RDP/client/rdpsnd_dsp.c@ 26437

最後變更 在這個檔案從26437是 11982,由 vboxsync 提交於 16 年 前

All: license header changes for 2.0 (OSE headers, add Sun GPL/LGPL disclaimer)
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1/*
2 rdesktop: A Remote Desktop Protocol client.
3 Sound DSP routines
4 Copyright (C) Michael Gernoth 2006-2007
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/*
22 * Sun GPL Disclaimer: For the avoidance of doubt, except that if any license choice
23 * other than GPL or LGPL is available it will apply instead, Sun elects to use only
24 * the General Public License version 2 (GPLv2) at this time for any software where
25 * a choice of GPL license versions is made available with the language indicating
26 * that GPLv2 or any later version may be used, or where a choice of which version
27 * of the GPL is applied is otherwise unspecified.
28 */
29
30#include <strings.h>
31
32#include "rdesktop.h"
33#include "rdpsnd.h"
34#include "rdpsnd_dsp.h"
35
36#ifdef HAVE_LIBSAMPLERATE
37#include <samplerate.h>
38
39#define SRC_CONVERTER SRC_SINC_MEDIUM_QUALITY
40#endif
41
42#define MAX_VOLUME 65535
43
44static uint16 softvol_left = MAX_VOLUME;
45static uint16 softvol_right = MAX_VOLUME;
46static uint32 resample_to_srate = 44100;
47static uint16 resample_to_bitspersample = 16;
48static uint16 resample_to_channels = 2;
49#ifdef HAVE_LIBSAMPLERATE
50static SRC_STATE *src_converter = NULL;
51#endif
52
53void
54rdpsnd_dsp_softvol_set(uint16 left, uint16 right)
55{
56 softvol_left = left;
57 softvol_right = right;
58 DEBUG(("rdpsnd_dsp_softvol_set: left: %u, right: %u\n", left, right));
59}
60
61void
62rdpsnd_dsp_softvol(unsigned char *buffer, unsigned int size, RD_WAVEFORMATEX * format)
63{
64 unsigned int factor_left, factor_right;
65 unsigned char *posin = buffer;
66 unsigned char *posout = buffer;
67
68 if ((softvol_left == MAX_VOLUME) && (softvol_right == MAX_VOLUME))
69 return;
70
71 factor_left = (softvol_left * 256) / MAX_VOLUME;
72 factor_right = (softvol_right * 256) / MAX_VOLUME;
73
74 if (format->nChannels == 1)
75 {
76 factor_left = factor_right = (factor_left + factor_right) / 2;
77 }
78
79 if (format->wBitsPerSample == 8)
80 {
81 sint8 val;
82
83 while (posout < buffer + size)
84 {
85 /* Left */
86 val = *posin++;
87 val = (val * factor_left) >> 8;
88 *posout++ = val;
89
90 /* Right */
91 val = *posin++;
92 val = (val * factor_right) >> 8;
93 *posout++ = val;
94 }
95 }
96 else
97 {
98 sint16 val;
99
100 while (posout < buffer + size)
101 {
102 /* Left */
103 val = *posin++;
104 val |= *posin++ << 8;
105 val = (val * factor_left) >> 8;
106 *posout++ = val & 0xff;
107 *posout++ = val >> 8;
108
109 /* Right */
110 val = *posin++;
111 val |= *posin++ << 8;
112 val = (val * factor_right) >> 8;
113 *posout++ = val & 0xff;
114 *posout++ = val >> 8;
115 }
116 }
117
118 DEBUG(("using softvol with factors left: %d, right: %d (%d/%d)\n", factor_left,
119 factor_right, format->wBitsPerSample, format->nChannels));
120}
121
122void
123rdpsnd_dsp_swapbytes(unsigned char *buffer, unsigned int size, RD_WAVEFORMATEX * format)
124{
125 int i;
126 uint8 swap;
127
128 if (format->wBitsPerSample == 8)
129 return;
130
131 if (size & 0x1)
132 warning("badly aligned sound data");
133
134 for (i = 0; i < (int) size; i += 2)
135 {
136 swap = *(buffer + i);
137 *(buffer + i) = *(buffer + i + 1);
138 *(buffer + i + 1) = swap;
139 }
140}
141
142RD_BOOL
143rdpsnd_dsp_resample_set(uint32 device_srate, uint16 device_bitspersample, uint16 device_channels)
144{
145#ifdef HAVE_LIBSAMPLERATE
146 int err;
147#endif
148
149 if (device_bitspersample != 16 && device_bitspersample != 8)
150 return False;
151
152 if (device_channels != 1 && device_channels != 2)
153 return False;
154
155 resample_to_srate = device_srate;
156 resample_to_bitspersample = device_bitspersample;
157 resample_to_channels = device_channels;
158
159#ifdef HAVE_LIBSAMPLERATE
160 if (src_converter != NULL)
161 src_converter = src_delete(src_converter);
162
163 if ((src_converter = src_new(SRC_CONVERTER, device_channels, &err)) == NULL)
164 {
165 warning("src_new failed: %d!\n", err);
166 return False;
167 }
168#endif
169
170 return True;
171}
172
173RD_BOOL
174rdpsnd_dsp_resample_supported(RD_WAVEFORMATEX * format)
175{
176 if (format->wFormatTag != WAVE_FORMAT_PCM)
177 return False;
178 if ((format->nChannels != 1) && (format->nChannels != 2))
179 return False;
180 if ((format->wBitsPerSample != 8) && (format->wBitsPerSample != 16))
181 return False;
182
183 return True;
184}
185
186uint32
187rdpsnd_dsp_resample(unsigned char **out, unsigned char *in, unsigned int size,
188 RD_WAVEFORMATEX * format, RD_BOOL stream_be)
189{
190#ifdef HAVE_LIBSAMPLERATE
191 SRC_DATA resample_data;
192 float *infloat, *outfloat;
193 int err;
194#else
195 int ratio1k = (resample_to_srate * 1000) / format->nSamplesPerSec;
196#endif
197 int innum, outnum;
198 unsigned char *tmpdata = NULL, *tmp = NULL;
199 int samplewidth = format->wBitsPerSample / 8;
200 int outsize = 0;
201 int i;
202
203 if ((resample_to_bitspersample == format->wBitsPerSample) &&
204 (resample_to_channels == format->nChannels) &&
205 (resample_to_srate == format->nSamplesPerSec))
206 return 0;
207
208#ifdef B_ENDIAN
209 if (!stream_be)
210 rdpsnd_dsp_swapbytes(in, size, format);
211#endif
212
213 if (resample_to_channels != format->nChannels)
214 {
215 int newsize = (size / format->nChannels) * resample_to_channels;
216 tmpdata = (unsigned char *) xmalloc(newsize);
217
218 for (i = 0; i < newsize / samplewidth; i++)
219 {
220 if (format->nChannels > resample_to_channels)
221 memcpy(tmpdata + (i * samplewidth),
222 in +
223 (((i * format->nChannels) / resample_to_channels) *
224 samplewidth), samplewidth);
225 else
226 memcpy(tmpdata + (i * samplewidth),
227 in +
228 (((i / resample_to_channels) * format->nChannels +
229 (i % format->nChannels)) * samplewidth), samplewidth);
230
231 }
232
233 in = tmpdata;
234 size = newsize;
235 }
236
237
238 /* Expand 8bit input-samples to 16bit */
239#ifndef HAVE_LIBSAMPLERATE /* libsamplerate needs 16bit samples */
240 if (format->wBitsPerSample != resample_to_bitspersample)
241#endif
242 {
243 /* source: 8 bit, dest: 16bit */
244 if (format->wBitsPerSample == 8)
245 {
246 tmp = tmpdata;
247 tmpdata = (unsigned char *) xmalloc(size * 2);
248 for (i = 0; i < (int) size; i++)
249 {
250 tmpdata[i * 2] = in[i];
251 tmpdata[(i * 2) + 1] = 0x00;
252 }
253 in = tmpdata;
254 samplewidth = 16 / 2;
255 size *= 2;
256
257 if (tmp != NULL)
258 xfree(tmp);
259 }
260 }
261
262 innum = size / samplewidth;
263
264 /* Do the resampling */
265#ifdef HAVE_LIBSAMPLERATE
266 if (src_converter == NULL)
267 {
268 warning("no samplerate converter available!!\n");
269 return 0;
270 }
271
272 outnum = ((float) innum * ((float) resample_to_srate / (float) format->nSamplesPerSec)) + 1;
273
274 infloat = (float *) xmalloc(sizeof(float) * innum);
275 outfloat = (float *) xmalloc(sizeof(float) * outnum);
276
277 src_short_to_float_array((short *) in, infloat, innum);
278
279 bzero(&resample_data, sizeof(resample_data));
280 resample_data.data_in = infloat;
281 resample_data.data_out = outfloat;
282 resample_data.input_frames = innum / resample_to_channels;
283 resample_data.output_frames = outnum / resample_to_channels;
284 resample_data.src_ratio = (double) resample_to_srate / (double) format->nSamplesPerSec;
285 resample_data.end_of_input = 0;
286
287 if ((err = src_process(src_converter, &resample_data)) != 0)
288 error("src_process: %s", src_strerror(err));
289
290 xfree(infloat);
291
292 outsize = resample_data.output_frames_gen * resample_to_channels * samplewidth;
293 *out = (unsigned char *) xmalloc(outsize);
294 src_float_to_short_array(outfloat, (short *) *out,
295 resample_data.output_frames_gen * resample_to_channels);
296 xfree(outfloat);
297
298#else
299 /* Michaels simple linear resampler */
300 if (resample_to_srate < format->nSamplesPerSec)
301 {
302 warning("downsampling currently not supported!\n");
303 return 0;
304 }
305
306 outnum = (innum * ratio1k) / 1000;
307
308 outsize = outnum * samplewidth;
309 *out = (unsigned char *) xmalloc(outsize);
310 bzero(*out, outsize);
311
312 for (i = 0; i < outsize / (resample_to_channels * samplewidth); i++)
313 {
314 int source = (i * 1000) / ratio1k;
315#if 0 /* Partial for linear resampler */
316 int part = (i * 100000) / ratio1k - source * 100;
317#endif
318 int j;
319
320 if (source * resample_to_channels + samplewidth > (int) size)
321 break;
322
323#if 0 /* Linear resampling, TODO: soundquality fixes (LP filter) */
324 if (samplewidth == 1)
325 {
326 sint8 cval1, cval2;
327 for (j = 0; j < resample_to_channels; j++)
328 {
329 memcpy(&cval1,
330 in + (source * resample_to_channels * samplewidth) +
331 (samplewidth * j), samplewidth);
332 memcpy(&cval2,
333 in + ((source + 1) * resample_to_channels * samplewidth) +
334 (samplewidth * j), samplewidth);
335
336 cval1 += (sint8) (cval2 * part) / 100;
337
338 memcpy(*out + (i * resample_to_channels * samplewidth) +
339 (samplewidth * j), &cval1, samplewidth);
340 }
341 }
342 else
343 {
344 sint16 sval1, sval2;
345 for (j = 0; j < resample_to_channels; j++)
346 {
347 memcpy(&sval1,
348 in + (source * resample_to_channels * samplewidth) +
349 (samplewidth * j), samplewidth);
350 memcpy(&sval2,
351 in + ((source + 1) * resample_to_channels * samplewidth) +
352 (samplewidth * j), samplewidth);
353
354 sval1 += (sint16) (sval2 * part) / 100;
355
356 memcpy(*out + (i * resample_to_channels * samplewidth) +
357 (samplewidth * j), &sval1, samplewidth);
358 }
359 }
360#else /* Nearest neighbor search */
361 for (j = 0; j < resample_to_channels; j++)
362 {
363 memcpy(*out + (i * resample_to_channels * samplewidth) + (samplewidth * j),
364 in + (source * resample_to_channels * samplewidth) +
365 (samplewidth * j), samplewidth);
366 }
367#endif
368 }
369 outsize = i * resample_to_channels * samplewidth;
370#endif
371
372 if (tmpdata != NULL)
373 xfree(tmpdata);
374
375 /* Shrink 16bit output-samples to 8bit */
376#ifndef HAVE_LIBSAMPLERATE /* libsamplerate produces 16bit samples */
377 if (format->wBitsPerSample != resample_to_bitspersample)
378#endif
379 {
380 /* source: 16 bit, dest: 8 bit */
381 if (resample_to_bitspersample == 8)
382 {
383 for (i = 0; i < outsize; i++)
384 {
385 *out[i] = *out[i * 2];
386 }
387 outsize /= 2;
388 }
389 }
390
391#ifdef B_ENDIAN
392 if (!stream_be)
393 rdpsnd_dsp_swapbytes(*out, outsize, format);
394#endif
395 return outsize;
396}
397
398STREAM
399rdpsnd_dsp_process(unsigned char *data, unsigned int size, struct audio_driver * current_driver,
400 RD_WAVEFORMATEX * format)
401{
402 static struct stream out;
403 RD_BOOL stream_be = False;
404
405 /* softvol and byteswap do not change the amount of data they
406 return, so they can operate on the input-stream */
407 if (current_driver->wave_out_volume == rdpsnd_dsp_softvol_set)
408 rdpsnd_dsp_softvol(data, size, format);
409
410#ifdef B_ENDIAN
411 if (current_driver->need_byteswap_on_be)
412 {
413 rdpsnd_dsp_swapbytes(data, size, format);
414 stream_be = True;
415 }
416#endif
417
418 out.data = NULL;
419
420 if (current_driver->need_resampling)
421 out.size = rdpsnd_dsp_resample(&out.data, data, size, format, stream_be);
422
423 if (out.data == NULL)
424 {
425 out.data = (unsigned char *) xmalloc(size);
426 memcpy(out.data, data, size);
427 out.size = size;
428 }
429
430 out.p = out.data;
431 out.end = out.p + out.size;
432
433 return &out;
434}
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