rdpsnd_dsp.c@ 53007

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RDP/client: fix OSE
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1	/*
2	rdesktop: A Remote Desktop Protocol client.
3	Sound DSP routines
4	Copyright (C) Michael Gernoth <[email protected]> 2006-2008
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 3 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, see <http://www.gnu.org/licenses/>.
18	*/
19
20	/*
21	* Oracle GPL Disclaimer: For the avoidance of doubt, except that if any license choice
22	* other than GPL or LGPL is available it will apply instead, Oracle elects to use only
23	* the General Public License version 2 (GPLv2) at this time for any software where
24	* a choice of GPL license versions is made available with the language indicating
25	* that GPLv2 or any later version may be used, or where a choice of which version
26	* of the GPL is applied is otherwise unspecified.
27	*/
28
29	#include <strings.h>
30
31	#include "rdesktop.h"
32	#include "rdpsnd.h"
33	#include "rdpsnd_dsp.h"
34
35	#ifdef HAVE_LIBSAMPLERATE
36	#include <samplerate.h>
37
38	#define SRC_CONVERTER SRC_SINC_MEDIUM_QUALITY
39	#endif
40
41	#define MAX_VOLUME 65535
42
43	static uint16 softvol_left = MAX_VOLUME;
44	static uint16 softvol_right = MAX_VOLUME;
45	static uint32 resample_to_srate = 44100;
46	static uint16 resample_to_bitspersample = 16;
47	static uint16 resample_to_channels = 2;
48	#ifdef HAVE_LIBSAMPLERATE
49	static SRC_STATE *src_converter = NULL;
50	#endif
51
52	void
53	rdpsnd_dsp_softvol_set(uint16 left, uint16 right)
54	{
55	softvol_left = left;
56	softvol_right = right;
57	DEBUG(("rdpsnd_dsp_softvol_set: left: %u, right: %u\n", left, right));
58	}
59
60	void
61	rdpsnd_dsp_softvol(unsigned char buffer, unsigned int size, RD_WAVEFORMATEX format)
62	{
63	unsigned int factor_left, factor_right;
64	unsigned char *posin = buffer;
65	unsigned char *posout = buffer;
66
67	if ((softvol_left == MAX_VOLUME) && (softvol_right == MAX_VOLUME))
68	return;
69
70	factor_left = (softvol_left * 256) / MAX_VOLUME;
71	factor_right = (softvol_right * 256) / MAX_VOLUME;
72
73	if (format->nChannels == 1)
74	{
75	factor_left = factor_right = (factor_left + factor_right) / 2;
76	}
77
78	if (format->wBitsPerSample == 8)
79	{
80	sint8 val;
81
82	while (posout < buffer + size)
83	{
84	/* Left */
85	val = *posin++;
86	val = (val * factor_left) >> 8;
87	*posout++ = val;
88
89	/* Right */
90	val = *posin++;
91	val = (val * factor_right) >> 8;
92	*posout++ = val;
93	}
94	}
95	else
96	{
97	sint16 val;
98
99	while (posout < buffer + size)
100	{
101	/* Left */
102	val = *posin++;
103	val \|= *posin++ << 8;
104	val = (val * factor_left) >> 8;
105	*posout++ = val & 0xff;
106	*posout++ = val >> 8;
107
108	/* Right */
109	val = *posin++;
110	val \|= *posin++ << 8;
111	val = (val * factor_right) >> 8;
112	*posout++ = val & 0xff;
113	*posout++ = val >> 8;
114	}
115	}
116
117	DEBUG(("using softvol with factors left: %d, right: %d (%d/%d)\n", factor_left,
118	factor_right, format->wBitsPerSample, format->nChannels));
119	}
120
121	void
122	rdpsnd_dsp_swapbytes(unsigned char buffer, unsigned int size, RD_WAVEFORMATEX format)
123	{
124	int i;
125	uint8 swap;
126
127	if (format->wBitsPerSample == 8)
128	return;
129
130	if (size & 0x1)
131	warning("badly aligned sound data");
132
133	for (i = 0; i < (int) size; i += 2)
134	{
135	swap = *(buffer + i);
136	(buffer + i) = (buffer + i + 1);
137	*(buffer + i + 1) = swap;
138	}
139	}
140
141	RD_BOOL
142	rdpsnd_dsp_resample_set(uint32 device_srate, uint16 device_bitspersample, uint16 device_channels)
143	{
144	#ifdef HAVE_LIBSAMPLERATE
145	int err;
146	#endif
147
148	if (device_bitspersample != 16 && device_bitspersample != 8)
149	return False;
150
151	if (device_channels != 1 && device_channels != 2)
152	return False;
153
154	resample_to_srate = device_srate;
155	resample_to_bitspersample = device_bitspersample;
156	resample_to_channels = device_channels;
157
158	#ifdef HAVE_LIBSAMPLERATE
159	if (src_converter != NULL)
160	src_converter = src_delete(src_converter);
161
162	if ((src_converter = src_new(SRC_CONVERTER, device_channels, &err)) == NULL)
163	{
164	warning("src_new failed: %d!\n", err);
165	return False;
166	}
167	#endif
168
169	return True;
170	}
171
172	RD_BOOL
173	rdpsnd_dsp_resample_supported(RD_WAVEFORMATEX * format)
174	{
175	if (format->wFormatTag != WAVE_FORMAT_PCM)
176	return False;
177	if ((format->nChannels != 1) && (format->nChannels != 2))
178	return False;
179	if ((format->wBitsPerSample != 8) && (format->wBitsPerSample != 16))
180	return False;
181
182	return True;
183	}
184
185	uint32
186	rdpsnd_dsp_resample(unsigned char *out, unsigned char in, unsigned int size,
187	RD_WAVEFORMATEX * format, RD_BOOL stream_be)
188	{
189	#ifdef HAVE_LIBSAMPLERATE
190	SRC_DATA resample_data;
191	float infloat, outfloat;
192	int err;
193	#else
194	int ratio1k = (resample_to_srate * 1000) / format->nSamplesPerSec;
195	#endif
196	int innum, outnum;
197	unsigned char tmpdata = NULL, tmp = NULL;
198	int samplewidth = format->wBitsPerSample / 8;
199	int outsize = 0;
200	int i;
201
202	if ((resample_to_bitspersample == format->wBitsPerSample) &&
203	(resample_to_channels == format->nChannels) &&
204	(resample_to_srate == format->nSamplesPerSec))
205	return 0;
206
207	#ifdef B_ENDIAN
208	if (!stream_be)
209	rdpsnd_dsp_swapbytes(in, size, format);
210	#endif
211
212	if (resample_to_channels != format->nChannels)
213	{
214	int newsize = (size / format->nChannels) * resample_to_channels;
215	tmpdata = (unsigned char *) xmalloc(newsize);
216
217	for (i = 0; i < newsize / samplewidth; i++)
218	{
219	if (format->nChannels > resample_to_channels)
220	memcpy(tmpdata + (i * samplewidth),
221	in +
222	(((i * format->nChannels) / resample_to_channels) *
223	samplewidth), samplewidth);
224	else
225	memcpy(tmpdata + (i * samplewidth),
226	in +
227	(((i / resample_to_channels) * format->nChannels +
228	(i % format->nChannels)) * samplewidth), samplewidth);
229
230	}
231
232	in = tmpdata;
233	size = newsize;
234	}
235
236
237	/* Expand 8bit input-samples to 16bit */
238	#ifndef HAVE_LIBSAMPLERATE /* libsamplerate needs 16bit samples */
239	if (format->wBitsPerSample != resample_to_bitspersample)
240	#endif
241	{
242	/* source: 8 bit, dest: 16bit */
243	if (format->wBitsPerSample == 8)
244	{
245	tmp = tmpdata;
246	tmpdata = (unsigned char ) xmalloc(size 2);
247	for (i = 0; i < (int) size; i++)
248	{
249	tmpdata[i * 2] = in[i];
250	tmpdata[(i * 2) + 1] = 0x00;
251	}
252	in = tmpdata;
253	samplewidth = 16 / 2;
254	size *= 2;
255
256	if (tmp != NULL)
257	xfree(tmp);
258	}
259	}
260
261	innum = size / samplewidth;
262
263	/* Do the resampling */
264	#ifdef HAVE_LIBSAMPLERATE
265	if (src_converter == NULL)
266	{
267	warning("no samplerate converter available!!\n");
268	return 0;
269	}
270
271	outnum = ((float) innum * ((float) resample_to_srate / (float) format->nSamplesPerSec)) + 1;
272
273	infloat = (float ) xmalloc(sizeof(float) innum);
274	outfloat = (float ) xmalloc(sizeof(float) outnum);
275
276	src_short_to_float_array((short *) in, infloat, innum);
277
278	bzero(&resample_data, sizeof(resample_data));
279	resample_data.data_in = infloat;
280	resample_data.data_out = outfloat;
281	resample_data.input_frames = innum / resample_to_channels;
282	resample_data.output_frames = outnum / resample_to_channels;
283	resample_data.src_ratio = (double) resample_to_srate / (double) format->nSamplesPerSec;
284	resample_data.end_of_input = 0;
285
286	if ((err = src_process(src_converter, &resample_data)) != 0)
287	error("src_process: %s", src_strerror(err));
288
289	xfree(infloat);
290
291	outsize = resample_data.output_frames_gen * resample_to_channels * samplewidth;
292	out = (unsigned char ) xmalloc(outsize);
293	src_float_to_short_array(outfloat, (short ) out,
294	resample_data.output_frames_gen * resample_to_channels);
295	xfree(outfloat);
296
297	#else
298	/* Michaels simple linear resampler */
299	if (resample_to_srate < format->nSamplesPerSec)
300	{
301	warning("downsampling currently not supported!\n");
302	return 0;
303	}
304
305	outnum = (innum * ratio1k) / 1000;
306
307	outsize = outnum * samplewidth;
308	out = (unsigned char ) xmalloc(outsize);
309	bzero(*out, outsize);
310
311	for (i = 0; i < outsize / (resample_to_channels * samplewidth); i++)
312	{
313	int source = (i * 1000) / ratio1k;
314	#if 0 /* Partial for linear resampler */
315	int part = (i * 100000) / ratio1k - source * 100;
316	#endif
317	int j;
318
319	if (source * resample_to_channels + samplewidth > (int) size)
320	break;
321
322	#if 0 /* Linear resampling, TODO: soundquality fixes (LP filter) */
323	if (samplewidth == 1)
324	{
325	sint8 cval1, cval2;
326	for (j = 0; j < resample_to_channels; j++)
327	{
328	memcpy(&cval1,
329	in + (source * resample_to_channels * samplewidth) +
330	(samplewidth * j), samplewidth);
331	memcpy(&cval2,
332	in + ((source + 1) * resample_to_channels * samplewidth) +
333	(samplewidth * j), samplewidth);
334
335	cval1 += (sint8) (cval2 * part) / 100;
336
337	memcpy(out + (i resample_to_channels * samplewidth) +
338	(samplewidth * j), &cval1, samplewidth);
339	}
340	}
341	else
342	{
343	sint16 sval1, sval2;
344	for (j = 0; j < resample_to_channels; j++)
345	{
346	memcpy(&sval1,
347	in + (source * resample_to_channels * samplewidth) +
348	(samplewidth * j), samplewidth);
349	memcpy(&sval2,
350	in + ((source + 1) * resample_to_channels * samplewidth) +
351	(samplewidth * j), samplewidth);
352
353	sval1 += (sint16) (sval2 * part) / 100;
354
355	memcpy(out + (i resample_to_channels * samplewidth) +
356	(samplewidth * j), &sval1, samplewidth);
357	}
358	}
359	#else /* Nearest neighbor search */
360	for (j = 0; j < resample_to_channels; j++)
361	{
362	memcpy(out + (i resample_to_channels * samplewidth) + (samplewidth * j),
363	in + (source * resample_to_channels * samplewidth) +
364	(samplewidth * j), samplewidth);
365	}
366	#endif
367	}
368	outsize = i * resample_to_channels * samplewidth;
369	#endif
370
371	if (tmpdata != NULL)
372	xfree(tmpdata);
373
374	/* Shrink 16bit output-samples to 8bit */
375	#ifndef HAVE_LIBSAMPLERATE /* libsamplerate produces 16bit samples */
376	if (format->wBitsPerSample != resample_to_bitspersample)
377	#endif
378	{
379	/* source: 16 bit, dest: 8 bit */
380	if (resample_to_bitspersample == 8)
381	{
382	for (i = 0; i < outsize; i++)
383	{
384	out[i] = out[i * 2];
385	}
386	outsize /= 2;
387	}
388	}
389
390	#ifdef B_ENDIAN
391	if (!stream_be)
392	rdpsnd_dsp_swapbytes(*out, outsize, format);
393	#endif
394	return outsize;
395	}
396
397	STREAM
398	rdpsnd_dsp_process(unsigned char data, unsigned int size, struct audio_driver current_driver,
399	RD_WAVEFORMATEX * format)
400	{
401	static struct stream out;
402	RD_BOOL stream_be = False;
403
404	/* softvol and byteswap do not change the amount of data they
405	return, so they can operate on the input-stream */
406	if (current_driver->wave_out_volume == rdpsnd_dsp_softvol_set)
407	rdpsnd_dsp_softvol(data, size, format);
408
409	#ifdef B_ENDIAN
410	if (current_driver->need_byteswap_on_be)
411	{
412	rdpsnd_dsp_swapbytes(data, size, format);
413	stream_be = True;
414	}
415	#endif
416
417	out.data = NULL;
418
419	if (current_driver->need_resampling)
420	out.size = rdpsnd_dsp_resample(&out.data, data, size, format, stream_be);
421
422	if (out.data == NULL)
423	{
424	out.data = (unsigned char *) xmalloc(size);
425	memcpy(out.data, data, size);
426	out.size = size;
427	}
428
429	out.p = out.data;
430	out.end = out.p + out.size;
431
432	return &out;
433	}

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source: vbox/trunk/src/VBox/RDP/client/rdpsnd_dsp.c@ 53007

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