rdpsnd_dsp.c@ 33656

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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	* Oracle 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, Oracle 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
44	static uint16 softvol_left = MAX_VOLUME;
45	static uint16 softvol_right = MAX_VOLUME;
46	static uint32 resample_to_srate = 44100;
47	static uint16 resample_to_bitspersample = 16;
48	static uint16 resample_to_channels = 2;
49	#ifdef HAVE_LIBSAMPLERATE
50	static SRC_STATE *src_converter = NULL;
51	#endif
52
53	void
54	rdpsnd_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
61	void
62	rdpsnd_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
122	void
123	rdpsnd_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
142	RD_BOOL
143	rdpsnd_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
173	RD_BOOL
174	rdpsnd_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
186	uint32
187	rdpsnd_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
398	STREAM
399	rdpsnd_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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source: vbox/trunk/src/VBox/RDP/client/rdpsnd_dsp.c@ 33656

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