rdesktop/trunk/rdpsnd_dsp.c

/*
   rdesktop: A Remote Desktop Protocol client.
   Sound DSP routines
   Copyright (C) Michael Gernoth 2006-2008

   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., 675 Mass Ave, Cambridge, MA 02139, USA.
*/

#include <strings.h>

#include "rdesktop.h"
#include "rdpsnd.h"
#include "rdpsnd_dsp.h"

#ifdef HAVE_LIBSAMPLERATE
#include <samplerate.h>

#define SRC_CONVERTER SRC_SINC_MEDIUM_QUALITY
#endif

#define MAX_VOLUME 65535

static uint16 softvol_left = MAX_VOLUME;
static uint16 softvol_right = MAX_VOLUME;
static uint32 resample_to_srate = 44100;
static uint16 resample_to_bitspersample = 16;
static uint16 resample_to_channels = 2;
#ifdef HAVE_LIBSAMPLERATE
static SRC_STATE *src_converter = NULL;
#endif

void
rdpsnd_dsp_softvol_set(uint16 left, uint16 right)
{
        softvol_left = left;
        softvol_right = right;
        DEBUG(("rdpsnd_dsp_softvol_set: left: %u, right: %u\n", left, right));
}

void
rdpsnd_dsp_softvol(unsigned char *buffer, unsigned int size, RD_WAVEFORMATEX * format)
{
        unsigned int factor_left, factor_right;
        unsigned char *posin = buffer;
        unsigned char *posout = buffer;

        if ((softvol_left == MAX_VOLUME) && (softvol_right == MAX_VOLUME))
                return;

        factor_left = (softvol_left * 256) / MAX_VOLUME;
        factor_right = (softvol_right * 256) / MAX_VOLUME;

        if (format->nChannels == 1)
        {
                factor_left = factor_right = (factor_left + factor_right) / 2;
        }

        if (format->wBitsPerSample == 8)
        {
                sint8 val;

                while (posout < buffer + size)
                {
                        /* Left */
                        val = *posin++;
                        val = (val * factor_left) >> 8;
                        *posout++ = val;

                        /* Right */
                        val = *posin++;
                        val = (val * factor_right) >> 8;
                        *posout++ = val;
                }
        }
        else
        {
                sint16 val;

                while (posout < buffer + size)
                {
                        /* Left */
                        val = *posin++;
                        val |= *posin++ << 8;
                        val = (val * factor_left) >> 8;
                        *posout++ = val & 0xff;
                        *posout++ = val >> 8;

                        /* Right */
                        val = *posin++;
                        val |= *posin++ << 8;
                        val = (val * factor_right) >> 8;
                        *posout++ = val & 0xff;
                        *posout++ = val >> 8;
                }
        }

        DEBUG(("using softvol with factors left: %d, right: %d (%d/%d)\n", factor_left,
               factor_right, format->wBitsPerSample, format->nChannels));
}

void
rdpsnd_dsp_swapbytes(unsigned char *buffer, unsigned int size, RD_WAVEFORMATEX * format)
{
        int i;
        uint8 swap;

        if (format->wBitsPerSample == 8)
                return;

        if (size & 0x1)
                warning("badly aligned sound data");

        for (i = 0; i < (int) size; i += 2)
        {
                swap = *(buffer + i);
                *(buffer + i) = *(buffer + i + 1);
                *(buffer + i + 1) = swap;
        }
}

RD_BOOL
rdpsnd_dsp_resample_set(uint32 device_srate, uint16 device_bitspersample, uint16 device_channels)
{
#ifdef HAVE_LIBSAMPLERATE
        int err;
#endif

        if (device_bitspersample != 16 && device_bitspersample != 8)
                return False;

        if (device_channels != 1 && device_channels != 2)
                return False;

        resample_to_srate = device_srate;
        resample_to_bitspersample = device_bitspersample;
        resample_to_channels = device_channels;

#ifdef HAVE_LIBSAMPLERATE
        if (src_converter != NULL)
                src_converter = src_delete(src_converter);

        if ((src_converter = src_new(SRC_CONVERTER, device_channels, &err)) == NULL)
        {
                warning("src_new failed: %d!\n", err);
                return False;
        }
#endif

        return True;
}

RD_BOOL
rdpsnd_dsp_resample_supported(RD_WAVEFORMATEX * format)
{
        if (format->wFormatTag != WAVE_FORMAT_PCM)
                return False;
        if ((format->nChannels != 1) && (format->nChannels != 2))
                return False;
        if ((format->wBitsPerSample != 8) && (format->wBitsPerSample != 16))
                return False;

        return True;
}

uint32
rdpsnd_dsp_resample(unsigned char **out, unsigned char *in, unsigned int size,
                    RD_WAVEFORMATEX * format, RD_BOOL stream_be)
{
#ifdef HAVE_LIBSAMPLERATE
        SRC_DATA resample_data;
        float *infloat, *outfloat;
        int err;
#else
        int ratio1k = (resample_to_srate * 1000) / format->nSamplesPerSec;
#endif
        int innum, outnum;
        unsigned char *tmpdata = NULL, *tmp = NULL;
        int samplewidth = format->wBitsPerSample / 8;
        int outsize = 0;
        int i;

        if ((resample_to_bitspersample == format->wBitsPerSample) &&
            (resample_to_channels == format->nChannels) &&
            (resample_to_srate == format->nSamplesPerSec))
                return 0;

#ifdef B_ENDIAN
        if (!stream_be)
                rdpsnd_dsp_swapbytes(in, size, format);
#endif

        if (resample_to_channels != format->nChannels)
        {
                int newsize = (size / format->nChannels) * resample_to_channels;
                tmpdata = (unsigned char *) xmalloc(newsize);

                for (i = 0; i < newsize / samplewidth; i++)
                {
                        if (format->nChannels > resample_to_channels)
                                memcpy(tmpdata + (i * samplewidth),
                                       in +
                                       (((i * format->nChannels) / resample_to_channels) *
                                        samplewidth), samplewidth);
                        else
                                memcpy(tmpdata + (i * samplewidth),
                                       in +
                                       (((i / resample_to_channels) * format->nChannels +
                                         (i % format->nChannels)) * samplewidth), samplewidth);

                }

                in = tmpdata;
                size = newsize;
        }


        /* Expand 8bit input-samples to 16bit */
#ifndef HAVE_LIBSAMPLERATE      /* libsamplerate needs 16bit samples */
        if (format->wBitsPerSample != resample_to_bitspersample)
#endif
        {
                /* source: 8 bit, dest: 16bit */
                if (format->wBitsPerSample == 8)
                {
                        tmp = tmpdata;
                        tmpdata = (unsigned char *) xmalloc(size * 2);
                        for (i = 0; i < (int) size; i++)
                        {
                                tmpdata[i * 2] = in[i];
                                tmpdata[(i * 2) + 1] = 0x00;
                        }
                        in = tmpdata;
                        samplewidth = 16 / 2;
                        size *= 2;

                        if (tmp != NULL)
                                xfree(tmp);
                }
        }

        innum = size / samplewidth;

        /* Do the resampling */
#ifdef HAVE_LIBSAMPLERATE
        if (src_converter == NULL)
        {
                warning("no samplerate converter available!!\n");
                return 0;
        }

        outnum = ((float) innum * ((float) resample_to_srate / (float) format->nSamplesPerSec)) + 1;

        infloat = (float *) xmalloc(sizeof(float) * innum);
        outfloat = (float *) xmalloc(sizeof(float) * outnum);

        src_short_to_float_array((short *) in, infloat, innum);

        bzero(&resample_data, sizeof(resample_data));
        resample_data.data_in = infloat;
        resample_data.data_out = outfloat;
        resample_data.input_frames = innum / resample_to_channels;
        resample_data.output_frames = outnum / resample_to_channels;
        resample_data.src_ratio = (double) resample_to_srate / (double) format->nSamplesPerSec;
        resample_data.end_of_input = 0;

        if ((err = src_process(src_converter, &resample_data)) != 0)
                error("src_process: %s", src_strerror(err));

        xfree(infloat);

        outsize = resample_data.output_frames_gen * resample_to_channels * samplewidth;
        *out = (unsigned char *) xmalloc(outsize);
        src_float_to_short_array(outfloat, (short *) *out,
                                 resample_data.output_frames_gen * resample_to_channels);
        xfree(outfloat);

#else
        /* Michaels simple linear resampler */
        if (resample_to_srate < format->nSamplesPerSec)
        {
                warning("downsampling currently not supported!\n");
                return 0;
        }

        outnum = (innum * ratio1k) / 1000;

        outsize = outnum * samplewidth;
        *out = (unsigned char *) xmalloc(outsize);
        bzero(*out, outsize);

        for (i = 0; i < outsize / (resample_to_channels * samplewidth); i++)
        {
                int source = (i * 1000) / ratio1k;
#if 0                           /* Partial for linear resampler */
                int part = (i * 100000) / ratio1k - source * 100;
#endif
                int j;

                if (source * resample_to_channels + samplewidth > (int) size)
                        break;

#if 0                           /* Linear resampling, TODO: soundquality fixes (LP filter) */
                if (samplewidth == 1)
                {
                        sint8 cval1, cval2;
                        for (j = 0; j < resample_to_channels; j++)
                        {
                                memcpy(&cval1,
                                       in + (source * resample_to_channels * samplewidth) +
                                       (samplewidth * j), samplewidth);
                                memcpy(&cval2,
                                       in + ((source + 1) * resample_to_channels * samplewidth) +
                                       (samplewidth * j), samplewidth);

                                cval1 += (sint8) (cval2 * part) / 100;

                                memcpy(*out + (i * resample_to_channels * samplewidth) +
                                       (samplewidth * j), &cval1, samplewidth);
                        }
                }
                else
                {
                        sint16 sval1, sval2;
                        for (j = 0; j < resample_to_channels; j++)
                        {
                                memcpy(&sval1,
                                       in + (source * resample_to_channels * samplewidth) +
                                       (samplewidth * j), samplewidth);
                                memcpy(&sval2,
                                       in + ((source + 1) * resample_to_channels * samplewidth) +
                                       (samplewidth * j), samplewidth);

                                sval1 += (sint16) (sval2 * part) / 100;

                                memcpy(*out + (i * resample_to_channels * samplewidth) +
                                       (samplewidth * j), &sval1, samplewidth);
                        }
                }
#else /* Nearest neighbor search */
                for (j = 0; j < resample_to_channels; j++)
                {
                        memcpy(*out + (i * resample_to_channels * samplewidth) + (samplewidth * j),
                               in + (source * resample_to_channels * samplewidth) +
                               (samplewidth * j), samplewidth);
                }
#endif
        }
        outsize = i * resample_to_channels * samplewidth;
#endif

        if (tmpdata != NULL)
                xfree(tmpdata);

        /* Shrink 16bit output-samples to 8bit */
#ifndef HAVE_LIBSAMPLERATE      /* libsamplerate produces 16bit samples */
        if (format->wBitsPerSample != resample_to_bitspersample)
#endif
        {
                /* source: 16 bit, dest: 8 bit */
                if (resample_to_bitspersample == 8)
                {
                        for (i = 0; i < outsize; i++)
                        {
                                *out[i] = *out[i * 2];
                        }
                        outsize /= 2;
                }
        }

#ifdef B_ENDIAN
        if (!stream_be)
                rdpsnd_dsp_swapbytes(*out, outsize, format);
#endif
        return outsize;
}

STREAM
rdpsnd_dsp_process(unsigned char *data, unsigned int size, struct audio_driver * current_driver,
                   RD_WAVEFORMATEX * format)
{
        static struct stream out;
        RD_BOOL stream_be = False;

        /* softvol and byteswap do not change the amount of data they
           return, so they can operate on the input-stream */
        if (current_driver->wave_out_volume == rdpsnd_dsp_softvol_set)
                rdpsnd_dsp_softvol(data, size, format);

#ifdef B_ENDIAN
        if (current_driver->need_byteswap_on_be)
        {
                rdpsnd_dsp_swapbytes(data, size, format);
                stream_be = True;
        }
#endif

        out.data = NULL;

        if (current_driver->need_resampling)
                out.size = rdpsnd_dsp_resample(&out.data, data, size, format, stream_be);

        if (out.data == NULL)
        {
                out.data = (unsigned char *) xmalloc(size);
                memcpy(out.data, data, size);
                out.size = size;
        }

        out.p = out.data;
        out.end = out.p + out.size;

        return &out;
}
1	stargo	1258	/*
2			rdesktop: A Remote Desktop Protocol client.
3			Sound DSP routines
4	jsorg71	1475	Copyright (C) Michael Gernoth 2006-2008
5	stargo	1258
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	stargo	1285	#include <strings.h>
22
23	stargo	1258	#include "rdesktop.h"
24			#include "rdpsnd.h"
25			#include "rdpsnd_dsp.h"
26
27	stargo	1278	#ifdef HAVE_LIBSAMPLERATE
28			#include <samplerate.h>
29
30			#define SRC_CONVERTER SRC_SINC_MEDIUM_QUALITY
31			#endif
32
33	stargo	1258	#define MAX_VOLUME 65535
34
35			static uint16 softvol_left = MAX_VOLUME;
36			static uint16 softvol_right = MAX_VOLUME;
37	stargo	1276	static uint32 resample_to_srate = 44100;
38			static uint16 resample_to_bitspersample = 16;
39			static uint16 resample_to_channels = 2;
40	stargo	1278	#ifdef HAVE_LIBSAMPLERATE
41			static SRC_STATE *src_converter = NULL;
42			#endif
43	stargo	1258
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	stargo	1260	void
53	jsorg71	1364	rdpsnd_dsp_softvol(unsigned char buffer, unsigned int size, RD_WAVEFORMATEX format)
54	stargo	1258	{
55			unsigned int factor_left, factor_right;
56	stargo	1259	unsigned char *posin = buffer;
57			unsigned char *posout = buffer;
58	stargo	1258
59	stargo	1259	if ((softvol_left == MAX_VOLUME) && (softvol_right == MAX_VOLUME))
60			return;
61
62	stargo	1270	factor_left = (softvol_left * 256) / MAX_VOLUME;
63			factor_right = (softvol_right * 256) / MAX_VOLUME;
64	stargo	1258
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	stargo	1262	sint8 val;
73	stargo	1258
74	stargo	1259	while (posout < buffer + size)
75	stargo	1258	{
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	stargo	1262	sint16 val;
90	stargo	1258
91	stargo	1259	while (posout < buffer + size)
92	stargo	1258	{
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	stargo	1263	DEBUG(("using softvol with factors left: %d, right: %d (%d/%d)\n", factor_left,
110			factor_right, format->wBitsPerSample, format->nChannels));
111	stargo	1258	}
112
113	stargo	1260	void
114	jsorg71	1364	rdpsnd_dsp_swapbytes(unsigned char buffer, unsigned int size, RD_WAVEFORMATEX format)
115	stargo	1260	{
116			int i;
117			uint8 swap;
118
119			if (format->wBitsPerSample == 8)
120			return;
121
122	ossman_	1431	if (size & 0x1)
123			warning("badly aligned sound data");
124
125	jsorg71	1394	for (i = 0; i < (int) size; i += 2)
126	stargo	1260	{
127			swap = *(buffer + i);
128			(buffer + i) = (buffer + i + 1);
129			*(buffer + i + 1) = swap;
130			}
131			}
132
133	jsorg71	1372	RD_BOOL
134	stargo	1276	rdpsnd_dsp_resample_set(uint32 device_srate, uint16 device_bitspersample, uint16 device_channels)
135			{
136	stargo	1278	#ifdef HAVE_LIBSAMPLERATE
137			int err;
138			#endif
139
140	stargo	1276	if (device_bitspersample != 16 && device_bitspersample != 8)
141			return False;
142
143			if (device_channels != 1 && device_channels != 2)
144			return False;
145
146			resample_to_srate = device_srate;
147			resample_to_bitspersample = device_bitspersample;
148			resample_to_channels = device_channels;
149
150	stargo	1278	#ifdef HAVE_LIBSAMPLERATE
151			if (src_converter != NULL)
152			src_converter = src_delete(src_converter);
153
154			if ((src_converter = src_new(SRC_CONVERTER, device_channels, &err)) == NULL)
155			{
156			warning("src_new failed: %d!\n", err);
157			return False;
158			}
159			#endif
160
161	stargo	1276	return True;
162			}
163
164	jsorg71	1372	RD_BOOL
165	jsorg71	1364	rdpsnd_dsp_resample_supported(RD_WAVEFORMATEX * format)
166	stargo	1276	{
167			if (format->wFormatTag != WAVE_FORMAT_PCM)
168			return False;
169			if ((format->nChannels != 1) && (format->nChannels != 2))
170			return False;
171			if ((format->wBitsPerSample != 8) && (format->wBitsPerSample != 16))
172			return False;
173
174			return True;
175			}
176
177			uint32
178			rdpsnd_dsp_resample(unsigned char *out, unsigned char in, unsigned int size,
179	jsorg71	1372	RD_WAVEFORMATEX * format, RD_BOOL stream_be)
180	stargo	1276	{
181	stargo	1278	#ifdef HAVE_LIBSAMPLERATE
182			SRC_DATA resample_data;
183			float infloat, outfloat;
184	stargo	1282	int err;
185	stargo	1278	#else
186	stargo	1283	int ratio1k = (resample_to_srate * 1000) / format->nSamplesPerSec;
187	stargo	1278	#endif
188	stargo	1283	int innum, outnum;
189	stargo	1288	unsigned char tmpdata = NULL, tmp = NULL;
190	stargo	1276	int samplewidth = format->wBitsPerSample / 8;
191	stargo	1278	int outsize = 0;
192	stargo	1281	int i;
193	stargo	1276
194	stargo	1277	if ((resample_to_bitspersample == format->wBitsPerSample) &&
195			(resample_to_channels == format->nChannels) &&
196			(resample_to_srate == format->nSamplesPerSec))
197			return 0;
198
199	stargo	1278	#ifdef B_ENDIAN
200			if (!stream_be)
201	stargo	1281	rdpsnd_dsp_swapbytes(in, size, format);
202	stargo	1278	#endif
203	stargo	1281
204	stargo	1288	if (resample_to_channels != format->nChannels)
205			{
206			int newsize = (size / format->nChannels) * resample_to_channels;
207	jsorg71	1394	tmpdata = (unsigned char *) xmalloc(newsize);
208	stargo	1288
209			for (i = 0; i < newsize / samplewidth; i++)
210			{
211			if (format->nChannels > resample_to_channels)
212			memcpy(tmpdata + (i * samplewidth),
213			in +
214			(((i * format->nChannels) / resample_to_channels) *
215			samplewidth), samplewidth);
216			else
217			memcpy(tmpdata + (i * samplewidth),
218			in +
219			(((i / resample_to_channels) * format->nChannels +
220			(i % format->nChannels)) * samplewidth), samplewidth);
221
222			}
223
224			in = tmpdata;
225			size = newsize;
226			}
227
228
229	stargo	1281	/* Expand 8bit input-samples to 16bit */
230			#ifndef HAVE_LIBSAMPLERATE /* libsamplerate needs 16bit samples */
231			if (format->wBitsPerSample != resample_to_bitspersample)
232			#endif
233			{
234			/* source: 8 bit, dest: 16bit */
235			if (format->wBitsPerSample == 8)
236	stargo	1278	{
237	stargo	1288	tmp = tmpdata;
238	jsorg71	1394	tmpdata = (unsigned char ) xmalloc(size 2);
239			for (i = 0; i < (int) size; i++)
240	stargo	1281	{
241			tmpdata[i * 2] = in[i];
242			tmpdata[(i * 2) + 1] = 0x00;
243			}
244			in = tmpdata;
245			samplewidth = 16 / 2;
246			size *= 2;
247	stargo	1288
248			if (tmp != NULL)
249			xfree(tmp);
250	stargo	1278	}
251	stargo	1281	}
252	stargo	1278
253	stargo	1283	innum = size / samplewidth;
254
255	stargo	1281	/* Do the resampling */
256	stargo	1278	#ifdef HAVE_LIBSAMPLERATE
257			if (src_converter == NULL)
258	stargo	1276	{
259	stargo	1278	warning("no samplerate converter available!!\n");
260			return 0;
261			}
262
263	stargo	1283	outnum = ((float) innum * ((float) resample_to_srate / (float) format->nSamplesPerSec)) + 1;
264	stargo	1278
265	jsorg71	1394	infloat = (float ) xmalloc(sizeof(float) innum);
266			outfloat = (float ) xmalloc(sizeof(float) outnum);
267	stargo	1278
268			src_short_to_float_array((short *) in, infloat, innum);
269
270			bzero(&resample_data, sizeof(resample_data));
271			resample_data.data_in = infloat;
272			resample_data.data_out = outfloat;
273			resample_data.input_frames = innum / resample_to_channels;
274			resample_data.output_frames = outnum / resample_to_channels;
275			resample_data.src_ratio = (double) resample_to_srate / (double) format->nSamplesPerSec;
276			resample_data.end_of_input = 0;
277
278	stargo	1282	if ((err = src_process(src_converter, &resample_data)) != 0)
279			error("src_process: %s", src_strerror(err));
280
281	stargo	1278	xfree(infloat);
282
283	stargo	1282	outsize = resample_data.output_frames_gen * resample_to_channels * samplewidth;
284	jsorg71	1394	out = (unsigned char ) xmalloc(outsize);
285	stargo	1283	src_float_to_short_array(outfloat, (short ) out,
286			resample_data.output_frames_gen * resample_to_channels);
287	stargo	1278	xfree(outfloat);
288
289			#else
290	stargo	1283	/* Michaels simple linear resampler */
291			if (resample_to_srate < format->nSamplesPerSec)
292	stargo	1278	{
293	stargo	1283	warning("downsampling currently not supported!\n");
294	stargo	1276	return 0;
295			}
296
297	stargo	1283	outnum = (innum * ratio1k) / 1000;
298
299			outsize = outnum * samplewidth;
300	jsorg71	1394	out = (unsigned char ) xmalloc(outsize);
301	stargo	1283	bzero(*out, outsize);
302	stargo	1277
303	stargo	1283	for (i = 0; i < outsize / (resample_to_channels * samplewidth); i++)
304	stargo	1276	{
305	stargo	1291	int source = (i * 1000) / ratio1k;
306	stargo	1292	#if 0 /* Partial for linear resampler */
307			int part = (i * 100000) / ratio1k - source * 100;
308			#endif
309	stargo	1287	int j;
310	stargo	1283
311	jsorg71	1394	if (source * resample_to_channels + samplewidth > (int) size)
312	stargo	1283	break;
313
314	stargo	1303	#if 0 /* Linear resampling, TODO: soundquality fixes (LP filter) */
315	stargo	1292	if (samplewidth == 1)
316			{
317			sint8 cval1, cval2;
318			for (j = 0; j < resample_to_channels; j++)
319			{
320			memcpy(&cval1,
321			in + (source * resample_to_channels * samplewidth) +
322			(samplewidth * j), samplewidth);
323			memcpy(&cval2,
324			in + ((source + 1) * resample_to_channels * samplewidth) +
325			(samplewidth * j), samplewidth);
326
327	astrand	1305	cval1 += (sint8) (cval2 * part) / 100;
328	stargo	1292
329			memcpy(out + (i resample_to_channels * samplewidth) +
330			(samplewidth * j), &cval1, samplewidth);
331			}
332			}
333			else
334			{
335			sint16 sval1, sval2;
336			for (j = 0; j < resample_to_channels; j++)
337			{
338			memcpy(&sval1,
339			in + (source * resample_to_channels * samplewidth) +
340			(samplewidth * j), samplewidth);
341			memcpy(&sval2,
342			in + ((source + 1) * resample_to_channels * samplewidth) +
343			(samplewidth * j), samplewidth);
344
345	astrand	1305	sval1 += (sint16) (sval2 * part) / 100;
346	stargo	1292
347			memcpy(out + (i resample_to_channels * samplewidth) +
348			(samplewidth * j), &sval1, samplewidth);
349			}
350			}
351			#else /* Nearest neighbor search */
352	stargo	1287	for (j = 0; j < resample_to_channels; j++)
353	stargo	1283	{
354	stargo	1287	memcpy(out + (i resample_to_channels * samplewidth) + (samplewidth * j),
355	stargo	1288	in + (source * resample_to_channels * samplewidth) +
356			(samplewidth * j), samplewidth);
357	stargo	1283	}
358	stargo	1292	#endif
359	stargo	1276	}
360	stargo	1283	outsize = i * resample_to_channels * samplewidth;
361	stargo	1278	#endif
362	stargo	1276
363	stargo	1281	if (tmpdata != NULL)
364			xfree(tmpdata);
365
366			/* Shrink 16bit output-samples to 8bit */
367			#ifndef HAVE_LIBSAMPLERATE /* libsamplerate produces 16bit samples */
368			if (format->wBitsPerSample != resample_to_bitspersample)
369			#endif
370			{
371			/* source: 16 bit, dest: 8 bit */
372			if (resample_to_bitspersample == 8)
373			{
374			for (i = 0; i < outsize; i++)
375			{
376			out[i] = out[i * 2];
377			}
378			outsize /= 2;
379			}
380			}
381
382	stargo	1278	#ifdef B_ENDIAN
383			if (!stream_be)
384	stargo	1281	rdpsnd_dsp_swapbytes(*out, outsize, format);
385	stargo	1278	#endif
386	stargo	1281	return outsize;
387	stargo	1276	}
388
389	stargo	1263	STREAM
390	ossman_	1342	rdpsnd_dsp_process(unsigned char data, unsigned int size, struct audio_driver current_driver,
391	jsorg71	1364	RD_WAVEFORMATEX * format)
392	stargo	1258	{
393	stargo	1263	static struct stream out;
394	jsorg71	1372	RD_BOOL stream_be = False;
395	stargo	1258
396	stargo	1263	/* softvol and byteswap do not change the amount of data they
397			return, so they can operate on the input-stream */
398	stargo	1258	if (current_driver->wave_out_volume == rdpsnd_dsp_softvol_set)
399	ossman_	1342	rdpsnd_dsp_softvol(data, size, format);
400	stargo	1258
401	stargo	1260	#ifdef B_ENDIAN
402			if (current_driver->need_byteswap_on_be)
403	stargo	1278	{
404	ossman_	1342	rdpsnd_dsp_swapbytes(data, size, format);
405	stargo	1278	stream_be = True;
406			}
407	stargo	1260	#endif
408
409	stargo	1276	out.data = NULL;
410	stargo	1263
411	stargo	1279	if (current_driver->need_resampling)
412	ossman_	1342	out.size = rdpsnd_dsp_resample(&out.data, data, size, format, stream_be);
413	stargo	1263
414	stargo	1276	if (out.data == NULL)
415			{
416	jsorg71	1394	out.data = (unsigned char *) xmalloc(size);
417	ossman_	1342	memcpy(out.data, data, size);
418			out.size = size;
419	stargo	1276	}
420
421	stargo	1263	out.p = out.data;
422			out.end = out.p + out.size;
423
424			return &out;
425	stargo	1258	}