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/* |
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rdesktop: A Remote Desktop Protocol client. |
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Sound DSP routines |
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Copyright (C) Michael Gernoth 2006 |
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|
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This program is free software; you can redistribute it and/or modify |
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it under the terms of the GNU General Public License as published by |
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the Free Software Foundation; either version 2 of the License, or |
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(at your option) any later version. |
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|
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This program is distributed in the hope that it will be useful, |
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but WITHOUT ANY WARRANTY; without even the implied warranty of |
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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GNU General Public License for more details. |
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|
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You should have received a copy of the GNU General Public License |
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along with this program; if not, write to the Free Software |
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Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. |
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*/ |
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|
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#include "rdesktop.h" |
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#include "rdpsnd.h" |
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#include "rdpsnd_dsp.h" |
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|
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#ifdef HAVE_LIBSAMPLERATE |
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#include <samplerate.h> |
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|
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#define SRC_CONVERTER SRC_SINC_MEDIUM_QUALITY |
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#endif |
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|
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#define MAX_VOLUME 65535 |
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|
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static uint16 softvol_left = MAX_VOLUME; |
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static uint16 softvol_right = MAX_VOLUME; |
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static uint32 resample_to_srate = 44100; |
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static uint16 resample_to_bitspersample = 16; |
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static uint16 resample_to_channels = 2; |
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#ifdef HAVE_LIBSAMPLERATE |
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static SRC_STATE *src_converter = NULL; |
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#endif |
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|
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void |
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rdpsnd_dsp_softvol_set(uint16 left, uint16 right) |
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{ |
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softvol_left = left; |
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softvol_right = right; |
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DEBUG(("rdpsnd_dsp_softvol_set: left: %u, right: %u\n", left, right)); |
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} |
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|
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void |
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rdpsnd_dsp_softvol(unsigned char *buffer, unsigned int size, WAVEFORMATEX * format) |
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{ |
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unsigned int factor_left, factor_right; |
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unsigned char *posin = buffer; |
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unsigned char *posout = buffer; |
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|
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if ((softvol_left == MAX_VOLUME) && (softvol_right == MAX_VOLUME)) |
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return; |
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|
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factor_left = (softvol_left * 256) / MAX_VOLUME; |
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factor_right = (softvol_right * 256) / MAX_VOLUME; |
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|
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if (format->nChannels == 1) |
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{ |
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factor_left = factor_right = (factor_left + factor_right) / 2; |
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} |
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|
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if (format->wBitsPerSample == 8) |
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{ |
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sint8 val; |
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|
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while (posout < buffer + size) |
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{ |
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/* Left */ |
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val = *posin++; |
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val = (val * factor_left) >> 8; |
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*posout++ = val; |
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|
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/* Right */ |
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val = *posin++; |
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val = (val * factor_right) >> 8; |
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*posout++ = val; |
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} |
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} |
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else |
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{ |
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sint16 val; |
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|
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while (posout < buffer + size) |
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{ |
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/* Left */ |
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val = *posin++; |
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val |= *posin++ << 8; |
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val = (val * factor_left) >> 8; |
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*posout++ = val & 0xff; |
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*posout++ = val >> 8; |
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|
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/* Right */ |
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val = *posin++; |
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val |= *posin++ << 8; |
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val = (val * factor_right) >> 8; |
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*posout++ = val & 0xff; |
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*posout++ = val >> 8; |
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} |
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} |
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|
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DEBUG(("using softvol with factors left: %d, right: %d (%d/%d)\n", factor_left, |
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factor_right, format->wBitsPerSample, format->nChannels)); |
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} |
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|
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void |
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rdpsnd_dsp_swapbytes(unsigned char *buffer, unsigned int size, WAVEFORMATEX * format) |
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{ |
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int i; |
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uint8 swap; |
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|
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if (format->wBitsPerSample == 8) |
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return; |
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|
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for (i = 0; i < size; i += 2) |
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{ |
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swap = *(buffer + i); |
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*(buffer + i) = *(buffer + i + 1); |
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*(buffer + i + 1) = swap; |
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} |
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} |
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|
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BOOL |
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rdpsnd_dsp_resample_set(uint32 device_srate, uint16 device_bitspersample, uint16 device_channels) |
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{ |
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#ifdef HAVE_LIBSAMPLERATE |
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int err; |
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#endif |
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|
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if (device_bitspersample != 16 && device_bitspersample != 8) |
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return False; |
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|
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if (device_channels != 1 && device_channels != 2) |
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return False; |
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|
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resample_to_srate = device_srate; |
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resample_to_bitspersample = device_bitspersample; |
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resample_to_channels = device_channels; |
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|
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#ifdef HAVE_LIBSAMPLERATE |
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if (src_converter != NULL) |
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src_converter = src_delete(src_converter); |
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|
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if ((src_converter = src_new(SRC_CONVERTER, device_channels, &err)) == NULL) |
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{ |
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warning("src_new failed: %d!\n", err); |
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return False; |
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} |
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#endif |
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|
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return True; |
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} |
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|
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BOOL |
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rdpsnd_dsp_resample_supported(WAVEFORMATEX * format) |
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{ |
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if (format->wFormatTag != WAVE_FORMAT_PCM) |
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return False; |
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if ((format->nChannels != 1) && (format->nChannels != 2)) |
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return False; |
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if ((format->wBitsPerSample != 8) && (format->wBitsPerSample != 16)) |
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return False; |
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|
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return True; |
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} |
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|
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uint32 |
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rdpsnd_dsp_resample(unsigned char **out, unsigned char *in, unsigned int size, |
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WAVEFORMATEX * format, BOOL stream_be) |
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{ |
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#ifdef HAVE_LIBSAMPLERATE |
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SRC_DATA resample_data; |
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float *infloat, *outfloat; |
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int err; |
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#else |
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int ratio1k = (resample_to_srate * 1000) / format->nSamplesPerSec; |
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#endif |
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int innum, outnum; |
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static BOOL warned = False; |
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unsigned char *tmpdata = NULL; |
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int samplewidth = format->wBitsPerSample / 8; |
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int outsize = 0; |
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int i; |
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|
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if ((resample_to_bitspersample == format->wBitsPerSample) && |
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(resample_to_channels == format->nChannels) && |
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(resample_to_srate == format->nSamplesPerSec)) |
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return 0; |
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|
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#ifdef B_ENDIAN |
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if (!stream_be) |
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rdpsnd_dsp_swapbytes(in, size, format); |
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#endif |
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|
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/* Expand 8bit input-samples to 16bit */ |
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#ifndef HAVE_LIBSAMPLERATE /* libsamplerate needs 16bit samples */ |
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if (format->wBitsPerSample != resample_to_bitspersample) |
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#endif |
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{ |
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/* source: 8 bit, dest: 16bit */ |
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if (format->wBitsPerSample == 8) |
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{ |
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tmpdata = xmalloc(size * 2); |
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for (i = 0; i < size; i++) |
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{ |
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tmpdata[i * 2] = in[i]; |
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tmpdata[(i * 2) + 1] = 0x00; |
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} |
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in = tmpdata; |
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samplewidth = 16 / 2; |
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size *= 2; |
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} |
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} |
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|
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if (resample_to_channels != format->nChannels) |
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{ |
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warning("unsupported resample-settings (%u -> %u/%u -> %u/%u -> %u), not resampling!\n", format->nSamplesPerSec, resample_to_srate, format->wBitsPerSample, resample_to_bitspersample, format->nChannels, resample_to_channels); |
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warned = True; |
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} |
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|
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innum = size / samplewidth; |
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|
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/* Do the resampling */ |
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#ifdef HAVE_LIBSAMPLERATE |
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if (src_converter == NULL) |
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{ |
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warning("no samplerate converter available!!\n"); |
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return 0; |
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} |
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|
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outnum = ((float) innum * ((float) resample_to_srate / (float) format->nSamplesPerSec)) + 1; |
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|
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infloat = xmalloc(sizeof(float) * innum); |
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outfloat = xmalloc(sizeof(float) * outnum); |
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|
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src_short_to_float_array((short *) in, infloat, innum); |
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|
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bzero(&resample_data, sizeof(resample_data)); |
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resample_data.data_in = infloat; |
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resample_data.data_out = outfloat; |
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resample_data.input_frames = innum / resample_to_channels; |
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resample_data.output_frames = outnum / resample_to_channels; |
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resample_data.src_ratio = (double) resample_to_srate / (double) format->nSamplesPerSec; |
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resample_data.end_of_input = 0; |
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|
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if ((err = src_process(src_converter, &resample_data)) != 0) |
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error("src_process: %s", src_strerror(err)); |
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|
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xfree(infloat); |
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|
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outsize = resample_data.output_frames_gen * resample_to_channels * samplewidth; |
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*out = xmalloc(outsize); |
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src_float_to_short_array(outfloat, (short *) *out, |
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resample_data.output_frames_gen * resample_to_channels); |
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xfree(outfloat); |
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|
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#else |
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/* Michaels simple linear resampler */ |
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if (resample_to_srate < format->nSamplesPerSec) |
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{ |
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warning("downsampling currently not supported!\n"); |
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return 0; |
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} |
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|
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outnum = (innum * ratio1k) / 1000; |
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|
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outsize = outnum * samplewidth; |
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*out = xmalloc(outsize); |
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bzero(*out, outsize); |
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|
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for (i = 0; i < outsize / (resample_to_channels * samplewidth); i++) |
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{ |
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int source = ((i + ratio1k / 1000 - 1) * 1000) / ratio1k; |
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|
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if (source * resample_to_channels + samplewidth > size) |
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break; |
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|
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if (resample_to_channels == 2) |
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{ |
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memcpy(*out + (i * resample_to_channels * samplewidth), |
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in + (source * resample_to_channels * samplewidth), samplewidth); |
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memcpy(*out + (i * resample_to_channels * samplewidth) + samplewidth, |
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in + (source * resample_to_channels * samplewidth) + samplewidth, |
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samplewidth); |
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} |
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else |
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{ |
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memcpy(*out + (i * samplewidth), in + (source * samplewidth), samplewidth); |
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} |
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} |
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outsize = i * resample_to_channels * samplewidth; |
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#endif |
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|
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if (tmpdata != NULL) |
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xfree(tmpdata); |
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|
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/* Shrink 16bit output-samples to 8bit */ |
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#ifndef HAVE_LIBSAMPLERATE /* libsamplerate produces 16bit samples */ |
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if (format->wBitsPerSample != resample_to_bitspersample) |
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#endif |
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{ |
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/* source: 16 bit, dest: 8 bit */ |
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if (resample_to_bitspersample == 8) |
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{ |
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for (i = 0; i < outsize; i++) |
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{ |
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*out[i] = *out[i * 2]; |
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} |
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outsize /= 2; |
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} |
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} |
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|
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#ifdef B_ENDIAN |
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if (!stream_be) |
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rdpsnd_dsp_swapbytes(*out, outsize, format); |
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#endif |
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return outsize; |
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} |
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|
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STREAM |
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rdpsnd_dsp_process(STREAM s, struct audio_driver * current_driver, WAVEFORMATEX * format) |
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{ |
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static struct stream out; |
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BOOL stream_be = False; |
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|
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/* softvol and byteswap do not change the amount of data they |
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return, so they can operate on the input-stream */ |
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if (current_driver->wave_out_volume == rdpsnd_dsp_softvol_set) |
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rdpsnd_dsp_softvol(s->data, s->size, format); |
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|
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#ifdef B_ENDIAN |
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if (current_driver->need_byteswap_on_be) |
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{ |
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rdpsnd_dsp_swapbytes(s->data, s->size, format); |
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stream_be = True; |
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} |
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#endif |
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|
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out.data = NULL; |
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|
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if (current_driver->need_resampling) |
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out.size = rdpsnd_dsp_resample(&out.data, s->data, s->size, format, stream_be); |
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|
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if (out.data == NULL) |
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{ |
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out.data = xmalloc(s->size); |
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memcpy(out.data, s->data, s->size); |
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out.size = s->size; |
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} |
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|
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out.p = out.data; |
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out.end = out.p + out.size; |
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|
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return &out; |
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} |