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/*
Audio format conversions
ARAnyM (C) 2008 Patrice Mandin
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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "sysdeps.h"
#include "audio_conv.h"
#include "host_audio.h"
#define DEBUG 0
#include "debug.h"
#define USE_SDL_RATECONVERSION 0
AudioConv::AudioConv(void)
: tmpBuf(NULL), srcRate(0), srcChan(0), srcOffset(0), srcSkip(0),
dstRate(0), tmpBufLen(0), volume(SDL_MIX_MAXVOLUME)
{
}
AudioConv::~AudioConv()
{
if (tmpBuf) {
free(tmpBuf);
tmpBuf=NULL;
}
}
/*
SDL 1.2 won't properly convert between different rates
so we use it only to convert formats
*/
void AudioConv::setConversion(Uint16 src_fmt, Uint8 src_chan, int src_rate, int src_offset, int src_skip,
Uint16 dst_fmt, Uint8 dst_chan, int dst_rate)
{
#if SDL_VERSION_ATLEAST(2, 0, 0) && USE_SDL_RATECONVERSION
SDL_BuildAudioCVT(&cvt,
src_fmt, src_chan, src_rate,
dst_fmt, dst_chan, dst_rate);
#else
SDL_BuildAudioCVT(&cvt,
src_fmt, src_chan, dst_rate,
dst_fmt, dst_chan, dst_rate);
#endif
srcRate = src_rate;
srcOffset = src_offset;
srcSkip = src_skip;
srcChan = src_chan;
dstRate = dst_rate;
D(bug("audio_conv: %s, %d chans, %d Hz to %s, %d chans, %d Hz",
HostAudio::FormatName(src_fmt), src_chan, src_rate,
HostAudio::FormatName(dst_fmt), dst_chan, dst_rate));
D(bug("audio_conv: offset %d bytes, skip %d bytes in source", src_offset, src_skip));
}
int AudioConv::rescaleFreq8(Uint8 *source, int *src_len, Uint8 *dest, int dst_len)
{
int srcSamplesRead=0, srcBytesRead=0;
int dstSamplesWritten=0, dstBytesWritten=0;
while ((srcBytesRead<*src_len) && (dstBytesWritten<dst_len)) {
switch(srcChan) {
case 2:
dest[dstBytesWritten++] = source[srcBytesRead++];
/* fall through */
case 1:
dest[dstBytesWritten++] = source[srcBytesRead];
break;
}
srcSamplesRead = (++dstSamplesWritten * srcRate) / dstRate;
srcBytesRead = srcSamplesRead*srcSkip+srcOffset;
}
*src_len = srcBytesRead;
return dstBytesWritten;
}
int AudioConv::rescaleFreq16(Uint16 *source, int *src_len, Uint16 *dest, int dst_len)
{
int srcSamplesRead=0, srcWordsRead=0;
int dstSamplesWritten=0, dstWordsWritten=0;
int curSkip = srcSkip>>1;
int curOffset = srcOffset>>1;
int srcLen = (*src_len)>>1;
dst_len >>= 1;
while ((srcWordsRead<srcLen) && (dstWordsWritten<dst_len)) {
switch(srcChan) {
case 2:
dest[dstWordsWritten++] = source[srcWordsRead++];
/* fall through */
case 1:
dest[dstWordsWritten++] = source[srcWordsRead];
break;
}
srcSamplesRead = (++dstSamplesWritten * srcRate) / dstRate;
srcWordsRead = srcSamplesRead*curSkip+curOffset;
}
*src_len = srcWordsRead<<1;
return dstWordsWritten<<1;
}
/*
Convert a block to host audio format
Set source and dest lengths to length really read and written
*/
void AudioConv::doConversion(Uint8 *source, int *src_len, Uint8 *dest, int *dst_len)
{
if ((srcRate==0) || (dstRate==0)) {
return;
}
D(bug("audioconv: from %p, %d -> %p, %d", source, *src_len, dest, *dst_len));
#if SDL_VERSION_ATLEAST(2, 0, 0) && USE_SDL_RATECONVERSION
/* Calc needed buffer size */
int neededBufSize = *src_len * cvt.len_mult;
if (tmpBufLen<neededBufSize) {
tmpBuf = (Uint8 *) realloc(tmpBuf, neededBufSize);
tmpBufLen = neededBufSize;
D(bug("audioconv: realloc tmpbuf, len: %d", neededBufSize));
}
/* Then convert to final format */
memcpy(tmpBuf, source, *src_len);
cvt.buf = tmpBuf;
cvt.len = *src_len;
SDL_ConvertAudio(&cvt);
#else
/* Calc needed buffer size */
int neededBufSize = *dst_len;
if (srcRate > dstRate) {
neededBufSize = (neededBufSize * srcRate) / dstRate;
}
if (tmpBufLen<neededBufSize) {
tmpBuf = (Uint8 *) realloc(tmpBuf, neededBufSize);
tmpBufLen = neededBufSize;
D(bug("audioconv: realloc tmpbuf, len: %d", neededBufSize));
}
/* First convert according to freq rates in a temp buffer */
int dstConvertedLen = 0;
int tmpLen = (int) (*dst_len / cvt.len_ratio);
neededBufSize = tmpLen * cvt.len_mult;
if (neededBufSize>tmpBufLen) {
neededBufSize = tmpBufLen;
}
switch(cvt.src_format & 0xff) {
case 8:
dstConvertedLen = rescaleFreq8(source, src_len, tmpBuf, tmpLen);
break;
case 16:
dstConvertedLen = rescaleFreq16((Uint16 *) source, src_len, (Uint16 *) tmpBuf, tmpLen);
break;
}
/* Then convert to final format */
cvt.buf = tmpBuf;
cvt.len = dstConvertedLen;
SDL_ConvertAudio(&cvt);
#endif
SDL_MixAudio(dest, cvt.buf, cvt.len_cvt, volume);
/* Set converted length */
*dst_len = cvt.len_cvt;
D(bug("audioconv: to %p, %d -> %p, %d", source, *src_len, dest, *dst_len));
}
void AudioConv::setVolume(int newVolume)
{
volume = newVolume;
}
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