/* the Music Player Daemon (MPD)
* (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
* This project's homepage is: http://www.musicpd.org
*
* 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 "pcm_utils.h"
#include "mpd_types.h"
#include "log.h"
#include <string.h>
#include <math.h>
#include <assert.h>
void pcm_changeBufferEndianness(char * buffer, int bufferSize, int bits) {
char temp;
switch(bits) {
case 16:
while(bufferSize) {
temp = *buffer;
*buffer = *(buffer+1);
*(buffer+1) = temp;
bufferSize-=2;
}
break;
}
}
void pcm_volumeChange(char * buffer, int bufferSize, AudioFormat * format,
int volume)
{
mpd_sint32 temp32;
mpd_sint8 * buffer8 = (mpd_sint8 *)buffer;
mpd_sint16 * buffer16 = (mpd_sint16 *)buffer;
if(volume>=1000) return;
if(volume<=0) {
memset(buffer,0,bufferSize);
return;
}
switch(format->bits) {
case 16:
while(bufferSize>0) {
temp32 = *buffer16;
temp32*= volume;
temp32/=1000;
*buffer16 = temp32>32767 ? 32767 :
(temp32<-32768 ? -32768 : temp32);
buffer16++;
bufferSize-=2;
}
break;
case 8:
while(bufferSize>0) {
temp32 = *buffer8;
temp32*= volume;
temp32/=1000;
*buffer8 = temp32>127 ? 127 :
(temp32<-128 ? -128 : temp32);
buffer8++;
bufferSize--;
}
break;
default:
ERROR("%i bits not supported by pcm_volumeChange!\n",
format->bits);
exit(EXIT_FAILURE);
}
}
void pcm_add(char * buffer1, char * buffer2, size_t bufferSize1,
size_t bufferSize2, int vol1, int vol2, AudioFormat * format)
{
mpd_sint32 temp32;
mpd_sint8 * buffer8_1 = (mpd_sint8 *)buffer1;
mpd_sint8 * buffer8_2 = (mpd_sint8 *)buffer2;
mpd_sint16 * buffer16_1 = (mpd_sint16 *)buffer1;
mpd_sint16 * buffer16_2 = (mpd_sint16 *)buffer2;
switch(format->bits) {
case 16:
while(bufferSize1>0 && bufferSize2>0) {
temp32 = (vol1*(*buffer16_1)+vol2*(*buffer16_2))/1000;
*buffer16_1 = temp32>32767 ? 32767 :
(temp32<-32768 ? -32768 : temp32);
buffer16_1++;
buffer16_2++;
bufferSize1-=2;
bufferSize2-=2;
}
if(bufferSize2>0) memcpy(buffer16_1,buffer16_2,bufferSize2);
break;
case 8:
while(bufferSize1>0 && bufferSize2>0) {
temp32 = (vol1*(*buffer8_1)+vol2*(*buffer8_2))/1000;
*buffer8_1 = temp32>127 ? 127 :
(temp32<-128 ? -128 : temp32);
buffer8_1++;
buffer8_2++;
bufferSize1--;
bufferSize2--;
}
if(bufferSize2>0) memcpy(buffer8_1,buffer8_2,bufferSize2);
break;
default:
ERROR("%i bits not supported by pcm_add!\n",format->bits);
exit(EXIT_FAILURE);
}
}
void pcm_mix(char * buffer1, char * buffer2, size_t bufferSize1,
size_t bufferSize2, AudioFormat * format, float portion1)
{
int vol1;
float s = sin(M_PI_2*portion1);
s*=s;
vol1 = s*1000+0.5;
vol1 = vol1>1000 ? 1000 : ( vol1<0 ? 0 : vol1 );
pcm_add(buffer1,buffer2,bufferSize1,bufferSize2,vol1,1000-vol1,format);
}
/* outFormat bits must be 16 and channels must be 2! */
void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
inSize, AudioFormat * outFormat, char * outBuffer)
{
static char * bitConvBuffer = NULL;
static int bitConvBufferLength = 0;
static char * channelConvBuffer = NULL;
static int channelConvBufferLength = 0;
char * dataChannelConv;
int dataChannelLen;
char * dataBitConv;
int dataBitLen;
assert(outFormat->bits==16);
assert(outFormat->channels==2);
/* converts */
switch(inFormat->bits) {
case 8:
dataBitLen = inSize << 1;
if(dataBitLen > bitConvBufferLength) {
bitConvBuffer = realloc(bitConvBuffer, dataBitLen);
bitConvBufferLength = dataBitLen;
}
dataBitConv = bitConvBuffer;
{
mpd_sint8 * in = (mpd_sint8 *)inBuffer;
mpd_sint16 * out = (mpd_sint16 *)dataBitConv;
int i;
for(i=0; i<inSize; i++) {
*out++ = (*in++) << 8;
}
}
break;
case 16:
dataBitConv = inBuffer;
dataBitLen = inSize;
break;
case 24:
/* put dithering code from mp3_decode here */
default:
ERROR("only 8 or 16 bits are supported for conversion!\n");
exit(EXIT_FAILURE);
}
/* converts only between 16 bit audio between mono and stereo */
switch(inFormat->channels) {
case 1:
dataChannelLen = (dataBitLen >> 1) << 2;
if(dataChannelLen > channelConvBufferLength) {
channelConvBuffer = realloc(channelConvBuffer,
dataChannelLen);
channelConvBufferLength = dataChannelLen;
}
dataChannelConv = channelConvBuffer;
{
mpd_sint16 * in = (mpd_sint16 *)dataBitConv;
mpd_sint16 * out = (mpd_sint16 *)dataChannelConv;
int i, inSamples = dataBitLen >> 1;
for(i=0;i<inSamples;i++) {
*out++ = *in;
*out++ = *in++;
}
}
break;
case 2:
dataChannelConv = dataBitConv;
dataChannelLen = dataBitLen;
break;
default:
ERROR("only 1 or 2 channels are supported for conversion!\n");
exit(EXIT_FAILURE);
}
if(inFormat->sampleRate == outFormat->sampleRate) {
memcpy(outBuffer,dataChannelConv,dataChannelLen);
}
else {
/* only works if outFormat is 16-bit stereo! */
/* resampling code blatantly ripped from XMMS */
const int shift = sizeof(mpd_sint16);
mpd_sint32 i, in_samples, out_samples, x, delta;
mpd_sint16 * inptr = (mpd_sint16 *)dataChannelConv;
mpd_sint16 * outptr = (mpd_sint16 *)outBuffer;
mpd_uint32 nlen = (((dataChannelLen >> shift) *
(mpd_uint32)(outFormat->sampleRate)) /
inFormat->sampleRate);
nlen <<= shift;
in_samples = dataChannelLen >> shift;
out_samples = nlen >> shift;
delta = (in_samples << 12) / out_samples;
for(x = 0, i = 0; i < out_samples; i++) {
int x1, frac;
x1 = (x >> 12) << 12;
frac = x - x1;
*outptr++ =
((inptr[(x1 >> 12) << 1] *
((1<<12) - frac) +
inptr[((x1 >> 12) + 1) << 1 ] *
frac) >> 12);
*outptr++ =
((inptr[((x1 >> 12) << 1) + 1] *
((1<<12) - frac) +
inptr[(((x1 >> 12) + 1) << 1) + 1] *
frac) >> 12);
x += delta;
}
}
return;
}
size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
char * inBuffer, size_t inSize, AudioFormat * outFormat)
{
const int shift = sizeof(mpd_sint16);
size_t outSize = inSize;
switch(inFormat->bits) {
case 8:
outSize = outSize << 1;
break;
case 16:
break;
default:
ERROR("only 8 or 16 bits are supported for conversion!\n");
exit(EXIT_FAILURE);
}
switch(inFormat->channels) {
case 1:
outSize = (outSize >> 1) << 2;
break;
case 2:
break;
default:
ERROR("only 1 or 2 channels are supported for conversion!\n");
exit(EXIT_FAILURE);
}
outSize = (((outSize >> shift) * (mpd_uint32)(outFormat->sampleRate)) /
inFormat->sampleRate);
outSize <<= shift;
return outSize;
}
/* vim:set shiftwidth=8 tabstop=8 expandtab: */
