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format conversion for 8->16 bis and mono->stereo

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git-svn-id: https://svn.musicpd.org/mpd/trunk@973 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This commit is contained in:
Warren Dukes 2004-05-10 17:08:46 +00:00
parent 872af20777
commit 4f76ba5a42
1 changed files with 75 additions and 12 deletions
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83
src/pcm_utils.c
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@ -138,27 +138,92 @@ void pcm_mix(char * buffer1, char * buffer2, size_t bufferSize1,
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)
{
/*int inSampleSize = inFormat->bits*inFormat->channels/8;
int outSampleSize = outFormat->bits*outFormat->channels/8;*/
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(inFormat->bits==16);
assert(outFormat->bits==16);
assert(inFormat->channels==2);
assert(outFormat->channels==2);
if(inFormat->sampleRate == outFormat->sampleRate) return;
/* 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 = dataChannelLen >> 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 *)inBuffer;
mpd_sint16 * inptr = (mpd_sint16 *)dataChannelConv;
mpd_sint16 * outptr = (mpd_sint16 *)outBuffer;
mpd_uint32 nlen = (((inSize >> shift) *
mpd_uint32 nlen = (((dataChannelLen >> shift) *
(mpd_uint32)(outFormat->sampleRate)) /
inFormat->sampleRate);
nlen <<= shift;
@ -197,9 +262,7 @@ size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
nlen <<= shift;
assert(inFormat->bits==16);
assert(outFormat->bits==16);
assert(inFormat->channels==2);
assert(outFormat->channels==2);
return nlen;