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authorWarren Dukes <warren.dukes@gmail.com>2004-05-10 17:08:46 +0000
committerWarren Dukes <warren.dukes@gmail.com>2004-05-10 17:08:46 +0000
commit4f76ba5a427b27c9d354f2fca762eebacea5ee0c (patch)
tree96c41a85dc4b4db04b5c5a269979e4f5e7a87def /src
parent872af2077706da859b5a64c2e06bfd06de80387b (diff)
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format conversion for 8->16 bis and mono->stereo
git-svn-id: https://svn.musicpd.org/mpd/trunk@973 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to 'src')
-rw-r--r--src/pcm_utils.c87
1 files changed, 75 insertions, 12 deletions
diff --git a/src/pcm_utils.c b/src/pcm_utils.c
index 8ef431841..5465af65e 100644
--- a/src/pcm_utils.c
+++ b/src/pcm_utils.c
@@ -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);
+ }
- /* only works if outFormat is 16-bit stereo! */
- /* resampling code blatantly ripped from XMMS */
- {
+ /* 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;