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-rw-r--r--mpg123_artsplugin/mpg123/decode_2to1.c233
1 files changed, 233 insertions, 0 deletions
diff --git a/mpg123_artsplugin/mpg123/decode_2to1.c b/mpg123_artsplugin/mpg123/decode_2to1.c
new file mode 100644
index 00000000..2eb5e07d
--- /dev/null
+++ b/mpg123_artsplugin/mpg123/decode_2to1.c
@@ -0,0 +1,233 @@
+/*
+ * Mpeg Layer-1,2,3 audio decoder
+ * ------------------------------
+ * copyright (c) 1995 by Michael Hipp, All rights reserved. See also 'README'
+ * version for slower machines .. decodes only every second sample
+ * sounds like 24000,22050 or 16000 kHz .. (depending on original sample freq.)
+ *
+ */
+
+#include <stdlib.h>
+#include <math.h>
+#include <string.h>
+
+#include "mpg123.h"
+
+#define WRITE_SAMPLE(samples,sum,clip) \
+ if( (sum) > 32767.0) { *(samples) = 0x7fff; (clip)++; } \
+ else if( (sum) < -32768.0) { *(samples) = -0x8000; (clip)++; } \
+ else { *(samples) = sum; }
+
+int synth_2to1_8bit(real *bandPtr,int channel,unsigned char *samples,int *pnt)
+{
+ short samples_tmp[32];
+ short *tmp1 = samples_tmp + channel;
+ int i,ret;
+ int pnt1 = 0;
+
+ ret = synth_2to1(bandPtr,channel,(unsigned char *) samples_tmp,&pnt1);
+ samples += channel + *pnt;
+
+ for(i=0;i<16;i++) {
+ *samples = conv16to8[*tmp1>>AUSHIFT];
+ samples += 2;
+ tmp1 += 2;
+ }
+ *pnt += 32;
+
+ return ret;
+}
+
+int synth_2to1_8bit_mono(real *bandPtr,unsigned char *samples,int *pnt)
+{
+ short samples_tmp[32];
+ short *tmp1 = samples_tmp;
+ int i,ret;
+ int pnt1 = 0;
+
+ ret = synth_2to1(bandPtr,0,(unsigned char *) samples_tmp,&pnt1);
+ samples += *pnt;
+
+ for(i=0;i<16;i++) {
+ *samples++ = conv16to8[*tmp1>>AUSHIFT];
+ tmp1 += 2;
+ }
+ *pnt += 16;
+
+ return ret;
+}
+
+
+int synth_2to1_8bit_mono2stereo(real *bandPtr,unsigned char *samples,int *pnt)
+{
+ short samples_tmp[32];
+ short *tmp1 = samples_tmp;
+ int i,ret;
+ int pnt1 = 0;
+
+ ret = synth_2to1(bandPtr,0,(unsigned char *) samples_tmp,&pnt1);
+ samples += *pnt;
+
+ for(i=0;i<16;i++) {
+ *samples++ = conv16to8[*tmp1>>AUSHIFT];
+ *samples++ = conv16to8[*tmp1>>AUSHIFT];
+ tmp1 += 2;
+ }
+ *pnt += 32;
+
+ return ret;
+}
+
+int synth_2to1_mono(real *bandPtr,unsigned char *samples,int *pnt)
+{
+ short samples_tmp[32];
+ short *tmp1 = samples_tmp;
+ int i,ret;
+ int pnt1=0;
+
+ ret = synth_2to1(bandPtr,0,(unsigned char *) samples_tmp,&pnt1);
+ samples += *pnt;
+
+ for(i=0;i<16;i++) {
+ *( (short *) samples) = *tmp1;
+ samples += 2;
+ tmp1 += 2;
+ }
+ *pnt += 32;
+
+ return ret;
+}
+
+int synth_2to1_mono2stereo(real *bandPtr,unsigned char *samples,int *pnt)
+{
+ int i,ret;
+
+ ret = synth_2to1(bandPtr,0,samples,pnt);
+ samples = samples + *pnt - 64;
+
+ for(i=0;i<16;i++) {
+ ((short *)samples)[1] = ((short *)samples)[0];
+ samples+=4;
+ }
+
+ return ret;
+}
+
+int synth_2to1(real *bandPtr,int channel,unsigned char *out,int *pnt)
+{
+ static real buffs[2][2][0x110];
+ static const int step = 2;
+ static int bo = 1;
+ short *samples = (short *) (out + *pnt);
+
+ real *b0,(*buf)[0x110];
+ int clip = 0;
+ int bo1;
+
+#ifndef NO_EQUALIZER
+ if(param.enable_equalizer)
+ do_equalizer(bandPtr,channel);
+#endif
+
+ if(!channel) {
+ bo--;
+ bo &= 0xf;
+ buf = buffs[0];
+ }
+ else {
+ samples++;
+ buf = buffs[1];
+ }
+
+ if(bo & 0x1) {
+ b0 = buf[0];
+ bo1 = bo;
+ dct64(buf[1]+((bo+1)&0xf),buf[0]+bo,bandPtr);
+ }
+ else {
+ b0 = buf[1];
+ bo1 = bo+1;
+ dct64(buf[0]+bo,buf[1]+bo+1,bandPtr);
+ }
+
+ {
+ register int j;
+ real *window = decwin + 16 - bo1;
+
+ for (j=8;j;j--,b0+=0x10,window+=0x30)
+ {
+ real sum;
+ sum = *window++ * *b0++;
+ sum -= *window++ * *b0++;
+ sum += *window++ * *b0++;
+ sum -= *window++ * *b0++;
+ sum += *window++ * *b0++;
+ sum -= *window++ * *b0++;
+ sum += *window++ * *b0++;
+ sum -= *window++ * *b0++;
+ sum += *window++ * *b0++;
+ sum -= *window++ * *b0++;
+ sum += *window++ * *b0++;
+ sum -= *window++ * *b0++;
+ sum += *window++ * *b0++;
+ sum -= *window++ * *b0++;
+ sum += *window++ * *b0++;
+ sum -= *window++ * *b0++;
+
+ WRITE_SAMPLE(samples,sum,clip); samples += step;
+#if 0
+ WRITE_SAMPLE(samples,sum,clip); samples += step;
+#endif
+ }
+
+ {
+ real sum;
+ sum = window[0x0] * b0[0x0];
+ sum += window[0x2] * b0[0x2];
+ sum += window[0x4] * b0[0x4];
+ sum += window[0x6] * b0[0x6];
+ sum += window[0x8] * b0[0x8];
+ sum += window[0xA] * b0[0xA];
+ sum += window[0xC] * b0[0xC];
+ sum += window[0xE] * b0[0xE];
+ WRITE_SAMPLE(samples,sum,clip); samples += step;
+#if 0
+ WRITE_SAMPLE(samples,sum,clip); samples += step;
+#endif
+ b0-=0x20,window-=0x40;
+ }
+ window += bo1<<1;
+
+ for (j=7;j;j--,b0-=0x30,window-=0x30)
+ {
+ real sum;
+ sum = -*(--window) * *b0++;
+ sum -= *(--window) * *b0++;
+ sum -= *(--window) * *b0++;
+ sum -= *(--window) * *b0++;
+ sum -= *(--window) * *b0++;
+ sum -= *(--window) * *b0++;
+ sum -= *(--window) * *b0++;
+ sum -= *(--window) * *b0++;
+ sum -= *(--window) * *b0++;
+ sum -= *(--window) * *b0++;
+ sum -= *(--window) * *b0++;
+ sum -= *(--window) * *b0++;
+ sum -= *(--window) * *b0++;
+ sum -= *(--window) * *b0++;
+ sum -= *(--window) * *b0++;
+ sum -= *(--window) * *b0++;
+
+ WRITE_SAMPLE(samples,sum,clip); samples += step;
+#if 0
+ WRITE_SAMPLE(samples,sum,clip); samples += step;
+#endif
+ }
+ }
+
+ *pnt += 64;
+
+ return clip;
+}
+
+