Correct phase interpolation to correct very minor error

pull/13/head
boblark 2 years ago
parent 1d39ef16ef
commit 69deebb2f2
  1. 14
      RadioIQMixer_F32.cpp

@ -20,6 +20,8 @@
* Inputs are either real or I-Q per bool twoChannel. Rev Apr 2021
*
* MIT License, Use at your own risk.
*
* Rev 28 Mar 2022 Correctedslight interpolation error. RSL
*/
#include "RadioIQMixer_F32.h"
@ -78,7 +80,7 @@ void RadioIQMixer_F32::update(void) {
a = sinTable512_f32[index];
b = sinTable512_f32[index+1];
// Linear interpolation and multiplying (DBMixer) with input
blockOut_i->data[i] = blockIn0->data[i] * (a + 0.001953125*(b-a)*deltaPhase);
blockOut_i->data[i] = blockIn0->data[i] * (a + (b-a)*deltaPhase);
/* Repeat for cosine by adding 90 degrees phase */
index = (index + 128) & 0x01ff;
@ -87,9 +89,9 @@ void RadioIQMixer_F32::update(void) {
b = sinTable512_f32[index+1];
/* deltaPhase will be the same as used for sin */
if(twoChannel)
blockOut_q->data[i] = blockIn1->data[i]*(a + 0.001953125*(b-a)*deltaPhase);
blockOut_q->data[i] = blockIn1->data[i]*(a + (b-a)*deltaPhase);
else
blockOut_q->data[i] = blockIn0->data[i]*(a + 0.001953125*(b-a)*deltaPhase);
blockOut_q->data[i] = blockIn0->data[i]*(a + (b-a)*deltaPhase);
}
}
else { // Do a more flexible update, i.e., not doSimple
@ -103,7 +105,7 @@ void RadioIQMixer_F32::update(void) {
b = sinTable512_f32[index+1];
// We now have a sine value, so multiply with the input data and save
// Linear interpolate sine and multiply with the input and amplitude (about 1.0)
blockOut_i->data[i] = amplitude_pk * blockIn0->data[i] * (a + 0.001953125*(b-a)*deltaPhase);
blockOut_i->data[i] = amplitude_pk * blockIn0->data[i] * (a + (b-a)*deltaPhase);
/* Shift forward phaseS_C and get cos. First, the calculation of index of the table */
phaseC = phaseS + phaseS_C;
@ -115,9 +117,9 @@ void RadioIQMixer_F32::update(void) {
b = sinTable512_f32[index+1];
// Same as sin, but leave amplitude of LO at +/- 1.0
if(twoChannel)
blockOut_q->data[i] = blockIn1->data[i]*(a + 0.001953125*(b-a)*deltaPhase);
blockOut_q->data[i] = blockIn1->data[i]*(a + (b-a)*deltaPhase);
else
blockOut_q->data[i] = blockIn0->data[i]*(a + 0.001953125*(b-a)*deltaPhase);
blockOut_q->data[i] = blockIn0->data[i]*(a + (b-a)*deltaPhase);
}
}
AudioStream_F32::release(blockIn0); // Done with this

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