Skip to content

Commit

Permalink
Drop wrapping_add in dsp_float math code
Browse files Browse the repository at this point in the history
  • Loading branch information
@PolyMeilex
PolyMeilex committed Dec 29, 2024
1 parent 755051b commit 0f67c33
Showing 1 changed file with 60 additions and 61 deletions.
121 changes: 60 additions & 61 deletions oxisynth/src/core/voice_pool/voice/dsp_float.rs
View file
Original file line number Diff line number Diff line change
Expand Up @@ -135,18 +135,18 @@ impl Voice {
let mut dsp_i: usize = 0;
loop {
// round to nearest point
let mut dsp_phase_index = (dsp_phase.wrapping_add(0x80000000) >> 32) as usize;
let mut dsp_phase_index = ((dsp_phase + 0x80000000) >> 32) as usize;

// interpolate sequence of sample points
while dsp_i < 64 && dsp_phase_index <= end_index {
dsp_buf[dsp_i] = dsp_amp * dsp_data[dsp_phase_index] as f32;

// increment phase and amplitude
dsp_phase = dsp_phase.wrapping_add(dsp_phase_incr);
dsp_phase += dsp_phase_incr;
// round to nearest point
dsp_phase_index = (dsp_phase.wrapping_add(0x80000000) >> 32) as usize;
dsp_phase_index = ((dsp_phase + 0x80000000) >> 32) as usize;
dsp_amp += dsp_amp_incr;
dsp_i = dsp_i.wrapping_add(1)
dsp_i += 1;
}
// break out if not looping (buffer may not be full)
if !looping {
Expand Down Expand Up @@ -217,20 +217,20 @@ impl Voice {

dsp_buf[dsp_i] = dsp_amp
* (coeffs[0] * dsp_data[dsp_phase_index] as f32
+ coeffs[1] * dsp_data[dsp_phase_index.wrapping_add(1)] as f32);
+ coeffs[1] * dsp_data[dsp_phase_index + 1] as f32);
// increment phase and amplitude
dsp_phase = dsp_phase.wrapping_add(dsp_phase_incr);
dsp_phase += dsp_phase_incr;
dsp_phase_index = (dsp_phase >> 32) as usize;
dsp_amp += dsp_amp_incr;
dsp_i = dsp_i.wrapping_add(1)
dsp_i += 1;
}

// break out if buffer filled
if dsp_i >= 64 {
break;
}
// we're now interpolating the last point
end_index = end_index.wrapping_add(1);
end_index += 1;

// interpolate within last point
while dsp_phase_index <= end_index && dsp_i < 64 {
Expand All @@ -240,11 +240,11 @@ impl Voice {
dsp_buf[dsp_i] = dsp_amp
* (coeffs[0] * dsp_data[dsp_phase_index] as f32 + coeffs[1] * point as f32);
// increment phase and amplitude
dsp_phase = dsp_phase.wrapping_add(dsp_phase_incr);
dsp_phase += dsp_phase_incr;
dsp_phase_index = (dsp_phase >> 32) as usize;
// increment amplitude
dsp_amp += dsp_amp_incr;
dsp_i = dsp_i.wrapping_add(1)
dsp_i += 1;
}

// break out if not looping (end of sample)
Expand Down Expand Up @@ -335,14 +335,14 @@ impl Voice {
dsp_buf[dsp_i] = dsp_amp
* (coeffs[0] * start_point as f32
+ coeffs[1] * dsp_data[dsp_phase_index] as f32
+ coeffs[2] * dsp_data[dsp_phase_index.wrapping_add(1)] as f32
+ coeffs[3] * dsp_data[dsp_phase_index.wrapping_add(2)] as f32);
+ coeffs[2] * dsp_data[dsp_phase_index + 1] as f32
+ coeffs[3] * dsp_data[dsp_phase_index + 2] as f32);

// increment phase and amplitude
dsp_phase = dsp_phase.wrapping_add(dsp_phase_incr);
dsp_phase += dsp_phase_incr;
dsp_phase_index = (dsp_phase >> 32) as usize;
dsp_amp += dsp_amp_incr;
dsp_i = dsp_i.wrapping_add(1)
dsp_i += 1;
}

// interpolate the sequence of sample points
Expand All @@ -353,14 +353,14 @@ impl Voice {
dsp_buf[dsp_i] = dsp_amp
* (coeffs[0] * dsp_data[dsp_phase_index.wrapping_sub(1)] as f32
+ coeffs[1] * dsp_data[dsp_phase_index] as f32
+ coeffs[2] * dsp_data[dsp_phase_index.wrapping_add(1)] as f32
+ coeffs[3] * dsp_data[dsp_phase_index.wrapping_add(2)] as f32);
+ coeffs[2] * dsp_data[dsp_phase_index + 1] as f32
+ coeffs[3] * dsp_data[dsp_phase_index + 2] as f32);

// increment phase and amplitude
dsp_phase = dsp_phase.wrapping_add(dsp_phase_incr);
dsp_phase += dsp_phase_incr;
dsp_phase_index = (dsp_phase >> 32) as usize;
dsp_amp += dsp_amp_incr;
dsp_i = dsp_i.wrapping_add(1)
dsp_i += 1;
}

// break out if buffer filled
Expand All @@ -369,7 +369,7 @@ impl Voice {
}

// we're now interpolating the 2nd to last point
end_index = end_index.wrapping_add(1);
end_index += 1;

// interpolate within 2nd to last point
while dsp_phase_index <= end_index && dsp_i < 64 {
Expand All @@ -379,17 +379,17 @@ impl Voice {
dsp_buf[dsp_i] = dsp_amp
* (coeffs[0] * dsp_data[dsp_phase_index.wrapping_sub(1)] as f32
+ coeffs[1] * dsp_data[dsp_phase_index] as f32
+ coeffs[2] * dsp_data[dsp_phase_index.wrapping_add(1)] as f32
+ coeffs[2] * dsp_data[dsp_phase_index + 1] as f32
+ coeffs[3] * end_point1 as f32);

// increment phase and amplitude
dsp_phase = dsp_phase.wrapping_add(dsp_phase_incr);
dsp_phase += dsp_phase_incr;
dsp_phase_index = (dsp_phase >> 32) as usize;
dsp_amp += dsp_amp_incr;
dsp_i = dsp_i.wrapping_add(1)
dsp_i += 1;
}
// we're now interpolating the last point
end_index = end_index.wrapping_add(1);
end_index += 1;

// interpolate within the last point
while dsp_phase_index <= end_index && dsp_i < 64 {
Expand All @@ -403,10 +403,10 @@ impl Voice {
+ coeffs[3] * end_point2 as f32);

// increment phase and amplitude
dsp_phase = dsp_phase.wrapping_add(dsp_phase_incr);
dsp_phase += dsp_phase_incr;
dsp_phase_index = (dsp_phase >> 32) as usize;
dsp_amp += dsp_amp_incr;
dsp_i = dsp_i.wrapping_add(1)
dsp_i += 1;
}

// break out if not looping (end of sample)
Expand Down Expand Up @@ -450,10 +450,9 @@ impl Voice {
// Convert playback "speed" floating point value to phase index/fract
let dsp_phase_incr = phase_set_float(phase_incr);

let dsp_phase = self.phase;
// add 1/2 sample to dsp_phase since 7th order interpolation is centered on
// the 4th sample point
let mut dsp_phase = dsp_phase.wrapping_add(0x80000000);
let mut dsp_phase = self.phase + 0x80000000;

// voice is currently looping?
let looping = SampleMode::from_val(self.gen[GeneratorType::SampleMode].val)
Expand Down Expand Up @@ -507,17 +506,17 @@ impl Voice {
+ coeffs[1] * start_points[1] as f32
+ coeffs[2] * start_points[0] as f32
+ coeffs[3] * dsp_data[dsp_phase_index] as f32
+ coeffs[4] * dsp_data[dsp_phase_index.wrapping_add(1)] as f32
+ coeffs[5] * dsp_data[dsp_phase_index.wrapping_add(2)] as f32
+ coeffs[6] * dsp_data[dsp_phase_index.wrapping_add(3)] as f32);
+ coeffs[4] * dsp_data[dsp_phase_index + 1] as f32
+ coeffs[5] * dsp_data[dsp_phase_index + 2] as f32
+ coeffs[6] * dsp_data[dsp_phase_index + 3] as f32);

// increment phase and amplitude
dsp_phase = dsp_phase.wrapping_add(dsp_phase_incr);
dsp_phase += dsp_phase_incr;
dsp_phase_index = (dsp_phase >> 32) as usize;
dsp_amp += dsp_amp_incr;
dsp_i = dsp_i.wrapping_add(1)
dsp_i += 1;
}
start_index = start_index.wrapping_add(1);
start_index += 1;

// interpolate 2nd to first sample point (start or loop start) if needed
while dsp_phase_index == start_index && dsp_i < 64 {
Expand All @@ -528,18 +527,18 @@ impl Voice {
+ coeffs[1] * start_points[0] as f32
+ coeffs[2] * dsp_data[dsp_phase_index.wrapping_sub(1)] as f32
+ coeffs[3] * dsp_data[dsp_phase_index] as f32
+ coeffs[4] * dsp_data[dsp_phase_index.wrapping_add(1)] as f32
+ coeffs[5] * dsp_data[dsp_phase_index.wrapping_add(2)] as f32
+ coeffs[6] * dsp_data[dsp_phase_index.wrapping_add(3)] as f32);
+ coeffs[4] * dsp_data[dsp_phase_index + 1] as f32
+ coeffs[5] * dsp_data[dsp_phase_index + 2] as f32
+ coeffs[6] * dsp_data[dsp_phase_index + 3] as f32);

// increment phase and amplitude
dsp_phase = dsp_phase.wrapping_add(dsp_phase_incr);
dsp_phase += dsp_phase_incr;
dsp_phase_index = (dsp_phase >> 32) as usize;
dsp_amp += dsp_amp_incr;
dsp_i = dsp_i.wrapping_add(1)
dsp_i += 1;
}

start_index = start_index.wrapping_add(1);
start_index += 1;

// interpolate 3rd to first sample point (start or loop start) if needed
while dsp_phase_index == start_index && dsp_i < 64 {
Expand All @@ -550,15 +549,15 @@ impl Voice {
+ coeffs[1] * dsp_data[dsp_phase_index.wrapping_sub(2)] as f32
+ coeffs[2] * dsp_data[dsp_phase_index.wrapping_sub(1)] as f32
+ coeffs[3] * dsp_data[dsp_phase_index] as f32
+ coeffs[4] * dsp_data[dsp_phase_index.wrapping_add(1)] as f32
+ coeffs[5] * dsp_data[dsp_phase_index.wrapping_add(2)] as f32
+ coeffs[6] * dsp_data[dsp_phase_index.wrapping_add(3)] as f32);
+ coeffs[4] * dsp_data[dsp_phase_index + 1] as f32
+ coeffs[5] * dsp_data[dsp_phase_index + 2] as f32
+ coeffs[6] * dsp_data[dsp_phase_index + 3] as f32);

// increment phase and amplitude
dsp_phase = dsp_phase.wrapping_add(dsp_phase_incr);
dsp_phase += dsp_phase_incr;
dsp_phase_index = (dsp_phase >> 32) as usize;
dsp_amp += dsp_amp_incr;
dsp_i = dsp_i.wrapping_add(1)
dsp_i += 1;
}

// set back to original start index
Expand All @@ -573,15 +572,15 @@ impl Voice {
+ coeffs[1] * dsp_data[dsp_phase_index.wrapping_sub(2)] as f32
+ coeffs[2] * dsp_data[dsp_phase_index.wrapping_sub(1)] as f32
+ coeffs[3] * dsp_data[dsp_phase_index] as f32
+ coeffs[4] * dsp_data[dsp_phase_index.wrapping_add(1)] as f32
+ coeffs[5] * dsp_data[dsp_phase_index.wrapping_add(2)] as f32
+ coeffs[6] * dsp_data[dsp_phase_index.wrapping_add(3)] as f32);
+ coeffs[4] * dsp_data[dsp_phase_index + 1] as f32
+ coeffs[5] * dsp_data[dsp_phase_index + 2] as f32
+ coeffs[6] * dsp_data[dsp_phase_index + 3] as f32);

// increment phase and amplitude
dsp_phase = dsp_phase.wrapping_add(dsp_phase_incr);
dsp_phase += dsp_phase_incr;
dsp_phase_index = (dsp_phase >> 32) as usize;
dsp_amp += dsp_amp_incr;
dsp_i = dsp_i.wrapping_add(1)
dsp_i += 1;
}

// break out if buffer filled
Expand All @@ -590,7 +589,7 @@ impl Voice {
}

// we're now interpolating the 3rd to last point
end_index = end_index.wrapping_add(1);
end_index += 1;

// interpolate within 3rd to last point
while dsp_phase_index <= end_index && dsp_i < 64 {
Expand All @@ -601,19 +600,19 @@ impl Voice {
+ coeffs[1] * dsp_data[dsp_phase_index.wrapping_sub(2)] as f32
+ coeffs[2] * dsp_data[dsp_phase_index.wrapping_sub(1)] as f32
+ coeffs[3] * dsp_data[dsp_phase_index] as f32
+ coeffs[4] * dsp_data[dsp_phase_index.wrapping_add(1)] as f32
+ coeffs[5] * dsp_data[dsp_phase_index.wrapping_add(2)] as f32
+ coeffs[4] * dsp_data[dsp_phase_index + 1] as f32
+ coeffs[5] * dsp_data[dsp_phase_index + 2] as f32
+ coeffs[6] * end_points[0] as f32);

// increment phase and amplitude
dsp_phase = dsp_phase.wrapping_add(dsp_phase_incr);
dsp_phase += dsp_phase_incr;
dsp_phase_index = (dsp_phase >> 32) as usize;
dsp_amp += dsp_amp_incr;
dsp_i = dsp_i.wrapping_add(1)
dsp_i += 1;
}

// we're now interpolating the 2nd to last point
end_index = end_index.wrapping_add(1);
end_index += 1;

// interpolate within 2nd to last point
while dsp_phase_index <= end_index && dsp_i < 64 {
Expand All @@ -624,19 +623,19 @@ impl Voice {
+ coeffs[1] * dsp_data[dsp_phase_index.wrapping_sub(2)] as f32
+ coeffs[2] * dsp_data[dsp_phase_index.wrapping_sub(1)] as f32
+ coeffs[3] * dsp_data[dsp_phase_index] as f32
+ coeffs[4] * dsp_data[dsp_phase_index.wrapping_add(1)] as f32
+ coeffs[4] * dsp_data[dsp_phase_index + 1] as f32
+ coeffs[5] * end_points[0] as f32
+ coeffs[6] * end_points[1] as f32);

// increment phase and amplitude
dsp_phase = dsp_phase.wrapping_add(dsp_phase_incr);
dsp_phase += dsp_phase_incr;
dsp_phase_index = (dsp_phase >> 32) as usize;
dsp_amp += dsp_amp_incr;
dsp_i = dsp_i.wrapping_add(1)
dsp_i += 1;
}

// we're now interpolating the last point
end_index = end_index.wrapping_add(1);
end_index += 1;

// interpolate within last point
while dsp_phase_index <= end_index && dsp_i < 64 {
Expand All @@ -652,10 +651,10 @@ impl Voice {
+ coeffs[6] * end_points[2] as f32);

// increment phase and amplitude
dsp_phase = dsp_phase.wrapping_add(dsp_phase_incr);
dsp_phase += dsp_phase_incr;
dsp_phase_index = (dsp_phase >> 32) as usize;
dsp_amp += dsp_amp_incr;
dsp_i = dsp_i.wrapping_add(1)
dsp_i += 1;
}

// break out if not looping (end of sample)
Expand Down

0 comments on commit 0f67c33

Please sign in to comment.