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basic feecback and sanity check
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@micsthepick
micsthepick committed Feb 28, 2024
1 parent a1b64ff commit 9a2e823
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Showing 2 changed files with 102 additions and 41 deletions.
104 changes: 82 additions & 22 deletions REAPERDenoiser
View file
Original file line number Diff line number Diff line change
Expand Up @@ -76,6 +76,9 @@ pdc_delay = SIZE;
pdc_bot_ch=0;
pdc_top_ch=2;

// tell the plugin not to output for the first two frames, as the buffers are still catching up
muted = 2;


@slider
// A simple function to zero out the noise buffers when switching mode to "Record Noise Sample"
Expand All @@ -92,22 +95,54 @@ slider1 > 0.5 ? (


@sample
// This is a function to store a noise profile from the fftBuffer (only argument)
// into the noiseBuffer (also global scope)
function store_noise_profile(buffer) global(SIZE, noiseBuffer) local(index, normSquareNew, normSquareOld) (
// for each band, compare the norm of the noise in this frame.
// If it is greater than what's already there for this band, then copy
// it into the noiseBuffer
index = 0;
loop(SIZE,
normSquareNew = sqr(buffer[2 * index + 0]) + sqr(buffer[2 * index + 1]);
normSquareOld = noiseBuffer[index];
normSquareNew >= normSquareOld ? (
noiseBuffer[index] = normSquareNew;
);
index += 1;
);
);

// calculate an attenuation given the yNorm, and nNorm, and kSquared to approximate sNorm
function denoise(yNorm, nNorm, kSquared) (
yNorm / (yNorm + kSquared * nNorm);
);

// helpful functions to get/set the right sample given an index
// ( used here because the EEL2 base language has no way of interpreting `spl(channel)` )
// assumes 2 channels!
function getspl(channel) (
channel == 0 ? spl0 : spl1
);
function setspl(channel, value) (
(channel == 0 ? spl0 : spl1) = value
);

// We'll write a function to denoise a single channel, and then we'll call this
// for each of the channels.
// In this case, we'll pass in the channel number, the four input and output
// tiles, and the current sample.
// We also need to specify which variables will be local to the function (i.e.
// which variables have local instead of global scope).
// Note that channels are zero-indexed (so the left channel is channel 0, and
// the right channel is channel 1).
// the right channel is channel 1).d
// Functions can return values, but this one won't return anything.
// Swapping tiles and resetting samplesCollected will be managed by the caller.
function denoiseChannel(channel tileI1 tileI2 tileO1 tileO2 noiseBuffer samplesCollected)
function denoise_channel(channel tileI1 tileI2 tileO1 tileO2 noiseBuffer samplesCollected)
local(sample tilePos1 tilePos2 hannWindowTile1 hannWindowTile2 index bandIndex
kSquared yNorm nNorm attenuationFactor)
(
// Read out input audio and write it into the input buffer.
sample = spl(channel); // You can also use spl0 or spl1.
sample = getspl(channel); // You could also use spl(channel) or spl0 and spl1 individually

// Compute our positions in tile 1 and tile 2 for conciseness
tilePos1 = samplesCollected + SIZE/2;
Expand All @@ -124,7 +159,10 @@ local(sample tilePos1 tilePos2 hannWindowTile1 hannWindowTile2 index bandIndex
tileI2[tilePos2] = sample * hannWindowTile2;

// For the output audio, read from the two tiles and sum their results.
spl(channel) = tileO1[tilePos1] + tileO2[tilePos2];
muted > 0 ? setspl(channel, 0) : (
setspl(channel, tileO1[tilePos1] + tileO2[tilePos2]);
channel == 0 ? muted -= 1;
);

// When we finish a tile, samplesCollected is equal to (SIZE/2) - 1
// When that happens, we transform the contents of tile 1 and write them to
Expand Down Expand Up @@ -173,18 +211,7 @@ local(sample tilePos1 tilePos2 hannWindowTile1 hannWindowTile2 index bandIndex
// If slider1 is greater than 0.5 (i.e. the user selected "Record Noise
// Sample", we store the FFTs of each of these buffers.
slider1 > 0.5? (
// for each band, compare the norm of the noise in this frame.
// If it is greater than what's already there for this band, then copy
// it into the noiseBuffer
index = 0;
loop(SIZE,
normSquareNew = sqr(fftBuffer[2 * index + 0]) + sqr(fftBuffer[2 * index + 1]);
normSquareOld = noiseBuffer[index];
normSquareNew >= normSquareOld ? (
noiseBuffer[index] = normSquareNew;
);
index += 1;
);
store_noise_profile(fftBuffer);
);

// Apply Norbert Weiner's filtering algorithm,
Expand All @@ -203,7 +230,7 @@ local(sample tilePos1 tilePos2 hannWindowTile1 hannWindowTile2 index bandIndex
// The same for the noise component:
nNorm = noiseBuffer[bandIndex];

attenuationFactor = yNorm / (SIZE * (yNorm + kSquared * nNorm));
attenuationFactor = denoise(yNorm, nNorm, kSquared);

fftBuffer[2 * bandIndex + 0] *= attenuationFactor;
fftBuffer[2 * bandIndex + 1] *= attenuationFactor;
Expand All @@ -218,15 +245,17 @@ local(sample tilePos1 tilePos2 hannWindowTile1 hannWindowTile2 index bandIndex
// Copy from the complex numbers in fftBuffer to the output tile:
index = 0;
loop(SIZE,
tileO1[index] = fftBuffer[2 * index + 0];
tileO1[index] = fftBuffer[2 * index + 0] / SIZE; // rescale due to FFT
index += 1;
);
)
);

// Now, call denoiseChannel for each of the channels.
denoiseChannel(0, bufferI1L, bufferI2L, bufferO1L, bufferO2L, noiseBufferL, samplesCollected);
denoiseChannel(1, bufferI1R, bufferI2R, bufferO1R, bufferO2R, noiseBufferR, samplesCollected);

//IFTEST function sample_code() (
// Now, call denoise_channel for each of the channels.
denoise_channel(0, bufferI1L, bufferI2L, bufferO1L, bufferO2L, noiseBufferL, samplesCollected);
denoise_channel(1, bufferI1R, bufferI2R, bufferO1R, bufferO2R, noiseBufferR, samplesCollected);
// Go to the next sample
samplesCollected += 1;
samplesCollected == SIZE/2 ? (
Expand All @@ -249,6 +278,7 @@ samplesCollected == SIZE/2 ? (
bufferO1R = bufferO2R;
bufferO2R = temp;
);
//IFTEST ); // function sample_code() (

/*IFNTEST{*/ // hide serialize
@serialize
Expand All @@ -261,7 +291,37 @@ file_mem(0, noiseBufferL, SIZE);
file_mem(0, noiseBufferR, SIZE);
/*}IFNTSET*/ // hide serialize

///*IFTEST{* // main test block
/*IFTEST{* // main test block

// helpers
function sum_first_pdc_samples(s0val, s1val) (
spl0sum = 0;
spl1sum = 0;
loop(pdc_delay,
spl0=s0val;
spl1=s1val;
sample_code();
spl0sum += abs(spl0);
spl1sum += abs(spl1);
);
);

//DEBUGPRINT("SAMPLE_1_1_STRESS_TEST\n");
// spl0=1 spl1=1 for many samples
sum_first_pdc_samples(1, 1);
assert_equal_exact(0, spl0sum, "SAMPLE_1_1_STRESS_TEST failed on initial spl0");
assert_equal_exact(0, spl1sum, "SAMPLE_1_1_STRESS_TEST failed on initail spl1");
drift_from_1_spl0 = 0;
drift_from_1_spl1 = 0;
loop(0,//SIZE*64,
spl0=1;
spl1=1;
sample_code();
drift_from_1_spl0 += abs(1-spl0);
drift_from_1_spl1 += abs(1-spl1);
);
assert_near_equal(0, 2^-14, drift_from_1_spl0, "SAMPLE_1_1_STRESS_TEST drifted from 1 on spl0");
assert_near_equal(0, 2^-14, drift_from_1_spl1, "SAMPLE_1_1_STRESS_TEST drifted from 1 on spl1");

test_summary();
//*}IFTEST*/ // main test block
39 changes: 20 additions & 19 deletions testing_defines.eel2
View file
Original file line number Diff line number Diff line change
Expand Up @@ -63,22 +63,23 @@ function test_summary() global(failed_asserts successful_asserts) local(total) (
);

// drop in for spl(channel)
function spl(channel) (
0 == channel ? spl0 :
1 == channel ? spl1 :
2 == channel ? spl2 :
3 == channel ? spl3 :
4 == channel ? spl4 :
5 == channel ? spl5 :
6 == channel ? spl6 :
7 == channel ? spl7 :
8 == channel ? spl8 :
9 == channel ? spl9 :
10 == channel ? spl10 :
11 == channel ? spl11 :
12 == channel ? spl12 :
13 == channel ? spl13 :
14 == channel ? spl14 :
15 == channel ? spl15 :
0;
);
// function spl(channel) (
// 0 == channel ? spl0 :
// 1 == channel ? spl1 :
// 2 == channel ? spl2 :
// 3 == channel ? spl3 :
// 4 == channel ? spl4 :
// 5 == channel ? spl5 :
// 6 == channel ? spl6 :
// 7 == channel ? spl7 :
// 8 == channel ? spl8 :
// 9 == channel ? spl9 :
// 10 == channel ? spl10 :
// 11 == channel ? spl11 :
// 12 == channel ? spl12 :
// 13 == channel ? spl13 :
// 14 == channel ? spl14 :
// 15 == channel ? spl15 :
// 0;
// );
// does not work for writes!

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