vocalrediso.jsfx

// Based on Vocalrediso.ny, a nyquist filter for audacity, currently released under the GNU GPL V3
// see https://www.gnu.org/licenses/gpl-3.0.en.html, https://www.audacityteam.org/
// credits to Neil Bickford for his denoiser, https://github.com/Nbickford/REAPERDenoiser,
// which was used as a starting point for this script
// which now uses the STFT template code by geraintluff: https://forum.cockos.com/showthread.php?t=225955

///IFTEST* //this section not used by testing code
desc: vocal removal/isolation
//tags: processing vocals stereo
//author: Michael Pannekoek

//slider1:fft_size_index=2<0,7,1{256,512,1024,2048,4096,8192,16384,32768}>FFT size
slider1:0<-100,100,0.1>dry mix
slider2:0<-100,100,0.1>C mix (Vocals)
slider3:0<-5,5,0.001>strength at Low Cut
slider4:0<-5,5,0.001>strength at High Cut
slider5:80<0,24000,10>Low Cut (Vocals)
slider6:24000<0,24000,10>High Cut (Vocals)
slider7:0<-90,90,0.1>Phase (Degrees)
slider8:0<-5,5,0.001>Phase width at Low Cut
slider9:0<-5,5,0.001>Phase width at High Cut
slider10:1<0,1,0.05>Attenuate if different volume
slider11:1<0,1,1{No,Yes}>undo input corrections
slider12:0<-180,180,0.05>Phase2 (Degrees)
//IFTEST*/ //this section not used by testing code


in_pin:left input
in_pin:right input
out_pin:left output
out_pin:right output




///IFTEST*
@init
//IFTEST*/
/*
slider1=0;
slider2=0;
slider3=0;
slider4=0;
slider5=80;
slider6=24000;
slider7=0;
slider8=0;
slider9=0;
slider10=1;
slider11=1;
slider12=0;
*/

//IFTEST srate = 24000;

_free_memory = 0;
function simple_alloc(amount)
local(_free_memory_old)
global(_free_memory)
(
  _free_memory_old = _free_memory;
  _free_memory += amount;
  _free_memory_old;
);

prev_fft_size = 0;

function setup_stft_state(fft_size, first_time) (
	//////////////////////// Setup block
	// This is called when playback starts, or when the FFT size is changed
	memset(output_buffer, 0, buffer_length*2);
	memset(input_buffer, 0, buffer_length*2);
	// reset indexes and counters
	buffer_index = 0;
	output_index = 0;
	fft_counter = 0;
	// force silence for first overlaps where fft not yet run
	silence = overlap_factor;
	////////////////////////
);

function get_weight(strength, phase_width, l_real, l_imag, r_real, r_imag) local(norm_left, norm_right, w1, weight) (
	// cacluate energy for this bin
	norm_left = sqrt(sqr(l_real) + sqr(l_imag));
	norm_right = sqrt(sqr(r_real) + sqr(r_imag));

	// calculate phase difference between left & right, divide by phase_width
	w1 = (l_real * r_real + l_imag * r_imag) / (norm_left * norm_right * phase_width * 2);
	weight = ((max(0, min(1, w1+0.5))) ^ strength);
	// calculate weight: truncate w1 to [0, 1] and apply strength, then take 1 - the result, and multiply
	// by 1 - the square of the difference between the two norm values divided by the sum of the two, moderated by strength * slider10
	slider10 > 0 ? weight *= (
		1 - sqr(norm_left - norm_right)/sqr(norm_left + norm_right)
	) ^ (slider10 / strength) * 0.5;
	weight;
);

setup_stft_state(fft_size, prev_fft_size == 0);

function process_stft_segment(fft_buffer, fft_size) local(fft_bin, left_real, left_imag, right_real, right_imag) (
	fft_bin = 0; // FFT bin number
	loop(fft_size/2+1,
		fft_bin2 = fft_bin ? (fft_size - fft_bin) : 0;

		// Unfold complex spectrum into two real spectra
		left_real = fft_buffer[2*fft_bin] + fft_buffer[2*fft_bin2];
		left_imag = fft_buffer[2*fft_bin + 1] - fft_buffer[2*fft_bin2 + 1];
		right_real = fft_buffer[2*fft_bin + 1] + fft_buffer[2*fft_bin2 + 1];
		right_imag = -fft_buffer[2*fft_bin] + fft_buffer[2*fft_bin2];

		// input corrections
		// first, change the phase of L based on phase2:
		l_real_twisted = left_real*cosine2 + left_imag*sine2;
		l_imag_twisted = left_imag*cosine2 - left_real*sine2;

		// now mix L&R together based on phase
		r_real_rotated = right_real*cosine + l_real_twisted*sine;
		r_imag_rotated = right_imag*cosine + l_imag_twisted*sine;
		l_real_rotated = l_real_twisted*cosine - right_real*sine;
		l_imag_rotated = l_imag_twisted*cosine - right_imag*sine;

		//////////////////////// Main STFT block
		// The 'meat' of the algorithm, the code in this block will most resemble the code from vocalrediso.ny

		weight = 0;
		// first, apply vocal reduction algorithm only in the right bands
		fft_bin >= low_bin && fft_bin <= high_bin ? (
			weight = get_weight(strength_buffer[fft_bin], phase_width_buffer[fft_bin], l_real_rotated, l_imag_rotated, r_real_rotated, r_imag_rotated);
			// find the c channel, the sum of the two complex numbers
			c_real = l_real_rotated + r_real_rotated;
			c_imag = l_imag_rotated + r_imag_rotated;

			// apply weight to c channel
			c_real *= weight;
			c_imag *= weight;
		) :
		(
			// let wet signal have 0 for fft coefficients when out of bounds
			c_real = 0;
			c_imag = 0;
		);
		//////////////////////// END MAIN STFT block

		// apply wet dry mix
		out_l_real = l_real_rotated * dry_mix + c_real * wet_mix;
		out_l_imag = l_imag_rotated * dry_mix + c_imag * wet_mix;
		out_r_real = r_real_rotated * dry_mix + c_real * wet_mix;
		out_r_imag = r_imag_rotated * dry_mix + c_imag * wet_mix;

		// output corrections
		slider11 > 0.5 ? (
			// if requested, undo input corrections

			// unmix by phase
			l_real_out_twisted = out_l_real*cosine - out_r_real*sine;
			l_imag_out_twisted = out_l_imag*cosine - out_r_imag*sine;
			right_real = out_r_real*cosine + out_l_real*sine;
			right_imag = out_r_imag*cosine + out_l_imag*sine;

			left_real = l_real_out_twisted * cosine2 - l_imag_out_twisted * sine2;
			left_imag = l_imag_out_twisted * cosine2 + l_real_out_twisted * sine2;
		) :
		(
			// else, just copy the values
			left_real = out_l_real;
			left_imag = out_l_imag;
			right_real = out_r_real;
			right_imag = out_r_imag;
		);

		// Re-fold back into complex spectrum
		fft_buffer[2*fft_bin] = (left_real - right_imag)*0.5;
		fft_buffer[2*fft_bin + 1] = (left_imag + right_real)*0.5;
		fft_buffer[2*fft_bin2] = (left_real + right_imag)*0.5;
		fft_buffer[2*fft_bin2 + 1] = (-left_imag + right_real)*0.5;

		fft_bin += 1;
	);
);

MAX_FFT_SIZE = 32768;
fft_size = 8192;

fft_buffer = simple_alloc(MAX_FFT_SIZE*2);
window_buffer = simple_alloc(MAX_FFT_SIZE);

strength_buffer = simple_alloc(MAX_FFT_SIZE);
phase_width_buffer = simple_alloc(MAX_FFT_SIZE);

buffer_length = srate;
buffer_index = 0;
input_buffer = simple_alloc(buffer_length*2);
output_buffer =  simple_alloc(buffer_length*2);

function window(r) (
	// When squared, the Hann window adds up perfectly for overlap >= 4, so it's suitable for perfect reconstruction
	//(0.5 - 0.5*cos(r*2*$pi))/sqrt(0.375);
	// the MLT sine window also appears to add up correctly, with sigma = sqrt(2).
	sin(r*$pi)*sqrt(2);
);

overlap_factor = 4;
fft_interval = fft_size/overlap_factor;
fft_scaling_factor = 1/overlap_factor/fft_size;

fft_size != prev_fft_size ? (
	setup_stft_state(fft_size, prev_fft_size == 0);
	prev_fft_size = fft_size;
	// Fill window buffer
	i = 0;
	loop(fft_size,
		r = i/fft_size;
		window_buffer[i] = window(r);
		i += 1;
	);
);

pdc_delay = fft_size;
pdc_bot_ch = 0;
pdc_top_ch = 2;


freembuf(_free_memory);

@slider

//IFTEST function slider_code() (
// convert low cut and high cut to bins every time a slider is changed
low_bin = min(slider5, slider6) / srate * fft_size;
high_bin = max(slider6, slider5) / srate * fft_size;
// convert to radians
rotation = slider7*$pi/180;
// convert percentage to raw scale factor
dry_mix = slider1/100;
wet_mix = slider2/100;
low_strength = slider3;
high_strength = slider4;
phase_width_low = slider8;
phase_width_high = slider9;
cosine = cos(rotation);
sine = sin(rotation);
cosine2 = cos(slider12*$pi/180);
sine2 = sin(slider12*$pi/180);
// fill strength_buffer and phase_width_buffer
band_index = 0;
loop(fft_size/2+1,
	band_index >= low_bin && band_index <= high_bin ?
	(
		// only set values for the appropriate frequency range
		frac = (band_index - low_bin)/(high_bin - low_bin + 1);
		// fraction of progress through range [low_bin, high_bin)
		strength = low_strength* (1 - frac) + high_strength * frac;
		strength_buffer[band_index] = exp(strength);
		// precaculate strength (actual value should be positive, so it makes
		// sense to take the power of ten, but only after the
		// linear mapping over the spectrum is done.
		// precalculate phase width
		phase_width = phase_width_low * (1 - frac) + phase_width_high * frac;
		phase_width_buffer[band_index] = exp(phase_width);
	);

	band_index += 1;
	// next index
);
//IFTEST ); // slider_code() (

///IFTEST*
@sample
//IFTEST*/
//IFTEST function sample_code() (
input_buffer[buffer_index*2] = spl0;
input_buffer[buffer_index*2 + 1] = spl1;

output_index = buffer_index - fft_size;
output_index < 0 ? output_index += buffer_length;
silence > 0 ? (
	spl0 = spl1 = 0;
	// silence for fft init
) : (
	spl0 = output_buffer[output_index*2];
	spl1 = output_buffer[output_index*2 + 1];
);
output_buffer[output_index*2] = 0; // clear the sample we just read
output_buffer[output_index*2 + 1] = 0;

fft_counter += 1;
fft_counter >= fft_interval ? (
	fft_counter = 0;

	// Copy input to buffer
	bi = buffer_index - fft_size + 1;
	i = 0;
	loop(fft_size,
		i2 = bi + i;
		i2 < 0 ? i2 += buffer_length;

		fft_buffer[2*i] = input_buffer[2*i2]*window_buffer[i];
		fft_buffer[2*i + 1] = input_buffer[2*i2 + 1]*window_buffer[i];

		i += 1;
	);

	// Process buffer
	fft(fft_buffer, fft_size);
	fft_permute(fft_buffer, fft_size);

	process_stft_segment(fft_buffer, fft_size);

	fft_ipermute(fft_buffer, fft_size);
	ifft(fft_buffer, fft_size);

	// Add to output
	bi = buffer_index - fft_size + 1;
	i = 0;
	loop(fft_size,
		i2 = bi + i;
		(i2 < 0) ? i2 += buffer_length;

		output_buffer[2*i2] += fft_buffer[2*i]*fft_scaling_factor*window_buffer[i];
		output_buffer[2*i2 + 1] += fft_buffer[2*i + 1]*fft_scaling_factor*window_buffer[i];

		i += 1;
	);
	silence > 0 ? silence -= 1;
);

buffer_index = (buffer_index + 1)%buffer_length;

//IFTEST ); // function sample_code()



@serialize

//IFTEST function file_var(file, val) (printf("FILE_VAR, FILE: %d, VAL: %g\n", file, val));
///IFEEL*
serial_version = 1.00;
file_var(0, serial_version);
//IFEEL*/
// nothing serialized yet, but keep track of the serial_version
// for the preset state of the original plugin, serial_version would now be euqal to 0.

/*IFTEST{ // main test block

// helpers
function sum_first_pdc_samples(s0val, s1val) (
	setup_stft_state(fft_size, 1);
	spl0sum = 0;
	spl1sum = 0;
	loop(pdc_delay,
		spl0=s0val;
		spl1=s1val;
		sample_code();
		spl0sum += abs(spl0);
		spl1sum += abs(spl1);
	);
	assert_equal_exact(0, spl0sum, "spl0sum for init");
	assert_equal_exact(0, spl1sum, "spl1sum for init");
);

printf("SETUP: fft range is [0, srate/2] dry=100 wet=100 attenuate_diff_vol=1\n");
slider1 = slider2 = 100;
slider5 = 0;
slider6 = srate/2;
slider10 = 1;
slider_code();
printf("SETUP: got low_bin=%g, high_bin=%g\n", low_bin, high_bin);

printf("get_weight_ALL_ZERO\n");
assert_equal_exact(0, get_weight(1, 1, 0, 0, 0, 0));

printf("get_weight_PERPENDICULAR\n");
assert_equal_exact(0.25, get_weight(1, 1, 0, 1, 1, 0));

printf("get_weight_EQUIVALENT\n");
assert_equal_exact(0.5, get_weight(1, 1, 1, 0, 1, 0));

printf("get_weight_INVERSE\n");
assert_equal_exact(0, get_weight(1, 1, 1, 0, -1, 0));

printf("get_weight_ONE_ONLY\n");
assert_equal_exact(0, get_weight(1, 1, 1, 0, 0, 0));

printf("get_weight_HALF\n");
assert_equal_exact(0.5, get_weight(1, 1, 0.5, 0, 0.5, 0));

printf("get_weight_STRANGE\n");
assert_near_equal(0.5, 0.0001, get_weight(1, 1, -33.28894848631978, -662.24573314730117, -33.28894848631978, -662.24573314730117), "STRANGE01");
assert_near_equal(0.5, 0.0001, get_weight(1, 1, 452.28715419177155, 53.67874245364300, 452.28715419177155, 53.67874245364300), "STRANGE02");
assert_near_equal(0.5, 0.0001, get_weight(1, 1, 267.79446776740104, 17.14616043377475, 267.79446776740104, 17.14616043377475), "STRANGE03");
assert_near_equal(0.5, 0.0001, get_weight(1, 1, -32.55549374859478, 261.41171292127513, -32.55549374859478, 261.41171292127513), "STRANGE04");
assert_near_equal(0.5, 0.0001, get_weight(1, 1, -2.75689966923304, -217.67592724424455, -2.75689966923304, -217.67592724424455), "STRANGE05");

printf("SAMPLE_0_0_TEST\n");
// spl0=0 spl1=0 for 100 output samples
sum_first_pdc_samples(0, 0);

loop(100,
	spl0=0;
	spl1=0;
	sample_code();
	spl0sum += abs(spl0);
	spl1sum += abs(spl1);
);
assert_near_equal(0, 0.001, spl0sum, "SAMPLE_0_0_TEST: spl0sum was not as expected");
assert_near_equal(0, 0.001, spl1sum, "SAMPLE_0_0_TEST: spl1sum was not as expected");
printf("spl0sum=%g, spl1sum=%g\n", spl0sum, spl1sum);

printf("SAMPLE_1_0_TEST\n");
// spl0=1 spl1=0 for for 100 output samples
sum_first_pdc_samples(1, 0);

loop(100,
	spl0=1;
	spl1=0;
	sample_code();
	spl0sum += abs(spl0);
	spl1sum += abs(spl1);
);
assert_near_equal(100, 0.001, spl0sum, "SAMPLE_1_0_TEST: spl0sum was not as expected");
assert_near_equal(0, 0.001, spl1sum, "SAMPLE_1_0_TEST: spl1sum was not as expected");
printf("spl0sum=%g, spl1sum=%g\n", spl0sum, spl1sum);

printf("SAMPLE_0_1_TEST\n");
// spl0=0 spl1=1 for for 100 output samples
sum_first_pdc_samples(0, 1);

loop(100,
	spl0=0;
	spl1=1;
	sample_code();
	spl0sum += abs(spl0);
	spl1sum += abs(spl1);
);
assert_near_equal(0, 0.001, spl0sum, "SAMPLE_0_1_TEST: spl0sum was not as expected");
assert_near_equal(100, 0.001, spl1sum, "SAMPLE_0_1_TEST: spl1sum was not as expected");
printf("spl0sum=%g, spl1sum=%g\n", spl0sum, spl1sum);

printf("SAMPLE_1_1_TEST\n");
// spl0=1 spl1=1 for for 100 output samples
sum_first_pdc_samples(1, 1);

loop(100,
	spl0=1;
	spl1=1;
	sample_code();
	spl0sum += abs(spl0);
	spl1sum += abs(spl1);
);
assert_near_equal(200, 0.001, spl0sum, "SAMPLE_1_1_TEST: spl0sum was not as expected");
assert_near_equal(200, 0.001, spl1sum, "SAMPLE_1_1_TEST: spl1sum was not as expected");
printf("spl0sum=%g, spl1sum=%g\n", spl0sum, spl1sum);


printf("SETUP: fft range is [srate/4, srate/2] dry=100 wet=100\n");
// check no values before first output sample
slider1 = slider2 = 100;
slider5 = srate/4;
slider6 = srate/2;
printf("PDC_SILENCE_ON_PARTIAL_RANGE_TEST\n");
slider_code();
sum_first_pdc_samples(1, 1);

test_summary();
}IFTEST*/ // main test block