Version 0.9

Added: Filter type "IIR", which allows to specify a user-defined IIR filter of arbitrary order with custom coefficients.
Improved: For playback devices, the APO is now installed both as GFX (post-mix) and LFX (pre-mix) APO. Normally the filtering happens in GFX, but the command "Stage" allows to select LFX to apply effects based on the number of input channels.
Added: Expression language. The commands "If", "ElseIf", "Else", "EndIf", "Eval" and inline expressions allow to change the filter behaviour based on runtime conditions.
Added: Command "Copy", which copies audio data between channels.
Added: Command "Delay", which delays the audio on the currently selected channels.
Improved: Configuration files are now locked exclusively for writing while reading them.
Improved: There is now a transition (10 ms) from old to new configuration after loading.
Improved: Substantial internal restructuring to improve extensibility and maintainability. Filtering commands now work exactly in the order they were specified.
Fixed: When the microphone input was in stereo but the application requested a mono stream, the audio was heavily distorted.
Fixed: The sine sweep generated by the Benchmark application had floating-point precision issues leading to severe artifacts when certain filters were applied.

filters/IIRFilter.cpp

@@ -0,0 +1,123 @@
+/*
+	This file is part of EqualizerAPO, a system-wide equalizer.
+	Copyright (C) 2014  Jonas Thedering
+
+	This program is free software; you can redistribute it and/or modify
+	it under the terms of the GNU General Public License as published by
+	the Free Software Foundation; either version 2 of the License, or
+	(at your option) any later version.
+
+	This program is distributed in the hope that it will be useful,
+	but WITHOUT ANY WARRANTY; without even the implied warranty of
+	MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+	GNU General Public License for more details.
+
+	You should have received a copy of the GNU General Public License along
+	with this program; if not, write to the Free Software Foundation, Inc.,
+	51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+*/
+
+#include "helpers/MemoryHelper.h"
+#include "IIRFilter.h"
+
+using namespace std;
+
+#define IS_DENORMAL(f) (((*(unsigned int *)&(f))&0x7f800000) == 0)
+
+IIRFilter::IIRFilter(const vector<double>& coefficients)
+{
+	order = (unsigned)coefficients.size() / 2 - 1;
+	a = (float*)MemoryHelper::alloc(order * sizeof(float));
+	b = (float*)MemoryHelper::alloc(order * sizeof(float));
+	x = NULL;
+	y = NULL;
+
+	double a0 = coefficients[order+1];
+	b0 = float(coefficients[0] / a0);
+	for(unsigned i=0; i<order; i++)
+	{
+		b[i] = float(coefficients[i+1] / a0);
+		a[i] = float(-coefficients[i+order+2] / a0);
+	}
+}
+
+IIRFilter::~IIRFilter()
+{
+	MemoryHelper::free(a);
+	MemoryHelper::free(b);
+
+	if(x != NULL)
+		MemoryHelper::free(x);
+	if(y != NULL)
+		MemoryHelper::free(y);
+}
+
+vector<wstring> IIRFilter::initialize(float sampleRate, unsigned maxFrameCount, vector<wstring> channelNames)
+{
+	channelCount = (unsigned)channelNames.size();
+
+	if(x != NULL)
+		MemoryHelper::free(x);
+	if(y != NULL)
+		MemoryHelper::free(y);
+
+	x = (float*)MemoryHelper::alloc(order * channelCount * sizeof(float));
+	y = (float*)MemoryHelper::alloc(order * channelCount * sizeof(float));
+	memset(x, 0, order * channelCount * sizeof(float));
+	memset(y, 0, order * channelCount * sizeof(float));
+
+	return channelNames;
+}
+
+#pragma AVRT_CODE_BEGIN
+void IIRFilter::process(float** output, float** input, unsigned frameCount)
+{
+	for(unsigned i=0; i<channelCount; i++)
+	{
+		float* inputChannel = input[i];
+		float* outputChannel = output[i];
+
+		unsigned channelOffset = i*order;
+		float* xo = x+channelOffset;
+		float* yo = y+channelOffset;
+		for(unsigned j=0; j<frameCount; j++)
+		{
+			float sample = inputChannel[j];
+			float sum = b0 * sample;
+
+			for(unsigned k=order-1; k>0; k--)
+			{
+				sum += b[k] * xo[k];
+			}
+
+			sum += b[0] * xo[0];
+
+			for(unsigned k=order-1; k>0; k--)
+			{
+				sum += a[k] * yo[k];
+			}
+
+			sum += a[0] * yo[0];
+
+			for(unsigned k=order-1; k>0; k--)
+			{
+				xo[k] = xo[k-1];
+				yo[k] = yo[k-1];
+			}
+
+			xo[0] = sample;
+			yo[0] = sum;
+
+			outputChannel[j] = sum;
+		}
+	}
+
+	for(unsigned i=0; i<channelCount*order; i++)
+	{
+		if(IS_DENORMAL(x[i]))
+			x[i] = 0.0f;
+		if(IS_DENORMAL(y[i]))
+			y[i] = 0.0f;
+	}
+}
+#pragma AVRT_CODE_END