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/****************************************************************************
**
** Copyright (C) 2022 The Qt Company Ltd.
** Contact: https://www.qt.io/licensing/
**
** This file is part of the Qt Toolkit.
**
** $QT_BEGIN_LICENSE:LGPL$
** Commercial License Usage
** Licensees holding valid commercial Qt licenses may use this file in
** accordance with the commercial license agreement provided with the
** Software or, alternatively, in accordance with the terms contained in
** a written agreement between you and The Qt Company. For licensing terms
** and conditions see https://www.qt.io/terms-conditions. For further
** information use the contact form at https://www.qt.io/contact-us.
**
** GNU Lesser General Public License Usage
** Alternatively, this file may be used under the terms of the GNU Lesser
** General Public License version 3 as published by the Free Software
** Foundation and appearing in the file LICENSE.LGPL3 included in the
** packaging of this file. Please review the following information to
** ensure the GNU Lesser General Public License version 3 requirements
** will be met: https://www.gnu.org/licenses/lgpl-3.0.html.
**
** GNU General Public License Usage
** Alternatively, this file may be used under the terms of the GNU
** General Public License version 2.0 or (at your option) the GNU General
** Public license version 3 or any later version approved by the KDE Free
** Qt Foundation. The licenses are as published by the Free Software
** Foundation and appearing in the file LICENSE.GPL2 and LICENSE.GPL3
** included in the packaging of this file. Please review the following
** information to ensure the GNU General Public License requirements will
** be met: https://www.gnu.org/licenses/gpl-2.0.html and
** https://www.gnu.org/licenses/gpl-3.0.html.
**
** $QT_END_LICENSE$
**
****************************************************************************/
#include "qambisonicdecoder_p.h"
#include "qambisonicdecoderdata_p.h"
#include <cmath>
QT_BEGIN_NAMESPACE
// Ambisonic decoding is described in detail in https://ambisonics.dreamhosters.com/BLaH3.pdf.
// We're using a phase matched band splitting filter to split the ambisonic signal into a low
// and high frequency component and apply matrix conversions to those components individually
// as described in the document.
//
// We are currently not using a near field compensation filter, something that could potentially
// improve sound quality further.
struct QAmbisonicDecoderData
{
QAudioFormat::ChannelConfig config;
const float *lf[3];
const float *hf[3];
};
static const QAmbisonicDecoderData decoderMap[] =
{
{ QAudioFormat::ChannelConfigMono,
{ decoderMatrix_mono_1_lf, decoderMatrix_mono_2_lf, decoderMatrix_mono_3_lf },
{ decoderMatrix_mono_1_hf, decoderMatrix_mono_2_hf, decoderMatrix_mono_3_hf }
},
{ QAudioFormat::ChannelConfigStereo,
{ decoderMatrix_stereo_1_lf, decoderMatrix_stereo_2_lf, decoderMatrix_stereo_3_lf },
{ decoderMatrix_stereo_1_hf, decoderMatrix_stereo_2_hf, decoderMatrix_stereo_3_hf }
},
{ QAudioFormat::ChannelConfig2Dot1,
{ decoderMatrix_2dot1_1_lf, decoderMatrix_2dot1_2_lf, decoderMatrix_2dot1_3_lf },
{ decoderMatrix_2dot1_1_hf, decoderMatrix_2dot1_2_hf, decoderMatrix_2dot1_3_hf }
},
{ QAudioFormat::ChannelConfigSurround5Dot0,
{ decoderMatrix_5dot0_1_lf, decoderMatrix_5dot0_2_lf, decoderMatrix_5dot0_3_lf },
{ decoderMatrix_5dot0_1_hf, decoderMatrix_5dot0_2_hf, decoderMatrix_5dot0_3_hf }
},
{ QAudioFormat::ChannelConfigSurround5Dot1,
{ decoderMatrix_5dot1_1_lf, decoderMatrix_5dot1_2_lf, decoderMatrix_5dot1_3_lf },
{ decoderMatrix_5dot1_1_hf, decoderMatrix_5dot1_2_hf, decoderMatrix_5dot1_3_hf }
},
{ QAudioFormat::ChannelConfigSurround7Dot0,
{ decoderMatrix_7dot0_1_lf, decoderMatrix_7dot0_2_lf, decoderMatrix_7dot0_3_lf },
{ decoderMatrix_7dot0_1_hf, decoderMatrix_7dot0_2_hf, decoderMatrix_7dot0_3_hf }
},
{ QAudioFormat::ChannelConfigSurround7Dot1,
{ decoderMatrix_7dot1_1_lf, decoderMatrix_7dot1_2_lf, decoderMatrix_7dot1_3_lf },
{ decoderMatrix_7dot1_1_hf, decoderMatrix_7dot1_2_hf, decoderMatrix_7dot1_3_hf }
}
};
// Implements a split second order IIR filter
// The audio data is split into a phase synced low and high frequency part
// This allows us to apply different factors to both parts for better sound
// localization when converting from ambisonic formats
//
// Details are described in https://ambisonics.dreamhosters.com/BLaH3.pdf, Appendix A.2.
class QAmbisonicDecoderFilter
{
public:
QAmbisonicDecoderFilter() = default;
void configure(float sampleRate, float cutoffFrequency = 380)
{
double k = tan(M_PI*cutoffFrequency/sampleRate);
a1 = float(2.*(k*k - 1.)/(k*k + 2*k + 1.));
a2 = float((k*k - 2*k + 1.)/(k*k + 2*k + 1.));
b0_lf = float(k*k/(k*k + 2*k + 1));
b1_lf = 2.f*b0_lf;
b0_hf = float(1./(k*k + 2*k + 1));
b1_hf = -2.f*b0_hf;
}
struct Output
{
float lf;
float hf;
};
Output next(float x)
{
float r_lf = x*b0_lf +
prevX[0]*b1_lf +
prevX[1]*b0_lf -
prevR_lf[0]*a1 -
prevR_lf[1]*a2;
float r_hf = x*b0_hf +
prevX[0]*b1_hf +
prevX[1]*b0_hf -
prevR_hf[0]*a1 -
prevR_hf[1]*a2;
prevX[1] = prevX[0];
prevX[0] = x;
prevR_lf[1] = prevR_lf[0];
prevR_lf[0] = r_lf;
prevR_hf[1] = prevR_hf[0];
prevR_hf[0] = r_hf;
return { r_lf, r_hf };
}
private:
float a1 = 0.;
float a2 = 0.;
float b0_hf = 0.;
float b1_hf = 0.;
float b0_lf = 0.;
float b1_lf = 0.;
float prevX[2] = {};
float prevR_lf[2] = {};
float prevR_hf[2] = {};
};
QAmbisonicDecoder::QAmbisonicDecoder(AmbisonicLevel ambisonicLevel, const QAudioFormat &format)
: level(ambisonicLevel)
{
auto outputConfiguration = format.channelConfig();
if (outputConfiguration == QAudioFormat::ChannelConfigUnknown)
outputConfiguration = format.defaultChannelConfigForChannelCount(format.channelCount());
for (const auto &d : decoderMap) {
if (d.config == outputConfiguration) {
decoderData = &d;
break;
}
}
if (!decoderData) {
// ### FIXME: use a stereo config, fill other channels with 0
}
inputChannels = (level+1)*(level+1);
outputChannels = format.channelCount();
filters = new QAmbisonicDecoderFilter[inputChannels];
for (int i = 0; i < inputChannels; ++i)
filters[i].configure(format.sampleRate());
}
void QAmbisonicDecoder::processBuffer(const float *input[], float *output, int nSamples)
{
float *o = output;
memset(o, 0, nSamples*outputChannels*sizeof(float));
const float *matrix_hi = decoderData->hf[level];
const float *matrix_lo = decoderData->hf[level];
for (int i = 0; i < nSamples; ++i) {
QAmbisonicDecoderFilter::Output buf[maxAmbisonicChannels];
for (int j = 0; j < inputChannels; ++j)
buf[j] = filters[j].next(input[j][i]);
for (int j = 0; j < inputChannels; ++j) {
for (int k = 0; k < outputChannels; ++k)
o[k] += matrix_lo[k*outputChannels + j]*buf[j].lf + matrix_hi[k*outputChannels + j]*buf[j].hf;
}
o += outputChannels;
}
}
void QAmbisonicDecoder::processBuffer(const float *input[], short *output, int nSamples)
{
if (level == 0) {
// ### copy W data
return;
}
const float *matrix_hi = decoderData->hf[level - 1];
const float *matrix_lo = decoderData->hf[level - 1];
for (int i = 0; i < nSamples; ++i) {
QAmbisonicDecoderFilter::Output buf[maxAmbisonicChannels];
for (int j = 0; j < inputChannels; ++j)
buf[j] = filters[j].next(input[j][i]);
float o[32]; // we can't support more than 32 channels from our API
memset(o, 0, 32*sizeof(short));
for (int j = 0; j < inputChannels; ++j) {
for (int k = 0; k < outputChannels; ++k)
o[k] += matrix_lo[k*outputChannels + j]*buf[j].lf + matrix_hi[k*outputChannels + j]*buf[j].hf;
}
for (int k = 0; k < outputChannels; ++k)
output[k] = static_cast<short>(o[k]*32768.);
output += outputChannels;
}
}
QT_END_NAMESPACE
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