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sdrangel/plugins/channelrx/freqscanner/freqscannersink.cpp
f4exb e4022031a9 Freq scanner VAD: Sonar fixes
2026-02-22 10:44:39 +01:00

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///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2023 Jon Beniston, M7RCE <jon@beniston.com> //
// //
// 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 as version 3 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 V3 for more details. //
// //
// You should have received a copy of the GNU General Public License //
// along with this program. If not, see <http://www.gnu.org/licenses/>. //
///////////////////////////////////////////////////////////////////////////////////
#include <QDebug>
#include <complex.h>
#include <cmath>
#ifndef M_PI
#define M_PI 3.14159265358979323846
#endif
#include "dsp/dspengine.h"
#include "dsp/fftfactory.h"
#include "util/db.h"
#include "freqscanner.h"
#include "freqscannersink.h"
FreqScannerSink::FreqScannerSink() :
m_channel(nullptr),
m_channelSampleRate(48000),
m_channelFrequencyOffset(0),
m_scannerSampleRate(33320),
m_centerFrequency(0),
m_messageQueueToChannel(nullptr),
m_fftSequence(-1),
m_fft(nullptr),
m_fftCounter(0),
m_fftSize(1024),
m_binsPerChannel(16),
m_averageCount(0),
m_cepstrumSequenceInverse(-1),
m_cepstrumSequenceForward(-1),
m_cepstrumFFTInverse(nullptr),
m_cepstrumFFTForward(nullptr),
m_cepstrumSize(0)
{
applySettings(m_settings, QStringList(), true);
applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, 16, 4, true);
}
FreqScannerSink::~FreqScannerSink()
{
FFTFactory* fftFactory = DSPEngine::instance()->getFFTFactory();
if (m_fftSequence >= 0) {
fftFactory->releaseEngine(m_fftSize, false, m_fftSequence);
}
if (m_cepstrumSequenceInverse >= 0) {
fftFactory->releaseEngine(m_cepstrumSize, true, m_cepstrumSequenceInverse);
}
if (m_cepstrumSequenceForward >= 0) {
fftFactory->releaseEngine(m_cepstrumSize, false, m_cepstrumSequenceForward);
}
}
void FreqScannerSink::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end)
{
Complex ci;
for (SampleVector::const_iterator it = begin; it != end; ++it)
{
Complex c(it->real(), it->imag());
c *= m_nco.nextIQ();
if (m_interpolatorDistance < 1.0f) // interpolate
{
while (!m_interpolator.interpolate(&m_interpolatorDistanceRemain, c, &ci))
{
processOneSample(ci);
m_interpolatorDistanceRemain += m_interpolatorDistance;
}
}
else // decimate (and filter)
{
if (m_interpolator.decimate(&m_interpolatorDistanceRemain, c, &ci))
{
processOneSample(ci);
m_interpolatorDistanceRemain += m_interpolatorDistance;
}
}
}
}
void FreqScannerSink::processOneSample(Complex &ci)
{
ci /= SDR_RX_SCALEF;
m_fft->in()[m_fftCounter] = ci;
m_fftCounter++;
if (m_fftCounter == m_fftSize)
{
// Apply windowing function
m_fftWindow.apply(m_fft->in());
// Perform FFT
m_fft->transform();
// Accumulate voice activity levels on individual FFT (before averaging)
// This captures sharp formant structure better than averaged spectrum
int freqCount = m_settings.m_frequencySettings.size();
if (m_voiceLevelSum.size() != freqCount) {
m_voiceLevelSum.resize(freqCount);
m_voiceLevelCount.resize(freqCount);
m_voiceLevelSum.fill(0.0);
m_voiceLevelCount.fill(0);
}
for (int i = 0; i < freqCount; i++)
{
if (m_settings.m_frequencySettings[i].m_enabled)
{
qint64 frequency = m_settings.m_frequencySettings[i].m_frequency;
qint64 startFrequency = m_centerFrequency - m_scannerSampleRate / 2;
qint64 diff = frequency - startFrequency;
float binBW = m_scannerSampleRate / (float)m_fftSize;
// avoid spectrum edges where there may be aliasing from half-band filters
if ((diff >= m_scannerSampleRate / 8) && (diff < m_scannerSampleRate * 7 / 8))
{
int bin = std::round(diff / binBW);
int channelBins;
if (m_settings.m_frequencySettings[i].m_channelBandwidth.isEmpty()) {
channelBins = m_binsPerChannel;
} else {
int channelBW = m_settings.getChannelBandwidth(&m_settings.m_frequencySettings[i]);
channelBins = m_fftSize / (m_scannerSampleRate / (float)channelBW);
}
Real voiceLevel = 0.0;
if (m_settings.m_voiceSquelchType == FreqScannerSettings::VoiceLsb) {
voiceLevel = voiceActivityLevel(bin, channelBins, true);
} else if (m_settings.m_voiceSquelchType == FreqScannerSettings::VoiceUsb) {
voiceLevel = voiceActivityLevel(bin, channelBins, false);
}
if (voiceLevel > 0.0) {
m_voiceLevelSum[i] += voiceLevel;
m_voiceLevelCount[i]++;
}
}
}
}
// Reorder (so negative frequencies are first) and average
int halfSize = m_fftSize / 2;
for (int i = 0; i < halfSize; i++) {
m_fftAverage.storeAndGetAvg(m_magSq[i], magSq(i + halfSize), i);
}
for (int i = 0; i < halfSize; i++) {
m_fftAverage.storeAndGetAvg(m_magSq[i + halfSize], magSq(i), i + halfSize);
}
if (m_fftAverage.nextAverage())
{
// Send results to channel
if (getMessageQueueToChannel() && (m_settings.m_channelBandwidth != 0) && (m_binsPerChannel != 0))
{
FreqScanner::MsgScanResult* msg = FreqScanner::MsgScanResult::create(m_fftStartTime);
QList<FreqScanner::MsgScanResult::ScanResult>& results = msg->getScanResults();
for (int i = 0; i < m_settings.m_frequencySettings.size(); i++)
{
if (m_settings.m_frequencySettings[i].m_enabled)
{
qint64 frequency = m_settings.m_frequencySettings[i].m_frequency;
qint64 startFrequency = m_centerFrequency - m_scannerSampleRate / 2;
qint64 diff = frequency - startFrequency;
float binBW = m_scannerSampleRate / (float)m_fftSize;
// Ignore results in upper and lower 12.5%, as there may be aliasing here from half-band filters
if ((diff >= m_scannerSampleRate / 8) && (diff < m_scannerSampleRate * 7 / 8))
{
int bin = std::round(diff / binBW); // Bin corresponding to the frequency
int channelBins; // Number of bins in the channel containing the frequency.
// This is either the default (m_binsPerChannel)
// or calculated based on the channel bandwidth if specified in settings for this frequency
if (m_settings.m_frequencySettings[i].m_channelBandwidth.isEmpty())
{
channelBins = m_binsPerChannel;
}
else
{
int channelBW = m_settings.getChannelBandwidth(&m_settings.m_frequencySettings[i]);
channelBins = m_fftSize / (m_scannerSampleRate / (float)channelBW);
}
// Calculate power at that frequency
Real power;
if (m_settings.m_measurement == FreqScannerSettings::PEAK) {
power = peakPower(bin, channelBins);
} else {
power = totalPower(bin, channelBins);
}
// Use averaged voice activity level from individual FFTs
Real voiceLevel = 0.0;
if ((m_settings.m_voiceSquelchType == FreqScannerSettings::VoiceLsb ||
m_settings.m_voiceSquelchType == FreqScannerSettings::VoiceUsb) &&
m_voiceLevelCount[i] > 0)
{
voiceLevel = m_voiceLevelSum[i] / m_voiceLevelCount[i];
if (voiceLevel > m_settings.m_voiceSquelchThreshold) {
qDebug() << "FreqScannerSink::processOneSample: freq"
<< frequency + (m_settings.m_voiceSquelchType == FreqScannerSettings::VoiceLsb ? 1500 : -1500)
<< "voiceLevel" << voiceLevel << "count" << m_voiceLevelCount[i];
}
}
FreqScanner::MsgScanResult::ScanResult result = {frequency, power, voiceLevel};
results.append(result);
}
}
}
getMessageQueueToChannel()->push(msg);
}
m_averageCount = 0;
m_fftStartTime = QDateTime::currentDateTime();
// Reset voice level accumulators for next averaging period
m_voiceLevelSum.fill(0.0);
m_voiceLevelCount.fill(0);
}
m_fftCounter = 0;
}
}
// Calculate total power in a channel containing the specified bin (i.e. sums adjacent bins in the same channel)
Real FreqScannerSink::totalPower(int bin, int channelBins) const
{
// Skip bin between halfway between channels
// Then skip first and last bins, to avoid spectral leakage (particularly at DC)
int startBin = bin - channelBins / 2 + 1 + 1;
Real magSqSum = 0.0f;
for (int i = 0; i < channelBins - 2 - 1; i++) {
int idx = startBin + i;
if ((idx < 0) || (idx >= m_fftSize)) {
continue;
}
magSqSum += m_magSq[idx];
}
Real db = CalcDb::dbPower(magSqSum);
return db;
}
// Calculate peak power in a channel containing the specified bin
Real FreqScannerSink::peakPower(int bin, int channelBins) const
{
// Skip bin between halfway between channels
// Then skip first and last bins, to avoid spectral leakage (particularly at DC)
int startBin = bin - channelBins/2 + 1 + 1;
Real maxMagSq = std::numeric_limits<Real>::min();
for (int i = 0; i < channelBins - 2 - 1; i++)
{
int idx = startBin + i;
if ((idx < 0) || (idx >= m_fftSize)) {
continue;
}
//qDebug() << "idx:" << idx << "power:" << CalcDb::dbPower(m_magSq[idx]);
maxMagSq = std::max(maxMagSq, m_magSq[idx]);
}
Real db = CalcDb::dbPower(maxMagSq);
return db;
}
Real FreqScannerSink::magSq(int bin) const
{
Complex c = m_fft->out()[bin];
Real v = c.real() * c.real() + c.imag() * c.imag();
Real magsq = v / (m_fftSize * m_fftSize);
return magsq;
}
// Compute magSq from raw FFT output with reordering (negative frequencies first)
Real FreqScannerSink::magSqFromRawFFT(int bin) const
{
// m_magSq is reordered: negative freqs first, then positive
// m_fft->out() is in standard FFT order: DC, positive freqs, negative freqs
int halfSize = m_fftSize / 2;
int fftBin;
if (bin < halfSize) {
// Negative frequencies: map to second half of FFT output
fftBin = bin + halfSize;
} else {
// Positive frequencies: map to first half of FFT output
fftBin = bin - halfSize;
}
return magSq(fftBin);
}
void FreqScannerSink::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, int scannerSampleRate, int fftSize, int binsPerChannel, bool force)
{
qDebug() << "FreqScannerSink::applyChannelSettings:"
<< " channelSampleRate: " << channelSampleRate
<< " channelFrequencyOffset: " << channelFrequencyOffset
<< " scannerSampleRate: " << scannerSampleRate
<< " fftSize: " << fftSize
<< " binsPerChannel: " << binsPerChannel;
if ((m_channelFrequencyOffset != channelFrequencyOffset) ||
(m_channelSampleRate != channelSampleRate) || force)
{
m_nco.setFreq(-channelFrequencyOffset, channelSampleRate);
}
if ((m_channelSampleRate != channelSampleRate) || (m_scannerSampleRate != scannerSampleRate) || force)
{
m_interpolator.create(16, channelSampleRate, scannerSampleRate / 2.2); // Filter potential aliasing resulting from half-band filters
m_interpolatorDistance = (Real) channelSampleRate / (Real)scannerSampleRate;
m_interpolatorDistanceRemain = m_interpolatorDistance;
}
if ((m_fftSize != fftSize) || force)
{
FFTFactory* fftFactory = DSPEngine::instance()->getFFTFactory();
if (m_fftSequence >= 0) {
fftFactory->releaseEngine(m_fftSize, false, m_fftSequence);
}
m_fftSequence = fftFactory->getEngine(fftSize, false, &m_fft);
m_fftCounter = 0;
m_fftStartTime = QDateTime::currentDateTime();
m_fftWindow.create(FFTWindow::Hanning, fftSize);
int averages = m_settings.m_scanTime * scannerSampleRate / 2 / fftSize;
m_fftAverage.resize(fftSize, averages);
m_magSq.resize(fftSize);
// Resize voice level accumulators to match frequency count
int freqCount = m_settings.m_frequencySettings.size();
m_voiceLevelSum.resize(freqCount);
m_voiceLevelCount.resize(freqCount);
m_voiceLevelSum.fill(0.0);
m_voiceLevelCount.fill(0);
// Allocate cepstral FFT engines for formant detection
// Size needs to be power-of-2 and >= channel bins (typically ~100-200 bins)
// Use conservative size to handle most SSB channels
int maxChannelBins = std::max(binsPerChannel * 2, 256);
int cepstrumSize = 1;
while (cepstrumSize < maxChannelBins) {
cepstrumSize <<= 1;
}
if (m_cepstrumSize != cepstrumSize) {
// Release old engines if size changed
if (m_cepstrumSequenceInverse >= 0) {
fftFactory->releaseEngine(m_cepstrumSize, true, m_cepstrumSequenceInverse);
}
if (m_cepstrumSequenceForward >= 0) {
fftFactory->releaseEngine(m_cepstrumSize, false, m_cepstrumSequenceForward);
}
// Allocate new engines
m_cepstrumSequenceInverse = fftFactory->getEngine(cepstrumSize, true, &m_cepstrumFFTInverse);
m_cepstrumSequenceForward = fftFactory->getEngine(cepstrumSize, false, &m_cepstrumFFTForward);
m_cepstrumSize = cepstrumSize;
qDebug() << "FreqScannerSink::applyChannelSettings: Allocated cepstral FFT engines, size:" << cepstrumSize;
}
}
m_channelSampleRate = channelSampleRate;
m_channelFrequencyOffset = channelFrequencyOffset;
m_scannerSampleRate = scannerSampleRate;
m_fftSize = fftSize;
m_binsPerChannel = binsPerChannel;
}
void FreqScannerSink::applySettings(const FreqScannerSettings& settings, const QStringList& settingsKeys, bool force)
{
qDebug() << "FreqScannerSink::applySettings:"
<< settings.getDebugString(settingsKeys, force)
<< " force: " << force;
if (settingsKeys.contains("scanTime") || force)
{
int averages = settings.m_scanTime * m_scannerSampleRate / 2 / m_fftSize;
m_fftAverage.resize(m_fftSize, averages);
}
if (force) {
m_settings = settings;
} else {
m_settings.applySettings(settingsKeys, settings);
}
}
// Voice activity detection for SSB signals
// Detects voice by looking for formant-like structure using spectral smoothing
// Returns a value from 0.0 (no voice) to 1.0 (strong voice signature)
Real FreqScannerSink::voiceActivityLevel(int bin, int channelBins, bool isLSB)
{
// Voice band in SSB is typically 100-3000 Hz from carrier
// We look for 2-4 formant peaks in the smoothed spectral envelope
int startBin = bin - channelBins / 2 + 1;
int endBin = startBin + channelBins - 1;
if (startBin < 0 || endBin >= m_fftSize) {
return 0.0;
}
// Calculate bin bandwidth in Hz
float binBW = m_scannerSampleRate / (float)m_fftSize;
// For LSB, spectrum is reversed - flip the search direction
int carrierBin = isLSB ? endBin : startBin;
// Get formant envelope using spectral smoothing
// This separates vocal tract resonances from pitch harmonics
QVector<Real> formantEnvelope;
Real pitchHz = 0.0;
getFormantEnvelope(startBin, endBin, formantEnvelope, &pitchHz);
if (formantEnvelope.isEmpty()) {
qDebug() << "FreqScannerSink::voiceActivityLevel formantEnvelope is empty!";
return 0.0;
}
// Calculate noise floor from formant envelope
Real noiseFloor = 0.0;
Real maxEnv = 0.0;
for (const Real env : formantEnvelope) {
noiseFloor += env;
maxEnv = std::max(maxEnv, env);
}
noiseFloor = formantEnvelope.size() > 0 ? noiseFloor / formantEnvelope.size() : 1e-12;
// For voice, we need reasonably high peaks relative to noise
// Use 4 dB (2.5x) above average as threshold for peak detection
// Additional validation: peak-to-noise ratio
// If max is not sufficiently higher than average, signal quality is poor
float peakToNoiseRatio = maxEnv / noiseFloor;
// Lower threshold: require at least 1.2x peak-to-noise ratio (only ~1.6dB)
// After heavy smoothing, formants appear as gentle bumps, not sharp peaks
if (peakToNoiseRatio < 1.2) {
// Signal is essentially all noise
return 0.0;
}
// For peak detection, use much lower threshold since smoothed spectral peaks are gentle
// Use 1.1x noise floor to catch formant peaks in the smoothed envelope
Real threshold = noiseFloor * 1.1;
// Find formant peaks in the smoothed envelope
QVector<int> formantBins;
QVector<Real> formantMags;
// Simple peak detection in formant envelope
const qsizetype formantEnvelopeSize = formantEnvelope.size();
for (qsizetype i = 1; i + 1 < formantEnvelopeSize; ++i)
{
Real prev = formantEnvelope[i - 1];
Real curr = formantEnvelope[i];
Real next = formantEnvelope[i + 1];
// Local maximum above threshold
if (curr > prev && curr > next && curr > threshold)
{
int absBin = startBin + static_cast<int>(i);
// Use SIGNED offset to distinguish USB from LSB and reject mistuned signals
// USB: formants at positive offset (100-3000 Hz above carrier)
// LSB: formants at negative offset (-3000 to -100 Hz below carrier)
float freqOffset = (absBin - carrierBin) * binBW;
// Check appropriate sideband for voice: USB uses positive, LSB uses negative
// This rejects signals tuned to wrong sideband (e.g., 1kHz offset)
bool inVoiceBand = isLSB ? (freqOffset <= -100.0 && freqOffset >= -3000.0)
: (freqOffset >= 100.0 && freqOffset <= 3000.0);
if (inVoiceBand)
{
formantBins.append(absBin);
formantMags.append(curr);
}
}
}
// Voice requires 2-4 formants
if (formantBins.size() < 2) {
return 0.0;
}
// Sort formants by frequency offset (not bin order) to handle LSB reversal
// Convert to absolute frequencies since F1/F2 validation expects positive values
QVector<float> formantFreqs;
QVector<int> formantIndices;
const qsizetype formantBinsSize = formantBins.size();
for (qsizetype i = 0; i < formantBinsSize; ++i)
{
// Use absolute value for formant frequency analysis (F1, F2 ranges are defined as positive)
float freqOffset = std::abs(formantBins[i] - carrierBin) * binBW;
formantFreqs.append(freqOffset);
formantIndices.append(static_cast<int>(i));
}
// Simple insertion sort by frequency
const qsizetype formantFreqsSize = formantFreqs.size();
for (qsizetype i = 1; i < formantFreqsSize; ++i)
{
for (qsizetype j = i; j > 0 && formantFreqs[j] < formantFreqs[j - 1]; --j)
{
std::swap(formantFreqs[j], formantFreqs[j - 1]);
std::swap(formantIndices[j], formantIndices[j - 1]);
}
}
// Merge peaks that are too close together (within 400 Hz)
// Real formants have minimum separation of 400-500 Hz in SSB voice
// So anything closer is ripple within a single formant
// We keep the peak with highest magnitude and remove others
QVector<float> mergedFormantFreqs;
QVector<int> mergedFormantIndices;
const float minFormantSpacing = 400.0; // Hz - minimum spacing between real formants
const qsizetype sortedFormantFreqsSize = formantFreqs.size();
for (qsizetype i = 0; i < sortedFormantFreqsSize; ++i)
{
if (i == 0 || formantFreqs[i] - mergedFormantFreqs.back() >= minFormantSpacing)
{
// This is a new formant (far enough from previous)
mergedFormantFreqs.append(formantFreqs[i]);
mergedFormantIndices.append(formantIndices[i]);
}
else
{
// This peak is too close to the previous one - merge by keeping highest magnitude
int prevIdx = mergedFormantIndices.back();
int currIdx = formantIndices[i];
if (formantMags[currIdx] > formantMags[prevIdx])
{
// Replace with higher magnitude peak
mergedFormantFreqs.back() = formantFreqs[i];
mergedFormantIndices.back() = currIdx;
}
}
}
formantFreqs = mergedFormantFreqs;
formantIndices = mergedFormantIndices;
// After merging, we should still have at least 2 formants for voice
if (formantFreqs.size() < 2) {
return 0.0;
}
// Check formant spacing (voice formants should be 400-1500 Hz apart)
// F1-F2 spacing is typically 600-1200 Hz
bool goodSpacing = false;
const qsizetype mergedFormantFreqsSize = formantFreqs.size();
for (qsizetype i = 0; i + 1 < mergedFormantFreqsSize; ++i)
{
float spacing = formantFreqs[i + 1] - formantFreqs[i];
if (spacing >= 400.0 && spacing <= 1500.0) {
goodSpacing = true;
break;
}
}
if (!goodSpacing) {
return 0.0; // Formants too close or too far apart
}
// Check for F1 formant in expected range (300-1000 Hz from carrier)
// This is critical for voice detection
// F1 should be the lowest frequency formant
bool hasF1 = false;
Real f1Mag = 0.0;
int f1Idx = -1;
if (formantFreqs.size() > 0 && formantFreqs[0] >= 300.0 && formantFreqs[0] <= 1000.0)
{
hasF1 = true;
f1Idx = formantIndices[0];
f1Mag = formantMags[f1Idx];
}
if (!hasF1) {
return 0.0; // No F1 formant - not voice or mistuned
}
// Check for F2 formant in expected range (900-2500 Hz from carrier)
// F2 should be higher frequency than F1 AND 400-1500 Hz away from F1
bool hasF2 = false;
Real f2Mag = 0.0;
int f2Idx = -1;
float f2Freq = 0.0;
const qsizetype validatedFormantFreqsSize = formantFreqs.size();
for (qsizetype i = 1; i < validatedFormantFreqsSize; ++i)
{
float freqOffset = formantFreqs[i];
float f1ToF2Spacing = freqOffset - formantFreqs[0];
// F2 must be: in range, higher than F1, and properly spaced from F1
if (freqOffset >= 900.0 && freqOffset <= 2500.0 &&
f1ToF2Spacing >= 400.0 && f1ToF2Spacing <= 1500.0)
{
hasF2 = true;
f2Idx = formantIndices[i];
f2Freq = freqOffset;
f2Mag = formantMags[f2Idx];
break; // Take the first valid F2
}
}
if (!hasF2) {
return 0.0; // No F2 formant - not voice or mistuned
}
// Reject strongly tonal spectra (typical CW / single-tone signals).
// Voice in SSB should have broader spectral spread, lower peak dominance,
// and non-negligible spectral flatness across the 100-3000 Hz voice band.
Real voiceBandTotalEnergy = 0.0;
Real voiceBandPeakEnergy = 0.0;
Real voiceBandSecondPeakEnergy = 0.0;
Real voiceBandLogEnergySum = 0.0;
Real voiceBandWeightedFreqSum = 0.0;
Real voiceBandWeightedFreqSqSum = 0.0;
QVector<Real> voiceBandEnergies;
voiceBandEnergies.reserve(std::max(0, endBin - startBin + 1));
int voiceBandBins = 0;
for (int absBin = startBin; absBin <= endBin; absBin++)
{
float signedOffset = (absBin - carrierBin) * binBW;
float voiceOffset = isLSB ? -signedOffset : signedOffset;
if (voiceOffset < 100.0f || voiceOffset > 3000.0f) {
continue;
}
Real binEnergy = magSqFromRawFFT(absBin);
Real safeEnergy = std::max(binEnergy, (Real) 1e-20);
voiceBandEnergies.append(safeEnergy);
voiceBandTotalEnergy += safeEnergy;
if (safeEnergy > voiceBandPeakEnergy)
{
voiceBandSecondPeakEnergy = voiceBandPeakEnergy;
voiceBandPeakEnergy = safeEnergy;
}
else if (safeEnergy > voiceBandSecondPeakEnergy)
{
voiceBandSecondPeakEnergy = safeEnergy;
}
voiceBandLogEnergySum += std::log(safeEnergy);
voiceBandWeightedFreqSum += safeEnergy * voiceOffset;
voiceBandWeightedFreqSqSum += safeEnergy * voiceOffset * voiceOffset;
voiceBandBins++;
}
if (voiceBandBins <= 0 || voiceBandTotalEnergy <= 0.0) {
return 0.0;
}
Real voiceBandMeanEnergy = voiceBandTotalEnergy / voiceBandBins;
Real voiceBandGeometricMean = std::exp(voiceBandLogEnergySum / voiceBandBins);
Real voiceBandMeanFreq = voiceBandWeightedFreqSum / voiceBandTotalEnergy;
Real voiceBandMeanFreqSq = voiceBandWeightedFreqSqSum / voiceBandTotalEnergy;
Real voiceBandVariance = std::max((Real) 0.0, voiceBandMeanFreqSq - voiceBandMeanFreq * voiceBandMeanFreq);
float voiceBandRmsSpreadHz = std::sqrt(voiceBandVariance);
float spectralFlatness = voiceBandMeanEnergy > 0.0 ? (voiceBandGeometricMean / voiceBandMeanEnergy) : 0.0f;
float peakFraction = voiceBandTotalEnergy > 0.0 ? (voiceBandPeakEnergy / voiceBandTotalEnergy) : 1.0f;
float peak12Ratio = voiceBandSecondPeakEnergy > 0.0 ? (voiceBandPeakEnergy / voiceBandSecondPeakEnergy) : 1000.0f;
int significantBins = 0;
Real significantThreshold = voiceBandMeanEnergy * 1.8;
for (const Real& binEnergy : voiceBandEnergies)
{
if (binEnergy > significantThreshold) {
significantBins++;
}
}
if ((peakFraction > 0.45f) || (spectralFlatness < 0.025f) || (significantBins < 5)) {
return 0.0;
}
// Additional strong-CW rejection:
// - very narrow energy spread in voice band
// - one dominant spectral line overwhelmingly larger than the second line
if ((voiceBandRmsSpreadHz < 220.0f) || ((peak12Ratio > 8.0f) && (significantBins < 9))) {
return 0.0;
}
float tonalPenalty = 1.0f;
if (peakFraction > 0.30f) {
tonalPenalty *= 0.6f;
}
if (spectralFlatness < 0.06f) {
tonalPenalty *= 0.7f;
}
if (voiceBandRmsSpreadHz < 350.0f) {
tonalPenalty *= 0.55f;
}
if (peak12Ratio > 4.0f) {
tonalPenalty *= 0.60f;
}
// CW rejection end
// Additional F1 plausibility checks.
// Prevent a tiny low-frequency ripple from being accepted as F1 when
// dominant formant energy is shifted high (e.g. around 2 kHz).
Real strongestFormantMag = 0.0;
float strongestFormantFreq = 0.0f;
for (int i = 0; i < formantFreqs.size(); i++)
{
int idx = formantIndices[i];
if (formantMags[idx] > strongestFormantMag)
{
strongestFormantMag = formantMags[idx];
strongestFormantFreq = formantFreqs[i];
}
}
const float minF1ToStrongestRatio = 0.35f;
const float dominantHighFormantHz = 1200.0f;
bool weakF1 = f1Mag < strongestFormantMag * minF1ToStrongestRatio;
bool dominantIsHigh = strongestFormantFreq >= dominantHighFormantHz;
if (weakF1 && dominantIsHigh) {
return 0.0;
}
// F1 must have a minimum contrast above noise floor.
if (f1Mag < noiseFloor * 1.25f) {
return 0.0;
}
// Additional F1 plausibility checks - end
// Calculate voice activity score based on formant characteristics
// Voice is indicated by presence of F1 and F2 formants - this is the primary voice signature
// Harmonics are less reliable in SSB due to spectral properties and noise
float score = 0.0;
// Base score from number of formants (2-4 formants typical for voice)
// Formants are the most reliable voice indicator
// Use merged formant count, not raw peak count
float formantScore = std::min(formantFreqs.size() / 3.0f, 1.0f);
score += formantScore * 0.6; // 60% weight
// Score from formant magnitude (strong formants = strong voice)
// Higher magnitudes indicate clearer voice detection
// Make this threshold appropriately high to prefer strong signals
float magnitudeScore = std::min((f1Mag + f2Mag) / (noiseFloor * 6.0f), 1.0f);
score += magnitudeScore * 0.4; // 40% weight
// Apply a soft pitch-based weighting. Pitch helps detect detuning, but should not gate voice.
const float minPitchHz = 70.0f;
const float maxPitchHz = 300.0f;
const float softMinPitchHz = 50.0f;
const float softMaxPitchHz = 400.0f;
float pitchScore = 0.7f;
float harmonicAlignmentScore = 0.75f;
float formantIndexScore = 0.8f;
if (pitchHz > 0.0f) {
if (pitchHz >= minPitchHz && pitchHz <= maxPitchHz) {
pitchScore = 1.0f;
} else if (pitchHz >= softMinPitchHz && pitchHz <= softMaxPitchHz) {
if (pitchHz < minPitchHz) {
pitchScore = 0.7f + 0.3f * (pitchHz - softMinPitchHz) / (minPitchHz - softMinPitchHz);
} else {
pitchScore = 0.7f + 0.3f * (softMaxPitchHz - pitchHz) / (softMaxPitchHz - maxPitchHz);
}
} else {
pitchScore = 0.6f;
}
// Check harmonic-comb alignment against the estimated pitch.
// A wrong carrier offset shifts all harmonics by a constant frequency,
// so they no longer align with integer multiples of pitch.
const float harmonicToleranceHz = std::max(2.0f * binBW, 0.18f * pitchHz);
Real rawNoiseFloor = 0.0;
int rawBinCount = 0;
for (int absBin = startBin; absBin <= endBin; absBin++)
{
float signedOffset = (absBin - carrierBin) * binBW;
float voiceOffset = isLSB ? -signedOffset : signedOffset;
if (voiceOffset >= 100.0f && voiceOffset <= 3000.0f)
{
rawNoiseFloor += magSqFromRawFFT(absBin);
rawBinCount++;
}
}
rawNoiseFloor = rawBinCount > 0 ? rawNoiseFloor / rawBinCount : 0.0;
Real alignedEnergy = 0.0;
Real totalEnergy = 0.0;
for (int absBin = startBin; absBin <= endBin; absBin++)
{
float signedOffset = (absBin - carrierBin) * binBW;
float voiceOffset = isLSB ? -signedOffset : signedOffset;
if (voiceOffset < 100.0f || voiceOffset > 3000.0f) {
continue;
}
Real binEnergy = magSqFromRawFFT(absBin);
if (binEnergy <= rawNoiseFloor * 1.2f) {
continue;
}
float residue = std::fmod(voiceOffset, pitchHz);
if (residue < 0.0f) {
residue += pitchHz;
}
float harmonicDistance = std::min(residue, pitchHz - residue);
Real weightedEnergy = std::max(binEnergy - rawNoiseFloor, (Real) 0.0);
totalEnergy += weightedEnergy;
if (harmonicDistance <= harmonicToleranceHz) {
alignedEnergy += weightedEnergy;
}
}
if (totalEnergy > 0.0)
{
float harmonicAlignment = alignedEnergy / totalEnergy;
float normalizedAlignment = (harmonicAlignment - 0.20f) / 0.45f;
normalizedAlignment = std::max(0.0f, std::min(normalizedAlignment, 1.0f));
harmonicAlignmentScore = 0.5f + 0.5f * normalizedAlignment;
}
// Check formant harmonic-index plausibility.
// Wrong carrier tuning that shifts spectrum down can make F2/F3 appear as F1/F2.
// In that case the implied harmonic indices become unusually high.
float f1HarmonicIndex = formantFreqs[0] / pitchHz;
float f2HarmonicIndex = f2Freq / pitchHz;
float harmonicGap = f2HarmonicIndex - f1HarmonicIndex;
float f1IndexScore = 1.0f;
if (f1HarmonicIndex < 2.0f) {
f1IndexScore = std::max(0.0f, (f1HarmonicIndex - 1.0f) / 1.0f);
} else if (f1HarmonicIndex > 18.0f) {
f1IndexScore = std::max(0.0f, (26.0f - f1HarmonicIndex) / 8.0f);
}
float f2IndexScore = 1.0f;
if (f2HarmonicIndex < 5.0f) {
f2IndexScore = std::max(0.0f, (f2HarmonicIndex - 3.0f) / 2.0f);
} else if (f2HarmonicIndex > 36.0f) {
f2IndexScore = std::max(0.0f, (44.0f - f2HarmonicIndex) / 8.0f);
}
float gapScore = 1.0f;
if (harmonicGap < 3.0f) {
gapScore = std::max(0.0f, (harmonicGap - 1.0f) / 2.0f);
} else if (harmonicGap > 22.0f) {
gapScore = std::max(0.0f, (28.0f - harmonicGap) / 6.0f);
}
formantIndexScore = std::max(0.4f, 0.25f + 0.75f * (0.40f * f1IndexScore + 0.40f * f2IndexScore + 0.20f * gapScore));
}
score *= tonalPenalty * pitchScore * harmonicAlignmentScore * formantIndexScore;
// Clamp to [0, 1]
score = std::max(0.0f, std::min(score, 1.0f));
return score;
}
// Compute formant envelope using cepstral liftering
// This separates vocal tract resonances (formants) from pitch harmonics
// Method: log spectrum → IFFT → lifter → FFT → exp
void FreqScannerSink::getFormantEnvelope(int startBin, int endBin, QVector<Real>& envelope, Real *pitchHz)
{
if (pitchHz) {
*pitchHz = 0.0;
}
if (startBin < 0 || endBin >= m_fftSize || startBin > endBin) {
envelope.clear();
return;
}
int numBins = endBin - startBin + 1;
envelope.resize(numBins);
// Check if cepstral FFT engines are available and large enough
if (!m_cepstrumFFTInverse || !m_cepstrumFFTForward || numBins > m_cepstrumSize) {
// Fallback: return simple log/exp without cepstral processing
for (int i = 0; i < numBins; i++) {
Real magSq = magSqFromRawFFT(startBin + i);
envelope[i] = std::sqrt(std::max(magSq, (Real)1e-12));
}
return;
}
// Step 1: Compute log magnitude spectrum
QVector<Real> logMag(numBins);
Real minLog = -10.0; // Floor to avoid log(0)
Real sumMagSq = 0.0;
for (int i = 0; i < numBins; i++)
{
Real magSq = magSqFromRawFFT(startBin + i);
sumMagSq += magSq;
Real mag = std::sqrt(std::max(magSq, (Real)1e-12));
logMag[i] = std::log(mag);
if (logMag[i] < minLog) {
logMag[i] = minLog;
}
}
// Step 2: Apply cepstral liftering for better source-filter separation
// Cepstral analysis separates:
// - Voice pitch (high quefrency peak)
// - Formants (low quefrency components)
// Liftering = low-pass filtering in quefrency domain
// Copy log magnitude to inverse FFT input (real data, symmetric spectrum)
// For real cepstrum, we need a symmetric spectrum:
// [DC, positive freqs, Nyquist, negative freqs (mirror)]
// DC component
m_cepstrumFFTInverse->in()[0] = Complex(logMag[0], 0.0);
// Positive frequencies
int halfBins = std::min(numBins, m_cepstrumSize / 2);
for (int i = 1; i < halfBins; i++) {
m_cepstrumFFTInverse->in()[i] = Complex(logMag[i], 0.0);
}
// Nyquist (if we have space)
if (m_cepstrumSize > halfBins) {
m_cepstrumFFTInverse->in()[halfBins] = Complex(numBins > halfBins ? logMag[halfBins] : logMag[halfBins-1], 0.0);
}
// Negative frequencies (mirror of positive)
for (int i = 1; i < halfBins; i++) {
m_cepstrumFFTInverse->in()[m_cepstrumSize - i] = Complex(logMag[i], 0.0);
}
// Zero-pad the middle if needed
for (int i = halfBins + 1; i < m_cepstrumSize - halfBins; i++) {
m_cepstrumFFTInverse->in()[i] = Complex(0.0, 0.0);
}
// Step 3: IFFT to get cepstrum (quefrency domain)
m_cepstrumFFTInverse->transform();
// Step 4: Estimate pitch from the cepstrum before liftering.
// Pitch appears as a peak at higher quefrencies (around 3-14 ms).
float binBW = m_scannerSampleRate / (float)m_fftSize;
float quefrencyResolution = 1.0f / (m_cepstrumSize * binBW); // seconds per bin in quefrency
if (pitchHz) {
const float minPitchHz = 70.0f;
const float maxPitchHz = 300.0f;
const float minQuefrency = 1.0f / maxPitchHz;
const float maxQuefrency = 1.0f / minPitchHz;
const int minBin = std::max(1, (int)std::ceil(minQuefrency / quefrencyResolution));
const int maxBin = std::min(m_cepstrumSize / 2, (int)std::floor(maxQuefrency / quefrencyResolution));
Real maxVal = 0.0;
int maxIdx = -1;
for (int i = minBin; i <= maxBin; i++) {
Real val = std::abs(m_cepstrumFFTInverse->out()[i].real());
if (val > maxVal) {
maxVal = val;
maxIdx = i;
}
}
if (maxIdx > 0) {
*pitchHz = 1.0f / (maxIdx * quefrencyResolution);
}
}
// Step 5: Apply lifter (low-pass filter in quefrency domain)
// Lifter cutoff: keep low quefrencies (formant envelope), remove high quefrencies (pitch harmonics)
// Typical pitch periods: 3-10 ms (100-330 Hz F0)
// We want to remove quefrencies corresponding to pitch harmonics
// Lifter cutoff in seconds (quefrency): keep components below this
// Use much lower cutoff - we only need to keep very low quefrencies for formant envelope
// Most formant information is in the first few quefrency bins
float lifterCutoffQuefrency = 0.002f; // 2 ms (reduced from 8 ms)
int lifterCutoffBin = std::max(1, (int)(lifterCutoffQuefrency / quefrencyResolution));
// Cap the lifter cutoff to reasonable maximum (1/4 of cepstrum size)
lifterCutoffBin = std::min(lifterCutoffBin, m_cepstrumSize / 4);
// Apply lifter: keep low quefrencies, zero out high quefrencies
// Use smooth transition (raised cosine) to reduce artifacts
int transitionBins = std::max(1, lifterCutoffBin / 4);
for (int i = 0; i < m_cepstrumSize; i++) {
Real lifterWeight = 1.0;
if (i > lifterCutoffBin + transitionBins) {
lifterWeight = 0.0; // Zero out high quefrencies
} else if (i > lifterCutoffBin) {
// Smooth transition using raised cosine
float t = (float)(i - lifterCutoffBin) / transitionBins;
lifterWeight = 0.5 * (1.0 + std::cos(M_PI * t));
}
// else: lifterWeight = 1.0 (keep low quefrencies)
m_cepstrumFFTForward->in()[i] = m_cepstrumFFTInverse->out()[i] * lifterWeight;
}
// Step 6: FFT back to frequency domain (smoothed log spectrum)
m_cepstrumFFTForward->transform();
// Step 7: Convert back to linear magnitude
// Take real part of FFT output and exponentiate
// IMPORTANT: Normalize by FFT size since FFT engines don't auto-normalize
Real normalization = 1.0 / m_cepstrumSize;
for (int i = 0; i < numBins; i++)
{
Real smoothedLog = m_cepstrumFFTForward->out()[i].real() * normalization;
envelope[i] = std::exp(smoothedLog);
}
}
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