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[AFV] Change output format and processing to 32 bit float

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32 bit float is what is used in C# reference and has a much higher
dynamic range. 16 bit integer was clipping very often with all the VHF
simulation applied.
This commit is contained in:
Roland Rossgotterer
2019-10-01 11:58:02 +02:00
committed by Mat Sutcliffe
parent fbb126370c
commit 240df93406
29 changed files with 77 additions and 80 deletions
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7
src/blackcore/afv/audio/callsignsampleprovider.cpp
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@@ -11,6 +11,7 @@
#include "callsignsampleprovider.h"
#include "callsigndelaycache.h"
#include "blacksound/sampleprovider/samples.h"
#include "blacksound/audioutilities.h"
#include "blackcore/afv/audio/receiversampleprovider.h"
#include <QtMath>
#include <QDebug>
@@ -60,7 +61,7 @@ namespace BlackCore
connect(&m_timer, &QTimer::timeout, this, &CallsignSampleProvider::timerElapsed);
}
int CallsignSampleProvider::readSamples(QVector<qint16> &samples, qint64 count)
int CallsignSampleProvider::readSamples(QVector<float> &samples, qint64 count)
{
int noOfSamples = m_mixer->readSamples(samples, count);
@@ -103,7 +104,7 @@ namespace BlackCore
if (delayMs > 0)
{
int phaseDelayLength = (m_audioFormat.sampleRate() / 1000) * delayMs;
QVector<qint16> phaseDelay(phaseDelayLength * 2, 0);
QVector<float> phaseDelay(phaseDelayLength * 2, 0);
m_audioInput->addSamples(phaseDelay);
}
}
@@ -131,7 +132,7 @@ namespace BlackCore
setEffects();
QVector<qint16> audio = decodeOpus(audioDto.audio);
m_audioInput->addSamples(audio);
m_audioInput->addSamples(BlackSound::convertFromShortToFloat(audio));
m_lastPacketLatch = audioDto.lastPacket;
if (audioDto.lastPacket && !m_underflow) { CallsignDelayCache::instance().success(m_callsign); }
m_lastSamplesAddedUtc = QDateTime::currentDateTimeUtc();