WSJT-X/Audio/soundin.cpp
Bill Somerville 60792182ad
Environment variables to set audio buffer sizes and fix Windows Rx timing
The two environment variables:

    WSJT_RX_AUDIO_BUFFER_FRAMES
    WSJT_TX_AUDIO_BUFFER_FRAMES

each can  be defined to  an integer number which  will be used  as the
suggested audio  buffer size for  Rx and Tx respectively.  Not setting
the variable  or setting  it to  zero or less  will cause  the default
buffer size to be used, which should be a good choice for most, if not
all, systems.
2020-12-07 20:34:56 +00:00

213 lines
5.8 KiB
C++

#include "soundin.h"
#include <cstdlib>
#include <cmath>
#include <iomanip>
#include <QAudioDeviceInfo>
#include <QAudioFormat>
#include <QAudioInput>
#include <QSysInfo>
#include <QDebug>
#include "Logger.hpp"
#include "moc_soundin.cpp"
bool SoundInput::checkStream ()
{
bool result (false);
if (m_stream)
{
switch (m_stream->error ())
{
case QAudio::OpenError:
Q_EMIT error (tr ("An error opening the audio input device has occurred."));
break;
case QAudio::IOError:
Q_EMIT error (tr ("An error occurred during read from the audio input device."));
break;
// case QAudio::UnderrunError:
// Q_EMIT error (tr ("Audio data not being fed to the audio input device fast enough."));
// break;
case QAudio::FatalError:
Q_EMIT error (tr ("Non-recoverable error, audio input device not usable at this time."));
break;
case QAudio::UnderrunError: // TODO G4WJS: stop ignoring this
// when we find the cause on macOS
case QAudio::NoError:
result = true;
break;
}
}
return result;
}
void SoundInput::start(QAudioDeviceInfo const& device, int framesPerBuffer, AudioDevice * sink
, unsigned downSampleFactor, AudioDevice::Channel channel)
{
Q_ASSERT (sink);
stop ();
m_sink = sink;
QAudioFormat format (device.preferredFormat());
// qDebug () << "Preferred audio input format:" << format;
format.setChannelCount (AudioDevice::Mono == channel ? 1 : 2);
format.setCodec ("audio/pcm");
format.setSampleRate (12000 * downSampleFactor);
format.setSampleType (QAudioFormat::SignedInt);
format.setSampleSize (16);
format.setByteOrder (QAudioFormat::Endian (QSysInfo::ByteOrder));
if (!format.isValid ())
{
Q_EMIT error (tr ("Requested input audio format is not valid."));
return;
}
else if (!device.isFormatSupported (format))
{
// qDebug () << "Nearest supported audio format:" << device.nearestFormat (format);
Q_EMIT error (tr ("Requested input audio format is not supported on device."));
return;
}
// qDebug () << "Selected audio input format:" << format;
m_stream.reset (new QAudioInput {device, format});
if (!checkStream ())
{
return;
}
connect (m_stream.data(), &QAudioInput::stateChanged, this, &SoundInput::handleStateChanged);
connect (m_stream.data(), &QAudioInput::notify, [this] () {checkStream ();});
//qDebug () << "SoundIn default buffer size (bytes):" << m_stream->bufferSize () << "period size:" << m_stream->periodSize ();
// the Windows MME version of QAudioInput uses 1/5 of the buffer
// size for period size other platforms seem to optimize themselves
if (framesPerBuffer > 0)
{
m_stream->setBufferSize (m_stream->format ().bytesForFrames (framesPerBuffer));
}
if (m_sink->initialize (QIODevice::WriteOnly, channel))
{
m_stream->start (sink);
checkStream ();
cummulative_lost_usec_ = -1;
LOG_DEBUG ("Selected buffer size (bytes): " << m_stream->bufferSize () << " period size: " << m_stream->periodSize ());
}
else
{
Q_EMIT error (tr ("Failed to initialize audio sink device"));
}
}
void SoundInput::suspend ()
{
if (m_stream)
{
m_stream->suspend ();
checkStream ();
}
}
void SoundInput::resume ()
{
// qDebug() << "Resume" << fmod(0.001*QDateTime::currentMSecsSinceEpoch(),6.0);
if (m_sink)
{
m_sink->reset ();
}
if (m_stream)
{
m_stream->resume ();
checkStream ();
}
}
void SoundInput::handleStateChanged (QAudio::State newState)
{
switch (newState)
{
case QAudio::IdleState:
Q_EMIT status (tr ("Idle"));
break;
case QAudio::ActiveState:
reset (false);
Q_EMIT status (tr ("Receiving"));
break;
case QAudio::SuspendedState:
Q_EMIT status (tr ("Suspended"));
break;
#if QT_VERSION >= QT_VERSION_CHECK (5, 10, 0)
case QAudio::InterruptedState:
Q_EMIT status (tr ("Interrupted"));
break;
#endif
case QAudio::StoppedState:
if (!checkStream ())
{
Q_EMIT status (tr ("Error"));
}
else
{
Q_EMIT status (tr ("Stopped"));
}
break;
}
}
void SoundInput::reset (bool report_dropped_frames)
{
if (m_stream)
{
auto elapsed_usecs = m_stream->elapsedUSecs ();
while (std::abs (elapsed_usecs - m_stream->processedUSecs ())
> 24 * 60 * 60 * 500000ll) // half day
{
// QAudioInput::elapsedUSecs() wraps after 24 hours
elapsed_usecs += 24 * 60 * 60 * 1000000ll;
}
// don't report first time as we don't yet known latency
if (cummulative_lost_usec_ != std::numeric_limits<qint64>::min () && report_dropped_frames)
{
auto lost_usec = elapsed_usecs - m_stream->processedUSecs () - cummulative_lost_usec_;
if (std::abs (lost_usec) > 48000 / 5)
{
LOG_WARN ("Detected dropped audio source samples: "
<< m_stream->format ().framesForDuration (lost_usec)
<< " (" << std::setprecision (4) << lost_usec / 1.e6 << " S)")
}
else if (std::abs (lost_usec) > 5 * 48000)
{
LOG_ERROR ("Detected excessive dropped audio source samples: "
<< m_stream->format ().framesForDuration (lost_usec)
<< " (" << std::setprecision (4) << lost_usec / 1.e6 << " S)")
}
}
cummulative_lost_usec_ = elapsed_usecs - m_stream->processedUSecs ();
}
}
void SoundInput::stop()
{
if (m_stream)
{
m_stream->stop ();
}
m_stream.reset ();
}
SoundInput::~SoundInput ()
{
stop ();
}