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https://github.com/saitohirga/WSJT-X.git
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60792182ad
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.
213 lines
5.8 KiB
C++
213 lines
5.8 KiB
C++
#include "soundin.h"
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#include <cstdlib>
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#include <cmath>
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#include <iomanip>
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#include <QAudioDeviceInfo>
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#include <QAudioFormat>
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#include <QAudioInput>
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#include <QSysInfo>
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#include <QDebug>
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#include "Logger.hpp"
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#include "moc_soundin.cpp"
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bool SoundInput::checkStream ()
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{
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bool result (false);
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if (m_stream)
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{
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switch (m_stream->error ())
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{
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case QAudio::OpenError:
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Q_EMIT error (tr ("An error opening the audio input device has occurred."));
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break;
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case QAudio::IOError:
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Q_EMIT error (tr ("An error occurred during read from the audio input device."));
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break;
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// case QAudio::UnderrunError:
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// Q_EMIT error (tr ("Audio data not being fed to the audio input device fast enough."));
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// break;
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case QAudio::FatalError:
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Q_EMIT error (tr ("Non-recoverable error, audio input device not usable at this time."));
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break;
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case QAudio::UnderrunError: // TODO G4WJS: stop ignoring this
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// when we find the cause on macOS
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case QAudio::NoError:
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result = true;
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break;
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}
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}
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return result;
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}
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void SoundInput::start(QAudioDeviceInfo const& device, int framesPerBuffer, AudioDevice * sink
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, unsigned downSampleFactor, AudioDevice::Channel channel)
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{
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Q_ASSERT (sink);
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stop ();
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m_sink = sink;
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QAudioFormat format (device.preferredFormat());
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// qDebug () << "Preferred audio input format:" << format;
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format.setChannelCount (AudioDevice::Mono == channel ? 1 : 2);
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format.setCodec ("audio/pcm");
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format.setSampleRate (12000 * downSampleFactor);
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format.setSampleType (QAudioFormat::SignedInt);
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format.setSampleSize (16);
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format.setByteOrder (QAudioFormat::Endian (QSysInfo::ByteOrder));
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if (!format.isValid ())
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{
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Q_EMIT error (tr ("Requested input audio format is not valid."));
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return;
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}
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else if (!device.isFormatSupported (format))
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{
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// qDebug () << "Nearest supported audio format:" << device.nearestFormat (format);
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Q_EMIT error (tr ("Requested input audio format is not supported on device."));
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return;
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}
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// qDebug () << "Selected audio input format:" << format;
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m_stream.reset (new QAudioInput {device, format});
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if (!checkStream ())
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{
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return;
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}
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connect (m_stream.data(), &QAudioInput::stateChanged, this, &SoundInput::handleStateChanged);
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connect (m_stream.data(), &QAudioInput::notify, [this] () {checkStream ();});
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//qDebug () << "SoundIn default buffer size (bytes):" << m_stream->bufferSize () << "period size:" << m_stream->periodSize ();
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// the Windows MME version of QAudioInput uses 1/5 of the buffer
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// size for period size other platforms seem to optimize themselves
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if (framesPerBuffer > 0)
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{
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m_stream->setBufferSize (m_stream->format ().bytesForFrames (framesPerBuffer));
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}
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if (m_sink->initialize (QIODevice::WriteOnly, channel))
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{
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m_stream->start (sink);
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checkStream ();
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cummulative_lost_usec_ = -1;
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LOG_DEBUG ("Selected buffer size (bytes): " << m_stream->bufferSize () << " period size: " << m_stream->periodSize ());
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}
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else
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{
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Q_EMIT error (tr ("Failed to initialize audio sink device"));
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}
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}
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void SoundInput::suspend ()
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{
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if (m_stream)
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{
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m_stream->suspend ();
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checkStream ();
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}
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}
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void SoundInput::resume ()
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{
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// qDebug() << "Resume" << fmod(0.001*QDateTime::currentMSecsSinceEpoch(),6.0);
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if (m_sink)
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{
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m_sink->reset ();
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}
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if (m_stream)
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{
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m_stream->resume ();
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checkStream ();
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}
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}
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void SoundInput::handleStateChanged (QAudio::State newState)
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{
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switch (newState)
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{
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case QAudio::IdleState:
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Q_EMIT status (tr ("Idle"));
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break;
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case QAudio::ActiveState:
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reset (false);
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Q_EMIT status (tr ("Receiving"));
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break;
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case QAudio::SuspendedState:
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Q_EMIT status (tr ("Suspended"));
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break;
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#if QT_VERSION >= QT_VERSION_CHECK (5, 10, 0)
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case QAudio::InterruptedState:
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Q_EMIT status (tr ("Interrupted"));
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break;
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#endif
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case QAudio::StoppedState:
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if (!checkStream ())
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{
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Q_EMIT status (tr ("Error"));
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}
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else
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{
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Q_EMIT status (tr ("Stopped"));
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}
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break;
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}
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}
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void SoundInput::reset (bool report_dropped_frames)
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{
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if (m_stream)
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{
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auto elapsed_usecs = m_stream->elapsedUSecs ();
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while (std::abs (elapsed_usecs - m_stream->processedUSecs ())
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> 24 * 60 * 60 * 500000ll) // half day
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{
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// QAudioInput::elapsedUSecs() wraps after 24 hours
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elapsed_usecs += 24 * 60 * 60 * 1000000ll;
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}
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// don't report first time as we don't yet known latency
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if (cummulative_lost_usec_ != std::numeric_limits<qint64>::min () && report_dropped_frames)
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{
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auto lost_usec = elapsed_usecs - m_stream->processedUSecs () - cummulative_lost_usec_;
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if (std::abs (lost_usec) > 48000 / 5)
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{
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LOG_WARN ("Detected dropped audio source samples: "
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<< m_stream->format ().framesForDuration (lost_usec)
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<< " (" << std::setprecision (4) << lost_usec / 1.e6 << " S)")
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}
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else if (std::abs (lost_usec) > 5 * 48000)
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{
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LOG_ERROR ("Detected excessive dropped audio source samples: "
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<< m_stream->format ().framesForDuration (lost_usec)
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<< " (" << std::setprecision (4) << lost_usec / 1.e6 << " S)")
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}
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}
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cummulative_lost_usec_ = elapsed_usecs - m_stream->processedUSecs ();
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}
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}
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void SoundInput::stop()
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{
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if (m_stream)
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{
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m_stream->stop ();
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}
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m_stream.reset ();
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}
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SoundInput::~SoundInput ()
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{
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stop ();
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}
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