void AudioOss::applyQualitySettings()
{
if( hqAudio() )
{
setSampleRate( Engine::mixer()->processingSampleRate() );
unsigned int value = sampleRate();
if ( ioctl( m_audioFD, SNDCTL_DSP_SPEED, &value ) < 0 )
{
perror( "SNDCTL_DSP_SPEED" );
printf( "Couldn't set audio frequency\n" );
return;
}
if( value != sampleRate() )
{
value = mixer()->baseSampleRate();
if ( ioctl( m_audioFD, SNDCTL_DSP_SPEED, &value ) < 0 )
{
perror( "SNDCTL_DSP_SPEED" );
printf( "Couldn't set audio frequency\n" );
return;
}
setSampleRate( value );
}
}
AudioDevice::applyQualitySettings();
}
bool FileInputDataSource::open()
{
if (!mFp) {
unsigned int channel;
unsigned int sampleRate;
audio_format_type_t pcmFormat;
audio_type_t audioType;
mFp = fopen(mDataPath.c_str(), "rb");
if (!mFp) {
medvdbg("file open failed error : %d\n", errno);
return false;
}
audioType = utils::getAudioTypeFromPath(mDataPath);
setAudioType(audioType);
switch (audioType) {
case AUDIO_TYPE_MP3:
case AUDIO_TYPE_AAC:
if (!utils::header_parsing(mFp, audioType, &channel, &sampleRate, NULL)) {
medvdbg("header parsing failed\n");
return false;
}
setSampleRate(sampleRate);
setChannels(channel);
break;
case AUDIO_TYPE_WAVE:
if (!utils::header_parsing(mFp, audioType, &channel, &sampleRate, &pcmFormat)) {
medvdbg("header parsing failed\n");
return false;
}
if (fseek(mFp, WAVE_HEADER_LENGTH, SEEK_SET) != 0) {
medvdbg("file seek failed error\n");
return false;
}
setSampleRate(sampleRate);
setChannels(channel);
setPcmFormat(pcmFormat);
break;
case AUDIO_TYPE_FLAC:
/* To be supported */
break;
default:
/* Don't set any decoder for unsupported formats */
break;
}
return true;
}
/** return true if mFp is not null, because it means it using now */
return true;
}
void GladiatorIMU::handleCommand(gladiator_cmd_t* cmd)
{
uint8_t* payload;
switch (cmd->command)
{
case CMD_LOADK:
printf("GladiatorIMU: Loading filter K: %f\n", *((float*)&cmd->payload));
loadFilterK(*((float*)&cmd->payload));
break;
case CMD_SETRATE:
printf("GladiatorIMU: Setting rate %dHz\n", cmd->payload);
switch (cmd->payload)
{
case 100:
setSampleRate(MODE_IMU_100HZ);
break;
case 200:
setSampleRate(MODE_IMU_200HZ);
break;
case 500:
setSampleRate(MODE_IMU_500HZ);
break;
case 1000:
setSampleRate(MODE_IMU_1000HZ);
break;
}
break;
case CMD_GETBOARD:
printf("GladiatorIMU: Returning board number bytes\n");
cmd->payload = 0;
payload = (uint8_t*)&(cmd->payload);
payload[0] = boardNumber[0];
payload[1] = boardNumber[1];
payload[2] = boardNumber[2];
payload[3] = boardNumber[3];
break;
case CMD_GETSTATUS:
printf("GladiatorIMU: Returning status bytes\n");
cmd->payload = 0;
payload = (uint8_t*)&(cmd->payload);
payload[0] = statusByte[0];
payload[1] = statusByte[1];
payload[2] = filterNumber;
payload[3] = 0;
break;
default:
printf("GladiatorIMU: Unknown command: 0x%d\n", cmd->command);
break;
}
cmd->signal();
}
void AudioJack::applyQualitySettings()
{
if( hqAudio() )
{
setSampleRate( engine::mixer()->processingSampleRate() );
if( jack_get_sample_rate( m_client ) != sampleRate() )
{
setSampleRate( jack_get_sample_rate( m_client ) );
}
}
AudioDevice::applyQualitySettings();
}
GenericTimeDataParams::GenericTimeDataParams(DataUiHandlerProperty * baseProperty, TimePlotParams* params, QObject *parent ) :
DataUiHandlerProperty(baseProperty,parent),
m_startTime(0),
m_name(QString("no name")),
m_curveEnabled(true)
{
if (params) {
setSampleRate(params->sampleRate());
setMaxDuration(params->maxDuration());
} else {
setSampleRate(TIMEDATA_DEFAULT_SR);
setMaxDuration(TIMEDATA_DEFAULT_PROJECT_TIME);
}
}
Result SoundSourceSndFile::open() {
#ifdef __WINDOWS__
// Pointer valid until string changed
LPCWSTR lpcwFilename = (LPCWSTR)getFilename().utf16();
fh = sf_wchar_open(lpcwFilename, SFM_READ, &info);
#else
const QByteArray qbaFilename(getFilename().toLocal8Bit());
fh = sf_open(qbaFilename.constData(), SFM_READ, &info);
#endif
if (fh == NULL) { // sf_format_check is only for writes
qWarning() << "libsndfile: Error opening file" << getFilename() << sf_strerror(fh);
return ERR;
}
if (sf_error(fh)>0) {
qWarning() << "libsndfile: Error opening file" << getFilename() << sf_strerror(fh);
return ERR;
}
channels = info.channels;
setSampleRate(info.samplerate);
// This is the 'virtual' filelength. No matter how many channels the file
// actually has, we pretend it has 2.
filelength = info.frames * 2; // File length with two interleaved channels
return OK;
}
AudioFileDevice::AudioFileDevice( OutputSettings const & outputSettings,
const ch_cnt_t _channels,
const QString & _file,
Mixer* _mixer ) :
AudioDevice( _channels, _mixer ),
m_outputFile( _file ),
m_outputSettings(outputSettings)
{
setSampleRate( outputSettings.getSampleRate() );
if( m_outputFile.open( QFile::WriteOnly | QFile::Truncate ) == false )
{
QString title, message;
title = ExportProjectDialog::tr( "Could not open file" );
message = ExportProjectDialog::tr( "Could not open file %1 "
"for writing.\nPlease make "
"sure you have write "
"permission to the file and "
"the directory containing the "
"file and try again!"
).arg( _file );
if( gui )
{
QMessageBox::critical( NULL, title, message,
QMessageBox::Ok,
QMessageBox::NoButton );
}
else
{
fprintf( stderr, "%s\n", message.toUtf8().constData() );
exit( EXIT_FAILURE );
}
}
}
void MMA7660::init()
{
initAccelTable();
setMode(MMA7660_STAND_BY);
setSampleRate(AUTO_SLEEP_32);
setMode(MMA7660_ACTIVE);
}
MainWindow::MainWindow()
{
setWindowTitle(tr("inspectrum"));
dock = new SpectrogramControls(tr("Controls"), this);
dock->setAllowedAreas(Qt::LeftDockWidgetArea | Qt::RightDockWidgetArea);
addDockWidget(Qt::LeftDockWidgetArea, dock);
input = new InputSource();
plots = new PlotView(input);
setCentralWidget(plots);
// Connect dock inputs
connect(dock, SIGNAL(openFile(QString)), this, SLOT(openFile(QString)));
connect(dock->sampleRate, SIGNAL(textChanged(QString)), this, SLOT(setSampleRate(QString)));
connect(dock, SIGNAL(fftOrZoomChanged(int, int)), plots, SLOT(setFFTAndZoom(int, int)));
connect(dock->powerMaxSlider, SIGNAL(valueChanged(int)), plots, SLOT(setPowerMax(int)));
connect(dock->powerMinSlider, SIGNAL(valueChanged(int)), plots, SLOT(setPowerMin(int)));
connect(dock->cursorsCheckBox, &QCheckBox::stateChanged, plots, &PlotView::enableCursors);
connect(dock->cursorBitsSpinBox, static_cast<void (QSpinBox::*)(int)>(&QSpinBox::valueChanged), plots, &PlotView::setCursorBits);
// Connect dock outputs
connect(plots, SIGNAL(timeSelectionChanged(float)), dock, SLOT(timeSelectionChanged(float)));
connect(plots, SIGNAL(zoomIn()), dock, SLOT(zoomIn()));
connect(plots, SIGNAL(zoomOut()), dock, SLOT(zoomOut()));
// Set defaults after making connections so everything is in sync
dock->setDefaults();
}
void Device::updateFormat( const Format &format )
{
size_t sampleRate = format.getSampleRate();
size_t framesPerBlock = format.getFramesPerBlock();
if( mSampleRate == sampleRate && mFramesPerBlock == framesPerBlock )
return;
auto deviceMgr = Context::deviceManager();
mSignalParamsWillChange.emit();
if( sampleRate && sampleRate != mSampleRate ) {
// set the samplerate to 0, forcing it to refresh on next get.
mSampleRate = 0;
deviceMgr->setSampleRate( shared_from_this(), sampleRate );
}
if( framesPerBlock && framesPerBlock != mFramesPerBlock ) {
// set the frames per block to 0, forcing it to refresh on next get
mFramesPerBlock = 0;
deviceMgr->setFramesPerBlock( shared_from_this(), framesPerBlock );
}
if( ! deviceMgr->isFormatUpdatedAsync() )
mSignalParamsDidChange.emit();
}
void AudioPortAudio::applyQualitySettings()
{
if( hqAudio() )
{
setSampleRate( Engine::mixer()->processingSampleRate() );
int samples = mixer()->framesPerPeriod();
PaError err = Pa_OpenStream(
&m_paStream,
supportsCapture() ? &m_inputParameters : NULL, // The input parameter
&m_outputParameters, // The outputparameter
sampleRate(),
samples,
paNoFlag, // Don't use any flags
_process_callback, // our callback function
this );
if( err != paNoError )
{
printf( "Couldn't open PortAudio: %s\n", Pa_GetErrorText( err ) );
return;
}
}
AudioDevice::applyQualitySettings();
}
void TrackInfoObject::parse() {
// Log parsing of header information in developer mode. This is useful for
// tracking down corrupt files.
const QString& canonicalLocation = m_fileInfo.canonicalFilePath();
if (CmdlineArgs::Instance().getDeveloper()) {
qDebug() << "TrackInfoObject::parse()" << canonicalLocation;
}
// Parse the information stored in the sound file.
SoundSourceProxy proxy(canonicalLocation, m_pSecurityToken);
Mixxx::SoundSource* pProxiedSoundSource = proxy.getProxiedSoundSource();
if (pProxiedSoundSource != NULL && proxy.parseHeader() == OK) {
// Dump the metadata extracted from the file into the track.
// TODO(XXX): This involves locking the mutex for every setXXX
// method. We should figure out an optimization where there are private
// setters that don't lock the mutex.
// If Artist, Title and Type fields are not blank, modify them.
// Otherwise, keep their current values.
// TODO(rryan): Should we re-visit this decision?
if (!(pProxiedSoundSource->getArtist().isEmpty())) {
setArtist(pProxiedSoundSource->getArtist());
}
if (!(pProxiedSoundSource->getTitle().isEmpty())) {
setTitle(pProxiedSoundSource->getTitle());
}
if (!(pProxiedSoundSource->getType().isEmpty())) {
setType(pProxiedSoundSource->getType());
}
setAlbum(pProxiedSoundSource->getAlbum());
setAlbumArtist(pProxiedSoundSource->getAlbumArtist());
setYear(pProxiedSoundSource->getYear());
setGenre(pProxiedSoundSource->getGenre());
setComposer(pProxiedSoundSource->getComposer());
setGrouping(pProxiedSoundSource->getGrouping());
setComment(pProxiedSoundSource->getComment());
setTrackNumber(pProxiedSoundSource->getTrackNumber());
setReplayGain(pProxiedSoundSource->getReplayGain());
setBpm(pProxiedSoundSource->getBPM());
setDuration(pProxiedSoundSource->getDuration());
setBitrate(pProxiedSoundSource->getBitrate());
setSampleRate(pProxiedSoundSource->getSampleRate());
setChannels(pProxiedSoundSource->getChannels());
setKeyText(pProxiedSoundSource->getKey(),
mixxx::track::io::key::FILE_METADATA);
setHeaderParsed(true);
} else {
qDebug() << "TrackInfoObject::parse() error at file"
<< canonicalLocation;
setHeaderParsed(false);
// Add basic information derived from the filename:
parseFilename();
}
}
OsStatus MpOss::attachReader()
{
OsStatus ret = OS_FAILED;
UtlBoolean threadSleep = (mStReader == FALSE) && (mStWriter == FALSE);
if (!((mReader == NULL) || (mStReader == TRUE)))
{
assert(mbReadCap == TRUE);
ret = setSampleRate(mReader->mSamplesPerSec, mReader->mSamplesPerFrame);
if (ret == OS_SUCCESS) {
mStReader = TRUE;
if (threadSleep)
{
threadWakeUp();
}
else
{
threadIoStatusChanged();
}
}
}
return ret;
}
LadderFilter<Type>::LadderFilter() : state (2)
{
setSampleRate (Type (1000)); // intentionally setting unrealistic default
// sample rate to catch missing initialisation bugs
setResonance (Type (0));
setDrive (Type (1.2));
setMode (Mode::LPF12);
}
void MMA7660::init(uint8_t interrupts)
{
initAccelTable();
setMode(MMA7660_STAND_BY);
setSampleRate(AUTO_SLEEP_32);
write(MMA7660_INTSU, interrupts);
setMode(MMA7660_ACTIVE);
}
SoundSource::OpenResult SoundSourceWV::tryOpen(
OpenMode /*mode*/,
const OpenParams& params) {
DEBUG_ASSERT(!m_wpc);
char msg[80]; // hold possible error message
int openFlags = OPEN_WVC | OPEN_NORMALIZE;
if ((params.channelCount() == 1) ||
(params.channelCount() == 2)) {
openFlags |= OPEN_2CH_MAX;
}
// We use WavpackOpenFileInputEx to support Unicode paths on windows
// http://www.wavpack.com/lib_use.txt
QString wavPackFileName = getLocalFileName();
m_pWVFile = new QFile(wavPackFileName);
m_pWVFile->open(QFile::ReadOnly);
QString correctionFileName(wavPackFileName + "c");
if (QFile::exists(correctionFileName)) {
// If there is a correction file, open it as well
m_pWVCFile = new QFile(correctionFileName);
m_pWVCFile->open(QFile::ReadOnly);
}
m_wpc = WavpackOpenFileInputEx(&s_streamReader, m_pWVFile, m_pWVCFile,
msg, openFlags, 0);
if (!m_wpc) {
kLogger.warning() << "failed to open file : " << msg;
return OpenResult::Failed;
}
setChannelCount(WavpackGetReducedChannels(m_wpc));
setSampleRate(WavpackGetSampleRate(m_wpc));
initFrameIndexRangeOnce(
mixxx::IndexRange::forward(
0,
WavpackGetNumSamples(m_wpc)));
if (WavpackGetMode(m_wpc) & MODE_FLOAT) {
m_sampleScaleFactor = CSAMPLE_PEAK;
} else {
const int bitsPerSample = WavpackGetBitsPerSample(m_wpc);
if ((bitsPerSample >= 8) && (bitsPerSample <= 32)) {
// Range of signed sample values: [-2 ^ (bitsPerSample - 1), 2 ^ (bitsPerSample - 1) - 1]
const uint32_t absSamplePeak = 1u << (bitsPerSample - 1);
DEBUG_ASSERT(absSamplePeak > 0);
// Scaled range of sample values: [-CSAMPLE_PEAK, CSAMPLE_PEAK)
m_sampleScaleFactor = CSAMPLE_PEAK / absSamplePeak;
} else {
kLogger.warning()
<< "Invalid bits per sample:"
<< bitsPerSample;
return OpenResult::Aborted;
}
}
m_curFrameIndex = frameIndexMin();
return OpenResult::Succeeded;
}
SpectralCentroid::SpectralCentroid(int history_size, int spectrum_size, int sample_rate) : Feature(history_size) {
initialized = false;
barkWeights = NULL;
barkUnits = NULL;
setSpectrumSize(spectrum_size);
setSampleRate(sample_rate);
}
GenericTimeDataParams::GenericTimeDataParams(QObject *object) :
DataUiHandlerProperty(object),
m_startTime(0),
m_name(QString("no name")),
m_curveEnabled(true)
{
setSampleRate(TIMEDATA_DEFAULT_SR);
setMaxDuration(TIMEDATA_DEFAULT_PROJECT_TIME);
}
// Constructor
Phaser::Phaser() : currentSampleRate(INIT_SAMPLE_RATE), zm1( 0.f )
{
// set params and sample rate
setParameters(Parameters());
setSampleRate(INIT_SAMPLE_RATE);
// set depth range
range(440.f, 1600.f);
}
Result SoundSourceOggVorbis::open() {
const QByteArray qBAFilename(getFilename().toLocal8Bit());
#ifdef __WINDOWS__
if(ov_fopen(qBAFilename.constData(), &vf) < 0) {
qDebug() << "oggvorbis: Input does not appear to be an Ogg bitstream.";
filelength = 0;
return ERR;
}
#else
FILE *vorbisfile = fopen(qBAFilename.constData(), "r");
if (!vorbisfile) {
qDebug() << "oggvorbis: cannot open" << getFilename();
return ERR;
}
if(ov_open(vorbisfile, &vf, NULL, 0) < 0) {
qDebug() << "oggvorbis: Input does not appear to be an Ogg bitstream.";
filelength = 0;
return ERR;
}
#endif
// lookup the ogg's channels and samplerate
vorbis_info * vi = ov_info(&vf, -1);
channels = vi->channels;
setSampleRate(vi->rate);
if (channels > 2) {
qDebug() << "oggvorbis: No support for more than 2 channels!";
ov_clear(&vf);
filelength = 0;
return ERR;
}
// ov_pcm_total returns the total number of frames in the ogg file. The
// frame is the channel-independent measure of samples. The total samples in
// the file is channels * ov_pcm_total. rryan 7/2009 I verified this by
// hand. a 30 second long 48khz mono ogg and a 48khz stereo ogg both report
// 1440000 for ov_pcm_total.
ogg_int64_t ret = ov_pcm_total(&vf, -1);
if (ret >= 0) {
// We pretend that the file is stereo to the rest of the world.
filelength = ret * 2;
}
else //error
{
if (ret == OV_EINVAL) {
//The file is not seekable. Not sure if any action is needed.
qDebug() << "oggvorbis: file is not seekable " << getFilename();
}
}
return OK;
}
bool AREngine::initHardware(IOService* inProvider)
{
bool theAnswer = false;
if(IOAudioEngine::initHardware(inProvider))
{
IOAudioSampleRate theInitialSampleRate = { 0, 0 };
UInt32 theNumberChannels = 0;
// create the streams
if(CreateStreams(&theInitialSampleRate, &theNumberChannels) && (theInitialSampleRate.whole != 0))
{
CreateControls(theNumberChannels);
// figure out how long each block is in microseconds
mBlockTimeoutMicroseconds = 1000000 * mBlockSize / theInitialSampleRate.whole;
setSampleRate(&theInitialSampleRate);
// Set the number of sample frames in each buffer
setNumSampleFramesPerBuffer(mBlockSize * mNumberBlocks);
// set up the timer
IOWorkLoop* theWorkLoop = getWorkLoop();
if(theWorkLoop != NULL)
{
mTimerEventSource = IOTimerEventSource::timerEventSource(this, TimerFired);
if(mTimerEventSource != NULL)
{
theWorkLoop->addEventSource(mTimerEventSource);
theAnswer = true;
}
}
// set the safety offset
// note that due to cache issues, it probably isn't wise to leave the safety offset at 0,
// we set it to 4 here, just to be safe.
setSampleOffset(4);
// set up the time stamp generator
mTimeStampGenerator.SetSampleRate(theInitialSampleRate.whole);
mTimeStampGenerator.SetFramesPerRingBuffer(mBlockSize * mNumberBlocks);
// nate that the rate scalar is a 4.28 fixed point number
// this means that each incremnt is 1/2^28
mTimeStampGenerator.SetRateScalar(1UL << 28);
// set the maximum jitter
// AbsoluteTime theMaximumJitter = { 0, 0 };
// nanoseconds_to_absolutetime(5ULL * 1000ULL, &theMaximumJitter);
// mTimeStampGenerator.SetMaximumJitter(theMaximumJitter.lo);
}
}
return theAnswer;
}
void AudioThread::run() {
#ifdef __APPLE__
pthread_t tID = pthread_self(); // ID of this thread
int priority = sched_get_priority_max( SCHED_RR) - 1;
sched_param prio = {priority}; // scheduling priority of thread
pthread_setschedparam(tID, SCHED_RR, &prio);
#endif
std::cout << "Audio thread initializing.." << std::endl;
if (dac.getDeviceCount() < 1) {
std::cout << "No audio devices found!" << std::endl;
return;
}
setupDevice((outputDevice.load() == -1) ? (dac.getDefaultOutputDevice()) : outputDevice.load());
std::cout << "Audio thread started." << std::endl;
inputQueue = (AudioThreadInputQueue *)getInputQueue("AudioDataInput");
threadQueueNotify = (DemodulatorThreadCommandQueue*)getOutputQueue("NotifyQueue");
while (!terminated) {
AudioThreadCommand command;
cmdQueue.pop(command);
if (command.cmd == AudioThreadCommand::AUDIO_THREAD_CMD_SET_DEVICE) {
setupDevice(command.int_value);
}
if (command.cmd == AudioThreadCommand::AUDIO_THREAD_CMD_SET_SAMPLE_RATE) {
setSampleRate(command.int_value);
}
}
if (deviceController[parameters.deviceId] != this) {
deviceController[parameters.deviceId]->removeThread(this);
} else {
try {
if (dac.isStreamOpen()) {
if (dac.isStreamRunning()) {
dac.stopStream();
}
dac.closeStream();
}
} catch (RtAudioError& e) {
e.printMessage();
}
}
if (threadQueueNotify != NULL) {
DemodulatorThreadCommand tCmd(DemodulatorThreadCommand::DEMOD_THREAD_CMD_AUDIO_TERMINATED);
tCmd.context = this;
threadQueueNotify->push(tCmd);
}
std::cout << "Audio thread done." << std::endl;
}
void DecoderBinaural::setPinnaSize(PinnaSize pinnaSize)
{
if(m_pinna_size != pinnaSize || getState() == 0)
{
m_pinna_size = pinnaSize;
unsigned int sample_rate = m_sample_rate;
m_sample_rate = 0;
setSampleRate(sample_rate);
}
}
UGen & UGen::patch( UGenInput & connectToInput )
{
connectToInput.setIncomingUGen( this );
mNumOutputs += 1;
setSampleRate( connectToInput.getOuterUGen().sampleRate() );
return connectToInput.getOuterUGen();
}
bool eqMac2DriverEngine::initHardware(IOService *provider)
{
bool result = false;
IOAudioSampleRate initialSampleRate;
IOWorkLoop *wl;
//IOLog("eqMac2DriverEngine[%p]::initHardware(%p)\n", this, provider);
duringHardwareInit = TRUE;
if (!super::initHardware(provider)) {
goto Done;
}
initialSampleRate.whole = 0;
initialSampleRate.fraction = 0;
if (!createAudioStreams(&initialSampleRate)) {
IOLog("eqMac2DriverEngine::initHardware() failed\n");
goto Done;
}
if (initialSampleRate.whole == 0) {
goto Done;
}
// calculate our timeout in nanosecs, taking care to keep 64bits
blockTimeoutNS = blockSize;
blockTimeoutNS *= 1000000000;
blockTimeoutNS /= initialSampleRate.whole;
setSampleRate(&initialSampleRate);
// Set the number of sample frames in each buffer
setNumSampleFramesPerBuffer(blockSize * numBlocks);
wl = getWorkLoop();
if (!wl) {
goto Done;
}
timerEventSource = IOTimerEventSource::timerEventSource(this, ourTimerFired);
if (!timerEventSource) {
goto Done;
}
workLoop->addEventSource(timerEventSource);
result = true;
Done:
duringHardwareInit = FALSE;
return result;
}
MonoDelay::MonoDelay( int maxTime , int sampleRate )
{
m_buffer = 0;
m_maxTime = maxTime;
m_maxLength = maxTime * sampleRate;
m_length = m_maxLength;
m_index = 0;
m_feedback = 0.0f;
setSampleRate( sampleRate );
}
void
ReSID::loadFromBuffer(uint8_t **buffer)
{
debug(2, " Loading ReSID state...\n");
setChipModel((chip_model)read8(buffer));
setAudioFilter((bool)read8(buffer));
setSamplingMethod((sampling_method)read8(buffer));
setSampleRate(read32(buffer));
setClockFrequency(read32(buffer));
}
void AudioSndio::applyQualitySettings( void )
{
if( hqAudio() )
{
setSampleRate( Engine::mixer()->processingSampleRate() );
/* change sample rate to sampleRate() */
}
AudioDevice::applyQualitySettings();
}
void CubicSDR::setDevice(SDRDeviceInfo *dev) {
if (!sdrThread->isTerminated()) {
sdrThread->terminate();
if (t_SDR) {
t_SDR->join();
delete t_SDR;
}
}
for (SoapySDR::Kwargs::const_iterator i = settingArgs.begin(); i != settingArgs.end(); i++) {
sdrThread->writeSetting(i->first, i->second);
}
sdrThread->setStreamArgs(streamArgs);
sdrThread->setDevice(dev);
DeviceConfig *devConfig = config.getDevice(dev->getDeviceId());
SoapySDR::Device *soapyDev = dev->getSoapyDevice();
if (soapyDev) {
if (long devSampleRate = devConfig->getSampleRate()) {
sampleRate = dev->getSampleRateNear(SOAPY_SDR_RX, 0, devSampleRate);
sampleRateInitialized.store(true);
}
if (!sampleRateInitialized.load()) {
sampleRate = dev->getSampleRateNear(SOAPY_SDR_RX, 0, DEFAULT_SAMPLE_RATE);
sampleRateInitialized.store(true);
} else {
sampleRate = dev->getSampleRateNear(SOAPY_SDR_RX, 0, sampleRate);
}
if (frequency < sampleRate/2) {
frequency = sampleRate/2;
}
setFrequency(frequency);
setSampleRate(sampleRate);
setPPM(devConfig->getPPM());
setOffset(devConfig->getOffset());
if (devConfig->getAGCMode()) {
setAGCMode(true);
} else {
setAGCMode(false);
}
t_SDR = new std::thread(&SDRThread::threadMain, sdrThread);
}
stoppedDev = nullptr;
}
UGen & UGen::patch( UGen & connectToUGen )
{
connectToUGen.addInput( this );
// TODO jam3: nOutputs should only increase when this chain will be ticked!
mNumOutputs += 1;
setSampleRate( connectToUGen.sampleRate() );
return connectToUGen;
}
