Small comment change
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@ -38,32 +38,32 @@ void fillRtAudioDeviceInfo(vector<DeviceInfo> &devinfolist) {
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// "Our device info struct"
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DeviceInfo d;
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switch (api) {
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case RtAudio::LINUX_ALSA:
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d.api = rtaudioAlsaApi;
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break;
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case RtAudio::LINUX_PULSE:
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d.api = rtaudioPulseaudioApi;
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break;
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case RtAudio::WINDOWS_WASAPI:
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d.api = rtaudioWasapiApi;
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break;
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case RtAudio::WINDOWS_DS:
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d.api = rtaudioDsApi;
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break;
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case RtAudio::WINDOWS_ASIO:
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d.api = rtaudioAsioApi;
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break;
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default:
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cerr << "Not implemented RtAudio API, skipping." << endl;
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continue;
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break;
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case RtAudio::LINUX_ALSA:
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d.api = rtaudioAlsaApi;
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break;
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case RtAudio::LINUX_PULSE:
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d.api = rtaudioPulseaudioApi;
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break;
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case RtAudio::WINDOWS_WASAPI:
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d.api = rtaudioWasapiApi;
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break;
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case RtAudio::WINDOWS_DS:
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d.api = rtaudioDsApi;
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break;
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case RtAudio::WINDOWS_ASIO:
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d.api = rtaudioAsioApi;
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break;
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default:
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cerr << "Not implemented RtAudio API, skipping." << endl;
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continue;
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break;
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}
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d.device_name = devinfo.name;
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d.api_specific_devindex = devno;
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/// We overwrite the default sample rate with the 48 kHz value, which
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/// is our preffered rate.
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/// When 48k is available we overwrite the default sample rate with the 48
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/// kHz value, which is our preffered rate,
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bool rate_48k_found = false;
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for (us j = 0; j < devinfo.sampleRates.size(); j++) {
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@ -72,13 +72,13 @@ void fillRtAudioDeviceInfo(vector<DeviceInfo> &devinfolist) {
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d.availableSampleRates.push_back((double)rate_int);
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if(!rate_48k_found) {
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if (!rate_48k_found) {
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if (devinfo.preferredSampleRate == rate_int) {
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d.prefSampleRateIndex = j;
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}
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if(rate_int == 48000) {
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if (rate_int == 48000) {
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d.prefSampleRateIndex = j;
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rate_48k_found = true;
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}
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@ -101,7 +101,7 @@ void fillRtAudioDeviceInfo(vector<DeviceInfo> &devinfolist) {
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}
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/* if (formats & RTAUDIO_SINT24) { *1/ */
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/* d.availableDataTypes.push_back(DataTypeDescriptor::DataType::dtype_int24);
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*/
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*/
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/* } */
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if (formats & RTAUDIO_SINT32) {
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d.availableDataTypes.push_back(
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@ -126,8 +126,8 @@ void fillRtAudioDeviceInfo(vector<DeviceInfo> &devinfolist) {
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}
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static int mycallback(void *outputBuffer, void *inputBuffer,
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unsigned int nFrames, double streamTime,
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RtAudioStreamStatus status, void *userData);
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unsigned int nFrames, double streamTime,
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RtAudioStreamStatus status, void *userData);
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static void myerrorcallback(RtAudioError::Type, const string &errorText);
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@ -144,101 +144,101 @@ class RtAudioDaq : public Daq {
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std::atomic<StreamStatus> _streamStatus{};
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public:
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public:
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RtAudioDaq(const DeviceInfo &devinfo, const DaqConfiguration &config)
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: Daq(devinfo, config),
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rtaudio(static_cast<RtAudio::Api>(devinfo.api.api_specific_subcode)),
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nFramesPerBlock(Daq::framesPerBlock()) {
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: Daq(devinfo, config),
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rtaudio(static_cast<RtAudio::Api>(devinfo.api.api_specific_subcode)),
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nFramesPerBlock(Daq::framesPerBlock()) {
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DEBUGTRACE_ENTER;
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DEBUGTRACE_ENTER;
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// We make sure not to run RtAudio in duplex mode. This seems to be buggy
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// and untested. Better to use a hardware-type loopback into the system.
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if (duplexMode()) {
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throw rte("RtAudio backend cannot run in duplex mode.");
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// We make sure not to run RtAudio in duplex mode. This seems to be buggy
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// and untested. Better to use a hardware-type loopback into the system.
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if (duplexMode()) {
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throw rte("RtAudio backend cannot run in duplex mode.");
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}
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assert(!monitorOutput);
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std::unique_ptr<RtAudio::StreamParameters> inParams, outParams;
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if (neninchannels() > 0) {
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inParams = std::make_unique<RtAudio::StreamParameters>();
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// +1 to get the count.
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inParams->nChannels = getHighestEnabledInChannel() + 1;
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if (inParams->nChannels < 1) {
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throw rte("Invalid input number of channels");
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}
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assert(!monitorOutput);
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inParams->firstChannel = 0;
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inParams->deviceId = devinfo.api_specific_devindex;
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std::unique_ptr<RtAudio::StreamParameters> inParams, outParams;
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} else {
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if (neninchannels() > 0) {
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outParams = std::make_unique<RtAudio::StreamParameters>();
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inParams = std::make_unique<RtAudio::StreamParameters>();
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// +1 to get the count.
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inParams->nChannels = getHighestEnabledInChannel() + 1;
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if (inParams->nChannels < 1) {
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throw rte("Invalid input number of channels");
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}
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inParams->firstChannel = 0;
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inParams->deviceId = devinfo.api_specific_devindex;
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} else {
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outParams = std::make_unique<RtAudio::StreamParameters>();
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outParams->nChannels = getHighestEnabledOutChannel() + 1;
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if (outParams->nChannels < 1) {
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throw rte("Invalid output number of channels");
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}
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outParams->firstChannel = 0;
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outParams->deviceId = devinfo.api_specific_devindex;
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}
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RtAudio::StreamOptions streamoptions;
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streamoptions.flags = RTAUDIO_HOG_DEVICE | RTAUDIO_NONINTERLEAVED;
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streamoptions.numberOfBuffers = 2;
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streamoptions.streamName = "LASP RtAudio DAQ stream";
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streamoptions.priority = 0;
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RtAudioFormat format;
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using Dtype = DataTypeDescriptor::DataType;
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const Dtype dtype = dataType();
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switch (dtype) {
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case Dtype::dtype_fl32:
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DEBUGTRACE_PRINT("Datatype float32");
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format = RTAUDIO_FLOAT32;
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break;
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case Dtype::dtype_fl64:
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DEBUGTRACE_PRINT("Datatype float64");
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format = RTAUDIO_FLOAT64;
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break;
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case Dtype::dtype_int8:
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DEBUGTRACE_PRINT("Datatype int8");
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format = RTAUDIO_SINT8;
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break;
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case Dtype::dtype_int16:
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DEBUGTRACE_PRINT("Datatype int16");
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format = RTAUDIO_SINT16;
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break;
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case Dtype::dtype_int32:
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DEBUGTRACE_PRINT("Datatype int32");
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format = RTAUDIO_SINT32;
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break;
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default:
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throw rte("Invalid data type specified for DAQ stream.");
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break;
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}
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// Copy here, as it is used to return the *actual* number of frames per
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// block.
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unsigned int nFramesPerBlock_copy = nFramesPerBlock;
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// Final step: open the stream.
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rtaudio.openStream(outParams.get(), inParams.get(), format,
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static_cast<us>(samplerate()), &nFramesPerBlock_copy,
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mycallback, (void *)this, &streamoptions,
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&myerrorcallback);
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if (nFramesPerBlock_copy != nFramesPerBlock) {
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throw rte("Got different number of frames per block back from RtAudio "
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"backend. Do not know what to do");
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outParams->nChannels = getHighestEnabledOutChannel() + 1;
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if (outParams->nChannels < 1) {
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throw rte("Invalid output number of channels");
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}
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outParams->firstChannel = 0;
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outParams->deviceId = devinfo.api_specific_devindex;
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}
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RtAudio::StreamOptions streamoptions;
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streamoptions.flags = RTAUDIO_HOG_DEVICE | RTAUDIO_NONINTERLEAVED;
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streamoptions.numberOfBuffers = 2;
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streamoptions.streamName = "LASP RtAudio DAQ stream";
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streamoptions.priority = 0;
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RtAudioFormat format;
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using Dtype = DataTypeDescriptor::DataType;
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const Dtype dtype = dataType();
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switch (dtype) {
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case Dtype::dtype_fl32:
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DEBUGTRACE_PRINT("Datatype float32");
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format = RTAUDIO_FLOAT32;
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break;
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case Dtype::dtype_fl64:
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DEBUGTRACE_PRINT("Datatype float64");
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format = RTAUDIO_FLOAT64;
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break;
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case Dtype::dtype_int8:
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DEBUGTRACE_PRINT("Datatype int8");
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format = RTAUDIO_SINT8;
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break;
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case Dtype::dtype_int16:
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DEBUGTRACE_PRINT("Datatype int16");
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format = RTAUDIO_SINT16;
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break;
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case Dtype::dtype_int32:
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DEBUGTRACE_PRINT("Datatype int32");
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format = RTAUDIO_SINT32;
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break;
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default:
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throw rte("Invalid data type specified for DAQ stream.");
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break;
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}
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// Copy here, as it is used to return the *actual* number of frames per
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// block.
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unsigned int nFramesPerBlock_copy = nFramesPerBlock;
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// Final step: open the stream.
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rtaudio.openStream(outParams.get(), inParams.get(), format,
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static_cast<us>(samplerate()), &nFramesPerBlock_copy,
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mycallback, (void *)this, &streamoptions,
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&myerrorcallback);
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if (nFramesPerBlock_copy != nFramesPerBlock) {
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throw rte("Got different number of frames per block back from RtAudio "
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"backend. Do not know what to do");
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}
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}
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virtual void start(InDaqCallback inCallback,
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OutDaqCallback outCallback) override final {
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OutDaqCallback outCallback) override final {
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DEBUGTRACE_ENTER;
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@ -251,7 +251,7 @@ class RtAudioDaq : public Daq {
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// Logical XOR
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if (inCallback && outCallback) {
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throw rte("Either input or output stream possible for RtAudio. "
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"Stream duplex mode not provided.");
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"Stream duplex mode not provided.");
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}
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if (neninchannels() > 0) {
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@ -294,7 +294,7 @@ class RtAudioDaq : public Daq {
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}
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int streamCallback(void *outputBuffer, void *inputBuffer,
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unsigned int nFrames, RtAudioStreamStatus status) {
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unsigned int nFrames, RtAudioStreamStatus status) {
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DEBUGTRACE_ENTER;
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@ -311,16 +311,16 @@ class RtAudioDaq : public Daq {
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};
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switch (status) {
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case RTAUDIO_INPUT_OVERFLOW:
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stopWithError(se::inputXRun);
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return 1;
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break;
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case RTAUDIO_OUTPUT_UNDERFLOW:
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stopWithError(se::outputXRun);
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return 1;
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break;
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default:
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break;
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case RTAUDIO_INPUT_OVERFLOW:
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stopWithError(se::inputXRun);
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return 1;
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break;
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case RTAUDIO_OUTPUT_UNDERFLOW:
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stopWithError(se::outputXRun);
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return 1;
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break;
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default:
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break;
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}
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const auto &dtype_descr = dtypeDescr();
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@ -331,7 +331,7 @@ class RtAudioDaq : public Daq {
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us sw = dtype_descr.sw;
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if (nFrames != nFramesPerBlock) {
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cerr << "RtAudio backend error: nFrames does not match block size!"
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<< endl;
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<< endl;
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stopWithError(se::logicError);
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return 1;
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}
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@ -347,7 +347,7 @@ class RtAudioDaq : public Daq {
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for (us ch = ch_min; ch <= ch_max; ch++) {
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if (inchannel_config.at(ch).enabled) {
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byte_t *ptr =
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static_cast<byte_t *>(inputBuffer) + sw * i * nFramesPerBlock;
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static_cast<byte_t *>(inputBuffer) + sw * i * nFramesPerBlock;
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DEBUGTRACE_PRINT((us)ptr);
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ptrs.push_back(ptr);
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}
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@ -377,7 +377,7 @@ class RtAudioDaq : public Daq {
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if (outchannel_config.at(ch).enabled) {
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ptrs.push_back(static_cast<byte_t *>(outputBuffer) +
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sw * i * nFramesPerBlock);
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sw * i * nFramesPerBlock);
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}
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i++;
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}
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@ -405,7 +405,7 @@ class RtAudioDaq : public Daq {
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};
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std::unique_ptr<Daq> createRtAudioDevice(const DeviceInfo &devinfo,
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const DaqConfiguration &config) {
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const DaqConfiguration &config) {
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return std::make_unique<RtAudioDaq>(devinfo, config);
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}
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@ -413,7 +413,7 @@ void myerrorcallback(RtAudioError::Type, const string &errorText) {
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cerr << "RtAudio backend stream error: " << errorText << endl;
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}
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int mycallback(void *outputBuffer, void *inputBuffer, unsigned int nFrames,
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double streamTime, RtAudioStreamStatus status, void *userData) {
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double streamTime, RtAudioStreamStatus status, void *userData) {
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return static_cast<RtAudioDaq *>(userData)->streamCallback(
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outputBuffer, inputBuffer, nFrames, status);
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