lasp/src/lasp/device/lasp_rtaudiodaq.cpp

#include <mutex>
#define DEBUGTRACE_ENABLED
#include "debugtrace.hpp"
#include "lasp_mathtypes.h"

#include "lasp_rtaudiodaq.h"
#if LASP_HAS_RTAUDIO == 1
#include "RtAudio.h"
#include "lasp_daq.h"
#include <atomic>
#include <cassert>

using std::atomic;
using std::cerr;
using std::endl;
using rte = std::runtime_error;
using std::vector;
using lck = std::scoped_lock<std::mutex>;

DEBUGTRACE_VARIABLES;

void fillRtAudioDeviceInfo(vector<DeviceInfo> &devinfolist) {
  DEBUGTRACE_ENTER;

  vector<RtAudio::Api> apis;
  RtAudio::getCompiledApi(apis);

  for (auto api : apis) {
    RtAudio rtaudio(api);
    us count = rtaudio.getDeviceCount();
    for (us devno = 0; devno < count; devno++) {

      RtAudio::DeviceInfo devinfo = rtaudio.getDeviceInfo(devno);
      if (!devinfo.probed) {
        // Device capabilities not successfully probed. Continue to next
        continue;
      }
      // "Our device info struct"
      DeviceInfo d;
      switch (api) {
      case RtAudio::LINUX_ALSA:
        d.api = rtaudioAlsaApi;
        break;
      case RtAudio::LINUX_PULSE:
        d.api = rtaudioPulseaudioApi;
        break;
      case RtAudio::WINDOWS_WASAPI:
        d.api = rtaudioWasapiApi;
        break;
      case RtAudio::WINDOWS_DS:
        d.api = rtaudioDsApi;
        break;
      case RtAudio::WINDOWS_ASIO:
        d.api = rtaudioAsioApi;
        break;
      default:
        cerr << "Not implemented RtAudio API, skipping." << endl;
        continue;
        break;
      }

      d.device_name = devinfo.name;
      d.api_specific_devindex = devno;

      for (us j = 0; j < devinfo.sampleRates.size(); j++) {
        us rate = devinfo.sampleRates[j];
        d.availableSampleRates.push_back((double)rate);
        if (devinfo.preferredSampleRate == rate) {
          d.prefSampleRateIndex = j;
        }
      }

      d.noutchannels = devinfo.outputChannels;
      d.ninchannels = devinfo.inputChannels;

      d.availableInputRanges = {1.0};

      RtAudioFormat formats = devinfo.nativeFormats;
      if (formats & RTAUDIO_SINT8) {
        d.availableDataTypes.push_back(
            DataTypeDescriptor::DataType::dtype_int8);
      }
      if (formats & RTAUDIO_SINT16) {
        d.availableDataTypes.push_back(
            DataTypeDescriptor::DataType::dtype_int16);
      }
      /* if (formats & RTAUDIO_SINT24) { *1/ */
      /*   d.availableDataTypes.push_back(DataTypeDescriptor::DataType::dtype_int24);
       */
      /* } */
      if (formats & RTAUDIO_SINT32) {
        d.availableDataTypes.push_back(
            DataTypeDescriptor::DataType::dtype_fl32);
      }
      if (formats & RTAUDIO_FLOAT64) {
        d.availableDataTypes.push_back(
            DataTypeDescriptor::DataType::dtype_fl64);
      }
      if (d.availableDataTypes.size() == 0) {
        std::cerr << "RtAudio: No data types found in device!" << endl;
      }

      d.prefDataTypeIndex = d.availableDataTypes.size() - 1;

      d.availableFramesPerBlock = {512, 1024, 2048, 4096, 8192};
      d.prefFramesPerBlockIndex = 1;

      devinfolist.push_back(d);
    }
  }
}

static int mycallback(void *outputBuffer, void *inputBuffer,
                      unsigned int nFrames, double streamTime,
                      RtAudioStreamStatus status, void *userData);

static void myerrorcallback(RtAudioError::Type, const string &errorText);

class RtAudioDaq : public Daq {

  RtAudio rtaudio;
  const us nFramesPerBlock;

  RtAudioDaq(const RtAudioDaq &) = delete;
  RtAudioDaq &operator=(const RtAudioDaq &) = delete;

  InDaqCallback _incallback;
  OutDaqCallback _outcallback;

  std::atomic<StreamStatus> _streamStatus{};

public:
  RtAudioDaq(const DeviceInfo &devinfo, const DaqConfiguration &config)
      : Daq(devinfo, config),
        rtaudio(static_cast<RtAudio::Api>(devinfo.api.api_specific_subcode)),
        nFramesPerBlock(Daq::framesPerBlock()) {

    DEBUGTRACE_ENTER;

    // We make sure not to run RtAudio in duplex mode. This seems to be buggy
    // and untested. Better to use a hardware-type loopback into the system.
    if (duplexMode()) {
      throw rte("RtAudio backend cannot run in duplex mode.");
    }
    assert(!monitorOutput);

    std::unique_ptr<RtAudio::StreamParameters> inParams, outParams;

    if (neninchannels() > 0) {

      inParams = std::make_unique<RtAudio::StreamParameters>();

      // +1 to get the count.
      inParams->nChannels = getHighestEnabledInChannel() + 1;
      if (inParams->nChannels < 1) {
        throw rte("Invalid input number of channels");
      }
      inParams->firstChannel = 0;
      inParams->deviceId = devinfo.api_specific_devindex;

    } else {

      outParams = std::make_unique<RtAudio::StreamParameters>();

      outParams->nChannels = getHighestEnabledOutChannel() + 1;
      if (outParams->nChannels < 1) {
        throw rte("Invalid output number of channels");
      }
      outParams->firstChannel = 0;
      outParams->deviceId = devinfo.api_specific_devindex;
    }

    RtAudio::StreamOptions streamoptions;
    streamoptions.flags = RTAUDIO_HOG_DEVICE | RTAUDIO_NONINTERLEAVED;

    streamoptions.numberOfBuffers = 2;
    streamoptions.streamName = "LASP RtAudio DAQ stream";
    streamoptions.priority = 0;

    RtAudioFormat format;
    using Dtype = DataTypeDescriptor::DataType;
    const Dtype dtype = dataType();
    switch (dtype) {
    case Dtype::dtype_fl32:
      DEBUGTRACE_PRINT("Datatype float32");
      format = RTAUDIO_FLOAT32;
      break;
    case Dtype::dtype_fl64:
      DEBUGTRACE_PRINT("Datatype float64");
      format = RTAUDIO_FLOAT64;
      break;
    case Dtype::dtype_int8:
      DEBUGTRACE_PRINT("Datatype int8");
      format = RTAUDIO_SINT8;
      break;
    case Dtype::dtype_int16:
      DEBUGTRACE_PRINT("Datatype int16");
      format = RTAUDIO_SINT16;
      break;
    case Dtype::dtype_int32:
      DEBUGTRACE_PRINT("Datatype int32");
      format = RTAUDIO_SINT32;
      break;
    default:
      throw rte("Invalid data type specified for DAQ stream.");
      break;
    }

    // Copy here, as it is used to return the *actual* number of frames per
    // block.
    unsigned int nFramesPerBlock_copy = nFramesPerBlock;

    // Final step: open the stream.
    rtaudio.openStream(outParams.get(), inParams.get(), format,
                       static_cast<us>(samplerate()), &nFramesPerBlock_copy,
                       mycallback, (void *)this, &streamoptions,
                       &myerrorcallback);

    if (nFramesPerBlock_copy != nFramesPerBlock) {
      throw rte("Got different number of frames per block back from RtAudio "
                "backend. Do not know what to do");
    }
  }

  virtual void start(InDaqCallback inCallback,
                     OutDaqCallback outCallback) override final {

    DEBUGTRACE_ENTER;
    lck lock(_mtx);

    assert(!monitorOutput);

    if (getStreamStatus().runningOK()) {
      throw rte("Stream already running");
    }

    // Logical XOR
    if (inCallback && outCallback) {
      throw rte("Either input or output stream possible for RtAudio. "
                "Stream duplex mode not provided.");
    }

    if (neninchannels() > 0) {
      if (!inCallback) {
        throw rte(

            "Input callback given, but stream does not provide input data");
      }

      _incallback = inCallback;
    }
    if (nenoutchannels() > 0) {
      if (!outCallback) {
        throw rte(
            "Output callback given, but stream does not provide output data");
      }
      _outcallback = outCallback;
    }

    // Start the stream. Throws on error.
    rtaudio.startStream();

    // If we are here, we are running without errors.
    StreamStatus status;
    status.isRunning = true;
    _streamStatus = status;
  }

  StreamStatus getStreamStatus() const override final { return _streamStatus; }

  void stop() override final {
    DEBUGTRACE_ENTER;
    if (getStreamStatus().runningOK()) {
      rtaudio.stopStream();
    }
    StreamStatus s = _streamStatus;
    s.isRunning = false;
    s.errorType = StreamStatus::StreamError::noError;
    _streamStatus = s;
  }

  int streamCallback(void *outputBuffer, void *inputBuffer,
                     unsigned int nFrames, RtAudioStreamStatus status) {

    DEBUGTRACE_ENTER;

    using se = StreamStatus::StreamError;

    int rval = 0;
    auto stopWithError = [&](se e) {
      DEBUGTRACE_PRINT("stopWithError");
      StreamStatus stat = _streamStatus;
      stat.errorType = e;
      stat.isRunning = false;
      _streamStatus = stat;
      rval = 1;
    };

    switch (status) {
    case RTAUDIO_INPUT_OVERFLOW:
      stopWithError(se::inputXRun);
      return 1;
      break;
    case RTAUDIO_OUTPUT_UNDERFLOW:
      stopWithError(se::outputXRun);
      return 1;
      break;
    default:
      break;
    }

    const auto &dtype_descr = dtypeDescr();
    const auto dtype = dataType();

    us neninchannels = this->neninchannels();
    us nenoutchannels = this->nenoutchannels();
    us sw = dtype_descr.sw;
    if (nFrames != nFramesPerBlock) {
      cerr << "RtAudio backend error: nFrames does not match block size!"
           << endl;
      stopWithError(se::logicError);
      return 1;
    }

    if (inputBuffer) {
      assert(_incallback);
      std::vector<byte_t *> ptrs;
      ptrs.reserve(neninchannels);

      const us ch_min = getLowestEnabledInChannel();
      const us ch_max = getHighestEnabledInChannel();
      us i = 0;
      for (us ch = ch_min; ch <= ch_max; ch++) {
        if (inchannel_config.at(ch).enabled) {
          byte_t *ptr =
              static_cast<byte_t *>(inputBuffer) + sw * i * nFramesPerBlock;
          DEBUGTRACE_PRINT((us)ptr);
          ptrs.push_back(ptr);
        }
        i++;
      }
      DaqData d{nFramesPerBlock, neninchannels, dtype};
      d.copyInFromRaw(ptrs);

      bool ret = _incallback(d);
      if (!ret) {
        stopWithError(se::noError);
        return 1;
      }
    }

    if (outputBuffer) {
      assert(_outcallback);
      std::vector<byte_t *> ptrs;
      ptrs.reserve(nenoutchannels);

      /* outCallback */

      const us ch_min = getLowestEnabledOutChannel();
      const us ch_max = getHighestEnabledOutChannel();
      us i = 0;
      for (us ch = ch_min; ch <= ch_max; ch++) {
        if (outchannel_config.at(ch).enabled) {

          ptrs.push_back(static_cast<byte_t *>(outputBuffer) +
                         sw * i * nFramesPerBlock);
        }
        i++;
      }
      DaqData d{nFramesPerBlock, nenoutchannels, dtype};

      bool ret = _outcallback(d);
      if (!ret) {
        stopWithError(se::noError);
        return 1;
      }
      us j = 0;
      for (auto ptr : ptrs) {
        d.copyToRaw(j, ptr);
        j++;
      }
    }

    return rval;
  }

  // RtAudio documentation says: if a stream is open, it will be stopped and
  // closed automatically on deletion. Therefore the destructor here is a
  // default one.
  ~RtAudioDaq() = default;
};

std::unique_ptr<Daq> createRtAudioDevice(const DeviceInfo &devinfo,
                                         const DaqConfiguration &config) {
  return std::make_unique<RtAudioDaq>(devinfo, config);
}

void myerrorcallback(RtAudioError::Type, const string &errorText) {
  cerr << "RtAudio backend stream error: " << errorText << endl;
}
int mycallback(void *outputBuffer, void *inputBuffer, unsigned int nFrames,
               double streamTime, RtAudioStreamStatus status, void *userData) {

  return static_cast<RtAudioDaq *>(userData)->streamCallback(
      outputBuffer, inputBuffer, nFrames, status);
}

#endif // LASP_HAS_RTAUDIO == 1