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// A tool from the FluCoMa project, funded by the European Research Council (ERC) under the European Union’s Horizon 2020 research and innovation programme (grant agreement No 725899)
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#include "SC_PlugIn.hpp"
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#include "data/FluidTensor.hpp"
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#include "clients/rt/BaseSTFTClient.hpp"
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//Using statements for fluidtensor
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using fluid::FluidTensor;
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using fluid::FluidTensorView;
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using fluid::audio::BaseSTFTClient;
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static InterfaceTable *ft;
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using audio_client = BaseSTFTClient<double, float>;
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using audio_signal_wrapper = audio_client::AudioSignal;
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using scalar_signal_wrapper = audio_client::ScalarSignal;
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using signal_wrapper = audio_client::Signal<float>;
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class FDSTFTPass: public SCUnit
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{
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public:
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FDSTFTPass()
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{
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//Order of args
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//Window size, Hop size, FFT Size
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//Get the window size
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const float window_size = in0(1);
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const float hop_size = in0(2);
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const float fft_size = in0(3);
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//Oh NO! Heap allocation! Make client object
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//This will do argument checking
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m_client = new audio_client(window_size, hop_size, fft_size);
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//and we can post strings back about the argument checking
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//Params get checked and adjusted during construction, we can pass on warning messages:
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std::istringstream param_feedback(m_client->getFeedbackString());
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std::string message;
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while(std::getline(param_feedback, message, '\n'))
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Print(message.c_str());
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m_client->set_host_buffer_size(bufferSize());
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m_client->reset();
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//Work out what signals we need. For now keep it simple:
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//in 0 => only audio
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//out 0 => only audio
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//out 1 => only audio (and only internal, used for OLA gain compensation)
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input_signals[0] = new audio_signal_wrapper();
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output_signals[0] = new audio_signal_wrapper();
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output_signals[1] = new audio_signal_wrapper();
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m_normalise = new float[bufferSize()];
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// set_calc_function<FDGain,&FDGain::next>();
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mCalcFunc = make_calc_function<FDSTFTPass,&FDSTFTPass::next>();
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Unit* unit = this;
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ClearUnitOutputs(unit,1);
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}
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~FDSTFTPass()
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{
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delete[] m_normalise;
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delete input_signals[0];
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delete output_signals[0];
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delete output_signals[1];
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delete m_client;
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}
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private:
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void next(int numsamples)
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{
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//TODO: Remove const_cast code smell by making input_signal type for const
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const float* input = in(0);
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const float inscalar = in0(0);
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float* output = out(0);
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input_signals[0]->set(const_cast<float*>(input), inscalar);
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output_signals[0]->set(out(0), out0(0));
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output_signals[1]->set(m_normalise,0);
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m_client->do_process(input_signals,output_signals,numsamples,1,2);
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//We need to normalise *after* the overlap add, because of the way we do it
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std::transform(output, output + numsamples, m_normalise, output, std::divides<double>());
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}
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audio_client* m_client;
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signal_wrapper* input_signals[1];
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signal_wrapper* output_signals[2];
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float* m_normalise;
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};
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PluginLoad(BoringMixer2UGens) {
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ft = inTable;
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registerUnit<FDSTFTPass>(ft, "FDSTFTPass");
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}
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