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    What is the process for writing my own module (not scriptnode)

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    • griffinboyG
      griffinboy @ustk
      last edited by griffinboy

      @ustk

      Right sure.
      Yes, you could drop a filter on the audio, but you misunderstand how aliasing works. Aliasing reflects downwards, creating low harmonics. It doesn't just create high harmonics. So even lowpass filtering the signal won't remove aliasing.

      You NEED to oversample if you want to filter out aliasing. It's not a case of it 'helping' but it's a way of raising the nyquist so that when you apply the filter, there are no alias harmonics already present, having reflected downwards.

      Also, filtering the drawn waveform is the same as filtering the buffer. The waveform is the buffer... Samples along the x axis, y value is the value of each sample.

      O 1 Reply Last reply Reply Quote 1
      • griffinboyG
        griffinboy @ustk
        last edited by griffinboy

        @ustk

        So vital works like this:

        Precompute the FFT, store the waveform in the frequency domain.
        When the user presses a note, clear harmonics that would alias, convert back to time domain (inverse fft) and play waveform. You only need to calculate the inverse fft once every time the pitch changes, or the user plays a new note. You could also use a threshold similar to mipmapping, say, ignore pitch bends until they go over a certain range. So then you don't do inverse fft constantly when stuff like vibrato is happening.

        For antiderivative techniques it becomes even more complicated for pitch varying signals you need to crossfade between mipmaps to avoid clicks.

        ustkU 1 Reply Last reply Reply Quote 1
        • ustkU
          ustk @griffinboy
          last edited by

          @griffinboy Thanks for all that!
          I do understand how aliasing works, nyquist, reflected harmonics... But I didn't understand how to prevent it in this particular context until you shine your knowledge on me! ☀ 😎
          I feel a few percents less idiot now ☺

          Can't help pressing F5 in the forum...

          1 Reply Last reply Reply Quote 2
          • griffinboyG
            griffinboy @Christoph Hart
            last edited by griffinboy

            @Christoph-Hart

            By the way, while we are on the topic of Wavetable synthesis, what is your process for scanning through a Wavetable?

            Are you doing anything to mitigate large jumps / clicks discontinuities when scanning through different frames? Do you generate any in-between frames using interpolation, or is it simply a case of 'snapping' to the next frame and allowing the realtime interpolator to interpolate between the previous and current sample (belonging to the previous and current frame)?

            I'm not in the loop when it comes to the 'common' approach. I'm going to do my own analysis but I thought I'd ask!

            1 Reply Last reply Reply Quote 0
            • O
              Orvillain @griffinboy
              last edited by

              @griffinboy said in What is the process for writing my own module (not scriptnode):

              @ustk

              Right sure.
              Yes, you could drop a filter on the audio, but you misunderstand how aliasing works. Aliasing reflects downwards, creating low harmonics. It doesn't just create high harmonics. So even lowpass filtering the signal won't remove aliasing.

              You NEED to oversample if you want to filter out aliasing. It's not a case of it 'helping' but it's a way of raising the nyquist so that when you apply the filter, there are no alias harmonics already present, having reflected downwards.

              Also, filtering the drawn waveform is the same as filtering the buffer. The waveform is the buffer... Samples along the x axis, y value is the value of each sample.

              @griffinboy out of interest... what about avoiding aliasing when downsampling?? I had assumed that a solid biquad lowpass would cover this, but maybe not??

              griffinboyG 1 Reply Last reply Reply Quote 0
              • griffinboyG
                griffinboy @Orvillain
                last edited by griffinboy

                @Orvillain

                At the moment I'm using custom fir filters and that seems to work. Again, following the design presented in 'quest for the perfect resampler'

                HISEnbergH 1 Reply Last reply Reply Quote 0
                • HISEnbergH
                  HISEnberg @griffinboy
                  last edited by

                  @griffinboy said in What is the process for writing my own module (not scriptnode):

                  quest for the perfect resampler

                  Found it online: Quest for the Perfect Resampler. Thanks for the suggestion @griffinboy

                  Graham Wakefield's book Generating Sound and Organizing Time also does a good job of covering wavetable synthesis and MipMapping in Max MSP's Gen~ environment, for those who are interested in the topic and need a more "digestible" explanation.

                  1 Reply Last reply Reply Quote 1
                  • HISEnbergH
                    HISEnberg @griffinboy
                    last edited by

                    @griffinboy My personal vision for this would be similar to Max MSP which allows you to install external packages, sort of like expansion packs. This would give full credits to the author of the external nodes (including the licensing structure) and allow building custom libraries for specific purposes. I envisage a system like this as HISE develops and grows to include more developers looking to do different things within the HISE framework.

                    Christoph HartC 1 Reply Last reply Reply Quote 0
                    • Christoph HartC
                      Christoph Hart @HISEnberg
                      last edited by

                      @HISEnberg empty room problem, but as soon as there starts to be demand for it I'll think about a good solution.

                      1 Reply Last reply Reply Quote 3
                      • O
                        Orvillain @Christoph Hart
                        last edited by Orvillain

                        @Christoph-Hart

                        Hey Christoph, when trying to compile your script here, I get:

                        > Create files
                        > Sorting include dependencies
                        > Copying third party files
                        > Compiling dll plugin
                        Re-saving file: C:\Users\avern\Desktop\blah\DspNetworks\Binaries\AutogeneratedProject.jucer
                        Finished saving: Visual Studio 2022
                        Finished saving: Xcode (macOS)
                        Finished saving: Linux Makefile
                        Compiling 64bit  blah ...
                        MSBuild version 17.10.4+10fbfbf2e for .NET Framework
                        
                          Main.cpp
                          include_hi_dsp_library_01.cpp
                          include_hi_dsp_library_02.cpp
                          include_hi_tools_01.cpp
                          include_hi_tools_02.cpp
                          include_hi_tools_03.cpp
                          include_juce_audio_formats.cpp
                          include_juce_core.cpp
                          include_juce_data_structures.cpp
                          include_juce_dsp.cpp
                          include_juce_events.cpp
                          include_juce_graphics.cpp
                        !C:\Users\avern\Desktop\blah\DspNetworks\ThirdParty\data_node.h(75,4): error C2593: 'operator []' is ambiguous [C:\Users\avern\Desktop\blah\DspNetworks\Binaries\Builds\VisualStudio2022\blah_DynamicLibrary.vcxproj]
                        !C:\Users\avern\Desktop\blah\DspNetworks\ThirdParty\data_node.h(76,4): error C2593: 'operator []' is ambiguous [C:\Users\avern\Desktop\blah\DspNetworks\Binaries\Builds\VisualStudio2022\blah_DynamicLibrary.vcxproj]
                        !C:\Users\avern\Desktop\blah\DspNetworks\ThirdParty\data_node.h(77,4): error C2593: 'operator []' is ambiguous [C:\Users\avern\Desktop\blah\DspNetworks\Binaries\Builds\VisualStudio2022\blah_DynamicLibrary.vcxproj]
                        !C:\Users\avern\Desktop\blah\DspNetworks\ThirdParty\data_node.h(78,4): error C2593: 'operator []' is ambiguous [C:\Users\avern\Desktop\blah\DspNetworks\Binaries\Builds\VisualStudio2022\blah_DynamicLibrary.vcxproj]
                        !C:\Users\avern\Desktop\blah\DspNetworks\ThirdParty\data_node.h(81,4): error C2593: 'operator []' is ambiguous [C:\Users\avern\Desktop\blah\DspNetworks\Binaries\Builds\VisualStudio2022\blah_DynamicLibrary.vcxproj]
                        	C:\Program Files\Microsoft Visual Studio\2022\Community\VC\Tools\MSVC\14.40.33807\include\bit(11): warning STL4038: The contents of <bit> are available only with C++20 or later. [C:\Users\avern\Desktop\blah\DspNetworks\Binaries\Builds\VisualStudio2022\blah_DynamicLibrary.vcxproj]
                        
                        

                        The lines causing this error are:

                        hise::JSONObject obj;
                        
                        // write the values into the JSON object
                        obj["magnitude"] = magnitude;
                        obj["RMS"] = rms;
                        obj["length"] = length;
                        obj["numChannels"] = channels;
                        
                        // just attach the current knob position so you know it's working.
                        obj["knobValue"] = value;
                        

                        I have just update to the latest develop. This SHA:
                        03c420c1d12f7a4457a8b497a6b78bc49d250e85

                        Explicitly casting like this did result in the compile working:

                        			// write the values into the JSON object
                        			obj[String("magnitude")] = magnitude;
                        			obj[String("RMS")] = rms;
                        			obj[String("length")] = length;
                        			obj[String("numChannels")] = channels;
                        			
                        
                        			// just attach the current knob position so you know it's working.
                        			obj[String("knobValue")] = value;
                        
                        griffinboyG 1 Reply Last reply Reply Quote 0
                        • griffinboyG
                          griffinboy @Orvillain
                          last edited by

                          @Orvillain

                          Oh yeah yeah, I found faults with the provided script too. I had to do some casting. I think that's correct.

                          1 Reply Last reply Reply Quote 0
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