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    Free Reverse Delay built in RNBO

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    • M
      Mighty23 @Lindon
      last edited by

      @Lindon I edited the code in the previous comment.
      The key is to make the delay time scale with the sample rate.

      Free Party, Free Tekno & Free Software too

      1 Reply Last reply Reply Quote 0
      • T
        treynterrio @Mighty23
        last edited by

        @Mighty23 thank you! what's the process to get 2 channels tried to copy paste the process but it still says only 1 Channel

        M 1 Reply Last reply Reply Quote 0
        • M
          Mighty23 @treynterrio
          last edited by

          @treynterrio you can use this code if you have 2 inputs and 2 outputs or use the previous one if you have one input and one output.

          The first code you can use in scriptnode in case you need to process, for example, mid and side separately.

          import("stdfaust.lib");
          
          MAX_DELAY = 192000;  
          
          // Phase-shifted phasor for reverse reading
          // dtime: delay time in samples
          // phase: phase offset for reading position
          phasor_phase(dtime, phase) = ((os.lf_rawsaw(dtime) + phase) % dtime) : int;
          phasor(dtime, phase) = phasor_phase(dtime*2, phase) <: <=(dtime), (*(-1) + dtime*2), _ : select2;
          
          // Single channel reverse delay processing
          reverse_delay = _ <: direct, delayed_and_reversed :> _
          with {
              // Get current sample rate for scaling controls
              current_sr = ma.SR;
              
              // User controls with sample rate scaling
              delay_base = hslider("Delay Time[style:knob]", 0.5, 0.02, 1, 0.01);  
              delay_time = min(MAX_DELAY-1, delay_base * current_sr);
              dry_wet = hslider("Dry/Wet[style:knob]", 0.5, 0, 1, 0.01);
              
              // Direct path with gain
              direct = *(1-dry_wet);
              
              // Delay into reverse buffer implementation
              delayed = de.delay(MAX_DELAY, delay_time);
              
              // Reverse buffer using rwtable
              reversed = rwtable(MAX_DELAY, 0.0, 
                                phasor(delay_time, 0) : int,    // Write index
                                _,                              // Input signal
                                phasor(delay_time, delay_time/2) : int   // Read index
                               );
                               
              // Process delayed signal through reverse buffer with gain
              delayed_and_reversed = delayed : reversed : *(dry_wet);
          };
          
          // Main stereo processing - apply reverse_delay to each channel independently
          process = reverse_delay, reverse_delay;
          

          Free Party, Free Tekno & Free Software too

          JulesVJ 1 Reply Last reply Reply Quote 1
          • JulesVJ
            JulesV @Mighty23
            last edited by JulesV

            @Mighty23 It works great, thanks!

            The delay value is in 0 - 1 range. What is it referring to?
            So what about time-based reverse? Like milliseconds, or 1/4, 1/8, 1/16?

            M 2 Replies Last reply Reply Quote 0
            • M
              Mighty23 @JulesV
              last edited by

              @JulesV
              the delay value (0-1 range) refers to delay time as a fraction of one second.

              • delay_base ranges from 0.02 to 1.0

              • This gets multiplied by current_sr (sample rate), so:
                at 44.1kHz: 0.5 = 22.050 samples = 500ms, 1.0 = 44.100 samples = 1 second

              To do this in milliseconds, I can modify the implementation (I just need to remember). If I don't remember or am unmotivated to do it, feel free to message me and send a reminder :).

              To switch from ms to tempo sync, there's a node for this:
              https://docs.hise.dev/scriptnode/list/control/tempo_sync.html

              Free Party, Free Tekno & Free Software too

              1 Reply Last reply Reply Quote 1
              • M
                Mighty23 @JulesV
                last edited by

                @JulesV

                import("stdfaust.lib");
                
                MAX_DELAY = 192000;  
                
                // Phase-shifted phasor for reverse reading
                // dtime: delay time in samples
                // phase: phase offset for reading position
                phasor_phase(dtime, phase) = ((os.lf_rawsaw(dtime) + phase) % dtime) : int;
                phasor(dtime, phase) = phasor_phase(dtime*2, phase) <: <=(dtime), (*(-1) + dtime*2), _ : select2;
                
                // Single channel reverse delay processing
                reverse_delay = _ <: direct, delayed_and_reversed :> _
                with {
                    // Get current sample rate for scaling controls
                    current_sr = ma.SR;
                    
                    // User controls in milliseconds 
                    delay_ms = hslider("Delay Time (ms)[style:knob]", 500, 20, 1000, 1);  
                    delay_time_raw = min(MAX_DELAY-1, delay_ms * current_sr / 1000);
                    delay_time = si.smooth(ba.tau2pole(0.05), delay_time_raw);
                    dry_wet = hslider("Dry/Wet[style:knob]", 0.5, 0, 1, 0.01);
                    
                    // Direct path with gain
                    direct = *(1-dry_wet);
                    
                    // Delay into reverse buffer implementation
                    delayed = de.delay(MAX_DELAY, delay_time);
                    
                    // Reverse buffer with windowing to reduce clicks
                    // Use a single buffer with smoothed reading
                    write_idx = phasor(delay_time, 0) : int;
                    read_idx = phasor(delay_time, delay_time/2) : int;
                    
                    // Apply Hann window to reduce discontinuities
                    window_pos = (phasor(delay_time, delay_time/2) / delay_time);
                    hann_window = 0.5 * (1.0 - cos(2.0 * ma.PI * window_pos));
                    
                    reversed = _ <: rwtable(MAX_DELAY, 0.0, write_idx, _, read_idx) * hann_window;
                                     
                    // Process delayed signal through reverse buffer with filtering and gain
                    // Add gentle low-pass filter to reduce aliasing artifacts
                    delayed_and_reversed = delayed : reversed : fi.lowpass(2, current_sr * 0.45) : *(dry_wet);
                };
                
                // Main stereo processing - apply reverse_delay to each channel independently
                process = reverse_delay, reverse_delay;
                
                

                I tried a few additions because the previous version worked decently with mostly percussive sounds, but with instruments like vocals or pads, it tended to create too many clicks/pops and glitchy audio.

                Test this new version (milliseconds > Delay time) included.

                Free Party, Free Tekno & Free Software too

                JulesVJ 1 Reply Last reply Reply Quote 2
                • JulesVJ
                  JulesV @Mighty23
                  last edited by

                  @Mighty23 Thanks for the update, it sounds better. I think you made the transitions smoother, right?

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                  • M
                    Mighty23 @JulesV
                    last edited by

                    @JulesV said in Free Reverse Delay built in RNBO:

                    Thanks for the update, it sounds better. I think you made the transitions smoother, right?

                    Yes, transitions are smoother now because of three changes:

                    1- Smoothed delay time

                    delay_time = si.smooth(ba.tau2pole(0.05), delay_time_raw);
                    

                    2 - Hann window on reverse buffer

                    hann_window = 0.5*(1 - cos(2*ma.PI*window_pos));
                    reversed = rwtable(...) * hann_window;
                    

                    3 - Gentle (?) low-pass filter:

                    ... : fi.lowpass(2, current_sr*0.45);
                    

                    this is to try to: avoid clicks, reduce aliasing

                    Free Party, Free Tekno & Free Software too

                    JulesVJ 2 Replies Last reply Reply Quote 1
                    • JulesVJ
                      JulesV @Mighty23
                      last edited by

                      @Mighty23 I tried it on vocals and pads, and rarely there're some clicks, but it's definitely better than the previous one.

                      Will there be a problem if we set the delay time higher than 1000 ms? Like 3000 ms?

                      1 Reply Last reply Reply Quote 0
                      • JulesVJ
                        JulesV @Mighty23
                        last edited by JulesV

                        @Mighty23 It seems 1000 ms is very short for low bpm and 1 bar.

                        Is it possible to increase the max delay time to 5000 ms?

                        1 Reply Last reply Reply Quote 0
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