#!/usr/bin/env python3 # -*- coding: utf-8 -*- """ Author: C0ffymachyne License: GPLv3 Version: 1.0.0 Description: Pad Synthesiser port of code from this article : https://zynaddsubfx.sourceforge.io/doc/PADsynth/PADsynth.htm#c_implementation """ import torch import math from typing import Tuple, Dict, Any, Union from ..core.io import audio_to_comfy_3d class SignalProcessingPadSynth: @classmethod def INPUT_TYPES(cls) -> Dict[str, Any]: return { "required": { "sample_rate": ( "INT", {"default": 44100, "min": 8000, "max": 96000, "step": 1}, ), "fundamental_freq": ( "FLOAT", {"default": 261.0, "min": 20.0, "max": 2000.0, "step": 1.0}, ), "bandwidth_cents": ( "FLOAT", {"default": 40.0, "min": 10.0, "max": 100.0, "step": 1.0}, ), "number_harmonics": ( "INT", {"default": 64, "min": 1, "max": 128, "step": 1}, ), } } RETURN_TYPES = ("AUDIO",) RETURN_NAMES = ("audio",) CATEGORY = "Signal Processing" FUNCTION = "process" def process( self, sample_rate: int, fundamental_freq: float, bandwidth_cents: float, number_harmonics: int, ) -> Tuple[Dict[str, Union[torch.Tensor, int]]]: """ Apply PADsynth algorithm to generate audio. Parameters: samplerate (int): Sampling rate in Hz. fundamental_freq (float): Fundamental frequency in Hz. bandwidth_cents (float): Bandwidth in cents for Gaussian profile. number_harmonics (int): Number of harmonics to generate. Returns: Tuple[Dict[str, torch.Tensor]]: Generated audio with waveform and sample rate. """ # Define FFT size N = 262144 # As per C++ code # Use default amplitude distribution A = torch.zeros(number_harmonics, dtype=torch.double) A[0] = 0.0 # A[0] is not used for i in range(1, number_harmonics): A[i] = 1.0 / i if (i % 2) == 0: A[i] *= 2.0 # Initialize frequency amplitude and phase arrays freq_amp = torch.zeros(N // 2, dtype=torch.double) freq_phase = ( torch.rand(N // 2, dtype=torch.double) * 2.0 * math.pi ) # Random phases between 0 and 2pi # Define Gaussian profile function def profile(fi: torch.Tensor, bwi: torch.Tensor) -> torch.Tensor: x = fi / bwi x_sq = x**2 # Avoid computing exp(-x^2) for x_sq > 14.71280603 mask = x_sq <= 14.71280603 result = torch.zeros_like(x_sq) result[mask] = torch.exp(-x_sq[mask]) / bwi[mask] return result # Convert bandwidth from cents to Hz # bw_Hz = (2^(bw/1200) -1) * f * nh # Convert bandwidth_cents to multiplier bw_multiplier = 2.0 ** (bandwidth_cents / 1200.0) - 1.0 # Populate frequency amplitude array for nh in range(1, number_harmonics): f_nh = fundamental_freq * nh bw_Hz = bw_multiplier * f_nh bwi = bw_Hz / (2.0 * sample_rate) fi = f_nh / sample_rate # Normalized frequency # Create tensors for frequency bins i = torch.arange(N // 2, dtype=torch.double) # Normalized frequency for each bin normalized_freq = ( i / N ) # Equivalent to i * (sample_rate / N) / sample_rate = i / N # Compute profile fi_tensor = torch.full_like(i, fi) bwi_tensor = torch.full_like(i, bwi) profile_values = profile(normalized_freq - fi_tensor, bwi_tensor) # Update frequency amplitude freq_amp += profile_values * A[nh] # Construct complex frequency domain tensor real = freq_amp * torch.cos(freq_phase) imag = freq_amp * torch.sin(freq_phase) freq_complex = torch.complex(real, imag) # Shape: (N//2,) # Perform IFFT using torch.fft.irfft smp = torch.fft.irfft(freq_complex, n=N) # Shape: (N,) # Normalize the signal to prevent clipping max_val = torch.max(torch.abs(smp)) if max_val < 1e-5: max_val = 1e-5 # Prevent division by zero smp = smp / (max_val * math.sqrt(2)) # Normalize to 1/sqrt(2) as in C++ code # Convert to float32 for saving smp = smp.float() # Prepare waveform tensor: (C, N) waveform_out = smp.unsqueeze(0) # Mono audio return audio_to_comfy_3d(waveform_out, sample_rate)