Soundhack phase vocoding tricks5/18/2023 ![]() The algorithm will look backwards at the previous input as well utilizing a ring buffer to maintain continuity and conserve space. ![]() Whenever the audio buffer is full we will then spawn Number of Samples / Hop Size streams on the gpu to process the input signal. In order to run this process in real time we must first use a thread based callback system on the CPU. Unfortunately do to bugs in the initial algorithm I was unable to finish it on time. I am currently in the process of developing the Real Time version of this algorithm. ![]() Developing the Algorithm for Real Time Audio Processing I have built a system into the code to test the overlap adding characteristics and believe that their may be an issue with windowing and shifting. I have yet to determine the cause of the bug On closer inspection it appears that their is a changing amplitude issue. Their is a bug that is smearing phase information across the frequency spectrum The current code outputs a distorted wave form when shifted When given an input wav at 440 Hz as shown by this spectrum diagram Resynthesis is simply the inverse fourier transform as well as proper scaling (by window size) and time shifting of the signal Current State of the system In the case of pitch modification careful measure must be taken to determine the continuous phase or phase difference between windows. Time shifting would feature modification of the hop/overlap size as well as scaling of phase information over time. The Processing that occurs here depends on the effect. The Fourier Transform transforms the time domain signal into polar coordinates which can then be changed into magnitude and phase information. Zero padding increases the accuracy of the next step which is the Fourier Transform. The next phase is to time alias and zero pad the input. The purpose of the windowing function is to spread power and phase information across multiple "snapshots" of the audio wave form. Graphically the algorithm can be described as followsĭuring the analysis phase we first take the input wave form and window it using a windowing function such as hanning or hamming. I hope to explore both options to determine which will work best in real time. These new vectors are then resynthesized using an Inverse Fourier Transform or Additive Synthesis. That allows us to view and manipulate not only the instantaneous phase but also the phase difference, representing the varying frequency components of a signal.Īfter completing the analysis, time stretching and pitch shifting can be applied to the frequency content through operations as simple as vector scaling. Taking individual fourier transforms of samples that take into account parts of the frame before them, we are able to create a Short Time Fourier Transform Analysis The algorithmĪt the core of Phase Vocoding is the Fast Fourier Transform, a complex mathematical tool separating time domain signals into their discrete frequency components. By utilizing a GPU to parallelize the fft and frequency domain operations, this project aims to put the power of Phase Vocoding into the hands of all musicians. These effects are commonly loaded offline, not allowing for real time manipulation or performance (Paul Stretch) or require computers to run the software (Autotune, the Messina, etc.). ![]() Phase Vocoding forms the basis of complex tools such as Autotune and the Prizmizer as well as Time stretching useful in quantization.
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