Timeline for How can the quality of a resampling algorithm be measured?
Current License: CC BY-SA 4.0
13 events
| when toggle format | what | by | license | comment | |
|---|---|---|---|---|---|
| Feb 11 at 1:27 | comment | added | sina bala | And if you have 32 bit MCU with fpu, then there's plenty available dsp libraries, "all the usual suspects" for c and c++, in fact. So, not quite sure you need much help. Just go and use what is out there under permissive licenses. | |
| Feb 11 at 1:26 | comment | added | sina bala | @TobiasHienzsch "quality is the most important metric" but quality means different things to different applications. You will have to be more specific. | |
| Feb 10 at 23:57 | comment | added | Tobias Hienzsch | @robertbristow-johnson I use Python + Jupyter notebooks for almost all prototypes. The final version will run on an ARM Cortex MCU with 700 Mhz. All the things around the resampling (wav-reading, convolution) already exists in C++. | |
| Feb 10 at 23:54 | comment | added | Tobias Hienzsch | @TimWescott Actually I'm interested in exactly the opposite. Quality is not the most important aspect I was after. The quality metrics where meant as a testing utility, so that I could verify whatever algorithm I decide to take on. | |
| Feb 10 at 22:44 | history | became hot network question | |||
| Feb 10 at 22:12 | comment | added | robert bristow-johnson | And if you're processing .wav files, what language are you writing this code with? It could also be done with MATLAB. If you're doing this in Python, you need to find someone else with Python code or maybe translate my MATLAB code. | |
| Feb 10 at 22:09 | comment | added | robert bristow-johnson | Do you have MATLAB? If you're writing your own resampler, I might be able to help you with MATLAB code to compute polyphase coefficients. It's about designing really good brickwall FIR low-pass filters. | |
| Feb 10 at 18:37 | comment | added | TimWescott | You seem to be thinking of quality metrics in terms of what is the absolute best one, or set, to use. I strongly suggest that you, instead, ask yourself what matters to your application. If you're feeding acoustic data to a machine learning algorithm you probably want different resampling than if you're resampling Led Zepplin songs, and you may want yet another set of metrics if you're resampling baroque-era string ensemble music. | |
| Feb 10 at 18:32 | comment | added | TimWescott | Linear interpolation is polyphase fltering -- it's just polyphase filtering with a 2-point parent anti-aliasing filter. | |
| Feb 10 at 18:02 | answer | added | Baddioes | timeline score: 1 | |
| Feb 10 at 16:44 | answer | added | Hilmar | timeline score: 6 | |
| Feb 10 at 15:46 | answer | added | robert bristow-johnson | timeline score: 2 | |
| Feb 10 at 14:41 | history | asked | Tobias Hienzsch | CC BY-SA 4.0 |