Best practices implementing live subsampling/resampling?

(I apologize if this has been asked many times already)

I’m hoping to make a plugin involving live subsampling and resampling, similar to a DJ turntable.
I want to make sure I’m on the right track and not overlooking any functionality of JUCE.

I’m thinking of interpolating manually using oversampling to reduce aliasing, as this seems to be common practice. However, I’m wondering if I can use filtered sinc interpolation directly to avoid oversampling.

I looked at ResamplingAudioSource, Oversampling and Interpolators, but I’m not sure which best applies.

ResamplingAudioSource only goes forwards and not backwards (ratio > 0), but the code seems extremely close to what I hope to achieve.
However, it seems to use linear subsampling.

Oversampling has a fixed sampling ratio, so I cannot rapidly change it without adding stages.

Interpolators have a fixed speed ratio, so I could not rapidly change the speed from fast to slow.
Also, the WindowedSinc interpolator doesn’t seem to include a filter.

Ideally I would like to seek any arbitrary point in a buffer like so:

Any thoughts on this would help me know the best way forward, as I am very new to DSP in general.