I wrote on android ndk and there is fastpath concept. Thus, I think that resampling can help me. Can you recommend me code example ? Can you give me an example of resampling ? for example from 48Khz to 8Khz and 8Khz to 48Khz. I found this: https://dspguru.com/dsp/faqs/multirate/resampling/ but it is not enough clear for me,
Thanks, Alex בתאריך יום ד׳, 3 באוק׳ 2018 ב-20:56 מאת Spencer Jackson < ssjackso...@gmail.com>: > > > On Wed, Oct 3, 2018 at 3:17 AM Alex Dashevski <alexd...@gmail.com> wrote: > >> >> if I do resampling before and after processing. for example, 48Khz -> >> 8Khz and then 8Khz -> 48Khz then will it help ? >> > > Lowering sample rate can help achieve lower latencies by giving you fewer > samples to process in the same amount of time but just downsampling and > then upsampling back doesn't really have any effect. > > > >> I don't understand why I need filter, This is to prevent alias but I >> can't understand why ? >> >> Technically you only need a filter if your signal has information above > the nyquist frequency of the lowest rate but this is not usually the case. > I think wikipedia explains aliasing pretty well: > https://en.wikipedia.org/wiki/Aliasing#Sampling_sinusoidal_functions . > Once the high frequency information aliases it cannot be recovered by > resampling back to the higher rate and your lower band information is now > mixed in with the aliased information. The filter removes this high > freqency data so that the low band stays clean through the whole process. > > > Is there option to decrease latency or delay ? >> > > The only way to reduce latency in your algorithm (unless there is some > error in the implementation) is to reduce the block size, so you process > 128 samples rather than 240. 240 isn't a very large amount of latency for a > pitch shifter which is typically a CPU intensive process and therefore most > implementations have relatively high latencies. > > I'm not sure I understand what you mean by the pitch duration requiring a > buffer-resize or sample-rate decrease. WSOLA creates a signal with more > samples than the input, you must resample that (usually by a non-integer > amount) to make it the correct number of samples then output that, and > reload your buffer with the next block of input data. Please clarify if you > mean some other issue. > > _Spencer > _______________________________________________ > dupswapdrop: music-dsp mailing list > music-dsp@music.columbia.edu > https://lists.columbia.edu/mailman/listinfo/music-dsp
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