Hi, I need to do resampling on android. Could you give me code on c/c++/Java?
On Tue, Jul 24, 2018, 08:56 Tom O'Hara <tom_i...@ticino.com> wrote: > I've done many resamplers over the decades (48<->32, 24,16,8) and always > used FIRs for these reasons. > > Tom > > On 7/23/2018 6:25 PM, Nigel Redmon wrote: > > Some articles on my website: > > > http://www.earlevel.com/main/category/digital-audio/sample-rate-conversion/, > > > especially the 2010 articles, but the Amp Sim article might be a > > helpful overview. > > > > 48k -> 8k: Filter with a lowpass with cutoff below 4k; keep 1 sample, > > throw away 5, repeat. > > > > 8k -> 48k: Use 1 sample, follow it with 5 new samples of 0 value, > > repeat; filter with a lowpass filter with cutoff below 4k. > > > > Nuances: > > > > A linear phase FIR is a popular choice for the lowpass filter (odd > > length, Kaiser windowed sinc is a good choice). In downsampling, you > > don’t have to calculate the samples you intend to discard, and in > > upsampling, you don’t need to do the operations for added 0-valued > > samples. > > > > You want the filter stop band (above 4k) to have suitable attenuation > > (Kaiser is nice for this, because you can specify it, trading off with > > transition sharpness). > > > > Advance topic: You can optimize performance by doing it in two stages > > (3x, 2x). You win by noting that the first stage doesn’t have to be > > perfect, and long as the second stage cleans up after it. > > > > _______________________________________________ > dupswapdrop: music-dsp mailing list > music-dsp@music.columbia.edu > https://lists.columbia.edu/mailman/listinfo/music-dsp
_______________________________________________ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
