where is low pass filter? On Sun, Jul 22, 2018, 23:22 Kjetil Matheussen <k.s.matheus...@gmail.com> wrote:
> Maybe this will give you an idea: > > 48khz -> 8khz: > float get_output_sample(get_input_sample){ > static int i=0; > static float sample; > > if (i % 6 == 0) > sample = get_input_sample(); > > i++; > > return sample; > } > > 8khz -> 48khz: > float get_output_sample(get_input_sample){ > float ret = get_input_sample(); > > for(int i=1;i<6;i++) > get_input_sample(); > > return ret; > } > > Not the best sound quality though. > > > On Sun, Jul 22, 2018 at 9:55 PM, Alex Dashevski <alexd...@gmail.com> > wrote: > >> real time >> >> On Sun, Jul 22, 2018, 22:52 jpff <j...@codemist.co.uk> wrote: >> >>> Were you expecting real-time/time-critical resampling or offline? >>> >>> _______________________________________________ >>> 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 >> > > _______________________________________________ > dupswapdrop: music-dsp mailing list > music-dsp@music.columbia.edu > https://lists.columbia.edu/mailman/listinfo/music-dsp
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