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May 28, 2017 at 18:30 vote accept Anup Agarwal
May 28, 2017 at 10:49 answer added Anup Agarwal timeline score: 0
May 28, 2017 at 7:49 vote accept Anup Agarwal
May 28, 2017 at 10:25
May 27, 2017 at 19:14 answer added Fusho timeline score: 0
May 27, 2017 at 6:07 comment added Anup Agarwal Could you please elaborate @MarcusMüller, I didn't quite understand what you meant by normalizing freq to sampling rate
May 26, 2017 at 15:17 comment added MimSaad @MarcusMüller seems a good answer , I think I have the same "Standard beginner's mistake" , I am also interested in full version your answer .
May 26, 2017 at 13:43 comment added MBaz Read the documentation of audiowrite. It expects the signal to be within a certain range of amplitudes. You may also find fftshift interesting.
May 26, 2017 at 12:52 comment added Dan Boschen @MarcusMüller good answer! Why not paste below so the question is not left open?
May 26, 2017 at 12:42 comment added Marcus Müller you forgot to normalize your frequency to the sampling rate. Standard beginner's mistake – always ask yourself how many samples should your period be long (and not how many seconds); there's no "time" or "frequency" info attached to samples; they're just a sequence of numbers. The same sequence played back ("interpreted") at a sampling rate of 44.1 kHz will sound different than if played at let's say 16 kHz.
May 26, 2017 at 12:20 review First posts
May 26, 2017 at 12:39
May 26, 2017 at 12:10 history asked Anup Agarwal CC BY-SA 3.0