Re: linear PCM
Re: linear PCM
- Subject: Re: linear PCM
- From: Roland Silver <email@hidden>
- Date: Tue, 15 Jan 2008 12:17:07 -0700
I proposed:
signal(t) = sin(2*pi*f*t), where t = 0, dt, 2*dt, 3*dt ...
where dt = 1/sampleRate.
In what way does that fail "to actually calculate [the] samples spaced
1/sampleRate apart"?
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Message: 1
Date: Mon, 14 Jan 2008 21:25:49 +0100
From: "tahome izwah" <email@hidden>
Subject: Re: linear PCM
To: "Core Audio" <email@hidden>
No, it does not, for the exact reason that you have stated - you need
to actually calculate your samples spaced 1/sampleRate apart.
So your sampled sine becomes: signal[t] = sin(2.*pi*f*t/sampleRate);
HTH
--th
2008/1/14, Roland Silver <email@hidden>:
Brian,
Thanks for your help with my questions about Audio Queue.
I'm puzzled by your last comment: "However, you should note that
(2*pi*f*t) does not account for sampling rate. You'll need to modify
that parameter if you want frequency f to be accurate when the
samples
are played back."
In my example, I compute the successive values of sin(2*pi*f*t) for
successive values of t which are 1/samplingRate apart. Does that not
account for sampling rate?
--RS
Roland Silver
email@hidden
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