sic-dsp-boun...@music.columbia.edu> on behalf of gm <g...@voxangelica.net>
Sent: Saturday, March 10, 2018 1:20 AM
To: music-dsp@music.columbia.edu
Subject: Re: [music-dsp] Clock drift and compensation
The problem I see is that your sine wave needs to have a precise amplitude for
the arcs
eck.
>
> -ben
>
>
>
> --
> *From:* music-dsp-boun...@music.columbia.edu <music-dsp-bounces@music.
> columbia.edu> on behalf of gm <g...@voxangelica.net>
> *Sent:* Monday, January 29, 2018 1:29 AM
>
> *To:* music-dsp@musi
eck.
-ben
*From:* music-dsp-boun...@music.columbia.edu
<music-dsp-boun...@music.columbia.edu> on behalf of gm
<g...@voxangelica.net>
*Sent:* Monday, January 29, 2018 1:29 AM
*To:* music-dsp@music.columbia.edu
*Subject:* Re: [music-dsp] Clock drift and co
robert bristow-johnson
> <r...@audioimagination.com>
> Sent: Monday, February 5, 2018 1:01 PM
> To: music-dsp@music.columbia.edu
> Subject: Re: [music-dsp] Clock drift and compensation
>
>
> Ben, can you confirm that what you want to do is Asynchronous Sample
> Rate Conversio
I've done a few different systems similar to what you're describing - a
radio front-end tuner that generates baseband I & Q at audio rates
that's then further processed by a DSP to extract true audio.
Normally what I do is slave the DSP rate to the tuner audio rate. That's
usually possible
ic.columbia.edu>
>> <music-dsp-boun...@music.columbia.edu
>> <mailto:music-dsp-boun...@music.columbia.edu>> on behalf of gm
>> <g...@voxangelica.net <mailto:g...@voxangelica.net>>
>> *Sent:* Saturday, January 27, 2018 5:20 PM
>> *To:
27, 2018 5:20 PM
*To:* music-dsp@music.columbia.edu <mailto:music-dsp@music.columbia.edu>
*Subject:* Re: [music-dsp] Clock drift and compensation
I don't understand your project at all so not sure if this is helpful,
probably not,
but you can calculate the drift or instantanous frequency of a sine wave
:* music-dsp@music.columbia.edu
*Subject:* Re: [music-dsp] Clock drift and compensation
diff gives you the phase step per sample,
basically the frequency.
However the phase will jump back to zero periodically when the phase exceeds
360°
(when it wraps around) in this case diff will get you a wrong result.
gt; on behalf of gm
<g...@voxangelica.net>
*Sent:* Saturday, January 27, 2018 5:20 PM
*To:* music-dsp@music.columbia.edu
*Subject:* Re: [music-dsp] Clock drift and compensation
I don't understand your project at all so not sure if this is helpful,
probably not,
but you can calculate the drift
an I Q signal
> diff-> gives what ?
> unwrap ?
>
> -ben
>
>
> From: music-dsp-boun...@music.columbia.edu
> <music-dsp-boun...@music.columbia.edu> on behalf of gm <g...@voxangelica.net>
> Sent: Saturday, January 27, 2018 5:20 PM
> To: music-dsp@music.columb
___
From: music-dsp-boun...@music.columbia.edu
<music-dsp-boun...@music.columbia.edu> on behalf of gm <g...@voxangelica.net>
Sent: Saturday, January 27, 2018 5:20 PM
To: music-dsp@music.columbia.edu
Subject: Re: [music-dsp] Clock drift and compensation
I don't understand your proje
I don't understand your project at all so not sure if this is helpful,
probably not,
but you can calculate the drift or instantanous frequency of a sine wave
on a per sample basis
using a Hilbert transform
HT -> Atan2 -> differenciate -> unwrap
___
, January 24, 2018 2:17 AM
To: music-dsp@music.columbia.edu
Subject: Re: [music-dsp] Clock drift and compensation
On Tue, Jan 23, 2018 at 04:17:40PM +, Benny Alexandar wrote:
> How to design a control system such that a digital baseband frame of duration
> 'T' ms is mapped to audio an
On Tue, January 23, 2018 7:17 pm, Benny Alexandar wrote:
> Now if the tuner xtal is drifting then the dsp audio streaming needs to
> adjust to that drift, else buffer overflow or underrun happens as the
> sample rates doesn't match.
Assuming you do not have the option of modifying the hardware,
On Tue, Jan 23, 2018 at 04:17:40PM +, Benny Alexandar wrote:
> How to design a control system such that a digital baseband frame of duration
> 'T' ms is mapped to audio and adjust the drift ?
A classic asynchronous resampling problem. Look at something like
SMPTE drop frame resampling using
Hi All,
I have a problem to solve,
I have a system which has a tuner chip and DSP chip, both are clocked by
independent
xtals. Tuner gives the baseband samples to DSP, and tuner is master and dsp is
slave.
Dsp does the demodulation of base band samples and does the audio decoding and
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