Hi,
Apparently there are no timing information being broadcast and no meta data
fields available for getting these like in mpeg TS.
The only way is to find the symbol boundary and time stamp it, since the tuner
is feeding the base band samples to
DSP, if any tuner variations will be reflected in the timing of symbols in DSP.
Monitor this difference and adjust audio clock
in DSP, which is possible in hardware of DSP.
This is what I'm planning to implement, please find the attached figure,
Now the timing of input side is after detecting the start of symbol. Every
symbol will be timestamped and measure the time deviation between two symbols.
d = t1 - t0,
where t0 - time of arrival of symbol (n)
t1 - time of arrival of symbol (n+1)
d - time deviation between two symbols.
D - time duration between two symbols according to digital radio standards,
then error = ( D / d ) - 1
Please send your suggestions feedback regarding this approach.
-ben
________________________________
From: music-dsp-boun...@music.columbia.edu
<music-dsp-boun...@music.columbia.edu> on behalf of Andy Farnell
<padawa...@obiwannabe.co.uk>
Sent: Wednesday, 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 +0000, 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 div/modulo to calculate the
number of frames of m samples over which to interpolate to get
some new number of n samples.
Real problem is that you need to know the difference/drift
in the clocks. Is there some feature in your signal that helps
with this?
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