Hi Robert,

Yes, I need to do ASRC, and the challenge is how to estimate the drift
and correct it.

As I mentioned in the earlier attached figure, DSP is slave and tuner chip 
feeds the baseband samples and is the master.
Now the question is where to do the timestamping for correctly estimate the 
drift.
The system is an embedded platform having a tuner chip and DSP chip, both have 
independent oscillators ( Xtal)  for providing the clock.
So my question is how to timestamp the audio data. After channel decoder the 
compressed audio will have variable decoding times
based on audio content. So this is not a good place to timestamp as it is very 
jittery.

Suppose every digital radio transmission frame duration T seconds  corresponds 
to T seconds of audio, can I timestamp the
baseband RF IQ samples when it arrives at the DSP ?   After demodulation and 
audio decoder calculate the max delay it can
have for worst case scenarios, and add that as target delay before playing out. 
Then while playing out each audio period
read the current time, the difference of current time - ( RF packet arrival 
time + Target Delay) should be ideally zero,
if audio plays out at the same rate as transmission of audio,.

-ben


________________________________
From: music-dsp-boun...@music.columbia.edu 
<music-dsp-boun...@music.columbia.edu> on behalf of 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 Conversion (ASRC)?  this is what Steffan is talking about and what
it looked like you were looking for in your first post.

If ASRC is what you wanna do, that is a combination of the SRC task
(like what is done to a sound file when you change the sample rate of
it) and servo systems like you get in control theory, like what is done
in a phase-lock loop (like hurrying up or slowing down based on the
delay).  If you wanna write your own code to do this, it's about those
two general DSP and digital control problems.  you will need to be able
to read a fast system clock (like what profilers read) to do this really
well.

r b-j

On 1/28/18 1:16 PM, STEFFAN DIEDRICHSEN wrote:
> Actually, there are SRC chips available from Texas Instruments, just
> take look at their website. They don’t cost too much and are found in
> countless digital mixing consoles.
>
> Best,
>
> Steffan
>
> Von meinem iPhone gesendet
>
>
>
> Von meinem iPhone gesendet
> Am 28.01.2018 um 17:19 schrieb Benny Alexandar <ben.a...@outlook.com
> <mailto:ben.a...@outlook.com>>:
>
>> Hi GM,
>>
>> >> HT -> Atan2 -> differenciate -> unwrap
>> Could you please explain how to find the drift using HT,
>>
>> HT -> gives real(I) & imaginary (Q) components of real signal
>> Atan2 -> the phase of an I Q signal
>> diff-> gives what ?
>> unwrap ?
>>
>> -ben
>>
>>
>> ------------------------------------------------------------------------
>> *From:* music-dsp-boun...@music.columbia.edu
>> <mailto:music-dsp-boun...@music.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:* 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
>> on a per sample basis
>> using a Hilbert transform
>> HT -> Atan2 -> differenciate -> unwrap
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--

r b-j                  r...@audioimagination.com

"Imagination is more important than knowledge."



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