[hpsdr] Ozy/Mercury Protocol

[Andrea Montefusco]

Phil Harman wrote:

> For a soundcard based SDR (e.g. SDR1000 or Softrock) then the clock in 
> the sound card determines the rate that samples are fed to the PC code 
> and also how fast they are played back.  Since these use the same clock 
> then it does not need to be exactly 48kHz etc and any drift is common to 
> both the ADC and DAC paths.   If you are using Jack and Linux then I 
> would hope that Jack gets its clock from the sound card.

According to the documentation, inside jackd there is a module that
receives hw  interrupts from the sound card and uses them to provide
the clock tick for the callback chain.
The application programs, like DttSP, Audacity and so on, are hooked
to the chain and do the processing when its callback is recalled.

A similar problem arises when one has to use multiple sound card:
to use jack, it is mandatory synchronize all cards from the same clock source.

> The only way I can see this working is to resample the data going to 
> Jack to a lower rate so it never runs out of data.  This will reduce the 
> bandwidth of the signals we can process to the new (lower) sampling rate.

Alternatively should be possible extract the clock from USB: to do that we need
to write an 'HPSDR' ALSA driver (ALSA is the main Linux audio subsytem).
The driver simply read (and write) the stream from/to the right USB endpoint.
In this manner the jack's syncronization source is just the stream coming out
from USB and all software that comes next would be in synch.
Of course, the output stream has to be sent to same USB stream to reach the
onboard DAC.


One question for Alberto I2PHD: when you wrote the WindRad DLL for Perseus,
did you got the same problem ?
The problem seems to me very similar, because you are getting data at the FPGA rate
and send them to audio card.

         *am*

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Andrea Montefusco iw0hdv        http://www.montefusco.com
tel: +393356992791                     fax: +390623318709
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