yeah you can pretty much get away with NO aliasing filter considering 
the source ( a band limited microphone) and the required voiceband SN.

still, good reason to have at least a single RC- from say switchmode 
power supply whistles etc getting back into the input, and aliasing down 
into the baseband (from say 50-300kHz) , or even the transmitter.

Yes-on 160m, I have had the RF aliasing back into the passband from the 
32kHz sampled mic audio.....

g
On 18/09/2015 7:35 PM, David Rowe wrote:
> Also there is very little speech energy out past 12 kHz, i.e. its
> already 50dB down, and this is communications quality speech so a little
> aliasing is lost in the codec artefacts.
>
> On the radios interface side the audio is band limited by the radio's
> frequency response, ie the xtal filter BW.
>
> - David
>
> On 18/09/15 15:43, glen english wrote:
>> Hi Stuart
>>
>> An elaboration on David's reply.
>>
>> We want our audio BW to go to at least say 3.4 kHz . That is what fixed
>> line telephones offer.
>>
>> If we were to sample at 8kHz, our nyquist frequency is 4kHz of course.
>> Any spectral information above the nyquist rate will be aliased bay into
>> the baseband, IE below the nyquist rate.
>>
>> IF we assume we don't care about aliases between 3.4 and 4kHz, then this
>> allows for signals up to 4.0 - 3.4 = 0.6,  and 0.6+Fyquist(4kHz) =
>> 4.6kHz. (as a 4.5kHz signal will get aliased to 3.5 kHz.)
>>
>> Assume we want our aliases 50dB down, so our aliasing filter must go
>> from 0dB down at 3.4kHz to 50 dB down at 4.6kHz
>>
>> Clearly alot of work in the analog department- probably needs an
>> elliptic filter.
>> a transition bandwidth factor of (4.6/3.4) = 1.35
>>
>> Now, if the sample rate was 16kHz, now, Fnyquist is 8 kHz. again, we
>> dont care about aliases above 3.4kHz,
>> and Fnyquist, now 8kHz - 3.4kHz is 4.6kHz.
>>
>> So, for freq up to Fnyquist + 4.6kHz - IE 12.6kHz  will not alias into
>> problem areas.
>>
>> so, now our analog filter spec if relaxed
>> 0dB down at 3.4kHz, and 50dB down at 12.6kHz
>> a transition bandwidth factor of (12.6/3.4) = 3.7
>>
>>
>>
>> QED
>>
>>
>>
>>
>> On 17/09/2015 7:53 PM, Stuart Longland wrote:
>>> Hi all,
>>>
>>> Just looking at the source code, it hit me.  When in analogue mode, we
>>> run 16kHz sample rate, fair enough.
>>>
>>> But in DV mode, we sample the ADC at 16kHz, downconvert to 8kHz, pass it
>>> through the modem and codec, then upconvert back to 16kHz for the DAC.
>>>
>>> We're dealing with voice frequencies, with SSB transmit bandwidths of
>>> less than 3kHz.
>>>
>>> Why not do the whole lot at 8kHz and save some CPU time?  Maybe some
>>> rate-switching when in analogue or DV mode might be an option too, so we
>>> run ADC/DAC at full-rate in analogue (for higher fidelity), then switch
>>> to half-rate when we're doing DV.
>>>
>>> Regards,
>>>
>>>
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>> --
>> -
>> Glen English
>> RF Communications and Electronics Engineer
>>
>> CORTEX RF
>> &
>> Pacific Media Technologies Pty Ltd
>>
>> ABN 40 075 532 008
>>
>> PO Box 5231 Lyneham ACT 2602, Australia.
>> au mobile : +61 (0)418 975077
>>
>>
>>
>> ------------------------------------------------------------------------------
>>
>>
>>
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-- 
-
Glen English
RF Communications and Electronics Engineer

CORTEX RF
&
Pacific Media Technologies Pty Ltd

ABN 40 075 532 008

PO Box 5231 Lyneham ACT 2602, Australia.
au mobile : +61 (0)418 975077



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