Re: [Freetel-codec2] Silly question regarding SM-1000 sample rate

2015-09-19 Thread Stuart Longland 
On 20/09/15 03:43, Tomas Härdin wrote:
> Couldn't you do further low-pass filtering in software, then decimate to
> 8 kHz? (caveat: I haven't checked if the code actually does this)

That's exactly what it does.  It samples at 16kHz from the ADC, filters
it to produce an 8kHz stream.

Actually a thought just occurred, could we just do nearest-neighbour
upsampling to get back to 16kHz?

I'll admit I havent looked at what algorithm is used to do the
resampling, however this is not a device that will be used by
audiophiles (it doesn't do 192kHz 32-bit FLAC for starters) so linear
interpolation or nearest-neighbour would probably work for getting it
back to 16kHz.
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  ...it's backed up on a tape somewhere.

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Re: [Freetel-codec2] Silly question regarding SM-1000 sample rate

2015-09-19 Thread Brady O'Brien 
I don't think a linear interpolator would work here. There's quite a bit of
attenuation in the 0-Fs/4 band (~8dB by Fs/8) and only around -28dB on the
aliases. -28dB might be enough for the aliases, but I don't think it's flat
enough in the Fs/4 band to be useful.

https://ccrma.stanford.edu/~jos/pasp/Linear_Interpolation_Frequency_Response.html

On Sat, Sep 19, 2015 at 4:03 PM, Stuart Longland  wrote:

> On 20/09/15 03:43, Tomas Härdin wrote:
> > Couldn't you do further low-pass filtering in software, then decimate to
> > 8 kHz? (caveat: I haven't checked if the code actually does this)
>
> That's exactly what it does.  It samples at 16kHz from the ADC, filters
> it to produce an 8kHz stream.
>
> Actually a thought just occurred, could we just do nearest-neighbour
> upsampling to get back to 16kHz?
>
> I'll admit I havent looked at what algorithm is used to do the
> resampling, however this is not a device that will be used by
> audiophiles (it doesn't do 192kHz 32-bit FLAC for starters) so linear
> interpolation or nearest-neighbour would probably work for getting it
> back to 16kHz.
> --
> Stuart Longland (aka Redhatter, VK4MSL)
>
> I haven't lost my mind...
>   ...it's backed up on a tape somewhere.
>
>
> --
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Re: [Freetel-codec2] Silly question regarding SM-1000 sample rate

2015-09-19 Thread Tomas Härdin 
Couldn't you do further low-pass filtering in software, then decimate to
8 kHz? (caveat: I haven't checked if the code actually does this)

/Tomas

On Fri, 2015-09-18 at 16:13 +1000, 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,
> > 
> > 
> > --
> > Monitor Your Dynamic Infrastructure at Any Scale With Datadog!
> > Get real-time metrics from all of your servers, apps and tools
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> > SourceForge users - Click here to start your Free Trial of Datadog now!
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> > 
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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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Re: [Freetel-codec2] Silly question regarding SM-1000 sample rate

2015-09-18 Thread glen english 

  
  
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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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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Re: [Freetel-codec2] Silly question regarding SM-1000 sample rate

2015-09-18 Thread Stuart Longland 
Hi David & Glen,
On 18/09/15 16:13, glen english wrote:
> 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.)

Fair enough.  I just wondered, seemed that we could save some CPU
processing power if we used a lower rate, but good point on the analogue
filtering, I wasn't thinking of that side of things.

Definitely we want to avoid nasty aliasing, and having a higher rate
does make the filter cutoff less critical.
-- 
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I haven't lost my mind...
  ...it's backed up on a tape somewhere.



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Re: [Freetel-codec2] Silly question regarding SM-1000 sample rate

2015-09-18 Thread David Rowe 
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,
>>
>>
>> --
>> Monitor Your Dynamic Infrastructure at Any Scale With Datadog!
>> Get real-time metrics from all of your servers, apps and tools
>> in one place.
>> SourceForge users - Click here to start your Free Trial of Datadog now!
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>>
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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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Re: [Freetel-codec2] Silly question regarding SM-1000 sample rate

2015-09-18 Thread glen english 
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,
>>>
>>>
>>> --
>>> Monitor Your Dynamic Infrastructure at Any Scale With Datadog!
>>> Get real-time metrics from all of your servers, apps and tools
>>> in one place.
>>> SourceForge users - Click here to start your Free Trial of Datadog now!
>>> http://pubads.g.doubleclick.net/gampad/clk?id=241902991=/4140
>>>
>>>
>>> ___
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>>> Freetel-codec2@lists.sourceforge.net
>>> https://lists.sourceforge.net/lists/listinfo/freetel-codec2
>> --
>> -
>> 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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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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Re: [Freetel-codec2] Silly question regarding SM-1000 sample rate

2015-09-17 Thread David Rowe 
Hi Stuart,

Running the ADC/DAC at 16 kHz largely removes the need for analog 
anti-aliasing/reconstruction filters, significantly simplifying the 
hardware.

- David

On 17/09/15 19:23, 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,
>
>
>
> --
> Monitor Your Dynamic Infrastructure at Any Scale With Datadog!
> Get real-time metrics from all of your servers, apps and tools
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> SourceForge users - Click here to start your Free Trial of Datadog now!
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