Getting the most out of onboard HDA

[Jens M Andreasen-2]

Assuming a 7.1/192K/24bit $3:- chip onboard ... What next?

First I will say, it actually will do the planned send and return
to/from the two pieces of stereo outboard equipment I have (that is not
going anywhere,) as well as do separate headphones and monitors. So I
should be a happy camper ...

Suppose one was to face the real world of playing live again? In my
experience this can be a torturous venture into the realms of cheap
lights and the evil thyristors and diacs of this world, all trying to
make their voices heard through my equipment ...


So, how about turning a single unbalanced stereo jack into a single
balanced mono jack by converting a single mono signal to a stereo signal
with inverted phase, would that be a good idea? If so, I believe this
might work in a simalar way on microphone input as well.

If the above holds and we then have two (instead of one) DAC's working
push/pull on the line, would it be possible to take advantage of this?

What I have in mind here is that, the lower you get in the "bittiness",
the more the systenatic errors of distortion will be apparent.

For starters, how about having one the phases at a level slightly below
the other? This should ideally trigger transisitions between absolute
levels at slightly different times for the two phases, giving us an
extra 6dB of useable headroom.


mvh // Jens M Andreasen






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[justin smith]

On Fri, Sep 12, 2008 at 11:28 AM, Jens M Andreasen
<jens.andreasen@...> wrote:

> Assuming a 7.1/192K/24bit $3:- chip onboard ... What next?
>
> First I will say, it actually will do the planned send and return
> to/from the two pieces of stereo outboard equipment I have (that is not
> going anywhere,) as well as do separate headphones and monitors. So I
> should be a happy camper ...
>
> Suppose one was to face the real world of playing live again? In my
> experience this can be a torturous venture into the realms of cheap
> lights and the evil thyristors and diacs of this world, all trying to
> make their voices heard through my equipment ...
>
>
> So, how about turning a single unbalanced stereo jack into a single
> balanced mono jack by converting a single mono signal to a stereo signal
> with inverted phase, would that be a good idea? If so, I believe this
> might work in a simalar way on microphone input as well.
>
> If the above holds and we then have two (instead of one) DAC's working
> push/pull on the line, would it be possible to take advantage of this?
>
> What I have in mind here is that, the lower you get in the "bittiness",
> the more the systenatic errors of distortion will be apparent.
>
> For starters, how about having one the phases at a level slightly below
> the other? This should ideally trigger transisitions between absolute
> levels at slightly different times for the two phases, giving us an
> extra 6dB of useable headroom.
>
>
> mvh // Jens M Andreasen
>
>
>
>
>
>
> _______________________________________________
> Linux-audio-dev mailing list
> Linux-audio-dev@...
> http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
>

It should not be so hard to make a ladspa plugin that takes a mono
signal and gives you a stereo signal meant to be composited into the
two sides of a balanced signal, giving you double the original
absolute amplitude in headroom. Or maybe this would be better as a
daemon that provided a jack sink to be placed before your system
outputs.

If you just did a simple phase inversion, you would lose half of the
information you could be sending to the dac, you could easily use an
algorithm that sets things up so you have full bit resolution (ie.
double the rated resolution of a single channel).
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[Jens M Andreasen-2]

On Fri, 2008-09-12 at 12:14 -0700, Justin Smith wrote:
> If you just did a simple phase inversion, you would lose half of the
> information you could be sending to the dac, you could easily use an
> algorithm that sets things up so you have full bit resolution (ie.
> double the rated resolution of a single channel).

So what it is the magic number then for improved resolution? At first I
thought intuitively: -3dB!

But now I am not so sure. I can't explain to an audiemce why -3 is a
better number than any other.

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[justin smith]

On Fri, Sep 12, 2008 at 12:33 PM, Jens M Andreasen
<jens.andreasen@...> wrote:

> On Fri, 2008-09-12 at 12:14 -0700, Justin Smith wrote:
>> If you just did a simple phase inversion, you would lose half of the
>> information you could be sending to the dac, you could easily use an
>> algorithm that sets things up so you have full bit resolution (ie.
>> double the rated resolution of a single channel).
>
> So what it is the magic number then for improved resolution? At first I
> thought intuitively: -3dB!
>
> But now I am not so sure. I can't explain to an audiemce why -3 is a
> better number than any other.
>
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> Linux-audio-dev@...
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>

I am not sure if I understand what you aare saying herer at all.

What I imagined was this: instead of sending one floating point
output, you send two, out of phase with one nother, except ot perfetly
out of phase, so tht you retin the potential of the full bit depth (if
each signal were just the inverse of the other, you are wasting half
of the bits you send). Now that I think about it more, this woul be
most useful if the full signal chain used this format, and to really
double your bit depth sending -29 on one side and +1 on the other
would have to be different from sending -30 on one side, and 0 on the
other. I think I was just confused, and maybe you had something in
mind other than my erronious speculation.
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[Jens M Andreasen-2]

On Fri, 2008-09-12 at 12:49 -0700, Justin Smith wrote:
> I am not sure if I understand what you aare saying herer at all.
>
Go on?

> What I imagined was this: instead of sending one floating point
> output, you send two, out of phase with one nother, except ot perfetly
> out of phase,

Yes, as in: L = mono, R=mono * -1; // this is the baseline

> so tht you retin the potential of the full bit depth (if
> each signal were just the inverse of the other, you are wasting half
> of the bits you send). Now that I think about it more, this woul be
> most useful if the full signal chain used this format,

I disagree unless you have other anolog signals in mind.

>  .. and to really
> double your bit depth sending -29 on one side and +1 on the other
> would have to be different from sending -30 on one side, and 0 on the
> other. I think I was just confused, and maybe you had something in
> mind other than my erronious speculation.

With a 24bit signal of +/- 15 integer values, we would be drowned in
noise. But since this is foremost a theoretical discussion, let us stick
to those numbers.

Let us say that our programs current examination of the incoming float
leaves us with the impression that i'15' is the the nearest intger
approxination, then we could also naively multiply by -1 and send both
signals - that is to say L == 15 and R == -15 - down the balanced line.
This should give us the advantage of reduced noice from external
sources, since we at the receiving end will do the inverse, that is to
say that for the signal we care about we would dol: NewMono = L + (R *
-1) which will come out just the way planned and expected. The noise we
encountered on the balanced line affects both wires equally and are
hence eliminated since - following the equetion above - it will collopse
into: x-x == 0


Whatever ...


What I had in mind regarding the +6dB notion was that a signal can have
an (integer) amplitude of 0.499.. which we would normally flush to zero
because that is all we'we got, but if you have two of them - one on each
side of the sending line - then we could average out your signal and
pretend that we have a higher (actual) resolution than speified.

/j

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[Jussi Laako-2]

Jens M Andreasen wrote:
> With a 24bit signal of +/- 15 integer values, we would be drowned in
> noise. But since this is foremost a theoretical discussion, let us stick
> to those numbers.

What might be more useful would be to use two DACs in parallel followed
by high quality mixing electronics. In a way some highend audio hardware
does. Then non-error signals get amplified and error-signals might get
even attenuated (given the error signal is not the same for both channels).


BR,

        - Jussi

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[Jens M Andreasen-2]

On Sat, 2008-09-13 at 01:25 +0300, Jussi Laako wrote:

> Jens M Andreasen wrote:
> > With a 24bit signal of +/- 15 integer values, we would be drowned in
> > noise. But since this is foremost a theoretical discussion, let us stick
> > to those numbers.
>
> What might be more useful would be to use two DACs in parallel followed
> by high quality mixing electronics. In a way some highend audio hardware
> does. Then non-error signals get amplified and error-signals might get
> even attenuated (given the error signal is not the same for both channels).
>

Hey, slow down, you are getting ahead of me now ... :-D

One of my (two) power amplifiers is starting to show it's age. spewing
out little bursts of noise at random intervals. So they therefore both
are in in for a replacement.

The target is a set of JBL Roadversion from early 80:s. They have a
steady drop of 3dB/octave north and south of 1000Hz, easily corected by
setting bass and treble on ones simple keyboard  mixer at two o clock.
They have been beaten and kicked by amateurs when hired out as well as
endured all of my own mistakes, but are still perfectly functional ...
In other words, these are the WW2 Willis Jeep of loudspeakers, not going
anywhere.

Anyways ...

Finding a replacement for the 250 Watt ACM1 with no fan (silent!) that I
am currently using proved to be a challenge though. People are not
building that kind of gear these days anymore, at least not within my
monetary budget.

Findig a silent poweramplifier in the 75 - 125W range is a lot easier,
so ...

Why not feed the two 12 inch's in each cabinet from each of their own
poweramp, using each a souurce of their own? This would be the
equivalence of - no, actually just like - using two DAC's in paralel
that you proposed above.




>
> BR,
>
> - Jussi
>

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