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Frank Stearns Frank Stearns is offline
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Default DSP Experts: HPF and DC Offset

Are we assured that a typical digital high pass filter, say of the type found in the
EQ of Protools 8 and later, removes DC offset?

There's a lengthy back story here which for brevity sake I'll not get into, but it
seems that a digital HPF is not removing DC offset -- at least in PT.

I'm in the process of setting up some tests to prove it one way or the
other for sure, but am swamped at the moment.

Curious if any who are well-versed in DSP might say one of the following:

- that's true, a typical digital HPF will not remove DC offset and here's why....

- only a poorly designed digital HPF fails to remove DC offset.

- you're out of your mind, of course by definition ANY digital HPF removes DC
offsets.

- (something else entirely.)


Curious folks want to know.....

Thanks in advance,
Frank
Mobile Audio
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Arny Krueger[_4_] Arny Krueger[_4_] is offline
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Default DSP Experts: HPF and DC Offset


"Frank Stearns" wrote in message
tion...
Are we assured that a typical digital high pass filter, say of the type
found in the
EQ of Protools 8 and later, removes DC offset?

There's a lengthy back story here which for brevity sake I'll not get
into, but it
seems that a digital HPF is not removing DC offset -- at least in PT.


PT has historically had some of the dumbest technical errors that I've ever
heard of, but to the best of my knowlege, they are generally fixed.

I'm in the process of setting up some tests to prove it one way or the
other for sure, but am swamped at the moment.


Save time and forget about PT?

Curious if any who are well-versed in DSP might say one of the following:


- that's true, a typical digital HPF will not remove DC offset and here's
why....


False.

- only a poorly designed digital HPF fails to remove DC offset.


True.

- you're out of your mind, of course by definition ANY digital HPF removes
DC
offsets.


Mostly true, but never underestimate the human ability to screw up.

In the digital domain, there are at least two ways to remove DC offsets, and
a HPF is just one of them. The second method is to calculate the average
value of the wave and subtract it from each and every sample. Good DAW
software offers both alternatives.


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Scott Dorsey Scott Dorsey is offline
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Default DSP Experts: HPF and DC Offset

Frank Stearns wrote:
Are we assured that a typical digital high pass filter, say of the type found in the
EQ of Protools 8 and later, removes DC offset?


I don't know about Protools. But a real high pass filter will remove DC
offset. There's no obvious error I can think of in a dsp filter that would
cause it not to.

There's a lengthy back story here which for brevity sake I'll not get into, but it
seems that a digital HPF is not removing DC offset -- at least in PT.


What makes you think so? The waveform not be centered around zero all the
time, sometimes that's just the way the waveform is.

Take every sample, sum them all together... and you won't necessarily get
zero either.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
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Neil Gould Neil Gould is offline
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Default DSP Experts: HPF and DC Offset

Scott Dorsey wrote:
Frank Stearns wrote:
Are we assured that a typical digital high pass filter, say of the
type found in the EQ of Protools 8 and later, removes DC offset?


I don't know about Protools. But a real high pass filter will remove
DC offset. There's no obvious error I can think of in a dsp filter
that would cause it not to.

There's a lengthy back story here which for brevity sake I'll not
get into, but it seems that a digital HPF is not removing DC offset
-- at least in PT.


What makes you think so? The waveform not be centered around zero
all the time, sometimes that's just the way the waveform is.

Take every sample, sum them all together... and you won't necessarily
get zero either.

Well... this is an interesting thought exercise, if nothing else. I would
think DC offset would be represented either as a steady-state level above or
below zero when no AM is present or as an asymmetrical waveform. If so,
perhaps DAW software "removes" DC offset in the absence of a steady-state
level by re-centering the waveform based on an average of the waveform
values, and that creates an opportunity for error, especially for asymmetric
waveforms lacking DC offsets.

--
Neil



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Richard Webb[_3_] Richard Webb[_3_] is offline
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Default DSP Experts: HPF and DC Offset

On Thu 2011-Nov-10 09:20, Scott Dorsey writes:
I don't know about Protools. But a real high pass filter will
remove DC offset. There's no obvious error I can think of in a dsp
filter that would cause it not to.


Much as I thought as well.

There's a lengthy back story here which for brevity sake I'll not get into, but
it seems that a digital HPF is not removing DC offset -- at least in PT.


What makes you think so? The waveform not be centered around zero
all the time, sometimes that's just the way the waveform is.


Much as I always believed or was taught as well.

Take every sample, sum them all together... and you won't
necessarily get zero either.
--scott


tHIs is true, which begs the question if the problem is
something audible that will negatively impact listenability
or further downstream processing.

Regards,
Richard
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Peter Larsen[_3_] Peter Larsen[_3_] is offline
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Default DSP Experts: HPF and DC Offset

Neil Gould wrote:

Scott Dorsey wrote:


What makes you think so? The waveform not be centered around zero
all the time, sometimes that's just the way the waveform is.


Take every sample, sum them all together... and you won't necessarily
get zero either.


Well... this is an interesting thought exercise, if nothing else. I
would think DC offset would be represented either as a steady-state
level above or below zero when no AM is present or as an asymmetrical
waveform. If so, perhaps DAW software "removes" DC offset in the
absence of a steady-state level by re-centering the waveform based on
an average of the waveform values, and that creates an opportunity
for error, especially for asymmetric waveforms lacking DC offsets.


Re-centering would be an error. The mechanism is that the unlinearity of air
causes 2 harmonic distortion and second harmonic distortion causes
asymmetry. The asymmetry is in itself audible as an ""intonation or
chord-weight change"": take a recording of asymmetric vox humana or trumpet
and invert it and do something untraditional: listen! - or take a recording
of a speaker with varying asymmetry and listen for the difference in
perceived voice tonality with difference in asymmetry.

The asymmetry is "known and described in the literature" and it is an
established practice, at least for those that run AM receivers, to insert a
contraption in the audio chain that watches signal asymmetry and inverts
signal polarity to maintain positive asymmetry for optimum transmitter
modulation. The Elton John and Kiki Dee duet was bad news for those systems
.... one singer on each side on a fig 8, my recollection of Studio Sound is
that it was an AKG "the everchanging model".

What is less easily agreed on is that some, myself included, feel that
getting the overall polarity right is important for imaging and perspective.
Interestingly however it appears that just as some types of microphones are
more likely to deliver asymmetric audio than others also the audibility of
it varies with amount of second harmonic distortion added by the playback
transducers.

My observations - I do NOT want to call them "findings" - are that high
quality omni microphones tend to deliver recordings that are more symmetric
than some cardioids, applause does however always seem to have positive
asymmetry, also high pass filtering in post and/or the use of multiband
processing does tend to de-naturalize the waveform and can cause the
polarity to appear inverted.

Kind regards

Peter Larsen







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Neil Gould Neil Gould is offline
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Default DSP Experts: HPF and DC Offset

Peter Larsen wrote:
Neil Gould wrote:

Scott Dorsey wrote:


What makes you think so? The waveform not be centered around zero
all the time, sometimes that's just the way the waveform is.


Take every sample, sum them all together... and you won't
necessarily get zero either.


Well... this is an interesting thought exercise, if nothing else. I
would think DC offset would be represented either as a steady-state
level above or below zero when no AM is present or as an asymmetrical
waveform. If so, perhaps DAW software "removes" DC offset in the
absence of a steady-state level by re-centering the waveform based on
an average of the waveform values, and that creates an opportunity
for error, especially for asymmetric waveforms lacking DC offsets.


Re-centering would be an error. The mechanism is that the unlinearity
of air causes 2 harmonic distortion and second harmonic distortion
causes asymmetry. The asymmetry is in itself audible as an
""intonation or chord-weight change"": take a recording of asymmetric
vox humana or trumpet and invert it and do something untraditional:
listen! - or take a recording of a speaker with varying asymmetry and
listen for the difference in perceived voice tonality with difference
in asymmetry.

The asymmetry is "known and described in the literature" and it is an
established practice, at least for those that run AM receivers, to
insert a contraption in the audio chain that watches signal asymmetry
and inverts signal polarity to maintain positive asymmetry for
optimum transmitter modulation. The Elton John and Kiki Dee duet was
bad news for those systems ... one singer on each side on a fig 8, my
recollection of Studio Sound is that it was an AKG "the everchanging
model".

What is less easily agreed on is that some, myself included, feel that
getting the overall polarity right is important for imaging and
perspective. Interestingly however it appears that just as some types
of microphones are more likely to deliver asymmetric audio than
others also the audibility of it varies with amount of second
harmonic distortion added by the playback transducers.

My observations - I do NOT want to call them "findings" - are that
high quality omni microphones tend to deliver recordings that are
more symmetric than some cardioids, applause does however always seem
to have positive asymmetry, also high pass filtering in post and/or
the use of multiband processing does tend to de-naturalize the
waveform and can cause the polarity to appear inverted.

I completely agree with your observations, which is why I'm curious about
trying to remove DC offset in post-processing if there is no point of
reference such as a steady state level without AM signal. Since an xformer
balanced input won't pass DC, I wondered how likely it would be that DC
offset would be passed through the A/D. So, in either case, HPF is not much
of a solution.

--
best,

Neil



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Arny Krueger[_4_] Arny Krueger[_4_] is offline
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Default DSP Experts: HPF and DC Offset

"Neil Gould" wrote in message
...

I completely agree with your observations, which is why I'm curious about
trying to remove DC offset in post-processing if there is no point of
reference such as a steady state level without AM signal. Since an xformer
balanced input won't pass DC, I wondered how likely it would be that DC
offset would be passed through the A/D. So, in either case, HPF is not
much
of a solution.


If you are editing audio, and any of the file(s) involved have a DC offset,
there is a very good chance of introducing a click when you cut and paste.
Not just theory, but a fact that many of us who actually edit audio (and not
just pontificate about it) have unfortunately experienced.

You can minimize clicks and pops due to editing by only cutting on zero
crossings, but that is not a 100% solution. You can also minimize clicks and
pops by merging the file segments with what amounts to be a cross fade or
other smoothing technique, but again these are not 100% solutions. Cross
fades and other smoothing techniques essentially turn the clicks and pops
into thumps. The good news is that a thump that is composed of only very low
frequencies can become very hard to hear.

Having a file whose DC component approaches zero over shorter periods of
time is therefore a good thing, because it helps minimize the possibility of
introducing thumps, clicks and pops when you are editing it.

If you apply a HPF to a file, you are ensuring that the average or DC
component of the file approaches zero over a period of time that decreases
as you increase the cut-off frequency. If you zero out the files average
value, then the DC component of the file is only zero for the exact segment
of the file that you processed.


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Default DSP Experts: HPF and DC Offset

Arny Krueger wrote:
"Neil Gould" wrote in message
...

I completely agree with your observations, which is why I'm curious
about trying to remove DC offset in post-processing if there is no
point of reference such as a steady state level without AM signal.
Since an xformer balanced input won't pass DC, I wondered how likely
it would be that DC offset would be passed through the A/D. So, in
either case, HPF is not much
of a solution.


If you are editing audio, and any of the file(s) involved have a DC
offset, there is a very good chance of introducing a click when you
cut and paste. Not just theory, but a fact that many of us who
actually edit audio (and not just pontificate about it) have
unfortunately experienced.

Others of us who actually edit audio realize that edits that occur at
locations other than zero-crossing points will result in a click, and in
such cases a DC component is irrelevant.

You can minimize clicks and pops due to editing by only cutting on
zero crossings, but that is not a 100% solution.

Please explain this, as it is not something I've experienced in over 25
years of editing digital audio. There is no amplitude in the signal at
zero-crossing points, so how can it possibly result in a click?

--
best,

Neil


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Arny Krueger[_4_] Arny Krueger[_4_] is offline
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Default DSP Experts: HPF and DC Offset


"Neil Gould" wrote in message
...

Please explain this, as it is not something I've experienced in over 25
years of editing digital audio. There is no amplitude in the signal at
zero-crossing points, so how can it possibly result in a click?


Just because you're merging two clips at a zero crossing doesn't mean that
there isn't a difference in the DC levels of the two clips. It is the
difference in the DC levels of the two clips that causes the click, pop, or
thump.

If your editing software gently merges clips which is common, then it is
probably still creating some sort of thumping, but if the thumps are small
enough and at a low enough frequency, you don't hear them.





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Scott Dorsey Scott Dorsey is offline
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Default DSP Experts: HPF and DC Offset

Neil Gould wrote:
I completely agree with your observations, which is why I'm curious about
trying to remove DC offset in post-processing if there is no point of
reference such as a steady state level without AM signal. Since an xformer
balanced input won't pass DC, I wondered how likely it would be that DC
offset would be passed through the A/D. So, in either case, HPF is not much
of a solution.


Most DC offset problems are actually caused by an imbalance in the A/D
itself.

Since DC offset is in fact a shift in the quiescent level, it means that
cutting in and out of silent parts will cause an abrupt change in level,
ie. a pop.

A good fix for this is low-pass filtering, ie. offset removal.

Another issue with offset is that it eats a tiny bit of your dynamic range
up if your waveform is symmetric.... and in the age of trying to squeeze
every bit of level out, making sure your dc offset is eliminated is a way
to get a little extra level for free.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
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Scott Dorsey Scott Dorsey is offline
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Default DSP Experts: HPF and DC Offset

Neil Gould wrote:
Please explain this, as it is not something I've experienced in over 25
years of editing digital audio. There is no amplitude in the signal at
zero-crossing points, so how can it possibly result in a click?


What do you do when there's a silent passage? There's no zero crossing
point to cut at because the waveform never comes down to zero.

The reason why you haven't experienced this as a problem is probably because
you have been using good quality converters that make offset a non-issue.
--scott


--
"C'est un Nagra. C'est suisse, et tres, tres precis."
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Neil Gould Neil Gould is offline
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Default DSP Experts: HPF and DC Offset

Scott Dorsey wrote:
Neil Gould wrote:
Please explain this, as it is not something I've experienced in over
25 years of editing digital audio. There is no amplitude in the
signal at zero-crossing points, so how can it possibly result in a
click?


What do you do when there's a silent passage? There's no zero
crossing point to cut at because the waveform never comes down to
zero.

If by "silent passage" you mean only down to ambient noise levels, there is
still AM, and where there is AM there are zero crossing points unless there
is DC offset present in the signal.

The reason why you haven't experienced this as a problem is probably
because you have been using good quality converters that make offset
a non-issue.

Probably so.

--
Neil



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Arny Krueger wrote:
"Neil Gould" wrote in message
...

Please explain this, as it is not something I've experienced in over
25 years of editing digital audio. There is no amplitude in the
signal at zero-crossing points, so how can it possibly result in a
click?


Just because you're merging two clips at a zero crossing doesn't mean
that there isn't a difference in the DC levels of the two clips. It
is the difference in the DC levels of the two clips that causes the
click, pop, or thump.

Perhaps we're talking about different kinds of "zero", Arny. 8-)

DC level would have to be represented as something other than zero. Just to
be clear, I'm not arguing that DC offsets are never problematic... they can
be. But, so can trying to remove it in a push-button fashion.

--
Neil





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Trevor Trevor is offline
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Default DSP Experts: HPF and DC Offset


"Scott Dorsey" wrote in message
...
Another issue with offset is that it eats a tiny bit of your dynamic range
up if your waveform is symmetric.... and in the age of trying to squeeze
every bit of level out, making sure your dc offset is eliminated is a way
to get a little extra level for free.


Actually in the age of hypercompression with minimal dynamic range, you'd
need a pretty mighty DC offset to matter :-)

Trevor.


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Peter Larsen[_3_] Peter Larsen[_3_] is offline
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Default DSP Experts: HPF and DC Offset

Scott Dorsey wrote:

A good fix for this is low-pass filtering, ie. offset removal.


?

Another issue with offset is that it eats a tiny bit of your dynamic
range up if your waveform is symmetric.... and in the age of trying
to squeeze every bit of level out, making sure your dc offset is
eliminated is a way to get a little extra level for free.


!

--scott


Kind regards

Peter Larsen



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Scott Dorsey Scott Dorsey is offline
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Default DSP Experts: HPF and DC Offset

Don Pearce wrote:

The DC offset vanishes when you apply the low end crud filter. You do
filter the low end crud from every track?


Well, that is the whole subject of the thread. The original poster
seemed to still have DC offset after doing the low pass filter, which
shouldn't happen.
--Scott
--
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Default DSP Experts: HPF and DC Offset

Neil Gould wrote:
Scott Dorsey wrote:
Neil Gould wrote:
Please explain this, as it is not something I've experienced in over
25 years of editing digital audio. There is no amplitude in the
signal at zero-crossing points, so how can it possibly result in a
click?


What do you do when there's a silent passage? There's no zero
crossing point to cut at because the waveform never comes down to
zero.

If by "silent passage" you mean only down to ambient noise levels, there is
still AM, and where there is AM there are zero crossing points unless there
is DC offset present in the signal.


No, let's say there is just ambient noise, and it's a few dB above the
noise floor of the converter.... but the DC offset is far more than the
level of ambient noise, so the waveform is shifted far enough that it
never comes down to zero. This was sadly a common thing in the eighties.
--scott

--
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Default DSP Experts: HPF and DC Offset

Peter Larsen wrote:
Scott Dorsey wrote:

A good fix for this is low-pass filtering, ie. offset removal.


?


Sorry, I meant high-pass, or low-cut. I'm currently in a mix-a-thon and
not really sleeping as much as I should...
--scott

--
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Default DSP Experts: HPF and DC Offset


Please explain this, as it is not something I've experienced in over
25 years of editing digital audio. There is no amplitude in the
signal at zero-crossing points, so how can it possibly result in a
click?




In theory you have to match not only the amplitudes but also the slopes. If
you butt spliced together 2 sine waves of different frequencies at the zero
crossing you will still get a click because the slopes do not match at the
splice and there will be a discontinuity in the waveform. In fact in
theory you have to match all the derivatives of the two signals to
completely eliminate any click. In other words, for there to be no click,
in theory, the waveform and all it's derivatives have to be continuous.

So the only practical way to do this is to splice two wave forms together
during silence, i.e. not just when the waveform crosses zero, but when it
has been zero for a time and will be zero for a time.

Mark




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"MarkK" wrote in message
...
In theory you have to match not only the amplitudes but also the slopes.
If
you butt spliced together 2 sine waves of different frequencies at the
zero
crossing you will still get a click because the slopes do not match at the
splice and there will be a discontinuity in the waveform. In fact in
theory you have to match all the derivatives of the two signals to
completely eliminate any click. In other words, for there to be no click,
in theory, the waveform and all it's derivatives have to be continuous.

So the only practical way to do this is to splice two wave forms together
during silence, i.e. not just when the waveform crosses zero, but when it
has been zero for a time and will be zero for a time.


And in practice as long as the signal level is low at the zero crossing, or
the slope change is not too great, any click will be practicaly or totally
inaudible.

Trevor.


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Les Cargill[_4_] Les Cargill[_4_] is offline
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Default DSP Experts: HPF and DC Offset

Peter Larsen wrote:
Neil Gould wrote:

Scott Dorsey wrote:


What makes you think so? The waveform not be centered around zero
all the time, sometimes that's just the way the waveform is.


Take every sample, sum them all together... and you won't necessarily
get zero either.


Well... this is an interesting thought exercise, if nothing else. I
would think DC offset would be represented either as a steady-state
level above or below zero when no AM is present or as an asymmetrical
waveform. If so, perhaps DAW software "removes" DC offset in the
absence of a steady-state level by re-centering the waveform based on
an average of the waveform values, and that creates an opportunity
for error, especially for asymmetric waveforms lacking DC offsets.


Re-centering would be an error. The mechanism is that the unlinearity of air
causes 2 harmonic distortion and second harmonic distortion causes
asymmetry. The asymmetry is in itself audible as an ""intonation or
chord-weight change"": take a recording of asymmetric vox humana or trumpet
and invert it and do something untraditional: listen! - or take a recording
of a speaker with varying asymmetry and listen for the difference in
perceived voice tonality with difference in asymmetry.


One whale of a lot depends on what the recording is to be used for. If
it's a solo thing, and it will neither be spliced or edited - then
maybe.

If it is to be mixed or edited, then no. I think you pretty much have
to zero it out. I have not examined in detail what CoolEdit does to
eliminate DC offset, but if there's much at all, I take it out.
Surprisingly, it almost never comes up.

The asymmetry is "known and described in the literature" and it is an
established practice, at least for those that run AM receivers, to insert a
contraption in the audio chain that watches signal asymmetry and inverts
signal polarity to maintain positive asymmetry for optimum transmitter
modulation. The Elton John and Kiki Dee duet was bad news for those systems
... one singer on each side on a fig 8, my recollection of Studio Sound is
that it was an AKG "the everchanging model".

What is less easily agreed on is that some, myself included, feel that
getting the overall polarity right is important for imaging and perspective.
Interestingly however it appears that just as some types of microphones are
more likely to deliver asymmetric audio than others also the audibility of
it varies with amount of second harmonic distortion added by the playback
transducers.


I can believe this.


When i have played with asymmetric distortion as an effect, I always
remove DC offset. This makes rather unexpected and radical changes in
waveform, so I think the overall premise that DC offset is to the
good holds, when you're doing "documentary" work.

My observations - I do NOT want to call them "findings" - are that high
quality omni microphones tend to deliver recordings that are more symmetric
than some cardioids, applause does however always seem to have positive
asymmetry, also high pass filtering in post and/or the use of multiband
processing does tend to de-naturalize the waveform and can cause the
polarity to appear inverted.


Bizarre.


Kind regards

Peter Larsen

snip
--
Les Cargill

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On Fri 2011-Nov-11 15:56, Trevor writes:
Another issue with offset is that it eats a tiny bit of your dynamic range
up if your waveform is symmetric.... and in the age of trying to squeeze
every bit of level out, making sure your dc offset is eliminated is a way
to get a little extra level for free.


Actually in the age of hypercompression with minimal dynamic range,
you'd need a pretty mighty DC offset to matter :-)


Um yeah, that's if it's most popular forms of music heard on mainstream radio, but if you're recording classical or jazz
you're actually trying to preserve some dynamics, hence you
need some dynamic range.
OTherwise, just slam that thing as close to the pin as you
can get it. Remember the principle of loud. "Make it loud
dammit!"

Regards,
Richard
--
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Default DSP Experts: HPF and DC Offset

Scott Dorsey wrote:
Neil Gould wrote:
Scott Dorsey wrote:
Neil Gould wrote:
Please explain this, as it is not something I've experienced in
over 25 years of editing digital audio. There is no amplitude in
the signal at zero-crossing points, so how can it possibly result
in a click?

What do you do when there's a silent passage? There's no zero
crossing point to cut at because the waveform never comes down to
zero.

If by "silent passage" you mean only down to ambient noise levels,
there is still AM, and where there is AM there are zero crossing
points unless there is DC offset present in the signal.


No, let's say there is just ambient noise, and it's a few dB above the
noise floor of the converter.... but the DC offset is far more than
the level of ambient noise, so the waveform is shifted far enough
that it never comes down to zero. This was sadly a common thing in
the eighties.

We said exactly the same thing, Scott! ;-)

--
Neil


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Neil Gould Neil Gould is offline
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MarkK wrote:
Please explain this, as it is not something I've experienced in
over 25 years of editing digital audio. There is no amplitude in
the signal at zero-crossing points, so how can it possibly result
in a click?



In theory you have to match not only the amplitudes but also the
slopes. If you butt spliced together 2 sine waves of different
frequencies at the zero crossing you will still get a click because
the slopes do not match at the splice and there will be a
discontinuity in the waveform. In fact in theory you have to match
all the derivatives of the two signals to completely eliminate any
click. In other words, for there to be no click, in theory, the
waveform and all it's derivatives have to be continuous.

So the only practical way to do this is to splice two wave forms
together during silence, i.e. not just when the waveform crosses
zero, but when it has been zero for a time and will be zero for a
time.

What you seem to be describing is a problem with the D/A or downstream
handling of the signal that results in overshoot. What I thought we were
talking about is an actual change in the level of the waveform at the edit
point due to DC offset.

--
Neil


Mark



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Les Cargill wrote:

When i have played with asymmetric distortion as an effect, I always
remove DC offset.


DC offset and asymmetry are different concepts, you will have asymmetry even
if the "silence" line is on the midline and it will remain even if you
remove the DC offset as Cooledit does it, ie. a "few hz" highpass.

This makes rather unexpected and radical changes in
waveform, so I think the overall premise that DC offset is to the
good holds, when you're doing "documentary" work.


Whatever you do it is - in my opinion - good practice to remove dc offset,
CoolEdit can do it when accepting input, but doesn't if the input is a file,
then remembering setting the toggle once and for all doesn't do.

I had stopped highpass filtering in post, because some clarity is lost for
every single modification of the audio - I have changed workflow and aim for
doing all the post, even for a stereo recording, in the Audition 3
multitrack worksspace, but after have a "thermic air rise" surprise due to
event lights recently it is back in the workflow.

Always remember to wonder whenever the recording meters are more stationary
than suggested by the audible signal envelope! - what made it "an issue"
was:

1 part microphones without internal highpass,
1 part microphones without windscreen and some 12 feet above the floor,
1 part "cardioid compensating bass boost" and finally
1 part a friends transmission line-loudspeakers that - at a guess -
enhanced the VLF by adding second harmonic distortion and then hitting a
room resonance in his listening room and air noise, it is only very
marginally audible on my big un's that just play it back as it is, a very
deep rumble.

High pass filtering is also in most contexts a good practice because it
keeps small loudspeaker membrane excursion low, and thus reduces actual
playback distortion for those that do not have full range loudspeakers.

Listening on extra playback systems is also a good practice.

Kind regards

Peter Larsen



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"Neil Gould" wrote in message
...
Arny Krueger wrote:
"Neil Gould" wrote in message
...

Please explain this, as it is not something I've experienced in over
25 years of editing digital audio. There is no amplitude in the
signal at zero-crossing points, so how can it possibly result in a
click?


Just because you're merging two clips at a zero crossing doesn't mean
that there isn't a difference in the DC levels of the two clips. It
is the difference in the DC levels of the two clips that causes the
click, pop, or thump.


Perhaps we're talking about different kinds of "zero", Arny. 8-)


There are two different concepts at work here - the instantaneous value of
the wave at its end points, and the average value of the audio clip.
Consider an audio clip that is zero at both ends, but entirely positive for
its duration. Its average value is clearly positive. If you splice that clip
between two other audio clips whose average value is zero, then there is a
DC pulse whose duration is that of the middle clip.

Audio signals are generally asymetrical, which means that only special cases
of them actually have an average value of zero. The easiest way to contrive
these special cases is to high pass the signal with a filter that has the
highest corner frequency possible.

When put high pass filters on microphones to control low frequency noise,
proximity effect, and popping, we accomplish the reasonable goal of having
signals that are easy to splice without adding clicks, pops, or thumps.


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"Scott Dorsey" wrote in message
...
Neil Gould wrote:
Please explain this, as it is not something I've experienced in over 25
years of editing digital audio. There is no amplitude in the signal at
zero-crossing points, so how can it possibly result in a click?


What do you do when there's a silent passage? There's no zero crossing
point to cut at because the waveform never comes down to zero.

The reason why you haven't experienced this as a problem is probably
because
you have been using good quality converters that make offset a non-issue.


Agreed.

Those of us who were forced to struggle with low quality converters in the
past, such as the too-cheap on-board computer audio interfaces from the late
1990s and early 2000's, obtained painful practical experience with
converters that had built-in DC offsets. There were converters with so much
DC offset that it caused a meaningful loss of dynamic range with symmetrical
signals. Its been a while since I've seen this with even commodity products.

If you go all the way back to the days when digital audio was new, it was
not unusual for the best available equipment to use converters with DC
offset adjustments. If they drifted or were badly adjusted, voila: DC
offsets in the recording.




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Neil Gould Neil Gould is offline
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Default DSP Experts: HPF and DC Offset

Arny Krueger wrote:
"Neil Gould" wrote in message
...
Arny Krueger wrote:
"Neil Gould" wrote in message
...

Please explain this, as it is not something I've experienced in
over 25 years of editing digital audio. There is no amplitude in
the signal at zero-crossing points, so how can it possibly result
in a click?

Just because you're merging two clips at a zero crossing doesn't
mean that there isn't a difference in the DC levels of the two
clips. It is the difference in the DC levels of the two clips that
causes the click, pop, or thump.


Perhaps we're talking about different kinds of "zero", Arny. 8-)


There are two different concepts at work here - the instantaneous
value of the wave at its end points, and the average value of the
audio clip. Consider an audio clip that is zero at both ends, but
entirely positive for its duration. Its average value is clearly
positive. If you splice that clip between two other audio clips whose
average value is zero, then there is a DC pulse whose duration is
that of the middle clip.

It sounds like you're considering the point at which a waveform reaches it's
average value to be "zero", but I would not call it so, because waveform
asymmetry is "normal" and in such cases the the average value can be
something other than 0.0000...dB. Merging a waveform with DC offset with one
without DC offset at 0.0000...dB should not result in an audible artifact
unless the downstream reproduction can't handle the transition without
overshoot, but then the problem isn't the DC offset.

Of course, if you have a waveform that is so offset that it never reaches
0.000...dB, IMO, HPF won't help. If I were faced with such a problem, I'd
probably write an applet to adjust the overall level of the file without
impacting the symmetry of its waveforms. If there's a good reference for the
offset, it's not that big of a task. Fortunately, I've never found it
necessary to do something like that.

--
Neil


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Scott Dorsey Scott Dorsey is offline
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Neil Gould wrote:

Of course, if you have a waveform that is so offset that it never reaches
0.000...dB, IMO, HPF won't help.


Sure it will! It removes all components below, say 1 Hz. That includes
zero hertz components. It fixes the problem right up.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
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Frank Stearns Frank Stearns is offline
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(Scott Dorsey) writes:

Neil Gould wrote:

Of course, if you have a waveform that is so offset that it never reaches
0.000...dB, IMO, HPF won't help.


Sure it will! It removes all components below, say 1 Hz. That includes
zero hertz components. It fixes the problem right up.


Don't you need some sort of sample value delta, even at 1 or 2 Hz, to have a HPF do
something? Doesn't a constant bias, such as DC, merely shift all sample values? And
if nothing's "moving", even at VLF, there's nothing for a digital HPF to find?

Arithmetically speaking, how does a digital HPF recognize "non-moving" DC offset and
not, say, do something inappropriate to a naturally very asymmetrical waveform, such
as a brass instrument? Often brass instrument waveforms look like they have DC in
them, when in fact that's what brass beasties produce in real life.

Why does the DC offset removal tool in Sound Forge, for example, require at minimum
5 seconds worth of samples (or a scan of the entire sample set of the file) to do DC
removal?

I'm not challenging, just curious as all get out, based on some recent experiences
with DC offset removal (and thus the original post).

As noted before, when I have time to make the stories really succinct, I'll post
them for comment. I have made some interesting observations...

Thanks in advance for any interim enlightenment!

Frank
Mobile Audio
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Frank Stearns wrote:

Why does the DC offset removal tool in Sound Forge, for example,
require at minimum 5 seconds worth of samples (or a scan of the
entire sample set of the file) to do DC removal?


What it does is then not - as I read you - DC removal. The requirement
(almost) only makes sense if the may the area above the midline equal to the
area below. They could also simply make max positive and max negative equal,
but that would be oversimplifying.

Also even the asumption of equal area above and below the midline is really
easy to break: if you make it with applause then the result is adding a DC
offset since that wave-shape is inherently asymmetric all of the time
because of the non-ideal properties of real air under compression.

IMO the optimal tool is a non-minimum phase highpass, ie. a phase linear
highpass and I would love to have it as a virtual button with selectable
frequency on each virtual channel in a multitrack view. And please not ever
so slightly grainy sounding .... O;-)

Mostly I advocate minimum phase eq, but crud removal highpass appears to me
to be a reasonable exception.

Frank
Mobile Audio


Kind regards

Peter Larsen



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"Neil Gould" wrote in message
...
Arny Krueger wrote:
"Neil Gould" wrote in message
...
Arny Krueger wrote:
"Neil Gould" wrote in message
...

Please explain this, as it is not something I've experienced in
over 25 years of editing digital audio. There is no amplitude in
the signal at zero-crossing points, so how can it possibly result
in a click?

Just because you're merging two clips at a zero crossing doesn't
mean that there isn't a difference in the DC levels of the two
clips. It is the difference in the DC levels of the two clips that
causes the click, pop, or thump.


Perhaps we're talking about different kinds of "zero", Arny. 8-)


There are two different concepts at work here - the instantaneous
value of the wave at its end points, and the average value of the
audio clip. Consider an audio clip that is zero at both ends, but
entirely positive for its duration. Its average value is clearly
positive. If you splice that clip between two other audio clips whose
average value is zero, then there is a DC pulse whose duration is
that of the middle clip.

It sounds like you're considering the point at which a waveform reaches
it's
average value to be "zero", but I would not call it so, because waveform
asymmetry is "normal" and in such cases the the average value can be
something other than 0.0000...dB.


No, I am speaking of the average value of the waveform. If you will, the sum
of the area under the curves that form the waveform, divided by its length.
The zeroeth term of the Fourier analysis of the wave. The DC component of
the wave. It has many names, but they are all descriptions of the same
thing.

Until you grasp these basic concepts, there really is no more that can be
said.




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Default DSP Experts: HPF and DC Offset

On Nov 12, 2:46*am, (Don Pearce) wrote:
On 11 Nov 2011 20:44:11 -0500, (Scott Dorsey) wrote:

Don Pearce wrote:


The DC offset vanishes when you apply the low end crud filter. You do
filter the low end crud from every track?


Well, that is the whole subject of the thread. *The original poster
seemed to still have DC offset after doing the low pass filter, which
shouldn't happen.
--Scott


I presume you mean high pass, and yes you are dead right. It sounds
like his high pass isn't entirely so. I have seen this in the past
where the number of bins treated in the FFT is wrong. It tends to
leave a small residue of DC rather than all of it though.

d


Not to mention that the sum of the filter coefficients will be zero in
a properly designed high-pass filter which of course removes DC and is
a necessary condition for any filter that has infinite attenuation at
zero frequency.
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Default DSP Experts: HPF and DC Offset


I presume you mean high pass, and yes you are dead right. It sounds
like his high pass isn't entirely so. I have seen this in the past
where the number of bins treated in the FFT is wrong. It tends to
leave a small residue of DC rather than all of it though.

d


lets say the system full scale is + 255 and - 255 for sake of discussion...

if you have a waveform like applause that is highly asymmetric,....
lets say it is +255 for 4 samples and 0 for 1000 samples, that waveform,
even though it is 0 for many samples DOES have a DC component.

If you remove the DC component then the result will be +253 for 4 samples
and -3 for 1000 samples. (those may not be the exact numbers, I'm just
trying to demonstrate a point)

Now if you looked at that waveform and saw the long string of -3's you
might incorrectly conclude that there was a DC offset.

The point I'm trying to get across is that the measure of the DC value
depends strongly on the time duration over which you make the measurement.

For example if you put the waveform through a true high pass filter with a
1 Hz cutoff, there may be an apparent short term DC value over the duration
of say 0.1 second but not over the duration of several seconds.

Mark





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Scott Dorsey Scott Dorsey is offline
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Default DSP Experts: HPF and DC Offset

Frank Stearns wrote:

Don't you need some sort of sample value delta, even at 1 or 2 Hz, to have a HPF do
something? Doesn't a constant bias, such as DC, merely shift all sample values? And
if nothing's "moving", even at VLF, there's nothing for a digital HPF to find?


Think about the digital IIR filter as just being like an analogue high pass,
like a blocking capacitor and a shunt resistor. If you apply a DC offset to
the input with no signal, the filter will eventually reduce it down to zero
in some time that depends on the time constant of the filter. So you may have
a couple seconds of time before the offset is actually removed.

Arithmetically speaking, how does a digital HPF recognize "non-moving" DC offset and
not, say, do something inappropriate to a naturally very asymmetrical waveform, such
as a brass instrument? Often brass instrument waveforms look like they have DC in
them, when in fact that's what brass beasties produce in real life.


The same way that an analogue HPF filter does... and that indeed can cause the
DC point to be shifted while the insrument is playing, but it doesn't
necessarily make anything more symmetric.

Why does the DC offset removal tool in Sound Forge, for example, require at minimum
5 seconds worth of samples (or a scan of the entire sample set of the file) to do DC
removal?


Probably because the filter has some settling time. But also, the DC offset
removal tool may be something other than a simple high pass as well. That
doesn't mean that a simple high pass isn't a reasonable solution though.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."
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