computing compressor automatic makeup gain (?)

[Ross Bencina-3]

Hi Everyone

This question was asked here a few years back but without a clear answer.

I'm wondering whether there's a standard (or preferred) way to calculate the
amount of automatic makeup gain required in a compressor, given a particular
threshold/ratio/compression curve. I realise that the concept of automatic
makeup is problematic, but enough compressors offer the feature that there
must at least be known approaches to implementing it.

I assume that its done by calculating the amount of gain required to make an
input of  XdB have the same output level. I'm not sure what X should be
though, 0db?

E.g. With a hard-knee compressor, threshold at -10dB, ratio of 2:1,
compressing a 0dB input would give a -5dB output, and therefore the
automatic makeup gain should be +5dB. Does that sound correct/incorrect to
anyone?

I'd be interested to hear about alternative approaches or whether anyone
knows the actual reference levels for auto makeup gain in hardware
compressors.

Thanks!

Ross.

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[Martin Eisenberg]

Ross Bencina wrote:

> I'm wondering whether there's a standard (or preferred) way to
> calculate the amount of automatic makeup gain required in a
> compressor, given a particular threshold/ratio/compression
> curve.

I'm not that good with compressors but FWIW, I always thought
that makeup gain basically converts downward compression into
upward compression, which is what you described with a 0 dB
fixpoint.


Martin

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[Tom Duffy-2]

The way it is implemented in TASCAM products:

Calculate the amount of gain reduction would be applied to a 0dBFS
signal.
Halve that.
Add it.

e.g. At Threshold = -20dB, with 2:1 compression, a 0dB signal would
be reduced by 10dB. Therefore the auto makeup gain applied would be
5dB.
This method was found empirically to maintain perceived loudness
over a range of thresholds and ratios.
In your example, making up the entire gain to bring 0dB back to
where it was results in an overly "loud" signal.

There are more complicated things you could do by including the
attack and release times into the equation. A faster attack means
the loudness will be reduced more for the same threshold and
ratio settings, therefore more automatic gain is OK.
at Attack = 0ms, maybe a 100% makeup would sound OK.

Tom

-------- Original Message --------
Subject: [music-dsp] computing compressor automatic makeup gain (?)
From: Ross Bencina <rossb-lists@...>
To: A discussion list for music-related DSP <music-dsp@...>
Date: 8/23/2009 5:35 AM

Hi Everyone

This question was asked here a few years back but without a clear answer.

I'm wondering whether there's a standard (or preferred) way to calculate
the
amount of automatic makeup gain required in a compressor, given a
particular
threshold/ratio/compression curve. I realise that the concept of automatic
makeup is problematic, but enough compressors offer the feature that there
must at least be known approaches to implementing it.

I assume that its done by calculating the amount of gain required to
make an
input of  XdB have the same output level. I'm not sure what X should be
though, 0db?

E.g. With a hard-knee compressor, threshold at -10dB, ratio of 2:1,
compressing a 0dB input would give a -5dB output, and therefore the
automatic makeup gain should be +5dB. Does that sound correct/incorrect to
anyone?

I'd be interested to hear about alternative approaches or whether anyone
knows the actual reference levels for auto makeup gain in hardware
compressors.

Thanks!

Ross.

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[Hannes Löschke]

Isn't the general aim of compression to get a louder signal?

The naive approach would be to adjust the gain in a way that 0dB input comes out at 0dB output by apllying the maximum gain reduction. In that case all the attacks would be pushed over 0dB into your (hopefully existing) headroom an possibly cause clipping.

Applying half the maximum gain reduction sounds like an attemt to prevent attacks from clipping for a reasonable range of attack times and signals. I don't know any TASCAM compresors, but maintaining percveived loudness doesn't sound like a sensible goal for automatic gain adjustments in compressors to me.

Taking attack time into account depends on the signal you use. Send an impulse through the compressor and any setting apart from 0ms schould essentially pass the impulse unchanged. Any automatic gain makeup would cause clipping in for an impulse.

 
Hannes



----- Ursprüngliche Mail ----
Von: Tom Duffy <tduffy@...>
An: A discussion list for music-related DSP <music-dsp@...>
Gesendet: Montag, den 31. August 2009, 18:27:02 Uhr
Betreff: Re: [music-dsp] computing compressor automatic makeup gain (?)

The way it is implemented in TASCAM products:

Calculate the amount of gain reduction would be applied to a 0dBFS
signal.
Halve that.
Add it.

e.g. At Threshold = -20dB, with 2:1 compression, a 0dB signal would
be reduced by 10dB. Therefore the auto makeup gain applied would be
5dB.
This method was found empirically to maintain perceived loudness
over a range of thresholds and ratios.
In your example, making up the entire gain to bring 0dB back to
where it was results in an overly "loud" signal.

There are more complicated things you could do by including the
attack and release times into the equation. A faster attack means
the loudness will be reduced more for the same threshold and
ratio settings, therefore more automatic gain is OK.
at Attack = 0ms, maybe a 100% makeup would sound OK.

Tom

-------- Original Message --------
Subject: [music-dsp] computing compressor automatic makeup gain (?)
From: Ross Bencina <rossb-lists@...>
To: A discussion list for music-related DSP <music-dsp@...>
Date: 8/23/2009 5:35 AM

Hi Everyone

This question was asked here a few years back but without a clear answer.

I'm wondering whether there's a standard (or preferred) way to calculate
the
amount of automatic makeup gain required in a compressor, given a
particular
threshold/ratio/compression curve. I realise that the concept of automatic
makeup is problematic, but enough compressors offer the feature that there
must at least be known approaches to implementing it.

I assume that its done by calculating the amount of gain required to
make an
input of  XdB have the same output level. I'm not sure what X should be
though, 0db?

E.g. With a hard-knee compressor, threshold at -10dB, ratio of 2:1,
compressing a 0dB input would give a -5dB output, and therefore the
automatic makeup gain should be +5dB. Does that sound correct/incorrect to
anyone?

I'd be interested to hear about alternative approaches or whether anyone
knows the actual reference levels for auto makeup gain in hardware
compressors.

Thanks!

Ross.

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[Tom Duffy-2]

(sorry for top-post style, that's all I'm set up to do)

A compressor's role is to change (generally reduce) the dynamic range
of a signal.
Without any gain, setting the threshold low (e.g. -40dB) and setting
the compression ratio high, e.g. 1:100, the output signal will be mostly
40dB lower than the input signal.  That's not a louder signal.
Adding 40dB of gain now gives you peaks over the prior signal signal
level, depending on the attack speed.

"Mastering Loudness" is achieved by making the peaks meet 0dBFS,
after you've reduced the dynamic range; it's a simple gain stage
after a compressor.
"Loudness" in the last 10 years or so has been also achieved by
letting the peaks go above 0dBFS and simply chopping them off, which
is a high distortion (therefore non-musical) transformation.

There is no sense in sending impulses through a compressor, it is
by definition a non-linear system, that would tell you nothing about
it.

Compression as a tool for managing high dynamic range signals during
recording (e.g. TASCAM products) is different from compression as a
component in "making things louder".

In general, you don't want to have to adjust the gain every time you
change the compressor settings, so keeping the perceived level about
the same eliminates some of the "louder sounds better, quieter
sounds worse" psychoacoustic effect, and lets you concentrate on the
dynamics of the signal you are working on.

Tom.

-------- Original Message --------
Subject: Re: [music-dsp] computing compressor automatic makeup gain (?)
From: Hannes Löschke <hannes_loeschke@...>
To: A discussion list for music-related DSP <music-dsp@...>
Date: 9/1/2009 7:18 AM

Isn't the general aim of compression to get a louder signal?

The naive approach would be to adjust the gain in a way that 0dB input
comes out at 0dB output by apllying the maximum gain reduction. In that
case all the attacks would be pushed over 0dB into your (hopefully
existing) headroom an possibly cause clipping.

Applying half the maximum gain reduction sounds like an attemt to
prevent attacks from clipping for a reasonable range of attack times and
signals. I don't know any TASCAM compresors, but maintaining percveived
loudness doesn't sound like a sensible goal for automatic gain
adjustments in compressors to me.

Taking attack time into account depends on the signal you use. Send an
impulse through the compressor and any setting apart from 0ms schould
essentially pass the impulse unchanged. Any automatic gain makeup would
cause clipping in for an impulse.

 
Hannes



----- Ursprüngliche Mail ----
Von: Tom Duffy <tduffy@...>
An: A discussion list for music-related DSP <music-dsp@...>
Gesendet: Montag, den 31. August 2009, 18:27:02 Uhr
Betreff: Re: [music-dsp] computing compressor automatic makeup gain (?)

The way it is implemented in TASCAM products:

Calculate the amount of gain reduction would be applied to a 0dBFS
signal.
Halve that.
Add it.

e.g. At Threshold = -20dB, with 2:1 compression, a 0dB signal would
be reduced by 10dB. Therefore the auto makeup gain applied would be
5dB.
This method was found empirically to maintain perceived loudness
over a range of thresholds and ratios.
In your example, making up the entire gain to bring 0dB back to
where it was results in an overly "loud" signal.

There are more complicated things you could do by including the
attack and release times into the equation. A faster attack means
the loudness will be reduced more for the same threshold and
ratio settings, therefore more automatic gain is OK.
at Attack = 0ms, maybe a 100% makeup would sound OK.

Tom

-------- Original Message --------
Subject: [music-dsp] computing compressor automatic makeup gain (?)
From: Ross Bencina <rossb-lists@...>
To: A discussion list for music-related DSP <music-dsp@...>
Date: 8/23/2009 5:35 AM

Hi Everyone

This question was asked here a few years back but without a clear answer.

I'm wondering whether there's a standard (or preferred) way to calculate
the
amount of automatic makeup gain required in a compressor, given a
particular
threshold/ratio/compression curve. I realise that the concept of automatic
makeup is problematic, but enough compressors offer the feature that there
must at least be known approaches to implementing it.

I assume that its done by calculating the amount of gain required to
make an
input of  XdB have the same output level. I'm not sure what X should be
though, 0db?

E.g. With a hard-knee compressor, threshold at -10dB, ratio of 2:1,
compressing a 0dB input would give a -5dB output, and therefore the
automatic makeup gain should be +5dB. Does that sound correct/incorrect to
anyone?

I'd be interested to hear about alternative approaches or whether anyone
knows the actual reference levels for auto makeup gain in hardware
compressors.

Thanks!

Ross.

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[Kevin Dixon-2]

Thanks for the insight Tom, as a bassist and engineer, I would agree
with your (and TASCAM's) assessment of how this should work

-Kevin

Tom Duffy wrote:

> (sorry for top-post style, that's all I'm set up to do)
>
> A compressor's role is to change (generally reduce) the dynamic range
> of a signal.
> Without any gain, setting the threshold low (e.g. -40dB) and setting
> the compression ratio high, e.g. 1:100, the output signal will be mostly
> 40dB lower than the input signal.  That's not a louder signal.
> Adding 40dB of gain now gives you peaks over the prior signal signal
> level, depending on the attack speed.
>
> "Mastering Loudness" is achieved by making the peaks meet 0dBFS,
> after you've reduced the dynamic range; it's a simple gain stage
> after a compressor.
> "Loudness" in the last 10 years or so has been also achieved by
> letting the peaks go above 0dBFS and simply chopping them off, which
> is a high distortion (therefore non-musical) transformation.
>
> There is no sense in sending impulses through a compressor, it is
> by definition a non-linear system, that would tell you nothing about
> it.
>
> Compression as a tool for managing high dynamic range signals during
> recording (e.g. TASCAM products) is different from compression as a
> component in "making things louder".
>
> In general, you don't want to have to adjust the gain every time you
> change the compressor settings, so keeping the perceived level about
> the same eliminates some of the "louder sounds better, quieter
> sounds worse" psychoacoustic effect, and lets you concentrate on the
> dynamics of the signal you are working on.
>
> Tom.
>
> -------- Original Message --------
> Subject: Re: [music-dsp] computing compressor automatic makeup gain (?)
> From: Hannes Löschke <hannes_loeschke@...>
> To: A discussion list for music-related DSP <music-dsp@...>
> Date: 9/1/2009 7:18 AM
>
> Isn't the general aim of compression to get a louder signal?
>
> The naive approach would be to adjust the gain in a way that 0dB input
> comes out at 0dB output by apllying the maximum gain reduction. In that
> case all the attacks would be pushed over 0dB into your (hopefully
> existing) headroom an possibly cause clipping.
>
> Applying half the maximum gain reduction sounds like an attemt to
> prevent attacks from clipping for a reasonable range of attack times and
> signals. I don't know any TASCAM compresors, but maintaining percveived
> loudness doesn't sound like a sensible goal for automatic gain
> adjustments in compressors to me.
>
> Taking attack time into account depends on the signal you use. Send an
> impulse through the compressor and any setting apart from 0ms schould
> essentially pass the impulse unchanged. Any automatic gain makeup would
> cause clipping in for an impulse.
>
>  
> Hannes
>
>
>
> ----- Ursprüngliche Mail ----
> Von: Tom Duffy <tduffy@...>
> An: A discussion list for music-related DSP <music-dsp@...>
> Gesendet: Montag, den 31. August 2009, 18:27:02 Uhr
> Betreff: Re: [music-dsp] computing compressor automatic makeup gain (?)
>
> The way it is implemented in TASCAM products:
>
> Calculate the amount of gain reduction would be applied to a 0dBFS
> signal.
> Halve that.
> Add it.
>
> e.g. At Threshold = -20dB, with 2:1 compression, a 0dB signal would
> be reduced by 10dB. Therefore the auto makeup gain applied would be
> 5dB.
> This method was found empirically to maintain perceived loudness
> over a range of thresholds and ratios.
> In your example, making up the entire gain to bring 0dB back to
> where it was results in an overly "loud" signal.
>
> There are more complicated things you could do by including the
> attack and release times into the equation. A faster attack means
> the loudness will be reduced more for the same threshold and
> ratio settings, therefore more automatic gain is OK.
> at Attack = 0ms, maybe a 100% makeup would sound OK.
>
> Tom
>
> -------- Original Message --------
> Subject: [music-dsp] computing compressor automatic makeup gain (?)
> From: Ross Bencina <rossb-lists@...>
> To: A discussion list for music-related DSP <music-dsp@...>
> Date: 8/23/2009 5:35 AM
>
> Hi Everyone
>
> This question was asked here a few years back but without a clear answer.
>
> I'm wondering whether there's a standard (or preferred) way to calculate
> the
> amount of automatic makeup gain required in a compressor, given a
> particular
> threshold/ratio/compression curve. I realise that the concept of automatic
> makeup is problematic, but enough compressors offer the feature that there
> must at least be known approaches to implementing it.
>
> I assume that its done by calculating the amount of gain required to
> make an
> input of  XdB have the same output level. I'm not sure what X should be
> though, 0db?
>
> E.g. With a hard-knee compressor, threshold at -10dB, ratio of 2:1,
> compressing a 0dB input would give a -5dB output, and therefore the
> automatic makeup gain should be +5dB. Does that sound correct/incorrect to
> anyone?
>
> I'd be interested to hear about alternative approaches or whether anyone
> knows the actual reference levels for auto makeup gain in hardware
> compressors.
>
> Thanks!
>
> Ross.
>
> --
> dupswapdrop -- the music-dsp mailing list and website:
> subscription info, FAQ, source code archive, list archive, book reviews,
> dsp links
> http://music.columbia.edu/cmc/music-dsp
> http://music.columbia.edu/mailman/listinfo/music-dsp
>
>
> NOTICE: This electronic mail message and its contents, including any
> attachments hereto (collectively, "this e-mail"), is hereby designated
> as "confidential and proprietary." This e-mail may be viewed and used
> only by the person to whom it has been sent and his/her employer solely
> for the express purpose for which it has been disclosed and only in
> accordance with any confidentiality or non-disclosure (or similar)
> agreement between TEAC Corporation or its affiliates and said employer,
> and may not be disclosed to any other person or entity.
>  
>
>
>
>
>  
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>
>
>
>  
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>  
>
>  
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>  
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[robert bristow-johnson]

On Sep 3, 2009, at 1:25 AM, Kevin Dixon wrote:

> Thanks for the insight Tom, as a bassist and engineer, I would agree
> with your (and TASCAM's) assessment of how this should work
>
>
> Tom Duffy wrote:
>>
>> A compressor's role is to change (generally reduce) the dynamic range
>> of a signal.
>> Without any gain, setting the threshold low (e.g. -40dB) and setting
>> the compression ratio high, e.g. 1:100, the output signal will be  
>> mostly
>> 40dB lower than the input signal.  That's not a louder signal.
>> Adding 40dB of gain now gives you peaks over the prior signal signal
>> level, depending on the attack speed.
>>
>> "Mastering Loudness" is achieved by making the peaks meet 0dBFS,
>> after you've reduced the dynamic range; it's a simple gain stage
>> after a compressor.
>> "Loudness" in the last 10 years or so has been also achieved by
>> letting the peaks go above 0dBFS and simply chopping them off, which
>> is a high distortion (therefore non-musical) transformation.
>>
>> There is no sense in sending impulses through a compressor, it is
>> by definition a non-linear system, that would tell you nothing about
>> it.
>>

this last statement, i cannot agree with.  i look at compressors and  
limiters as being the same species of animal.

if your compressor or limiter is acting on peak amplitude rather than  
r.m.s., sending in impulses that exceed the threshold or knee will  
tell you much more than nothing about what the compressor is doing.

being a non-linear system means that the impulse response cannot tell  
you how the device will respond to all other inputs.  for a linear  
time-invariant system, the impulse response is sufficient to tell you  
how the system will behave for any general input.


>> Compression as a tool for managing high dynamic range signals during
>> recording (e.g. TASCAM products) is different from compression as a
>> component in "making things louder".

how are they different *qualitatively*?  different attack and release  
times, different mapping curve, different overall gain.

a multiband compressor is qualitatively different, but i might  
imagine such used in either situation.


>> In general, you don't want to have to adjust the gain every time you
>> change the compressor settings, so keeping the perceived level about
>> the same eliminates some of the "louder sounds better, quieter
>> sounds worse" psychoacoustic effect, and lets you concentrate on the
>> dynamics of the signal you are working on.

it seems to me that the issue is whether you want to tack down the  
knee or corner to a certain level, or if you want to tack down 0 dB  
FS, or somewhere in between.  but those are settings; i don't see the  
qualitative difference.


--

r b-j                  rbj@...

"Imagination is more important than knowledge."




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[Didier Dambrin]

IMHO the only difference between a compressor and a limiter is the
lookahead/latency. A limiter absolutely requires one to do its job properly,
a compressor can exist without (but will then start acting as an expander).
A limiter must also work with peaks, not RMS. So I'd class a limiter as a
compressor with more strict requirements.

>
> On Sep 3, 2009, at 1:25 AM, Kevin Dixon wrote:
>
>> Thanks for the insight Tom, as a bassist and engineer, I would agree
>> with your (and TASCAM's) assessment of how this should work
>>
>>
>> Tom Duffy wrote:
>>>
>>> A compressor's role is to change (generally reduce) the dynamic range
>>> of a signal.
>>> Without any gain, setting the threshold low (e.g. -40dB) and setting
>>> the compression ratio high, e.g. 1:100, the output signal will be
>>> mostly
>>> 40dB lower than the input signal.  That's not a louder signal.
>>> Adding 40dB of gain now gives you peaks over the prior signal signal
>>> level, depending on the attack speed.
>>>
>>> "Mastering Loudness" is achieved by making the peaks meet 0dBFS,
>>> after you've reduced the dynamic range; it's a simple gain stage
>>> after a compressor.
>>> "Loudness" in the last 10 years or so has been also achieved by
>>> letting the peaks go above 0dBFS and simply chopping them off, which
>>> is a high distortion (therefore non-musical) transformation.
>>>
>>> There is no sense in sending impulses through a compressor, it is
>>> by definition a non-linear system, that would tell you nothing about
>>> it.
>>>
>
> this last statement, i cannot agree with.  i look at compressors and
> limiters as being the same species of animal.
>
> if your compressor or limiter is acting on peak amplitude rather than
> r.m.s., sending in impulses that exceed the threshold or knee will
> tell you much more than nothing about what the compressor is doing.
>
> being a non-linear system means that the impulse response cannot tell
> you how the device will respond to all other inputs.  for a linear
> time-invariant system, the impulse response is sufficient to tell you
> how the system will behave for any general input.
>
>
>>> Compression as a tool for managing high dynamic range signals during
>>> recording (e.g. TASCAM products) is different from compression as a
>>> component in "making things louder".
>
> how are they different *qualitatively*?  different attack and release
> times, different mapping curve, different overall gain.
>
> a multiband compressor is qualitatively different, but i might
> imagine such used in either situation.
>
>
>>> In general, you don't want to have to adjust the gain every time you
>>> change the compressor settings, so keeping the perceived level about
>>> the same eliminates some of the "louder sounds better, quieter
>>> sounds worse" psychoacoustic effect, and lets you concentrate on the
>>> dynamics of the signal you are working on.
>
> it seems to me that the issue is whether you want to tack down the
> knee or corner to a certain level, or if you want to tack down 0 dB
> FS, or somewhere in between.  but those are settings; i don't see the
> qualitative difference.
>
>
> --
>
> r b-j                  rbj@...
>
> "Imagination is more important than knowledge."
>
>
>
>
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[Bogac Topaktas]

> On Sep 3, 2009, at 6:04 PM, robert bristow-johnson wrote:
>
> if your compressor or limiter is acting on peak amplitude rather than  
> r.m.s., sending in impulses that exceed the threshold or knee will  
> tell you much more than nothing about what the compressor is doing.

But it can not reveal any clue about program dependent behavior.

> being a non-linear system means that the impulse response cannot tell  
> you how the device will respond to all other inputs.

The key word here is "memory". Dynamic convolution can capture static
(i.e. memory-less) non-linear behavior but becomes totally useless when
it comes to dynamic non-linearities.

> how are they different *qualitatively*?  different attack and release  
> times, different mapping curve, different overall gain.
 
All of them plus program dependent behavior, see the following articles for
full details:

"Analysis of Dynamic Range Control (DRC) Devices"
http://www.uaudio.com/webzine/2006/september/index2.html

"What is it about the 1176LN and LA2A that makes them
distinctive-sounding?
How can the distinctive properties of these compressors be captured in a
digital emulation?"
http://www.uaudio.com/webzine/2004/february/index2.html




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[robert bristow-johnson]

On Sep 3, 2009, at 4:34 PM, Bogac Topaktas wrote:

>> On Sep 3, 2009, at 6:04 PM, robert bristow-johnson wrote:
>>
>> if your compressor or limiter is acting on peak amplitude rather than
>> r.m.s., sending in impulses that exceed the threshold or knee will
>> tell you much more than nothing about what the compressor is doing.
>
> But it can not reveal any clue about program dependent behavior.

no clue at all??

i am not saying (nor have ever said) that the complete system  
description can be obtained by driving the compressor/limiter with an  
impulse or even a string of impulses (perhaps with different  
heights).  but you *can* get a clue about where the knee might be and  
the compressor starts kicking in.

if you build a compressor/limiter (say, in software as a plug-in) you  
have to test it with signals to see, regarding those test signals, if  
the compressor does as it is expected to do.  many test signals are  
necessary, but if the compressor/limiter is supposed to prevent  
spikes from exceeding the rails and getting clipped, to test that  
behavior, what test signal do you suggest to try at first?


>> being a non-linear system means that the impulse response cannot tell
>> you how the device will respond to all other inputs.
>
> The key word here is "memory". Dynamic convolution can capture static
> (i.e. memory-less) non-linear behavior but becomes totally useless  
> when
> it comes to dynamic non-linearities.

saying "totally useless" is a very strong statement and hard to  
defend.  all i have to do is find a single use and it is refuted.


>> how are they different *qualitatively*?  different attack and release
>> times, different mapping curve, different overall gain.
>
> All of them plus program dependent behavior,

of course, there is program dependent behavior.  it's a compressor.  
i still do not get the point that the response of the compressor to a  
well-defined impulse tells you *nothing* about how the compressor is  
working.  if it is peak-level detecting, it should detect that  
impulse even if it is only one sample wide.  if the height of that  
impulse exceeds the knee threshold, what comes out should be  
predictably adjusted in height.  if you have a string of impulses of  
increasing amplitude that are spaced far enough apart that the  
compressor attack and release recover to their quiescent states, that  
string of impulses can at least give you a *clue* about what the dB-
in vs. dB-out mapping function might be.

i think you may have reacted to something i never said (like an  
impulse response completely describes the input-output behavior of  
the box when it is not LTI), but i continue to maintain that claiming  
that the response to an impulse provides *no* clue at all is an  
overstatement.


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[Tom Duffy-2]

Of course a series of impulses, properly calibrated and calculated,
would start
to characterize a compressor/limiter.

But, a single impulse tells you nothing.
When I am developing, debugging or analyzing an existing dynamics processor,
impulse responses are the last tool I would use, after a tone generator
and storage
scope, and probably a load of test wave files, .e.g. 1kHz tone at
-10dBFS for 5 seconds,
then 5 seconds silence, repeated, then different levels and then
different ramped
levels.

Tom.

robert bristow-johnson wrote:

> On Sep 3, 2009, at 4:34 PM, Bogac Topaktas wrote:
>
>  
>>> On Sep 3, 2009, at 6:04 PM, robert bristow-johnson wrote:
>>>
>>> if your compressor or limiter is acting on peak amplitude rather than
>>> r.m.s., sending in impulses that exceed the threshold or knee will
>>> tell you much more than nothing about what the compressor is doing.
>>>      
>> But it can not reveal any clue about program dependent behavior.
>>    
>
> no clue at all??
>
> i am not saying (nor have ever said) that the complete system  
> description can be obtained by driving the compressor/limiter with an  
> impulse or even a string of impulses (perhaps with different  
> heights).  but you *can* get a clue about where the knee might be and  
> the compressor starts kicking in.
>
> if you build a compressor/limiter (say, in software as a plug-in) you  
> have to test it with signals to see, regarding those test signals, if  
> the compressor does as it is expected to do.  many test signals are  
> necessary, but if the compressor/limiter is supposed to prevent  
> spikes from exceeding the rails and getting clipped, to test that  
> behavior, what test signal do you suggest to try at first?
>
>
>  
>>> being a non-linear system means that the impulse response cannot tell
>>> you how the device will respond to all other inputs.
>>>      
>> The key word here is "memory". Dynamic convolution can capture static
>> (i.e. memory-less) non-linear behavior but becomes totally useless  
>> when
>> it comes to dynamic non-linearities.
>>    
>
> saying "totally useless" is a very strong statement and hard to  
> defend.  all i have to do is find a single use and it is refuted.
>
>
>  
>>> how are they different *qualitatively*?  different attack and release
>>> times, different mapping curve, different overall gain.
>>>      
>> All of them plus program dependent behavior,
>>    
>
> of course, there is program dependent behavior.  it's a compressor.  
> i still do not get the point that the response of the compressor to a  
> well-defined impulse tells you *nothing* about how the compressor is  
> working.  if it is peak-level detecting, it should detect that  
> impulse even if it is only one sample wide.  if the height of that  
> impulse exceeds the knee threshold, what comes out should be  
> predictably adjusted in height.  if you have a string of impulses of  
> increasing amplitude that are spaced far enough apart that the  
> compressor attack and release recover to their quiescent states, that  
> string of impulses can at least give you a *clue* about what the dB-
> in vs. dB-out mapping function might be.
>
> i think you may have reacted to something i never said (like an  
> impulse response completely describes the input-output behavior of  
> the box when it is not LTI), but i continue to maintain that claiming  
> that the response to an impulse provides *no* clue at all is an  
> overstatement.
>
>
> --
>
> r b-j                  rbj@...
>
> "Imagination is more important than knowledge."
>
>
>
>
> --
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>
>  

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[Steffan Diedrichsen-2]

Having developed some dynamic processors, I can say, that a response  
to an impulse is not more than a simple test case. As you said,  
Robert, it's great to check a Limiter. But it doesn't help to  
characterize the compressor. As Tom said, a sine  wave with variying  
amplitudes reveals more. A rectangular wave might have the advantage,  
that the level detection converts it to DC. Might be useful, too.  
'Nuff said. ;-)

Anyway, to get back to the automatic gain, Tom's approach is quite  
nice. But even a "fixed" automatic makeup (e.g. -12dB input level  
results in -12dB output level) makes live easy. Setting the fix point  
to 0dB causes the compressor to get louder, if you dive into the  
signal with the threshold control. At -12dB, it's seems to be a good  
compromise.

Best,

Steffan


Am 04.09.2009|KW36 um 17:18 schrieb robert bristow-johnson:

> but i continue to maintain that claiming
> that the response to an impulse provides *no* clue at all is an
> overstatement.

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[Ross Bencina-3]

That's interesting Steffan

I was playing with the Logic 5 Platinum compressor recently and it did seem
that the automatic makeup gain had the fix point set lower than 0dBFS...

On the other hand I received a message off list from someone who had worked
on some well-reputed mastering compression systems where the automatic
makeup gain fix point was 0dBFS.

On the subject of automatic analysis of compressor characteristics, the
following is one of the few papers I've found. It only presents preliminary
results but has an interesting methodology:

IDENTIFYING AND ANALYZING RELEVANT CHARACTERISTICS
OF DYNAMIC RANGE COMPRESSION
Andrés Cabrera
DAFx-06
http://www.dafx.ca/proceedings/papers/p_061.pdf

The UA web zine stuff that Bogac already posted is also good...


Bests

Ross.



----- Original Message -----
From: "Steffan Diedrichsen" <sdiedrichsen@...>
To: "A discussion list for music-related DSP" <music-dsp@...>
Sent: Saturday, September 05, 2009 6:18 AM
Subject: Re: [music-dsp] computing compressor automatic makeup gain (?)

> Having developed some dynamic processors, I can say, that a response
> to an impulse is not more than a simple test case. As you said,
> Robert, it's great to check a Limiter. But it doesn't help to
> characterize the compressor. As Tom said, a sine  wave with variying
> amplitudes reveals more. A rectangular wave might have the advantage,
> that the level detection converts it to DC. Might be useful, too.
> 'Nuff said. ;-)
>
> Anyway, to get back to the automatic gain, Tom's approach is quite
> nice. But even a "fixed" automatic makeup (e.g. -12dB input level
> results in -12dB output level) makes live easy. Setting the fix point
> to 0dB causes the compressor to get louder, if you dive into the
> signal with the threshold control. At -12dB, it's seems to be a good
> compromise.
>
> Best,
>
> Steffan
>
>
> Am 04.09.2009|KW36 um 17:18 schrieb robert bristow-johnson:
>
>> but i continue to maintain that claiming
>> that the response to an impulse provides *no* clue at all is an
>> overstatement.
>
> --
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[robert bristow-johnson]

On Sep 5, 2009, at 12:19 AM, Ross Bencina wrote:

> That's interesting Steffan

no shit.


> On Sep 4, 2009, at 4:18 PM, Steffan Diedrichsen wrote:
>>  A rectangular wave might have the advantage,
>> that the level detection converts it to DC. Might be useful, too.
>> 'Nuff said. ;-)

i get it.  thank you.

--

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[Steffan Diedrichsen-2]

So, you guys,

who'll be @ 0xffth AES convention? I know, that Robert will be there,  
won't you?
But who else?

Best,

Steffan


Am 05.09.2009|KW36 um 08:33 schrieb robert bristow-johnson:

>
> On Sep 5, 2009, at 12:19 AM, Ross Bencina wrote:
>
>> That's interesting Steffan
>
> no shit.
>
>
>> On Sep 4, 2009, at 4:18 PM, Steffan Diedrichsen wrote:
>>> A rectangular wave might have the advantage,
>>> that the level detection converts it to DC. Might be useful, too.
>>> 'Nuff said. ;-)
>
> i get it.  thank you.
>
> --
>
> r b-j                  rbj@...
>
> "Imagination is more important than knowledge."
>
>
>
>
> --
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[bastian.schnuerle]

we gonna get a logic9 nfr or a macbook off50% if we are there ?

Am 08.09.2009 um 18:24 schrieb Steffan Diedrichsen:

> So, you guys,
>
> who'll be @ 0xffth AES convention? I know, that Robert will be there,
> won't you?
> But who else?
>
> Best,
>
> Steffan
>
>
> Am 05.09.2009|KW36 um 08:33 schrieb robert bristow-johnson:
>
>>
>> On Sep 5, 2009, at 12:19 AM, Ross Bencina wrote:
>>
>>> That's interesting Steffan
>>
>> no shit.
>>
>>
>>> On Sep 4, 2009, at 4:18 PM, Steffan Diedrichsen wrote:
>>>> A rectangular wave might have the advantage,
>>>> that the level detection converts it to DC. Might be useful, too.
>>>> 'Nuff said. ;-)
>>
>> i get it.  thank you.
>>
>> --
>>
>> r b-j                  rbj@...
>>
>> "Imagination is more important than knowledge."
>>
>>
>>
>>
>> --
>> dupswapdrop -- the music-dsp mailing list and website:
>> subscription info, FAQ, source code archive, list archive, book
>> reviews, dsp links
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>
> --
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[Steffan Diedrichsen-2]

Am I Santa Claus? ;-)

At least the US AES members can take advantage of a sale special.
http://www.aes.org/e-news/2009/Sep3.cfm#apple

Best,

Steffan




Am 08.09.2009|KW37 um 18:30 schrieb bastian.schnuerle:

> we gonna get a logic9 nfr or a macbook off50% if we are there ?
>
> Am 08.09.2009 um 18:24 schrieb Steffan Diedrichsen:
>
>> So, you guys,
>>
>> who'll be @ 0xffth AES convention? I know, that Robert will be there,
>> won't you?
>> But who else?
>>
>> Best,
>>
>> Steffan
>>
>>
>> Am 05.09.2009|KW36 um 08:33 schrieb robert bristow-johnson:
>>
>>>
>>> On Sep 5, 2009, at 12:19 AM, Ross Bencina wrote:
>>>
>>>> That's interesting Steffan
>>>
>>> no shit.
>>>
>>>
>>>> On Sep 4, 2009, at 4:18 PM, Steffan Diedrichsen wrote:
>>>>> A rectangular wave might have the advantage,
>>>>> that the level detection converts it to DC. Might be useful, too.
>>>>> 'Nuff said. ;-)
>>>
>>> i get it.  thank you.
>>>
>>> --
>>>
>>> r b-j                  rbj@...
>>>
>>> "Imagination is more important than knowledge."
>>>
>>>
>>>
>>>
>>> --
>>> dupswapdrop -- the music-dsp mailing list and website:
>>> subscription info, FAQ, source code archive, list archive, book
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>>
>> --
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>
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[bastian.schnuerle]

hahaha
+
Am 08.09.2009 um 18:37 schrieb Steffan Diedrichsen:

> Am I Santa Claus? ;-)
>
> At least the US AES members can take advantage of a sale special.
> http://www.aes.org/e-news/2009/Sep3.cfm#apple
>
> Best,
>
> Steffan
>
>
>
>
> Am 08.09.2009|KW37 um 18:30 schrieb bastian.schnuerle:
>
>> we gonna get a logic9 nfr or a macbook off50% if we are there ?
>>
>> Am 08.09.2009 um 18:24 schrieb Steffan Diedrichsen:
>>
>>> So, you guys,
>>>
>>> who'll be @ 0xffth AES convention? I know, that Robert will be  
>>> there,
>>> won't you?
>>> But who else?
>>>
>>> Best,
>>>
>>> Steffan
>>>
>>>
>>> Am 05.09.2009|KW36 um 08:33 schrieb robert bristow-johnson:
>>>
>>>>
>>>> On Sep 5, 2009, at 12:19 AM, Ross Bencina wrote:
>>>>
>>>>> That's interesting Steffan
>>>>
>>>> no shit.
>>>>
>>>>
>>>>> On Sep 4, 2009, at 4:18 PM, Steffan Diedrichsen wrote:
>>>>>> A rectangular wave might have the advantage,
>>>>>> that the level detection converts it to DC. Might be useful, too.
>>>>>> 'Nuff said. ;-)
>>>>
>>>> i get it.  thank you.
>>>>
>>>> --
>>>>
>>>> r b-j                  rbj@...
>>>>
>>>> "Imagination is more important than knowledge."
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> dupswapdrop -- the music-dsp mailing list and website:
>>>> subscription info, FAQ, source code archive, list archive, book
>>>> reviews, dsp links
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>>>
>>> --
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>>
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[Al Clark]

Steffan Diedrichsen wrote:

> So, you guys,
>
> who'll be @ 0xffth AES convention? I know, that Robert will be there,  
> won't you?
> But who else?
>
> Best,
>
> Steffan
>  

Robert is giving a presentation on Monday. I think he is covering the
cookbook stuff. (Robert, this might be a good time to clarify if I have
it wrong)
I'll be there.

Al Clark
Danville Signal Processing









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[robert bristow-johnson]

-----Original Message-----
From: "Al Clark" [aclark@...]
Date: 09/08/2009 13:17
To: "A discussion list for music-related DSP" <music-dsp@...>
Subject: Re: [music-dsp] computing 0xFFth AES convention?

>
> Robert is giving a presentation on Monday. I think he is covering the
> cookbook stuff.

not exactly.  it's a tutorial, so it's about pretty basic stuff that i might have thought would be more commonly employed.

> (Robert, this might be a good time to clarify if I have it wrong)

it's about how (and why bother) to design finite order polynomial approximations to functions, as opposed to look-up table (LUT) (with or without linear interpolation) which appears to be much more commonly employed.  the use of these mappings could either be for the common "cook the coefficients" process between knob twists and what the DSP alg sees (like what is in the biquad filter cookbook), or it could be about memoryless non-linear functions that full-bandwidth audio signals are applied to.  we know that simple LUT (especially without any interpolation) has the advantage of speed when little other consideration is made (lot'sa memory, maybe some index quantization error).  but i would think that there it would be more common in hardware applications (where there might not be your trusty std C math lib or floating-point) for implementing functions with short finite power series.  and truncated Taylor series is not such a good way to derive it.

> I'll be there.

i guess i have to be, now.

--

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"Imagination is more important than knowledge."








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dupswapdrop -- the music-dsp mailing list and website:
subscription info, FAQ, source code archive, list archive, book reviews, dsp links
http://music.columbia.edu/cmc/music-dsp 
http://music.columbia.edu/mailman/listinfo/music-dsp