Question about Epilog

BenOBS · Aug 29, 2007, 06:49 PM

Hi all!

I was using k2lt since the beginning and now switched to epilog to have access to the last 2 channels of my audigy 2 zs (side channels whit the 4-poles jack connectors).

I noticed at full volume a way lower output, arround 10db to my ear. Every gains/volume i know has been checked. I tryed Epilog and Epiloglt_k2 whit the same result. Peak show me the same output (0db) from my src and if i boost this signal to get the same output i get into clipping. I know that k2lt max volume isnt clipping because i tunned my amplifier gains whit a scope..

So im wondering if this is normal or not? Does epilog have some kind of attenuation?
Also, if your want, i would like to know what are the difference between the epilog and K2lt other than the ressources needed. I know master volume now work whit the epilog but is there any processing done by the epilog that k2lt dont?

The main reason why I want to be sure to have the full output from my A2 is because its use in a car environment and higher pre-out voltage help lower the background noise. If i cant, i can adjust my amplifiers gains whit the new max voltage but this isnt the best option.

Thanks all for you invaluable help.

Russ · Aug 29, 2007, 08:36 PM

With the individual speaker volumes (Ins and Outs) at 0 dB, and the Master Volume at full, epilog should be the same as kxlt.

BenOBS · Aug 30, 2007, 05:56 PM

Thanks to confirm. I will investigate more. Maybe it is in my VST host...

Maybe one more question on the fly... According to the peak meter, where the A2 start clipping? 0db/3db/6db ????

Maddogg6 · Aug 30, 2007, 06:59 PM

I don't have a scope to measure - but audibly - I *usually* get it just above 0db.

I have played around quite a bit with 'mastered' levels - and have concurred that 0db is the max level to use.

But I can record a guitar solo that peaks at nearly +3 dbs and not notice it audibly (or in the audio editor - the 'peak' is more of a high freq spike, instead of a low freq - flatting) - and kept the 'take' with out regret.

I hope that made some sense....

BenOBS · Aug 30, 2007, 08:24 PM

Yeah, thanks Maddogg6. Ill stick to 0db max level.

Related to this. What type of algorithm does epilog use to lower the volume?
Im thinking it multiply each sample by a binary constant but cant confirm...

This is related to this article:

Quote:

[COLOR=Navy]Digital volume control with a PCM bitstream is actually very simple, you simply multiply
each sample by a binary constant. This apparent simplicity is a trap for improperly
implemented digital volume controls that can really mess up a digital signal. The result
length in bits of any binary multiplication is equal to the sum of the word lengths of both
input terms. For example a 16 bit pcm signal multiplied by a 16 bit volume control
constant results in a 32 bit binary number. This conversion is completely loss-less, the
original signal can be completely reconstructed from the 32 bit output data. However as
there are no 32 bit DACs a 32 bit bitstream must truncated and possibly dithered or
noise shaped to a manageable word length before it is converted into analog audio.
Truncation is always a lossy process, once a signal is truncated no amount of processing
may restore the original, it may be possible to to make any truncation artifacts inaudible
with dither or noise shaping but the key to good digital volume control is to avoid any
lossy translations. Instead of multiplying the pcm bitstream by a 16 bit constant you
may multiply it by constant with an 8 bit or less word length, a 16 bit data stream and 8
bit volume coefficient will result in a 24 bit word, which is attainable by many high
quality DAC designs so no truncation is required. However the loss-less resolution of the
result comes at the price of volume control steps that do not accurately track the ideal
logarithmic attenuation curve, an 8 bit word my only attenuate a signal by a maximum
of 48db and the last few steps will be very large making a reasonable attenuation limit
of an 8 bit volume coefficient about 30db or so.
The best digital volume controls are those that use a table that attempts to limit the
amount of truncation needed. Such a volume control may use 4 bit constants for the
first few attenuation levels, then gradually increase the word length as needed for
greater amounts of attenuation. This table should include as many "magic" attenuation
values as possible, -6.02db is only a 1 bit coefficient, -12.04db is only 2 bits ect... This
approach maximizes the signal quality but also minimizes the linearity of the attenuation
steps so that each volume step will be slightly different, however as volume control is more of a
bulk attenuation and human hearing can easy adapt, a linear volume control is not really
necessary.
[COLOR=Black]It should also be apparent that the least amount of attenuation possible should be used in a
system with digital volume control, in an ideal system you should always be listening as near as
possible full volume, this usually requires amplifiers with low gain.[/COLOR] This can be a real problem since
most amplifiers are designed with as much gain as possible to give the customer the illusion of
"Power". Using amplifiers with too much gain requires too much digital attenuation at normal
listening levels leading to very low resolution (too much truncation of the digital signal before it is
converted in the DAC.) Digital volume control should also never be used with DACs of insufficient
word length, a 16 bit DAC will require truncating a CD resolution digital input for any attenuation
value at all so digital volume controls should only be used with DACs capable of converting word
lengths of 20 bits or more. Of course better, higher resolution DACs result in better sound quality
with digital volume control than lesser DACs. An acceptable DAC used without digital volume
control may become unlistenable when used with digital volume control due to linearity errors that
are normally masked when converting full level data, delta sigma DAC designs are perhaps the
worst in this respect while sign-magnitude DAC designs are the best. A sign-magnitude DAC always
increases its linearity with decreasing signal level making it ideal for digital volume control use. A delta-sigma DAC on the other hand has a noise level that increases with decreasing signal level
making most of DACs of this type very poor choices to use with digital volume control.[/COLOR]

Russ · Aug 30, 2007, 08:28 PM

0 dB is the clipping point, but there is some overhead in the DSP that will allow you to exceed 0 dB without clipping, provided that the signal is brought back down below 0 dB before the output plugin (assuming output plugin is at optimal level).

Russ · Aug 30, 2007, 08:42 PM

Volume control (epilog) is done by multiplying every sample by the same amount (amount is set using sliders).

Individual speaker volumes are applied, then Master Volume is applied, and then the signal is multiplied by 16.

i.e.
(Signala are divided by 4 when entering the DSP to give some overhead).
Signal = 0.25 (max after div 4)
Speaker volume is 0 dB (= 0.25 in DSP)
Master volume is full (= 1)

0.25 * 0.25 * 1 * 16 = 1 (0 dB)

kxlt just multiples the signal by 4:
0.25 * 4 = 1 (0 dB)

BenOBS · Aug 30, 2007, 08:46 PM

Thats what I expected, thanks alot.

Russ · Aug 30, 2007, 09:51 PM

BTW: Just in case it went unnoticed, epilog can do +12 dB gain, which obviously can cause clipping.

i.e.
The volume sliders range from 0 to 1, and if both sliders are set to 1 (full):
0.25 * 1 * 1 * 16 = 4 (+12 dB)

Aug 29, 2007, 06:49 PM	#1
BenOBS DriverHeaven Newbie Join Date: Oct 2006 Posts: 17 Rep Power: 0	Question about Epilog Hi all! I was using k2lt since the beginning and now switched to epilog to have access to the last 2 channels of my audigy 2 zs (side channels whit the 4-poles jack connectors). I noticed at full volume a way lower output, arround 10db to my ear. Every gains/volume i know has been checked. I tryed Epilog and Epiloglt_k2 whit the same result. Peak show me the same output (0db) from my src and if i boost this signal to get the same output i get into clipping. I know that k2lt max volume isnt clipping because i tunned my amplifier gains whit a scope.. So im wondering if this is normal or not? Does epilog have some kind of attenuation? Also, if your want, i would like to know what are the difference between the epilog and K2lt other than the ressources needed. I know master volume now work whit the epilog but is there any processing done by the epilog that k2lt dont? The main reason why I want to be sure to have the full output from my A2 is because its use in a car environment and higher pre-out voltage help lower the background noise. If i cant, i can adjust my amplifiers gains whit the new max voltage but this isnt the best option. Thanks all for you invaluable help.

Aug 29, 2007, 08:36 PM	#2
Russ HardwareHeaven Extreme Member Join Date: Jan 2005 Posts: 5,563 Rep Power: 62	With the individual speaker volumes (Ins and Outs) at 0 dB, and the Master Volume at full, epilog should be the same as kxlt.

Aug 30, 2007, 05:56 PM	Thread Starter #3
BenOBS DriverHeaven Newbie Join Date: Oct 2006 Posts: 17 Rep Power: 0	Thanks to confirm. I will investigate more. Maybe it is in my VST host... Maybe one more question on the fly... According to the peak meter, where the A2 start clipping? 0db/3db/6db ????

Aug 30, 2007, 06:59 PM	#4
Maddogg6 Tail Razer Join Date: Jun 2005 Location: Bernyurass, AZ - USA Posts: 4,027 Rep Power: 0	I don't have a scope to measure - but audibly - I usually get it just above 0db. I have played around quite a bit with 'mastered' levels - and have concurred that 0db is the max level to use. But I can record a guitar solo that peaks at nearly +3 dbs and not notice it audibly (or in the audio editor - the 'peak' is more of a high freq spike, instead of a low freq - flatting) - and kept the 'take' with out regret. I hope that made some sense....

Aug 30, 2007, 08:28 PM	#6
Russ HardwareHeaven Extreme Member Join Date: Jan 2005 Posts: 5,563 Rep Power: 62	0 dB is the clipping point, but there is some overhead in the DSP that will allow you to exceed 0 dB without clipping, provided that the signal is brought back down below 0 dB before the output plugin (assuming output plugin is at optimal level).

Aug 30, 2007, 08:42 PM	#7
Russ HardwareHeaven Extreme Member Join Date: Jan 2005 Posts: 5,563 Rep Power: 62	Volume control (epilog) is done by multiplying every sample by the same amount (amount is set using sliders). Individual speaker volumes are applied, then Master Volume is applied, and then the signal is multiplied by 16. i.e. (Signala are divided by 4 when entering the DSP to give some overhead). Signal = 0.25 (max after div 4) Speaker volume is 0 dB (= 0.25 in DSP) Master volume is full (= 1) 0.25 * 0.25 * 1 * 16 = 1 (0 dB) kxlt just multiples the signal by 4: 0.25 * 4 = 1 (0 dB)

Aug 30, 2007, 08:46 PM	Thread Starter #8
BenOBS DriverHeaven Newbie Join Date: Oct 2006 Posts: 17 Rep Power: 0	Thats what I expected, thanks alot.

Aug 30, 2007, 09:51 PM	#9
Russ HardwareHeaven Extreme Member Join Date: Jan 2005 Posts: 5,563 Rep Power: 62	BTW: Just in case it went unnoticed, epilog can do +12 dB gain, which obviously can cause clipping. i.e. The volume sliders range from 0 to 1, and if both sliders are set to 1 (full): 0.25 * 1 * 1 * 16 = 4 (+12 dB)