Hi..
I'm wondering something and I know some people here could give me an answer.
When an audio signal goes above 0db in digital world, you get a digital click, right ? But this click is made by, for example, a vsti. Do you prevent the click by puting a limiter (an another vst plugin) in an insert of this vsti ?
I mean : the digital click goes out of the vsti and enter in the limiter -> so the limiter 'sees', the signal carries a click, how could it erase it ? The limiter (for me) is going to try to limit the signal but the click sound is still present ? Am I right ?
For me again, a limiter in digital sound can work only with several track , like in a bus channel : each channel doesn't have any clipping signal but the addition of all in the bus is above 0db, in this case the limiter can do a good work...(?)
Limiter in digital world
Re: Limiter in digital world
If the signal is clipped then the limiter can't do anything about that. You apply wave restoration tools in that case.
To properly limit a signal, feed the limiter with your non-clipped signal, lower the limiter's threshold so to catch the peaks, apply a small boost so to compensate for the volume attenuation. Use the sidechain "look-ahead" delay so the sidechain signal advances in relation to the signal that is limited and the limiter gets some time to limit the peaks correctly. (without artifacts)
For peak limiting fast attack and decay values are commonly used.
"For me again, a limiter in digital sound can work only with several track , like in a bus channel : each channel doesn't have any clipping signal but the addition of all in the bus is above 0db, in this case the limiter can do a good work...(?)"
That is not the case. You should lower the individual sounds gain so to get them properly mixed and be able to apply the neccesary tools on them. Stay away from 0 dB. -24 -12 dB are ok for the mixing - production stage. (even lower). Always boost your signals at the end of the chain, and use softclip and limiters to get them nicely at 0dB.
Most people regard digital sound harsh because they mix loud. Try lower volumes to get superior sound, smoothness and ability to actually use these tools.
To properly limit a signal, feed the limiter with your non-clipped signal, lower the limiter's threshold so to catch the peaks, apply a small boost so to compensate for the volume attenuation. Use the sidechain "look-ahead" delay so the sidechain signal advances in relation to the signal that is limited and the limiter gets some time to limit the peaks correctly. (without artifacts)
For peak limiting fast attack and decay values are commonly used.
"For me again, a limiter in digital sound can work only with several track , like in a bus channel : each channel doesn't have any clipping signal but the addition of all in the bus is above 0db, in this case the limiter can do a good work...(?)"
That is not the case. You should lower the individual sounds gain so to get them properly mixed and be able to apply the neccesary tools on them. Stay away from 0 dB. -24 -12 dB are ok for the mixing - production stage. (even lower). Always boost your signals at the end of the chain, and use softclip and limiters to get them nicely at 0dB.
Most people regard digital sound harsh because they mix loud. Try lower volumes to get superior sound, smoothness and ability to actually use these tools.
Re: Limiter in digital world
Thanks ! That confirms what I think.
This is how I try to do and it's true that I get much better result !Try lower volumes to get superior sound, smoothness and ability to actually use these tools.
Re: Limiter in digital world
dB is a reference for analogue world, dB=20log(vo/vi), but in digital world it is emulated to recreate the same behavior.
In fact the limit before distortion depends on word lenght(bit resolution) of the processing algorithm, the higher the bit resolution, the higher headroom we have, so it is possible to go beyond the 0 dB limit on a single channel, but a good practice is to always start with mixer faders at lower levels, such as -6dB to avoid the bottleneck on the master bus.
Then a good mixdown must not exceed -12dB rms so that it will be good for the mastering session.
About your click question, it depends on what is causing the click, usually when the volume is too high, you get distortion, if you get a click, it could be between a frequency range such as too much resonance on a synth, so it's better to apply some eq, or multiband, but if you start mixing at low levels everything is much clear.
In fact the limit before distortion depends on word lenght(bit resolution) of the processing algorithm, the higher the bit resolution, the higher headroom we have, so it is possible to go beyond the 0 dB limit on a single channel, but a good practice is to always start with mixer faders at lower levels, such as -6dB to avoid the bottleneck on the master bus.
Then a good mixdown must not exceed -12dB rms so that it will be good for the mastering session.
About your click question, it depends on what is causing the click, usually when the volume is too high, you get distortion, if you get a click, it could be between a frequency range such as too much resonance on a synth, so it's better to apply some eq, or multiband, but if you start mixing at low levels everything is much clear.
Re: Limiter in digital world
djmicron there is no reason to confuse the discussion. dB is a unit of measurement. It applies to both analog and digital world.
Mixer headroom and word length is different from signal amplitude. You are reffering to the bits of the mixing engine. In any case if your signal has a peak that reaches 0dB if you boost that signal even with 0.000001 of gain that peak is going to be clipped. Always use a oscilloscope so to know what you are saying. In general audio engineers work with an oscilloscope or have spend so many time with one, so to actually know what is happening. Never rely on theory. Its in the practice the things get screwed up.
A click is always a click. It doesn't matters what causes it. You should avoid clicks at all costs. Clicks are digital lost information.
An amateur usage of equipment.
In general don't be afraid to go down in volumes. For an analog signal there is a certain point where the signal is loud and the noise level is low. With digital signals be free to go down in levels.
Mixer headroom and word length is different from signal amplitude. You are reffering to the bits of the mixing engine. In any case if your signal has a peak that reaches 0dB if you boost that signal even with 0.000001 of gain that peak is going to be clipped. Always use a oscilloscope so to know what you are saying. In general audio engineers work with an oscilloscope or have spend so many time with one, so to actually know what is happening. Never rely on theory. Its in the practice the things get screwed up.
RMS is useless if you have peaks that ruin your mix. Your peaks should reach -12dB.Then a good mixdown must not exceed -12dB rms so that it will be good for the mastering session.
About your click question, it depends on what is causing the click,
A click is always a click. It doesn't matters what causes it. You should avoid clicks at all costs. Clicks are digital lost information.
An amateur usage of equipment.
Eq and multiband whatever, are useless with clicks. There is only one way to avoid them. Learn about the digital levels.if you get a click, it could be between a frequency range such as too much resonance on a synth, so it's better to apply some eq, or multiband,
They are not just "much clear". They are totally clear. Gosh...but if you start mixing at low levels everything is much clear.
In general don't be afraid to go down in volumes. For an analog signal there is a certain point where the signal is loud and the noise level is low. With digital signals be free to go down in levels.
Re: Limiter in digital world
fra77x, i forgot to ask your permission before writing my opinion.
When i started writing there weren't any answers.
I avoid replying to your wrong statements, at least talk with a mastering engineer and he will tell you that all it counts is rms not the peaks before mastering.
You are acting as a troll everytime i reply to a thread, you called me nazist on another thread, now you are sayng that i have not to reply, don't know what you have against me, but i'm not interested in your approvation, i don't know you, you don't know me.
When i started writing there weren't any answers.
I avoid replying to your wrong statements, at least talk with a mastering engineer and he will tell you that all it counts is rms not the peaks before mastering.
You are acting as a troll everytime i reply to a thread, you called me nazist on another thread, now you are sayng that i have not to reply, don't know what you have against me, but i'm not interested in your approvation, i don't know you, you don't know me.
Re: Limiter in digital world
Sorry i didn't want to sound that way. I don't remember calling you anything like that in any other thread. Perhaps you confused me with someone else.
I just thought that your contribution confused that matter and i wanted to state some facts clear for people's help.
Of course everyone posts whatever he things that is suitable in each case but sometimes is helpful to correct misguiding statements.
By the way, i am a mastering engineer.
I just thought that your contribution confused that matter and i wanted to state some facts clear for people's help.
Of course everyone posts whatever he things that is suitable in each case but sometimes is helpful to correct misguiding statements.
By the way, i am a mastering engineer.
Re: Limiter in digital world
ok, i'll post a video to show you what happens when going over 0dB on digital.
Re: Limiter in digital world
If you insist... i'm all eyes.
Re: Limiter in digital world
Hum.. That's right for me.. When I worked with 16bit / 44.1 khz, the clicks above 0db were terrible and with 32bit / same samplerate, I don't have the problem.In fact the limit before distortion depends on word lenght(bit resolution) of the processing algorithm, the higher the bit resolution, the higher headroom we have,
I must agree with that : it's the subject of this thread : if the click goes out of any device / track, it's too late, I believe there is no device to erase it exept some restauration plugin.Quote:
if you get a click, it could be between a frequency range such as too much resonance on a synth, so it's better to apply some eq, or multiband,
Eq and multiband whatever, are useless with clicks. There is only one way to avoid them. Learn about the digital levels.
Anyway, thank you both for your advices.. I'm NOT a mastering engineer... But I try to treat my case.
Re: Limiter in digital world
So the clicks are less terrible?"Hum.. That's right for me.. When I worked with 16bit / 44.1 khz, the clicks above 0db were terrible and with 32bit / same samplerate, I don't have the problem."
Word length is the number of binary digits necessary to represent an integer. 0dB is the maximum amplitude for any digital signal regardless of the wordlength. A signal above 0dB will clip at 16bit, 32 bit or 256 bit...
There is no magical procedure to restore lost information, and that is clipping. A signal that exceeds the specific medium's specifications by misusage of the equipment.
Re: Limiter in digital world
Well.. You must be right for the example you give us.
But via asio / 32bit float recording, in Cubase, the manual says (sorry, it's in frensh) :
"Si vous enregistrez au format 32 bits à virgule flottante,
la résolution en bits ne sera pas réduite – ce qui élimine
tout risque d’écrêtage à ce stade.
En outre, ceci préserve intégralement la qualité du signal."
Passed thru Reverso, it gives :
"If you record in the size(format) 32 bits with floating decimal point, the resolution in bits will not be reduced - what eliminates any risk of peak clipping at this stage.
Besides, this protects entirely the quality of the signal."
Maybe your example and 32bit float recording are two different things... And there is some misunderstanding in the air...
Hum...
But via asio / 32bit float recording, in Cubase, the manual says (sorry, it's in frensh) :
"Si vous enregistrez au format 32 bits à virgule flottante,
la résolution en bits ne sera pas réduite – ce qui élimine
tout risque d’écrêtage à ce stade.
En outre, ceci préserve intégralement la qualité du signal."
Passed thru Reverso, it gives :
"If you record in the size(format) 32 bits with floating decimal point, the resolution in bits will not be reduced - what eliminates any risk of peak clipping at this stage.
Besides, this protects entirely the quality of the signal."
Maybe your example and 32bit float recording are two different things... And there is some misunderstanding in the air...
Hum...
Re: Limiter in digital world
I will not try to convince you more. That was the reason i insisted in the difference between mixer headroom and signal amplitude.
4 24.08 dB 16 −8 to +7
8 48.16 dB 256 −128 to +127
11[15] 66.22 dB 2048 −1024 to +1023
16 96.33 dB 65,536 −32,768 to +32,767
20 120.41 dB 1,048,576 −524,288 to +524,287
24 144.49 dB 16,777,216 −8,388,608 to +8,388,607
32 192.66 dB 4,294,967,296 −2,147,483,648 to +2,147,483,647
48 288.99 dB 281,474,976,710,656 −140,737,488,355,328 to +140,737,488,355,327
64 385.32 dB 18,446,744,073,709,551,616 −9,223,372,036,854,775,808 to +9,223,372,036,854,775,807
That is the signal to noise ratio.
In any case 0dB is the maximum amplitude of the signal.
There is no misunderstanding, it is straightforward. How to add some values to a signal that uses your whole amplitude range?
It's impossible. You have to lower that signal or make the addition in a way that it incorporates more values. (mixing engines that work with more bits than the signals being added. When you sum these sources back to one file the values should get mapped to the normal bit length or get clipped.
If i take a 16 bit signal and route it through a 32 bit mixer, it actually becomes 32 bit in the process. Hence 0dB still applies.
Anyway, all these are tricks to make the users believe that it's more difficult to clip their signals but finally they confuse the proper way of working and most people complain that after they mixed down their tracks they got clipping.
If it was a matter of bits how loud a signal could get, then the loudness war would use more bits so to get louder sounds. But this is not the case. The loudness war incorporates careful usage of limiting and controlled saturation so to get loud in a specific medium specifications.
Everything else is misunderstanding.
4 24.08 dB 16 −8 to +7
8 48.16 dB 256 −128 to +127
11[15] 66.22 dB 2048 −1024 to +1023
16 96.33 dB 65,536 −32,768 to +32,767
20 120.41 dB 1,048,576 −524,288 to +524,287
24 144.49 dB 16,777,216 −8,388,608 to +8,388,607
32 192.66 dB 4,294,967,296 −2,147,483,648 to +2,147,483,647
48 288.99 dB 281,474,976,710,656 −140,737,488,355,328 to +140,737,488,355,327
64 385.32 dB 18,446,744,073,709,551,616 −9,223,372,036,854,775,808 to +9,223,372,036,854,775,807
That is the signal to noise ratio.
In any case 0dB is the maximum amplitude of the signal.
There is no misunderstanding, it is straightforward. How to add some values to a signal that uses your whole amplitude range?
It's impossible. You have to lower that signal or make the addition in a way that it incorporates more values. (mixing engines that work with more bits than the signals being added. When you sum these sources back to one file the values should get mapped to the normal bit length or get clipped.
If i take a 16 bit signal and route it through a 32 bit mixer, it actually becomes 32 bit in the process. Hence 0dB still applies.
Anyway, all these are tricks to make the users believe that it's more difficult to clip their signals but finally they confuse the proper way of working and most people complain that after they mixed down their tracks they got clipping.
If it was a matter of bits how loud a signal could get, then the loudness war would use more bits so to get louder sounds. But this is not the case. The loudness war incorporates careful usage of limiting and controlled saturation so to get loud in a specific medium specifications.
Everything else is misunderstanding.
Re: Limiter in digital world
I record in 32bit float with Scope asio 64bit float. And I always take care to never have any clipping anywhere : red leds are fought with the last fury !
It was just a question to improve my knowledge ! And you did !
It was just a question to improve my knowledge ! And you did !
Re: Limiter in digital world
One last thing.
Floats, bits or any representation system can't make your audio pump the speaker more.
Loud signals = Big amplitude and loudness signals, more watts
No other way.
Regards,
John
Floats, bits or any representation system can't make your audio pump the speaker more.
Loud signals = Big amplitude and loudness signals, more watts
No other way.
Regards,
John
Re: Limiter in digital world
here is the demo video:
http://goo.gl/HHoOow
bit depth is essential for more headroom under digital domain.
dB measurement is a standard created on analog gear, what we see on digital software is an emulation of that, on digital we have binary values, the more the bit resolution, the more we can sum together.
http://goo.gl/HHoOow
bit depth is essential for more headroom under digital domain.
dB measurement is a standard created on analog gear, what we see on digital software is an emulation of that, on digital we have binary values, the more the bit resolution, the more we can sum together.
Re: Limiter in digital world
also, more bit depth equal to more tracks to sum together in the mix befiore reaching the limit before distortion.
-
jksuperstar
- Posts: 1638
- Joined: Mon Nov 15, 2010 12:57 pm
Re: Limiter in digital world
The biggest problem is that it is true: "0db" (or any db for that matter) doesn't mean anything in the digital world. It's an arbitrary designation...it can mean something different for any piece of software that shows you the amplitude.
It's one of the reasons I like Wavelab's bit-view, which simply shows the bit-amplitude...you can always see what headroom (in bits) you have...or how a 16-bit signal is being padded with zeros or interpolated when it's converted to 32-bit. I'd love a view like that for scope!
Back to the issue...one DAW can define 0db = 24-bit max (leaving 8 bits headroom-and allows you to add 256 tracks together, each with 24-bit maxed out). Another can define 26-bit max = 0db. And therefore has less headroom...but maybe has more resolution when it converts 16-bit audio. None of it is a standard, and the last time I looked at ASIO, I don't think it was defined there, either. And I haven't found info about coreaudio on MAC either.
So having more bits is helpful, but depends on how it's used in software. It is more helpful for a mono synth to have more bits as resolution (smaller values for accuracy), but more helpful for a mixer with lots of tracks to have headroom (larger values to avoid saturation).
It's one of the reasons I like Wavelab's bit-view, which simply shows the bit-amplitude...you can always see what headroom (in bits) you have...or how a 16-bit signal is being padded with zeros or interpolated when it's converted to 32-bit. I'd love a view like that for scope!
Back to the issue...one DAW can define 0db = 24-bit max (leaving 8 bits headroom-and allows you to add 256 tracks together, each with 24-bit maxed out). Another can define 26-bit max = 0db. And therefore has less headroom...but maybe has more resolution when it converts 16-bit audio. None of it is a standard, and the last time I looked at ASIO, I don't think it was defined there, either. And I haven't found info about coreaudio on MAC either.
So having more bits is helpful, but depends on how it's used in software. It is more helpful for a mono synth to have more bits as resolution (smaller values for accuracy), but more helpful for a mixer with lots of tracks to have headroom (larger values to avoid saturation).
Re: Limiter in digital world
Guys please
Floating numbers let you go up i.e +3dB etc
If you render the file it gets clipped. (haven't you tried?)
A small story. When companies provided software daws they observed that their stupid users didn't even know how to lower the faders so to mix their sounds. (me included). By having an established misconception that loud means hot they were afraid to lower their signals so to not lose "dynamic range". That lead the common amateur user to clip their signals and reach the conclusion that digital daws have harsh or bad sound. So gradualy acompaning their other software development they tried to fix that issue so the common amateur would be able to apply equalizer without clipping the signal. That lead to the "more bits more headroom" misunderstanding. Because there is no extra headroom at all. In digital audio as i said there is NO headroom at all. 0dB is the top. Period.
Bits mean the signal dynamic ratio, the relation between noise and louder signal.
The whole subject is to have the user leave his faders at "0dB" and "mix".
wiki:
Headroom in digital audio
In digital audio, headroom is defined as the amount by which digital full scale (FS) exceeds the permitted maximum level (PML) in dB (decibels). The European Broadcasting Union (EBU) specifies a PML of 9 dB below 0 dBFS (-9 dBFS), thus giving 9 dB of headroom. An alternative EBU recommendation allows 24 dB of headroom, which might be used for 24-bit master recordings where it is useful to allow more room for unexpected peaks during live recording.
Failure to provide adequate headroom can bring about clipping of brief, higher-level transients.
All Daws use the same reference for dB.The biggest problem is that it is true: "0db" (or any db for that matter) doesn't mean anything in the digital world. It's an arbitrary designation...
Floating numbers let you go up i.e +3dB etc
If you render the file it gets clipped. (haven't you tried?)
A small story. When companies provided software daws they observed that their stupid users didn't even know how to lower the faders so to mix their sounds. (me included). By having an established misconception that loud means hot they were afraid to lower their signals so to not lose "dynamic range". That lead the common amateur user to clip their signals and reach the conclusion that digital daws have harsh or bad sound. So gradualy acompaning their other software development they tried to fix that issue so the common amateur would be able to apply equalizer without clipping the signal. That lead to the "more bits more headroom" misunderstanding. Because there is no extra headroom at all. In digital audio as i said there is NO headroom at all. 0dB is the top. Period.
Bits mean the signal dynamic ratio, the relation between noise and louder signal.
The whole subject is to have the user leave his faders at "0dB" and "mix".
wiki:
Headroom in digital audio
In digital audio, headroom is defined as the amount by which digital full scale (FS) exceeds the permitted maximum level (PML) in dB (decibels). The European Broadcasting Union (EBU) specifies a PML of 9 dB below 0 dBFS (-9 dBFS), thus giving 9 dB of headroom. An alternative EBU recommendation allows 24 dB of headroom, which might be used for 24-bit master recordings where it is useful to allow more room for unexpected peaks during live recording.
Failure to provide adequate headroom can bring about clipping of brief, higher-level transients.
Re: Limiter in digital world
Of course i don't want to convince you. Everyone can believe anything that he likes.
There are people around that believe in ghosts. So there can be people that believe in "headroom" too.
I have no problem at all with that.
Just stay away from my audio!
There are people around that believe in ghosts. So there can be people that believe in "headroom" too.
I have no problem at all with that.
Just stay away from my audio!