ableton sound engine and new monolake album

[newboss]

Do you notice that you are ruining your reputation more and more with each post ?

You are not even being able to read, it seems:

1. my studio is a tiny room in my apartment. See also: http://www.monolake.de/technology/studio.html
2. i used to work on PT systems for various professional non monolake things in the past.
3. the pic are from the mastering studio, which is also a production house, however the monolake album was not produced there but in my little home 'studio'.

and yes, my idea of friendship is oldschool. and it works quite well in my private life. i can actually trust my friends.

and as far as your 'questions' are concerned, I and others did answer them at length in the past. hundreds of times.

robert

What reputation? the guy that finds bugs that are not allowed to be there reputation?

So no direct answer again. no surprise really.
But dont you think that this might damage your reputation at one point?
You are a professor.. a man of science.
You know, I am a bit old school about the truth and it hurts when this gets bended all the time for marketing reasons.

Do all daws sound the same? A simple yes or no would do.

[monolake]

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[monolake]

Do you notice that you are ruining your reputation more and more with each post ?

You are not even being able to read, it seems:

1. my studio is a tiny room in my apartment. See also: http://www.monolake.de/technology/studio.html
2. i used to work on PT systems for various professional non monolake things in the past.
3. the pic are from the mastering studio, which is also a production house, however the monolake album was not produced there but in my little home 'studio'.

and yes, my idea of friendship is oldschool. and it works quite well in my private life. i can actually trust my friends.

and as far as your 'questions' are concerned, I and others did answer them at length in the past. hundreds of times.

robert

What reputation? the guy that finds bugs that are not allowed to be there reputation?

So no direct answer again. no surprise really.
But dont you think that this might damage your reputation at one point?
You are a professor.. a man of science.
You know, I am a bit old school about the truth and it hurts when this gets bended all the time for marketing reasons.

Do all daws sound the same? A simple yes or no would do.

There is no simple yes or no here. The 'direct answer' you are asking for is exactly the kind of answer that can only be politicians or marketing talk.I am not avoiding it, I am avoiding an answer that is wrong, because it simplifies reality !

We did tests with 80 or more tracks running on different DAWs and they cancel. So, summing is the same. Which is no surprise since it is supposed to be transparent and this is a clear definition. Let's ignore everything below - 180dB in the sum ( before rounding to 24 bit in your soundcard) here because I really do believe it does not matter, or it will not be present in the signal anyway. Summing in the digital world is adding and multiplying. The most basic thing a computer can do. You'll might get different results on the very last bits, but this is why we all do 32 or 64 bit floating point these days: To move that potential error way below the lowest possible signals. Therefor if you truncate at the very end of the mixing process down to 24 bit, the result is bitwise identical even in a worse case scenario. The summing is not and has not been an issue for digital audio since at least 20 years.

And everything else, EQs, Effects, etc. of course are different. The question is: Do you like working with your environment and is it inspiring you? If no, the best sound quality will not help you. If yes, it can be an 8 bit sampler and you are king. It's all out there, just choose what you need and enjoy.

Workflow is often the most important part: I am not using the ICON or any other mixing surface, because for my working method it is not needed. It would just eat up space. Composing music is a slow process for me, and I am perfectly fine with drawing an envelope or changing a parameter with the mouse. Because most of the time I am listening and thinking. If I were to work in a production house / film post pro situation etc... I would not want to do this. But this has nothing to do with sound quality and it never had. It is about the workflow for specific types of work. In my work flow for making music, my current setup with Live and my old synthesizers is actually quite perfect. The only thing I would want is a slightly bigger room in my flat and the ability to treat the room acoustics a bit more. Because this has a magnitude more influence on sound than anything else.


And as far as my reputation goes: I have no problems to state what I think. I have no problems admitting flaws, personal ones and known ones of Live. I cannot see how this could ruin my reputation.


A summary of this thread: an anonymous 'friend' of mine basically says I am lying because it is not possible in general that a good sounding album has been produced entirely in Live. I explain that his assumptions are wrong, and give evidence for that. And now we are back to: Live sounds shit. Awesome.

Robert

[newboss]

There is no simple yes or no here.

We did tests with 80 or more tracks running on different DAWs and they cancel. So, summing is the same. Which is no surprise since it is supposed to be transparent and this is a clear definition. Let's ignore everything below - 150dB in the sum here because I really do believe it does not matter.

And everything else, EQs, Effects, etc. of course are different.
Robert

ok.. thanks.. no easy yes or no already contradicts the all daws sound the same myth...

But lets assume that you only use wave plug ins ..no daw internal ones..

is it possible that different daws can get different results in the way the data gets to and from the plug ins? And level changes.. is it possible that different daws create different artefacts on automation changes? Is it possible that daws like logic or protools dither their output to 24 bit while others just drive the DA with 32 bit float data?

And last but not least the ableton speciality of seamlessly getting into a samplerate conversion and out without any clicks what so ever. Is this samplerate conversion really always off when clip tempo and project tempo are matching? or is it possible that live engages samplerate conversion for other reasons than a tempo missmatch? maybe in moments of high cpu loads or hardisk overloads aso? or when using track delays?

In any case i would say that the sound of a daw is the combination of all aspects that add to the sound. in case of ableton live the warping is an essential part of the functionality and therefore the sound of the samplerate conversion is part of the sonic footprint when working with live.
You cant really avoid it when you use all features.

So for me there is a special ableton sound.. maybe other daws sound the same but live is different.

[newboss]

A summary of this thread: an anonymous 'friend' of mine basically says I am lying because it is not possible in general that a good sounding album has been produced entirely in Live. I explain that his assumptions are wrong, and give evidence for that. And now we are back to: Live sounds shit. Awesome.

Robert

i dont question that you can have a good sounding mix in live, i had some myself Sometimes the special ableton sound suits a track very well.. But on other occasions i had difficulties..especially with a samplerate conversion that was on when it was supposed to be off...
But that was related to hopefully already fixed bugs..

[monolake]

(...) But lets assume that you only use wave plug ins ..no daw internal ones..

is it possible that different daws can get different results in the way the data gets to and from the plug ins?

Since I am not having access to their code, the only answer can be: sure. If you are interested, the only thing you can do is: try it out.
But in general, the host is just taking the audio signal as pair of floats and handles it over to the plugin, which in return sends back a pair of floats. I cannot see why this should be different in different DAWs. But I am sure this has been tested numerous times, so there must be some trustworthy info on this out there.

And level changes.. is it possible that different daws create different artefacts on automation changes?

It is absolutely possible that the same ramp in different DAWs sound slightly different, since we all use our own interpolations here, and there is no 'standard' for that. The question is again: is it a) audible and b) if it is, does it color the sound in a good or bad way ( for the average of all users - we are entering psychoacoustics area here, there is no mathematical way to determine good or bad sound anymore, it comes down to empirical tests and statistics )

Is it possible that daws like logic or protools dither their output to 24 bit while others just drive the DA with 32 bit float data?

I am not sure about this, but since the noise floor of the best DA converters and the rest connected to the output of the converter is certainly higher than -144dB the dithering would not make any difference since it must be in that order if it is done right.

And last but not least the ableton speciality of seamlessly getting into a samplerate conversion and out without any clicks what so ever. Is this samplerate conversion really always off when clip tempo and project tempo are matching?

We had problems with this in early versions, but since no manufacture wants the reputation of having bad sound quality, we took great care that the SR / timestrech is off when not needed. This has been tested by numerous users and it works.

or is it possible that live engages samplerate conversion for other reasons than a tempo missmatch?

Well, this is where I believe the problems start: of course, you can transpose things, you can have warp markers without noticing, you can render in 48k whilst the files are 44.1k, there are a lot of ways to fuck things up.. This is in my opinion the only real sound quality problem.

maybe in moments of high cpu loads or hardisk overloads aso? or when using track delays?

Not that I know. Track delays are static, they are not linked to songtime or anything else, and they do not interpolate. that's why it sounds so harsh if you change them. HD overload causes dropouts, but that should be it. High CPU load just means that things take longer, but *all* things, so it should not cause a problem here too.


In any case i would say that the sound of a daw is the combination of all aspects that add to the sound. in case of ableton live the warping is an essential part of the functionality and therefore the sound of the samplerate conversion is part of the sonic footprint when working with live.
You cant really avoid it when you use all features.

Exactly, and this is what the users need to understand. I never use warped files, unless I want the effect. And then I love it. But I do not expect anything to sound 'good' when timestreched. Regardless of the algorithm. It is a massive change of the signal.




So for me there is a special ableton sound.. maybe other daws sound the same but live is different.

I am not sure if this can stand as a general statement since there are numerous ways to use Live without warping, without SR conversion etc. and then it is as transparent as anything else.



Robert

[newboss]

Is it possible that daws like logic or protools dither their output to 24 bit while others just drive the DA with 32 bit float data?

I am not sure about this, but since the noise floor of the best DA converters and the rest connected to the output of the converter is certainly higher than -144dB the dithering would not make any difference since it must be in that order if it is done right.

-144 sounds way to low to be relevant.. but why do some daw manufacturers pay for appogees uv22 algo? just for rendering cd masters? or is there some use for it in the internal daw mechanics?
i experiance the sound of logic and cubase as finer and more detailed on the very low levels.. quite like a properly dithred 16 bit file versus a truncated one.. therefor i got the idea that dithering might be an issue..

The test i ve done to confirm that finer sound to myself was a 12 track realtime recording that was simultaniously recorded in live and cubase and played back again on unity level..there cubase sounded finer.. live a bit rougher and less open in the ambience.. not ugly.. had its charme aswell but it wanst indentical.. when such recordings would null the sound differene only can be explained by what happens between daw and converter..and than dithering is maybe a factor..
But i havent checked wether the daw outputs would null.. Its a waste of time to investigate in that without propper mesuring tools and methods. Sofar i just trust my ears.. However.. havent bought cubase because of that.

We had problems with this in early versions, but since no manufacture wants the reputation of having bad sound quality, we took great care that the SR / timestrech is off when not needed. This has been tested by numerous users and it works.

Sorry but theory and praxis..L8 had many problems.. and some SR shit was part of them.
There was at least 2 SR related bugs metioned in the changelogs of L8..
Thanks to the improoved samplerate conversion not allways so easy to get when it kicked in.
Some called the appearance of the unwanted samplerate conversion brown henke dither.. showing up like a ghost and disappearing again..
would be actually great when there would be a indicator or recalable info box that shows whether samplerate is converted or locked on a running clip

Well, this is where I believe the problems start: of course, you can transpose things, you can have warp markers without noticing, you can render in 48k whilst the files are 44.1k, there are a lot of ways to fuck things up.. This is in my opinion the only real sound quality problem.

maybe you have noted that with L8 the ableton sound quality discussion has popped up again.. or at least has been heavier discussed again.. after a relative silence in the L6/L7 times..
I think that this was mainly caused by SR related bugs. There was versions of L8 where it was enough to import some longer external recorded files and change the master tempo around .. than SR was on and only went off when you typed in the song tempo into each clip individually again.. or reloaded the project.. A situation that especially showed up when people transfered external projects into live, and than a unwanted SR can really fuck up the idea about lives transparent sound quality because people knew what the files were supposed to sound in a mix already.

[monolake]

Is it possible that daws like logic or protools dither their output to 24 bit while others just drive the DA with 32 bit float data?

I am not sure about this, but since the noise floor of the best DA converters and the rest connected to the output of the converter is certainly higher than -144dB the dithering would not make any difference since it must be in that order if it is done right.

-144 sounds way to low to be relevant.. but why do some daw manufacturers pay for appogees uv22 algo? just for rendering cd masters? or is there some use for it in the internal daw mechanics?
i experiance the sound of logic and cubase as finer and more detailed on the very low levels.. quite like a properly dithred 16 bit file versus a truncated one.. therefor i got the idea that dithering might be an issue..

The test i ve done to confirm that finer sound to myself was a 12 track realtime recording that was simultaniously recorded in live and cubase and played back again on unity level..there cubase sounded finer.. live a bit rougher and less open in the ambience.. not ugly.. had its charme aswell but it wanst indentical.. when such recordings would null the sound differene only can be explained by what happens between daw and converter..and than dithering is maybe a factor..
But i havent checked wether the daw outputs would null.. Its a waste of time to investigate in that without propper mesuring tools and methods. Sofar i just trust my ears.. However.. havent bought cubase because of that.

We had problems with this in early versions, but since no manufacture wants the reputation of having bad sound quality, we took great care that the SR / timestrech is off when not needed. This has been tested by numerous users and it works.

Sorry but theory and praxis..L8 had many problems.. and some SR shit was part of them.
There was at least 2 SR related bugs metioned in the changelogs of L8..
Thanks to the improoved samplerate conversion not allways so easy to get when it kicked in.
Some called the appearance of the unwanted samplerate conversion brown henke dither.. showing up like a ghost and disappearing again..

The henke brown dither term was coined by myself, but i cannot remember in which of the numerous threads. But as far as I remember, there was nothing ghosty there, if the sample would have been unwarped, no warp artefacts would have occurred by the wrong import. That caused some samples to play at eg. 119.97 bpm when the song was 120 bpm. Not that I say it is good that such things are happening, but if I as a user want best sound and I do not use warp intentionally for beat matching, the first thing I do is turning warp of for all clips.

would be actually great when there would be a indicator or recalable info box that shows whether samplerate is converted or locked on a running clip

I agree 100% and I would love to have this in. I would also want a global project sample rate and a warning dialog if the set runs at other sample rates and so on. Open doors here as far as I am concerned.

Well, this is where I believe the problems start: of course, you can transpose things, you can have warp markers without noticing, you can render in 48k whilst the files are 44.1k, there are a lot of ways to fuck things up.. This is in my opinion the only real sound quality problem.

maybe you have noted that with L8 the ableton sound quality discussion has popped up again.. or at least has been heavier discussed again.. after a relative silence in the L6/L7 times..
I think that this was mainly caused by SR related bugs. There was versions of L8 where it was enough to import some longer external recorded files and change the master tempo around .. than SR was on and only went off when you typed in the song tempo into each clip individually again.. or reloaded the project.. A situation that especially showed up when people transfered external projects into live, and than a unwanted SR can really fuck up the idea about lives transparent sound quality because people knew what the files were supposed to sound in a mix already.

The biggest problem these days is not the quality of the tools anymore, it is the way people expect them to work. And this is certainly also a result of marketing. For quality one needs training and experience. As numerous people pointed out before, if something lacks 'clarity', the first thing I would check is the mix, not in a technical sense but from an artistic point of view. Maybe it is a wrong idea to have a compressor and five EQis in each track and a mastering plugin in the master channel whilst composing. The first thing a lot of pro's do when getting a mix from other people: bypass, bypass, bypass,... and then see what's there. And suddenly there is definition, there is air, there is music. No software can help you here. In the old days of hardware, people thought twice before putting the one single compressor they had into a signal chain, and they learned how to operate it. I bet you can ask any 1980s producer about what a compressor does and you'll get a precise and experienced answer. Ask a random producer who uses 20 instance of 10 different brands in his or her Live set about what exactly the definition of threshold or ratio is... These are the sound issues in 2012 if you ask me. After all, we are talking about the final product called mp3 or overdriven club PA at 3:00 am or iPhone headphones. Or what difference does UV22 versus truncate at 12 bit make for a (instert name) US dubpstep producer who sells millions? Nothing at all. The people listen to it via Youtube and the dynamic range is probably 6dB if that. And I am not judging the artistic value here, just put it into relation with the sound quality discussion. I personally do concerts as Robert Henke where i have a dynamic range of say 60dB or more. So, MIDI fader resolution becomes a real issue and i found workarounds using a dedicated master volume fader plus channel faders plus faders at other gain stages to achieve such high dynamic in a live situation. But still, this would work perfectly fine with 16 bit, CD resolution. Any DAW is far superior here. It is not the problem!

First generation of Sony PCM F1 converters for archiving digital audio 14 bit- How many top 100s productions were stored on that ? High end vinyl cutting with a Neumann VMS 70 cutting lathe - audio goes through a EMT digital delay to compensate for the automatic groove width adjustment. A digital delay with 16 bit from the early 1980s !!!
Given the logic of the threads here, all this must sound totally unbearable.

Robert

[Nilus]

When can we expect a Monolake showing in the US with some of your new material? Would you care to come to Austin, TX?

Oh yes... this is about sound quality. Can we get that Live4/5 sound back (maybe a preferences option) in a 64bit version with Live9 - I understand you are no longer working in a full time capacity at Ableton, but you got the connections still?

[JuanSOLO]

I would definitely drive to Austin to see some Monolake action.

[barryalva]

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[anybody human]

An obvious internet discussion problem is tone, you can't start out accusatory or conspiratorial and expect to have a reasoned discourse. Plugin delay compensation, digital summing etc. - these are often discussed but valid questions. Thanks for the replies, interesting & informative. Workflow sustaining inspiration is the biggest thing for me, that's why I predominantly use Live.

As an aside, it's an audio discussion; Robert Henke would have to have a Phil Spector like melt down to damage his musical reputation.

[anybody human]

Edit: double post

[simmerdown]

dl'd 3 or 4 of your 'track of the month'...the sound you are getting is so excellent, however you do it...thanks, something to aspire to

[antarktika]

hey Robert, stop spending time arguing with these kids, and use the time you save to book a show in San Francisco.