What is the best sounding DAW??

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fforet wrote:why? isnt this always a matter of how a software is coded or how an instrument is built? if you change the case material or form of your guitar,piano,whatever it will sound different. daws wont probably better with pine wood , cause their function is to output audio at an exactly defined audio file format.
DAWs and Synths both sound better with wood! GarageBand has a wood master bus and Reason has a wood rack. So there you have it.
Last edited by Michael L on Fri Dec 01, 2017 6:23 am, edited 1 time in total.
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https://en.m.wikipedia.org/wiki/Compari ... io_editors

I guess it's more likely to assume that it comes down to a DAW's native plug ins.

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Mine :clown:
I wonder what happens if I press this button...

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That daw.. or was this (?) ahh can't remember

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replicant X wrote:There will be many arguments but I feel every DAW have different sounds.

Ableton sounds wide, loud and clear.
Bitwig sounds detailed on quiet part but darker than Ableton.
I think Cubase have something like psychoacoustic technics.
It sounds very different.
Logic have a artifact on high-frequency when it reached the certain point.
Mutools Mulab sounds very clear, sounds like Protools.
I didn't like Reaper 4's week thin sound, but Reaper 5 have very natural sound.
And I really feel FL studio sounds darker than others.

I don't know why and some of these maybe placebo effects.
But I feel differences.
It’s all in you mind. It’s all 0s and 1s.
Orion Platinum, Muzys 2

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Harrison Mixbuss reminds me of the smell of dirty stinky faders and the sound you hear when sitting in front of a dirty beaten up 64 channel analogue Console. It makes any laptop sound like it has a soul.
Orion Platinum, Muzys 2

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Jbravo wrote:"Spend time on any forum devoted to any Digital Audio Workstation (DAW) software or music production and you are guaranteed to see users making claims about the superior audio quality of this or that DAW application. Protagonists will say a given program is clearly and audibly superior to another. To be frank, that's just nonsense. Any DAW application that uses, at least, 32 Bit floating point calculations (and today, that's all software), will process audio without introducing unwanted distortions, frequency response alterations or any other effect that would be 'clearly audible' so as to influence opinion. This ability to process audio without making unintended, audible changes is called 'transparency'. From a transparency perspective all DAW software is created equal."

http://www.image-line.com/support/FLHel ... _audio.htm
That’s interesting. Everyone is aiming for absolute transparency with 32 but floating point audio.

There’s room on the market for a 12 bit daw. With 12 bit summing abased on the the Sp1200 and MPC 60.
Orion Platinum, Muzys 2

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solomute wrote:This is the most huge lies that all Daws sound the same. They can't sound the same because they use different algos for summing up channels. And those algos try to avoid phase cancellation at different degrees of success. That's why you should choose the best one which is DP. As for pros they buy summators and analog mixers to avoid that problem in which case all daws can sound the same since mixing is done on external hardware.
Summing is literally just adding numbers like 1 + 1. The only way a DAW can sound different is if it gets basic maths wrong.
Orion Platinum, Muzys 2

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Fake maths
dedication to flying

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They do sound different for a single reason. You use them to treat the sound.
As long as you just mix, I would not expect a difference, but that is not all you use a DAW for.
Start a single edit, and differences will appear.
Ableton for example is treating without telling you (sounds bad), as warp is on by default. (I couldn't believe it on a first listen)

I think Ardour sounds best, I don't hear the cry in the wallet...
Professionals think ProTools is the best, it can't be good if you didn't justify your professionalism with expensive gear...

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solomute wrote:This is the most huge lies that all Daws sound the same. They can't sound the same because they use different algos for summing up channels. And those algos try to avoid phase cancellation at different degrees of success. That's why you should choose the best one which is DP. As for pros they buy summators and analog mixers to avoid that problem in which case all daws can sound the same since mixing is done on external hardware.
The summing part is the easy one. I expect a serious DAW to just use the obvious and one and only algorithm called "+". I f a DAW claimes to avoid phase cancelation (clearly a sound engineers duty), I would avoid that DAW, because it is treating the sound where I don't want it to be treated and in a way I don't know what the hell they do.
Avoiding phase cancelation would destroy timing and change the sound. I bet DP is not doing that btw...(unless its their "catch the pros" marketing blah)
Mixing analog is treating the sound and can of course enhance it. Analog gear really sounds different...

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Maybe is subjective but i always find Mulab free as very clean and nice sounding DAW so that's my suggestion about 'best' if there is such thing at all when it comes to the sound - it's more like 'favorite' than best.Same line best sound designer,best songwriter,best beatmaker...no such thing it's just your preferences :)Cheerz :)

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If you make dance music or electronic music you do not need to read the following. All this refers to acoustic music.
I will repeat that daws use different resampling algos and by that reason they can't sound the same. And it's not 1+1=2 but an equation where tracks' sound is affected by a resampling algo at least. Currently I recommend sequoia\samplitude with ignoring non-magix's vsts except hofa, vb-audio and airwindows. In this case you preserve initial quality of soundlibs and soundfonts. If you process sound thru most other vsts you lose highs, clarity, space, resolution, details, timbre and finally get mud. Soundlibs already have highest quality in them and additional processing only spoils them ie your first rule must be the less processing the better. Instead of applying any processing you should try to point out beauty of sound of soundlibs by composition ie composition should be your goal #1 because software is made in such a way that you can't get higher or better quality than already present in soundlibs or your live recordings (of course you can improve sound by eqing but quality does not improve). Maybe this is made for you to buy expensive external hardware mixers. Because harrison mixbuss shows that hw can be nicely emulated. The drawback of that harrison mixbus is that it applies saturation all the time without you having control over saturation ie you can't switch off saturation. And awful midi editor. To sum it up your goal #2 is to try to emulate studio sound quality and for that you should avoid any additional processing to preserve the ultimate quality already present in soundlibs instead of applying emulators of analog gear. In this case sound preserves some rigid digital tint but at least clarity and details remain untouched which is an integral feature of studio sound. And it's complete bullshit that you should use compressors and other effects for improving the sound. Instead of them use high-quality eq like bba genwave eq (if you can find it!) where needed. Linearphase eqs are not usable due to latency they produce.
It may take years of tests for you to start to detect the moment where you start losing quality. Internet is a wastebin and gives a lot of wrong, evil, distracting and contradictory recommendations because too many individuals want to earn money and you are encouraged to buy more and more magical vsts from the people whose level of expertise, intentions and motives are totally unknown to you.
What I wanted to say is that it's not only daw itself which contributes to sound quality but also intentional and nonintentional limitations of digital technology + your personal workflow (choice of plugins, avoiding phase cancellation, correct\incorrect use of effects). It also means there is no daw which would protect soundquality against user's mistakes such as phase cancellation, artifacts caused by latency, bad algos of vsts and etc. Your task is to protect samplitude's high quality from agressive vsts which spoil that quality. Since digital technology supresses highs you should always have an eq boosting highs on master in ON mode, otherwise you won't be able to mix anything successfully. Your last vst on master must be hofa goniometer in correlator mode to see phase cancellation ie you do not need using mono mode. All Midside processing is waste of efforts which will destroy clear location of sound sources along stereo field. The best MS processor available is nugen monofilter and it supresses highs. So if you intend to apply any stereo-widening vst on master hoping to return lows to the center by further applying MS processing solution like nugen monofilter I will dissapoint you here - there is no MS vst I know of that would do it without ruining stereo scene and timbre. IE if stereowidening is needed it should be applied to tracks individually. Think, how can it be that low frequencies of an instrument come from one place and high frequences of the same instrument come from another place. And it's that totally crazy thing which you are advised doing all over the internet. And it's what those MS vsts exactly do. So if you follow those advices you will never get clean stereo scene with spacially pleseantly detectable instruments. There also other evil traps like reverbs and compressors, limiters, amp emulators and etc accompanied with the correspoding evil advices or just lack of correct instructions. As you may figure out great number of improvers are actually spoilers and you won't be able to get studio space and quality by default by applying this or that magical vst which boastingly lies that it produces vynil sound or other sweet thing. Don't let them deceive you!
Adding emulation of analog distortion only adds distortion and nothing more while quality and purity of highs degrade. For that reason try to preserve the initial quality of soundlibs with all their details instead of adding antialiasing-like emulating vsts.
samplitude is the best daw for me. To have studio like sound before asking questions on any audio forums in the internet please read the book by alex unlocking fx creative potential

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solomute wrote:If you make dance music or electronic music you do not need to read the following. All this refers to acoustic music.
I will repeat that daws use different resampling algos and by that reason they can't sound the same. And it's not 1+1=2 but an equation where tracks' sound is affected by a resampling algo at least. Currently I recommend sequoia\samplitude with ignoring non-magix's vsts except hofa, vb-audio and airwindows. In this case you preserve initial quality of soundlibs and soundfonts. If you process sound thru most other vsts you lose highs, clarity, space, resolution, details, timbre and finally get mud. Soundlibs already have highest quality in them and additional processing only spoils them ie your first rule must be the less processing the better. Instead of applying any processing you should try to point out beauty of sound of soundlibs by composition ie composition should be your goal #1 because software is made in such a way that you can't get higher or better quality than already present in soundlibs or your live recordings (of course you can improve sound by eqing but quality does not improve). Maybe this is made for you to buy expensive external hardware mixers. Because harrison mixbuss shows that hw can be nicely emulated. The drawback of that harrison mixbus is that it applies saturation all the time without you having control over saturation ie you can't switch off saturation. And awful midi editor. To sum it up your goal #2 is to try to emulate studio sound quality and for that you should avoid any additional processing to preserve the ultimate quality already present in soundlibs instead of applying emulators of analog gear. In this case sound preserves some rigid digital tint but at least clarity and details remain untouched which is an integral feature of studio sound. And it's complete bullshit that you should use compressors and other effects for improving the sound. Instead of them use high-quality eq like bba genwave eq (if you can find it!) where needed. Linearphase eqs are not usable due to latency they produce.
It may take years of tests for you to start to detect the moment where you start losing quality. Internet is a wastebin and gives a lot of wrong, evil, distracting and contradictory recommendations because too many individuals want to earn money and you are encouraged to buy more and more magical vsts from the people whose level of expertise, intentions and motives are totally unknown to you.
What I wanted to say is that it's not only daw itself which contributes to sound quality but also intentional and nonintentional limitations of digital technology + your personal workflow (choice of plugins, avoiding phase cancellation, correct\incorrect use of effects). It also means there is no daw which would protect soundquality against user's mistakes such as phase cancellation, artifacts caused by latency, bad algos of vsts and etc. Your task is to protect samplitude's high quality from agressive vsts which spoil that quality. Since digital technology supresses highs you should always have an eq boosting highs on master in ON mode, otherwise you won't be able to mix anything successfully. Your last vst on master must be hofa goniometer in correlator mode to see phase cancellation ie you do not need using mono mode. All Midside processing is waste of efforts which will destroy clear location of sound sources along stereo field. The best MS processor available is nugen monofilter and it supresses highs. So if you intend to apply any stereo-widening vst on master hoping to return lows to the center by further applying MS processing solution like nugen monofilter I will dissapoint you here - there is no MS vst I know of that would do it without ruining stereo scene and timbre. IE if stereowidening is needed it should be applied to tracks individually. Think, how can it be that low frequencies of an instrument come from one place and high frequences of the same instrument come from another place. And it's that totally crazy thing which you are advised doing all over the internet. And it's what those MS vsts exactly do. So if you follow those advices you will never get clean stereo scene with spacially pleseantly detectable instruments. There also other evil traps like reverbs and compressors, limiters, amp emulators and etc accompanied with the correspoding evil advices or just lack of correct instructions. As you may figure out great number of improvers are actually spoilers and you won't be able to get studio space and quality by default by applying this or that magical vst which boastingly lies that it produces vynil sound or other sweet thing. Don't let them deceive you!
Adding emulation of analog distortion only adds distortion and nothing more while quality and purity of highs degrade. For that reason try to preserve the initial quality of soundlibs with all their details instead of adding antialiasing-like emulating vsts.
I read it and I LOLed, what a bunch of non sense.
dedication to flying

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Somehow I am reminded of U.S. Air Force General Jack Ripper. He was into purity of essence, too.

I don't give a toss whether or not DAWS sound different, better, worse, whatever. Whatever sound is generated from within my pc gets pumped through my RME AIO card, along cheap, cheap cables into a pair of not quite so cheap, but still cheap, speakers. :shrug:

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