What is the best sounding DAW??

[fmr]

All DAWS do sound alike, except Harrison Mixbus, and that's only because it is designed to sound like an Harrison mixing desk.

So, I wonder why they try so hard to sell us those accessory plug-ins they have (with insane prices and that don't run anywhere else). Does the Harrison consoles need those also? Or am I missing something?

[Googly Smythe]

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.

You know more than the developers of the DAW's then. https://www.image-line.com/support/FLHe ... _audio.htm

Good read.

[Xandie]

I totally agree what replicant X and solomute wrote. There is A difference. I used to work in FL studio for about 4 or 5 months, when I started to create music. But then I found a lot of troubles and time-consuming about simple functions, such as mixer (you need type additional buttons to get access) and so on. So, I shifted to Ableton (thanks to my friend).

I also familiar with Logic and Cubase and they works fine. Many of my mates (producers) using it and its trully quality stuff. Of course, there is always 'war of brands' because its business. But I would say Ableton is a great DAW to work with PLEASURE.

Hope that helps

[borealfarmer]

cockos reaper

[Pyrotek45]

How can people say one daw Sounds better than another? They all use the same algorithm to add their tracks. Its called addition. Theres no such thing as a super specialized ableton addition mathematics or fl studio sound engine(sound engines don't exist btw its all a marketing sham). Just normal bland math for all daws. If your daw is doing more to your sound than summing your tracks into one(simple addition) then there is something wrong.

[Googly Smythe]

All DAWS do sound alike, except Harrison Mixbus, and that's only because it is designed to sound like an Harrison mixing desk.

So, I wonder why they try so hard to sell us those accessory plug-ins they have (with insane prices and that don't run anywhere else). Does the Harrison consoles need those also? Or am I missing something?

They gots to make a profit somehow.

[solomute]

Of course daws sound different. For example magix's daws produce kind of softened sound while DP more neutral and masculine like. And the most important thing according to the words' of openmpt's developers is the resampling algorithm. By the way they have confessed that resampling algos they use are not the best and not optimal and that they even are not going to implement the best algorithms. So you should keep in mind that all opensource daws, trackers and etc will always have inferior algorithms just as per desire of those "kind-hearted" developers of open-source software secretly sponsored by corporations. And when someone appears like Hannu Savolainen who wanted to develop pro quality audio driver for nix systems he is fastly neutralized. Alsa has no bypass for its mixer, so you can't get kernel streaming or wavert as on windows. That's why you will ALWAYS have larger latency on linux as well as worse sound. Keep it in mind. Openmpt supports wavert output and you can check that its faster than asio with at least the same quality. Kernel streaming is not available on foobar2000 and they have not even developed a wavert output plugin, that's why you should use albumplayer instead of foobar2000 with which you can have kernel streaming and bypass any mixers. Early versions of sequoia had wavert support which is absent now. Unlike openmpt's developers the developer of schism tracker confessed that his product produces studio like quality without need for buying expensive hardware. Unfortunately his product is only 32-bit for windows and for linux with low-quality driver it makes no sense whatever bitness it may have. As for openmpt's sound you may find that it gives most natural sound for libs run inside kontakt vst. All other daws somehow change hue of kontakt libs. I was impressed how they sound in openmpt but as I have said above all this will be spoiled on the stage of rendering when medium quality resamplers will be applied. That's why currently I would use DP provided it did not cause reboots which it does.
Also daws have plugin delay compensation feature which detects compensation values differently in different daws and this might cause different time shifts which are also phase shifts and phase shifts cause appearance of new harmonics. You can read about this rs-met plugins' developer's educating pdfs. Those harmonics may contribute to own hue of a daw's sound. But I am sure it's not the only cause. Another one is the resampler and perhaps other factors like use of hardware acceleration, use of OS's libs like ms visual c which may have mediocre quality algos and etc. By the way kmixer of windows XP is said to produce very noisy SNR of 80db according to ms' documentation officially. So whichever direction you look at you see there is something not perfect which will hardly be fixed. And we wonder why sound quality is not like in hw studios. Every time you lower sound volume by a fader you lose detail in sound. That's the main drawback of the digital technology because its best quality is provided at the maximum volume. But how could it be solved? We need to have some headroom in daws to only increase sound volumes and never decrease. But do you know any such daw which can do it? On the contrary they make you monitor volume all the time to stay lower than 0 db throughout the entire signal processing chain. Though frankly speaking I am not very knowledgeable to this subject ie if they allow mixing at volumes much higher than 0 like +6db for example. I saw a video where a guy increased volume by 1000 times in a track in sequoia and decrased volume by 1000 times on master and played back and quality remained the same. Can it mean it's a kind of non-destructive feature where sequia just have calculated that it should not change volume at all and just played it back with original volume or does it mean that sequia allows mixing at almost infinitely larger volume than 0db I don't know...
I has been kind of disappointed with DP recently because it reboots my pc and consumes a lot of cpu. That's why I won't use anymore. Samplitude seems to be the way to go because it's most stable, has retrospective recording and other valuable features like good manuals and generally it looks like a porsche for sound compared vs other daws. Perhaps someone can hack openmpt to change its resamplers by the highest quality ones in which case it will be the way to go. Or sponsor schism development to make it 64-bit and there are confessions already that it sounds better than openmpt.
Another pitfall concerns vst format. It uses floating-point calculations which are less exact than integer and produce softening and antialiasing effect which should be noticeable in reverb tails especially. Openmpt is only one which I know to use 32bit integer. Protools is said to use 48bit integer and own plugin format to match that value.
But as it seems impossible to get studio sound on pc perhaps it makes sense to focus on the choice of plugins and just think more about composition than about sound quality trying to mask low quality by skillful equalizing and panning. And the main foe to struggle against is lack of highs which is an major drawback of the digital technology perhaps because in the case of highs daws have to calculate much larger values which spoils RT capabilities.

[whyterabbyt]

Of course daws sound different. For example magix's daws produce kind of softened sound while DP more neutral and masculine like. And the most important thing according to the words' of openmpt's developers is the resampling algorithm. By the way they have confessed that resampling algos they use are not the best and not optimal and that they even are not going to implement the best algorithms. So you should keep in mind that all opensource daws, trackers and etc will always have inferior algorithms just as per desire of those "kind-hearted" developers of open-source software secretly sponsored by corporations. And when someone appears like Hannu Savolainen who wanted to develop pro quality audio driver for nix systems he is fastly neutralized. Alsa has no bypass for its mixer, so you can't get kernel streaming or wavert as on windows. That's why you will ALWAYS have larger latency on linux as well as worse sound. Keep it in mind. Openmpt supports wavert output and you can check that its faster than asio with at least the same quality. Kernel streaming is not available on foobar2000 and they have not even developed a wavert output plugin, that's why you should use albumplayer instead of foobar2000 with which you can have kernel streaming and bypass any mixers. Early versions of sequoia had wavert support which is absent now. Unlike openmpt's developers the developer of schism tracker confessed that his product produces studio like quality without need for buying expensive hardware. Unfortunately his product is only 32-bit for windows and for linux with low-quality driver it makes no sense whatever bitness it may have. As for openmpt's sound you may find that it gives most natural sound for libs run inside kontakt vst. All other daws somehow change hue of kontakt libs. I was impressed how they sound in openmpt but as I have said above all this will be spoiled on the stage of rendering when medium quality resamplers will be applied. That's why currently I would use DP provided it did not cause reboots which it does.
Also daws have plugin delay compensation feature which detects compensation values differently in different daws and this might cause different time shifts which are also phase shifts and phase shifts cause appearance of new harmonics. You can read about this rs-met plugins' developer's educating pdfs. Those harmonics may contribute to own hue of a daw's sound. But I am sure it's not the only cause. Another one is the resampler and perhaps other factors like use of hardware acceleration, use of OS's libs like ms visual c which may have mediocre quality algos and etc. By the way kmixer of windows XP is said to produce very noisy SNR of 80db according to ms' documentation officially. So whichever direction you look at you see there is something not perfect which will hardly be fixed. And we wonder why sound quality is not like in hw studios. Every time you lower sound volume by a fader you lose detail in sound. That's the main drawback of the digital technology because its best quality is provided at the maximum volume. But how could it be solved? We need to have some headroom in daws to only increase sound volumes and never decrease. But do you know any such daw which can do it? On the contrary they make you monitor volume all the time to stay lower than 0 db throughout the entire signal processing chain. Though frankly speaking I am not very knowledgeable to this subject ie if they allow mixing at volumes much higher than 0 like +6db for example. I saw a video where a guy increased volume by 1000 times in a track in sequoia and decrased volume by 1000 times on master and played back and quality remained the same. Can it mean it's a kind of non-destructive feature where sequia just have calculated that it should not change volume at all and just played it back with original volume or does it mean that sequia allows mixing at almost infinitely larger volume than 0db I don't know...
I has been kind of disappointed with DP recently because it reboots my pc and consumes a lot of cpu. That's why I won't use anymore. Samplitude seems to be the way to go because it's most stable, has retrospective recording and other valuable features like good manuals and generally it looks like a porsche for sound compared vs other daws. Perhaps someone can hack openmpt to change its resamplers by the highest quality ones in which case it will be the way to go. Or sponsor schism development to make it 64-bit and there are confessions already that it sounds better than openmpt.
Another pitfall concerns vst format. It uses floating-point calculations which are less exact than integer and produce softening and antialiasing effect which should be noticeable in reverb tails especially. Openmpt is only one which I know to use 32bit integer. Protools is said to use 48bit integer and own plugin format to match that value.
But as it seems impossible to get studio sound on pc perhaps it makes sense to focus on the choice of plugins and just think more about composition than about sound quality trying to mask low quality by skillful equalizing and panning. And the main foe to struggle against is lack of highs which is an major drawback of the digital technology perhaps because in the case of highs daws have to calculate much larger values which spoils RT capabilities.

your theory is cock.

[solomute]

I don't care for a troll's opinion. (:

[Jace-BeOS]

I care for readable text. Like, paragraph spacing and everything...

[audiosabre]

Sorry to ruin the laughs. People should check previous threads on this very topic. Those claiming that different DAWs sound different because of 'summing engines' and various other things have literally never given any evidence to support the claim. Summing (the clue is in the name) is what's known in mathematics as addition. So 2+2 = 4. Analog & digital summing are different because of the differences between discrete and continuous signals (this should be very obvious). If anyone can prove that another DAW sums differently, then you have evidence of a broken calculator. If you're talking about negligible rounding errors through differences between floating point & integer, or similar, present evidence that people can reliably identify a specific DAW through it's sound alone in a double blind test. If you can hear those errors above the probability of chance, I will personally petition for you to be in the Guinness book of world records

[Pyrotek45]

Sorry to ruin the laughs. People should check previous threads on this very topic. Those claiming that different DAWs sound different because of 'summing engines' and various other things have literally never given any evidence to support the claim. Summing (the clue is in the name) is what's known in mathematics as addition. So 2+2 = 4. Analog & digital summing are different because of the differences between discrete and continuous signals (this should be very obvious). If anyone can prove that another DAW sums differently, then you have evidence of a broken calculator. If you're talking about negligible rounding errors through differences between floating point & integer, or similar, present evidence that people can reliably identify a specific DAW through it's sound alone in a double blind test. If you can hear those errors above the probability of chance, I will personally petition for you to be in the Guinness book of world records

If only people could get this through their head.

[Pyrotek45]

I don't care for a troll's opinion. (:

Idk man, it looks to me like you're trolling here and doing a pretty good job. it's hard for me to believe someone could be so ignorant. The proof is there, you just have to read. Don't fall for marketing talk.

[whyterabbyt]

I don't care for a troll's opinion. (:

or facts, it would seem.

[elassi]

I care for readable text. Like, paragraph spacing and everything...

This!

Stopped reading at line three or four...