Actually this is a great tip for the OP, ( or anyone )

Get Bitwig 1 and Bitwig 2 <•> render the same song, and hear the difference

This is a proven case where the company actually improved their sound engine after loads of complaints <•> and you will be easily be able to hear if you can hear the difference and that there is a difference

 

They don't. Everyone buys timestretching/tuning engine from zplane called Elastique, just sometimes different tiers of the same thing.

 

The only kind of "filter" a DAW might do is either the stock plugins it comes with (of course these will sound different between DAWs) or dither (which you don't hear anyway and the point of it is to add noise so the conversion to 16-bit doesn't include artifacts - of course, the noise "will sound different" but this is also true for different bounces in the same DAW and you won't hear it anyway - this is why your exports might not null out by the way, make sure you're using 24-bit float WAVs for comparisons)

Yes, there is. 16-bit audio (which is CD quality) doesn't even use floating-point so there are no errors during processing and summing is literally a = 2, b = 3, master = a + b = 2 + 3 = 5. For floating point rounding errors are introduced at like the 10th decimal point which human ears can't recognize anyway. And if the DAWs are set to use the same precision there's no difference.

Do you even know what FIR means? FIR stands for Finite Impulse Response, it's the kind of filters that's not usable for realtime audio and uses convolution. So your linear phase EQs and whatnot use it. Why in your mind a DAW might inject a sneaky filter that introduces plenty of latency is beyond me.

Stop spreading the bullshit and first find out what gain means:

Gain is a ratio. It's literally volume of output / volume of input. No "absolute values of the coefficients here".
There's no such thing as "DC Gain" and "AC Gain". You'd have to isolate DC offset from the signal and amplify them separately, which nobody does, because first of all we're not using DC in audio and it gets removed before you even record since your audio interface blocks all DC.

But since you insist, the Fourier Transform is a way to calculate the spectrum from a piece of audio. If you process a FIR (impulse response of a FIR filter, which I still don't see the point of talking about), you get the "EQ curve" of said filter. Again, there's no "complex coefficients" to be talking about - they are indeed there, but don't mean anything and are only an intermediate value that you get while computing the transform. Anyway, it has absolutely nothing to do with "gain", and it most certainly has nothing to do with the audio engine of any DAW.

 

this is a dangerous thread to be resurrecting :D

 

OK obviously you don't get it so here you go ..

YES fir filtering can and is done in real time

I won't even tear your sorry reply a new one . Just a explanation of how sample rate is determined...


And honestly the fact i'm using this simple explanation of practical use should be all the proof you need to there importance.

****************************************************************************************************************************************************************************

The sampling frequency or sampling rate, fs, is the average number of samples obtained in one second (samples per second), thus fs = 1/T.

Reconstructing a continuous function from samples is done by interpolation algorithms. The Whittaker–Shannon interpolation formula is mathematically equivalent to an ideal lowpass filter whose input is a sequence of Dirac delta functions that are modulated (multiplied) by the sample values. When the time interval between adjacent samples is a constant (T), the sequence of delta functions is called a Dirac comb. Mathematically, the modulated Dirac comb is equivalent to the product of the comb function with s(t). That purely mathematical abstraction is sometimes referred to as impulse sampling.[2]

Most sampled signals are not simply stored and reconstructed. But the fidelity of a theoretical reconstruction is a customary measure of the effectiveness of sampling. That fidelity is reduced when s(t) contains frequency components whose periodicity is smaller than two samples; or equivalently the ratio of cycles to samples exceeds ½ (see Aliasing). The quantity ½ cycles/sample × fs samples/sec = fs/2 cycles/sec (hertz) is known as the Nyquist frequency of the sampler. Therefore, s(t) is usually the output of a lowpass filter, functionally known as an anti-aliasing filter. Without an anti-aliasing filter, frequencies higher than the Nyquist frequency will influence the samples in a way that is misinterpreted by the interpolation process.[3]
****************************************************************************************************************************************************************************

And NO not every company is using the same pieces of code for this.

 

It's an all-time greatest, getting a rebirth from time to time in phases of low creativity levels...
It's always the DAW's fault anyway :P

 

here is a link to discussion of the Fabfilter and DMG eq's and how they implemented the FIR
https://www.gearslutz.com/board/mastering-forum/1074140-non-linear-fir-vs-iir.html

 

If you take the time to read that you'll understand the importance of the point i'm making and why It's the difference between daws as much as it's the difference between plugins.
Dave from DMG had 12 years of EQ research go into the FIR in equilibrium ...

definitely not some stock code companies all use and pass around..

 

https://masters-of-music.com/proof-that-all-daws-sound-the-same-video/

 

A major self test: take the same project and load on every single DAW that you have (clean without stock plugs) and ask a friend to put PLAY, you, close your eyes and listen paying attention and take note the name of DAW that you think are run, you'll be surprised.

 

in this whole argument not one test actually tested the argument in question.

They even prove the point with this statement:
as long as all the plugins and settings are identical, and EQ presets are used and not just copied.

Which I agree with 100% if everything is identical they better be the same ..
But that's not what debate was originally about now was it..

So I wrote them

HI guys, I have some input in regards to all daws sound the same debate. I'm not trying to pick some sort of fight or dredge up shit from the murky depths .But ....you guys even prove the argument of them sounding different by stating ( as long as all the plugins and settings are identical, and EQ presets are used and not just copied.) Well that means you have chosen a group of effects and EQ's that can be used in each daw .. And yes 100 % I agree they better be identical. But now your not really testing each daw. But rather a group of effects transferred between them. Doing the test with strictly native fx in each daw and a 1;1 match in settings doesn't sound the same. One would have to dial them in to match the other, and even then pulling off a null test on a FX / EQ heavy mix...well good luck. Even if the original tracks were identical they won't be after that.

Anyway Have a good one

and for anyone interested ...
But by all means go ahead prove they sound the same, because nulling identical files without actually doing anything to them doesn't prove anything in regards to actual freq manipulation.

which as far as I understood is what the original statement is about .

 

☀️

 

Last edited: Nov 30, 2018

☀️

 

Last edited: Nov 30, 2018

If someone cannot hear the differences between DAW summing engines... then they should change their inaccurate speakers or listening environment or keep music production strictly as a hobby. And stop posting null tests as proof. utter horse bollocks.
Pro Tools 12 still sounds better than all DAWS and still kicks ableton 10's ass in sound quality and summing. I dont need a white paper to tell me different.

 

+1. Totally agree.

 

Last edited by a moderator: Nov 30, 2018

My dick is bigger than your

 

-1. Totally disagree

So it's back to 0 (aka horseshit)

 

This question again...

 

No, it's not. My is bigger and even if we will measure our dicks and they will turn out to be exactly the same, I will consider my dick to be longer and superior, because it's my dick after all.

(This thread in a nutshell.)