I have a correction/clarification (perspective: I am a research scientist working with, among other things, signal processing - so it's my domain, though I work mostly with images and videos): All filters always introduce extra latency. It can be either the latency of individual frequencies (phase shift) when working with minimum phase and similar filters or the latency of the whole signal (linear filters or mixed phase filters). People have different preferences. I don't want to get into it now, but this is why software like Sonarworks gives an option for mode selection and some people prefer higher latency (and pre-ringing) linear phase filters for some applications. But even a minimum phase, or analog filter will add some phase shifts/latency. So, it's not that placing it in the audio interface will magically reduce this latency. It will be present no matter where it happens. I would make an opposite argument, actually. It's better to place filtering and correction in the speakers. Speakers already do filtering. Unless someone is using a literally driver/speaker single cabinet, all multi-way speakers need to filter to split the signal to different drivers. Many modern speakers do DSP filtering for this purpose (arbitrary precision etc). So adding extra filters to existing filters can make this latency or phase shift even smaller (if it's in the opposite direction). For this reason, I bought Adam A8Hs. It works with Sonarworks, has onboard DSP, and allows to conveniently upload any profiles (or switch them from PC). To me, it's easier than doing this from a soundcard driver UI. And, as an additional benefit - once it's uploaded, I can drive hardware synths and mixers through it, I don't need to launch my PC and route stuff through an audio interface (which WILL add extra latency as compared to going straight from a cheap mixer to my speakers).
Wouldn't a USB audio interface, due to the fact that it receives the audio data in frames (packets, buffers) introduce a much higher delay than any filter? And it's not a problem of USB per se; it's perhaps an inherent problem computers ran by non-real-time OS (i.e. all typical computers). Also audio programs output audio data in chunks. Or are these more advanced audio interfaces that can pass audio from inputs to outputs in real-time, bypassing the computer?
@@khi787ghid I agree almost with all you said, except for a) the linear phase filters. Very high quality very steep/precise ones can add even 50ms latency. Some audio interfaces can have just a few ms latency. b) the nature of phase shift of very low frequencies. Imagine you shift a 50Hz wave by half cycle (180 shift). Full cycle is 20ms, so here you'd introduce a 10ms phase latency to low frequencies. And it's not some abstract example, by high passing kicks in electronic music production you can change their tonal time envelope and even the duration a lot - even when you filter only around barely audible frequencies!
They also added it to the Fireface UCX II. I own both the Fireface UCX II and the RME ADI-2 Pro fs AD/DA converter. The ADI-2 Pro already had these features as well as crossfeed, but I'm still happy that they included it in the higher end audio interfaces as well as I ofter travel with my Fireface UCX and leave my ADI- 2 Pro at home. I also mix and master in headphones primarily so having the eq and crossfeed for headphone correction is also helpful. It replaced my need for my Sonarworks software.
@Acoustic Insider Other manufacturers already had this functionality. Apogee (Symphony MK II only), Lynx (Aurora), AVID's MTRX and MTRX Studio, and others. But it's very cool that RME has added it as well. But of course - most others are very very expensive for home studio use. The Audient Oria is probably the least expensive device that does 'everything'.
Asked for this ages ago when we first got our UFX+ and got finally implemented. It is great - also the cross-feed option for headhpones. Its fair enough to say that they rule for quite some time now 🙂Thank you for putting this video out Jesko!
RME gets more into the world of the DADman Software. Really excited! We just Need an „atmos source“ function now to Stream atmos content over RME interfaces
i have the minidsp SHD studio, i use it for home stereo. think that unit has many function that you speak of. the whole SHD series would work great for studio use. i do not know what a "interface" is but i use mine as a volume control with built in dsp. adding dac's and power amps as needed. or directly into plate amps if it got digital input, otherwise i would need a dac in-between.
I got tired of waiting and went DIY. I use the PEACE GUI for REW to make eq and delay adjustments. Combine that with VB Audio Audio Matrix and I've been very happy with the control and result.
I've been using Voicemeeter for the last... Almost a decade I guess to accomplish per channel EQ, it's very limited with only a few EQ bands per channel but still very handy
Excellent video as always! That is decently where the industry is going. Neumann MT48 offers this and works perfectly with their Neumann studio monitors and well as their headphones. But the Neumann solution locks you into their ecosystem. So I hope Sonarworks is busy negotiating with interface manufacturers. I would love to see a Sonarworks plugin for the UA Apollo and other interfaces!
That's exactly what is frustrating me for couple of years now and why I'm seriously considering RME interface. Individual calibration for every output is essential... I have Sonarworks but it is too aggressive and cause latency so in my opinion only option is to have dsp tied directly into audio interface and tools like Sonarworks can be used as useful guidelines and then just use microphone and your ears... But apart from RME until recently I had very hard time to find any audio interface with built in EQ with enough precision.
As far as I understood, you can import SoundID correction profiles into Audient ORIA, including custom eq as well. So it should be possible to do high and low pass.
Exactly my thoughts. Why don't they do it? It's almost as if the thought hadn't crossed their mind yet, or perhaps the ones who make semi-pro (or smaller) interfaces don't see it as a necessity. They likely think the pros will have enough money to buy a Trinnov system. But yeah, it would be game-changer, even if it's just a manual EQ curve you can apply yourself for correction (as well as delay, phase, etc.)
Using RME about 30 years and have been asking for them to use the DSP for at least delay on the outputs, now they’ve done it, but at the same time Audient have brought out their new immersive interface, but anyway I just need it to integrate with my AFMG SysTune and I’ll be a very happy chap…..
We’ve had these in car audio for years, probably due to the more demanding “room” and speaker placement limitations. Why include it in the master software when they can sell you another piece of gear? 😅
You can get it another way also.No need any software to play with master audio reproduction. Just read the book called #Get Better Sound# by Jim Smith.
Wow, I had no idea this was an issue, I run an evo4 into a digital hifi pre as it has all the calibration I need, never occurred to me that people with a PC and interface don’t get to change delay, balance etc. I check calibration about once a year.
In Studio One you can utilise the Listen Bus which allows you to add speaker correction plugins that go direct to your interface but they don't affect bounce down. Similar thing really. P.S. Dolby Atmos is finished, dead in the water, it's a dead parrot!
You make a good point saying those things should be included in the Audio interface. But, since the speaker is amplified by itself. Couldn't it add some difference between the two speakers if the signals going into their preamp is not the same volume? Wouldn't that affect how the preamp will amplify this signal and then possibly give a slightly different tone to the sound of that speaker? Or is that negligeable because those preamps and usually very clean?
Hi! Long time lurker first time commenter :) how does that compare to what trinov nova is capable of (ignoring the automated process)? Please if you could give a piece of your mind on that comparison.
about the alignment and phase adjustment between the subwoofer and the speakers, can the DSP included in the Neumann KH 80 and the KH 750 sub solve this problem? thank you for your reply. Dennis
I have a correction/clarification (perspective: I am a research scientist working with, among other things, signal processing - so it's my domain, though I work mostly with images and videos):
All filters always introduce extra latency. It can be either the latency of individual frequencies (phase shift) when working with minimum phase and similar filters or the latency of the whole signal (linear filters or mixed phase filters). People have different preferences. I don't want to get into it now, but this is why software like Sonarworks gives an option for mode selection and some people prefer higher latency (and pre-ringing) linear phase filters for some applications. But even a minimum phase, or analog filter will add some phase shifts/latency.
So, it's not that placing it in the audio interface will magically reduce this latency.
It will be present no matter where it happens.
I would make an opposite argument, actually. It's better to place filtering and correction in the speakers.
Speakers already do filtering. Unless someone is using a literally driver/speaker single cabinet, all multi-way speakers need to filter to split the signal to different drivers.
Many modern speakers do DSP filtering for this purpose (arbitrary precision etc). So adding extra filters to existing filters can make this latency or phase shift even smaller (if it's in the opposite direction).
For this reason, I bought Adam A8Hs. It works with Sonarworks, has onboard DSP, and allows to conveniently upload any profiles (or switch them from PC). To me, it's easier than doing this from a soundcard driver UI.
And, as an additional benefit - once it's uploaded, I can drive hardware synths and mixers through it, I don't need to launch my PC and route stuff through an audio interface (which WILL add extra latency as compared to going straight from a cheap mixer to my speakers).
Wouldn't a USB audio interface, due to the fact that it receives the audio data in frames (packets, buffers) introduce a much higher delay than any filter?
And it's not a problem of USB per se; it's perhaps an inherent problem computers ran by non-real-time OS (i.e. all typical computers). Also audio programs output audio data in chunks.
Or are these more advanced audio interfaces that can pass audio from inputs to outputs in real-time, bypassing the computer?
@@khi787ghid I agree almost with all you said, except for a) the linear phase filters. Very high quality very steep/precise ones can add even 50ms latency. Some audio interfaces can have just a few ms latency.
b) the nature of phase shift of very low frequencies. Imagine you shift a 50Hz wave by half cycle (180 shift). Full cycle is 20ms, so here you'd introduce a 10ms phase latency to low frequencies. And it's not some abstract example, by high passing kicks in electronic music production you can change their tonal time envelope and even the duration a lot - even when you filter only around barely audible frequencies!
@@BartWronsk, interesting, thank you! TIL.
Merging Anubis also includes EQ, level and delay per output, bass management, and SoundID integration.
Can you make a tutorial for RME on how to calibrate your speakers, headphones and other systems?
yessss! REW + Measurement mic. Setting volume balance, setting delay for phase, setting eq (not sure about the correct order) Please make this happen!
Should be on RME’s website / channel
They also added it to the Fireface UCX II. I own both the Fireface UCX II and the RME ADI-2 Pro fs AD/DA converter. The ADI-2 Pro already had these features as well as crossfeed, but I'm still happy that they included it in the higher end audio interfaces as well as I ofter travel with my Fireface UCX and leave my ADI- 2 Pro at home. I also mix and master in headphones primarily so having the eq and crossfeed for headphone correction is also helpful. It replaced my need for my Sonarworks software.
@Acoustic Insider Other manufacturers already had this functionality. Apogee (Symphony MK II only), Lynx (Aurora), AVID's MTRX and MTRX Studio, and others. But it's very cool that RME has added it as well. But of course - most others are very very expensive for home studio use. The Audient Oria is probably the least expensive device that does 'everything'.
Yea, not sure why this is a game changer. Looks like he was just unaware that this was an option with other brands.
Asked for this ages ago when we first got our UFX+ and got finally implemented. It is great - also the cross-feed option for headhpones. Its fair enough to say that they rule for quite some time now 🙂Thank you for putting this video out Jesko!
RME gets more into the world of the DADman Software. Really excited! We just Need an „atmos source“ function now to Stream atmos content over RME interfaces
i have the minidsp SHD studio, i use it for home stereo. think that unit has many function that you speak of.
the whole SHD series would work great for studio use.
i do not know what a "interface" is but i use mine as a volume control with built in dsp. adding dac's and power amps as needed. or directly into plate amps if it got digital input, otherwise i would need a dac in-between.
I got tired of waiting and went DIY. I use the PEACE GUI for REW to make eq and delay adjustments. Combine that with VB Audio Audio Matrix and I've been very happy with the control and result.
I've been using Voicemeeter for the last... Almost a decade I guess to accomplish per channel EQ, it's very limited with only a few EQ bands per channel but still very handy
Excellent video as always! That is decently where the industry is going. Neumann MT48 offers this and works perfectly with their Neumann studio monitors and well as their headphones. But the Neumann solution locks you into their ecosystem. So I hope Sonarworks is busy negotiating with interface manufacturers. I would love to see a Sonarworks plugin for the UA Apollo and other interfaces!
That's exactly what is frustrating me for couple of years now and why I'm seriously considering RME interface.
Individual calibration for every output is essential...
I have Sonarworks but it is too aggressive and cause latency so in my opinion only option is to have dsp tied directly into audio interface and tools like Sonarworks can be used as useful guidelines and then just use microphone and your ears...
But apart from RME until recently I had very hard time to find any audio interface with built in EQ with enough precision.
As far as I understood, you can import SoundID correction profiles into Audient ORIA, including custom eq as well. So it should be possible to do high and low pass.
I love that your name means "peaceful."
how appropriate!
DADman also has this incorporated in their software.
Sounds great. I hope they will release these feature on the fireface 802 fs.
Exactly my thoughts. Why don't they do it? It's almost as if the thought hadn't crossed their mind yet, or perhaps the ones who make semi-pro (or smaller) interfaces don't see it as a necessity. They likely think the pros will have enough money to buy a Trinnov system.
But yeah, it would be game-changer, even if it's just a manual EQ curve you can apply yourself for correction (as well as delay, phase, etc.)
You can always get a separate box like minidsp for this
Using RME about 30 years and have been asking for them to use the DSP for at least delay on the outputs, now they’ve done it, but at the same time Audient have brought out their new immersive interface, but anyway I just need it to integrate with my AFMG SysTune and I’ll be a very happy chap…..
My son has one, we were able to measure and EQ the room easily.
Nice product
We’ve had these in car audio for years, probably due to the more demanding “room” and speaker placement limitations. Why include it in the master software when they can sell you another piece of gear? 😅
You can get it another way also.No need any software to play with master audio reproduction. Just read the book called #Get Better Sound# by Jim Smith.
Wow, I had no idea this was an issue, I run an evo4 into a digital hifi pre as it has all the calibration I need, never occurred to me that people with a PC and interface don’t get to change delay, balance etc. I check calibration about once a year.
I have an Evo 16 but I don't know how to calibrate it. How can I do that? Can you explain to me?
Hi there, Antelope has these features in all its new interfaces , even for Atmos integration on the hi end ones.cheers
Nice video, very well explained 👍
In Studio One you can utilise the Listen Bus which allows you to add speaker correction plugins that go direct to your interface but they don't affect bounce down. Similar thing really.
P.S. Dolby Atmos is finished, dead in the water, it's a dead parrot!
Why is Dolby Atmos dead?
Any solutions for us Babyface Pro users??
Audient just released the ORIA that does ATMOS, calibration and DSP in one.
Are there any measurement plugins that analyze within a daw? Just measurements, not corrective.
my studioliive series iii 32 can be eq'd and delayed per channel on output ..
not available for RME UFX mKI users ? :'(
My neumann mt48 has eq built in on the monitor out. But only 4-5 bands :/
excellent video..!Some of the things you said were always in the back of my head and i thought i was crazy or regularly getting faulty equipment!
plant b lookin fresh now boss
dont‘ think gettin a haircut is a longtime solution though… plant needa eat!!
Great news, now the others vendors have to follow.
Genelc had it for years.
What?? I have an RME UFX+ and had ZERO idea... Time to investigate!
You make a good point saying those things should be included in the Audio interface. But, since the speaker is amplified by itself. Couldn't it add some difference between the two speakers if the signals going into their preamp is not the same volume? Wouldn't that affect how the preamp will amplify this signal and then possibly give a slightly different tone to the sound of that speaker? Or is that negligeable because those preamps and usually very clean?
I would not expect any kind of measurable difference in the response of the kind of amplification stages that you'll find in monitoring setups.
@@Bolt_241 It's probably because I'm a guitarist, where it can have a big impact how hot the instrument signal goes into the amp :)
Hi! Long time lurker first time commenter :) how does that compare to what trinov nova is capable of (ignoring the automated process)? Please if you could give a piece of your mind on that comparison.
For UFX+, UFX II, UFX III and UCX II. Rathe expensive.
I’m sure it’s more to facilitate atmos but whatever helps us get a better sound 😉
game changer for RME
about the alignment and phase adjustment between the subwoofer and the speakers, can the DSP included in the Neumann KH 80 and the KH 750 sub solve this problem? thank you for your reply. Dennis