Best Way To Output DAW Audio During Writing, Mixing, Producing, Mastering

JudeWei

JudeWei

New member
Sorry if this sounds like a stupid question, but I've had some bad luck experimenting on my own to find the best playback quality for writing and mixing on my own at my home work station.
I'm sure I'm not the only person to find these days it feels like any search engine listing is only interested in reviewing products rather than education.

That's the reason I joined and so thank you for allowing me into your forum
I search but not turning up anything thats overly helpiing my specific need for info so if anyone with more experience than I have would give me your best advice for what to do gonig forward, would love the most opinions i can get just to see what the general feeling is because I have no clue, its hard to really find people in real life and everything I find just feels like a direct and then othertimes often some secret underhanded marketing campasign in disguise.

I currently have a generic type of setup, brands are not important at my price point but I'll describe as best I can and if you need more info feel free I'll be more specific if neccesary.

AMD Ryzen PC 8cores @ 4.2ghz (Its driving my projects for the most part)
64gB Ram (I dont think any issues here or CPU but I like Omniverse and Kontakt so I push my rersources)
Presonus AudioBox USB Older 48000khz version (Output works half the time since unsupported drivers, can sometimes record no problem)
MouKey Usb3.0 24bit Interface Total Garbage (Generic drivers - it has problems that make the output not an option & recordings is less quality than presonus despite better specs listed).

Both of these I've "scored" by trading local people who didnt want them, for the price I got a good deal but had no power of selection, however, better than not having at all.
I also have a cheap direct USB condensor its not the best, and some other XLR condensors and XLR dynamic mics, doesnt really matter in this case however.

My Motherboard's green stereo output is my current output in use audio output is running so called "Hi Def Audio Device" of the Microsoft Driver variety
This runs with the Stereo 1/8 unbalanced audio line to 4" Studio Monitors Mackie CRX brand One active 1 passive (better than nothing)

Before I was using a Samsung HDMI Home Stereo Reciever and Polk Audio 5.1 Speaker set which was good except not flat in the sense of moitors and would often lead to Stereo Pairing issues if something would fall out of the wirie clips and just all aorund was not a good set up for producing so thats not what I'm using here anymore, plus I neeeded space under my desk.

Mainly because the presonus interface doesnt function properly most of the time on output.

GPU hosts HDMI Nvidea 1.3.38.35 HD Audio Drivers I could but don't use since I got the monitors

So I have a couple hundred CDN and I really wanna do myself a solid for my audio playback situation.
My instinct tellsme to find something that can use the balanced input option on the monitors at the full definition, I'm of the understanding the green output on nmy motherboard is only unbalanced, and my presonus interface is thwe only one that couple have that option but its not working kind of over ready to move on.

Ive looked at a number of things including what I think is probably the best interface in my price range.
Options like Tube Headphone Preamps ....

PCIe Soundcards that seems to be mostly desined for gaming but dop offer comparible specs for what I'd like, however, nothing with the "Balanced" R/L output.Unfortunatyely due to my financial situation Amazon is my only real option for a store front right now. I understand if we dont wanna mention brands and etc, my bad if thats the case sorry.
Other wise, I apologize if this was far to specific, don't eman to lose you in details but I'm just putting all out in hopes for the best advice.
Hope you an help even me!

Thanks for having me!
 
Hey Jude! (sorry, I bet you get that a lot!) welcome to the forum. What you seem to have there is a 'horse and hare stew' (I can explain the metaphor if required) in that you have a stonking computer which must have set you back a decent wedge but are now arsing about with cheap, slightly defective interfaces.

But first let us get a couple of technical 'facts' straight. Whilst it is always desirable to feed monitors from a balanced source it is in no way essential. Moreover, a balanced feed give NO improvement in the quality of the sound...ALL it does is reduce the impact of interfering electrical signals, hum in other words. If the signal from the sound card is free from noise it does not matter that it is not balanced.

Internal Sound Cards: The 'hi def' thing. These are rarely of much use for the business of recording with its associated tasks of over dubbing and they rarely deliver low latency. However, in terms of basic sound quality they are usually, in my experience very good. Their main drawback being that they don't deliver a very strong signal*. but even this is rarely a problem.

I shall suggest to you three interfaces, two of which are I am sure out of your pocket range (but many Cos like Sweetwater do credit)
Top of the line for my money is the MOTU M2 (the M4 which I have is the same but with two extra line inputs) Mic pre amp and general audio and converter quality is close to state of art. Drivers are very stable and return very low latency.

Native Instruments Komplete Audio 6 Mkll: Different beast from the M2 but again audio quality beyond reproach and drivers rock solid with low latency.

Bargain basement is the Behringer UMC 204HD. Still has decent mic pres, you could run a 57 for speech though the interface is perhaps not quite up to an SM7b. Sound quality leaves nothing to be desired.

All these interfaces have balanced outputs on TRS jacks. I have bought all three interfaces and still have the M4 and KA6 (Mk1) the Berry went to son in France and he was very happy with it.

*Most 'budget' monitors seem to have unnecessarily high input sensitivity for use with 'pro' level sources but I guess this is because people DO use them with on board sound cards?


Lastly Jude, if there is anything in the above you don't understand, come back to me and I can walk you through it.


Dave.
 
Dave has given some very sensible advice. Nothing wrecks creativity more than a temperamental computer. In all honesty, the damn things need to work reliably, every time, so you can forget the damn things. The reality on the term quality is that even cheap computers produce excellent quality. What people get worked up about are hums and buzzes, especially those tha5 change when you move a mouse or the screen display changes. The changes between any typical home recording, or indeed studio recording device are tiny with just a few exceptions, and often these are talked about as if they are quality changes, but are really tonal changes. Microphones and speakers ALL sound different. You need to pick ones that are appropriate for the job and your own liking. Interfaces, gadgets and gizmos rarely make much difference and it can be depressing for people to spend a lot of money and not hear anything. In general, you get more or different facilities. I did a show a few weeks back where the clever multi channel interface failed. The audience heard the music through the ten pound or less audio chop via the green socket. There must have been a difference, but for all practical purposes, we didn’t hear it.
 
Heh! Put a £25 Behrry UCA202 in the gigbag Rob. Rock solid and comes out on RCAs at low, 100 Ohm impedance.

Dave.
 
ecc83 said:
Hey Jude! (sorry, I bet you get that a lot!) welcome to the forum. What you seem to have there is a 'horse and hare stew' (I can explain the metaphor if required) in that you have a stonking computer which must have set you back a decent wedge but are now arsing about with cheap, slightly defective interfaces.

But first let us get a couple of technical 'facts' straight. Whilst it is always desirable to feed monitors from a balanced source it is in no way essential. Moreover, a balanced feed give NO improvement in the quality of the sound...ALL it does is reduce the impact of interfering electrical signals, hum in other words. If the signal from the sound card is free from noise it does not matter that it is not balanced.

Internal Sound Cards: The 'hi def' thing. These are rarely of much use for the business of recording with its associated tasks of over dubbing and they rarely deliver low latency. However, in terms of basic sound quality they are usually, in my experience very good. Their main drawback being that they don't deliver a very strong signal*. but even this is rarely a problem.

I shall suggest to you three interfaces, two of which are I am sure out of your pocket range (but many Cos like Sweetwater do credit)
Top of the line for my money is the MOTU M2 (the M4 which I have is the same but with two extra line inputs) Mic pre amp and general audio and converter quality is close to state of art. Drivers are very stable and return very low latency.

Native Instruments Komplete Audio 6 Mkll: Different beast from the M2 but again audio quality beyond reproach and drivers rock solid with low latency.

Bargain basement is the Behringer UMC 204HD. Still has decent mic pres, you could run a 57 for speech though the interface is perhaps not quite up to an SM7b. Sound quality leaves nothing to be desired.

All these interfaces have balanced outputs on TRS jacks. I have bought all three interfaces and still have the M4 and KA6 (Mk1) the Berry went to son in France and he was very happy with it.

*Most 'budget' monitors seem to have unnecessarily high input sensitivity for use with 'pro' level sources but I guess this is because people DO use them with on board sound cards?


Lastly Jude, if there is anything in the above you don't understand, come back to me and I can walk you through it.


Dave.
Click to expand...
I've got long lasting Behr bias, probably irrational, but would sooner develop my own gear from a hacky cicuit-bend type aesthetic, simply in spite ..that said, you hit the nail so thank you so much for mentioning the NI interface i cant believe that name didn't cross my mind, weird. they have on AMZ Can listing so I'm reading through the KA2 model product page and its looking good for delivery tomorrow by 5pm. You recommended Mk2 which is a bit out of my price range but between the two they're both 192kHz 24 bit conversion mk2 = 4io a + 2io d and ka2 is just 2ioa .
My main concern is the brand of ASIO's reputation (reliability etc), and the interface's 2-channel mono outputs for the NI KA2 . Do you have anything to share, with functionality with specifically the audio output? Im seeking a product that I can connect from the outputs of my interface, signaled directly into the speaker-monitors, at least for some meantime waiting while I daydream window shop for my next speaker-monitor & interface setup, but I wont be able to afford that for some time, probably until someone purchases my kidneys on the dark-net. But jjust to be very specific, I want an interface with stereo output that is capable of being used for stereo output even outwise production, for example when I'm doing homework and listening to music or working on other stuff. I think this is, but just keeping it 100% to be specific on my use case being supported.
In case I sound trivial and confused I am, but in my defense I often feel like sales language is misleading or just not direct and confrontational enough to leave me confident. One specific thing I feel like it might be saying the KA2 is only direct monitoring from the inputs to output, and will not directly output from my computer otherwise. But truly I'm just confusing myself I'm sure. Furthermore, even if the output is from interface, when the inputs are not in use, can the monitoribng be turned off to not mionitor microsphobnes, or channels directly from the line of recording, say if I'm on webcam will te oter person be patched through feedback lol, Ive had these types of weird issues and I dont know how to determine this.... But maybe you probably know exactly what I mean Im hoping anyways.... Thanks very much

I have some other similar related questions all towards places im stuck in my development as a producer and artist mainly with hardware and software. should i start a new a thread for different questions of fire off more here after I figure the interface
 
The KA2 looks a good choice. It lacks the 3/4 line inputs and S/PDIF i/o of the KA6 and MIDI which is a shame but I have no reason to believe the the sound quality, converter quality and drivers are any less brilliant than the KA6.

It has a continuous pot to go from computer (DAW) sound to input sound whereas many interfaces of comparable cost make you do with a switch.

The outputs are balanced (probably 'impedance' balanced but no matter) and, although I don't quite understand some of your issues Jade, I would suggest you treat the KA2 as your ONLY playback device, even disabling OB sound in Control Panel/Devices.

Now, forgive me J' but you come across as a bit of a 'dabbler'? When you get the NI box I implore you to read TFM and follow the driver and setup instructions TO THE LETTER. Do NOT try to 'wing it'!

Dave.
 
I've got a little confused here. Your vocabulary and descriptions suggest a more advanced understanding than a beginner, but there's a suggestion of a little less understanding in some areas. I'd like to think that after all my years doing this lark, that the more outrageous claims some products use to support them get seen through, but in other areas, I make little effort to pay attention. You mention ASIO reputation? Steinberg's design has in all the years I have used it been totally transparent and not even worthy of comment. It works and has low enough latency I simply don't notice it. It's widespread enough to just work? ASIO4all is often spoken about, mainly because it does seem to work for people with systems that for whatever reason, can't keep up with the real thing. That's a bit misleading, I guess, because when you simply cannot get an interface device to work with your computer, ASIO4all often can - but it's latency can be pretty daunting to work with.

However - I've got a number of interfaces from simple to complex, via USB and Firewire with two or more inputs and outputs. My computer and Cubase just work with any of them. No tweaking buffers or parameters. It's sort of a non-event. I'm not certain if I even understand what you mean about two mono vs stereo? Most interfaces have individual outputs - so have I misunderstood? The interfaces usually have the mix control to blend the input stream with the computer output stream, but mine always sit turned 100% to the computer and I use the channel strips to turn on the monitor of the input if I need to - I have no need at all for the mix facility. If I had lots of latency I guess I'd use them, but I don't and haven't had for at least ten years now. You mentioned you MUST have a stereo output. Every interface has this? It might be on two separate sockets (and that's been the case as long as recording in stereo has been done) or it could be via an unbalanced 3 circuit headphone type jack. Makes no difference apart from the cables.

All the interfaces I have work pretty much the same - the differences are in two areas. The number of ins and outs and the performance of the preamps. If you record quiet things at a distance then preamp performance is important, and the specs are always pretty accurate on this. With all my interfaces, if I want to do clever routing, the restrictions come from the software not the interface. Almost universally, what they have appears in the software - so it's a software patch to connect input 1 to output 2, but send output 3 to a streaming device and so on. restrictions usually come from the other devices. A webcam, for example might be built and designed to use channels 1 and 2 only - because that is what every computer has. You might then need to use an interface patched differently to enable the webcam outputs and inputs to appear elsewhere. You won't find this info the manufacturers sites - they have no clue what you are pairing the kit with. They will simply tell you it has 8 in or 2 in or whatever because it has them. Some software has patching limitations of it's design - All I can say is that conflicts are rare. I use Cubase and also like Sound Forge - Soundforge occasionally tries to seek private use of the ins and outs and removes them from cubase grrrrrrr. Other times, it doesn't. The cure is simply closing cubase and restarting it and it grabs back the 'private' inputs. Neither Cubase or Sound Forge are at fault, nor is it a problem with ASIO - they just annoy each other. For me the work around is to start Sound Forge first - then it always works. Of course I usually forget.

You also need to determine your own definition for quality - mine sits at 48K 24 or 32 bit. Everything I do is at this setting. I have tried 96 and 192KHz sampling and on my old ears, and visually on the screens, I can detect no useful content above 20K, probably 16K really. I therefore reject the higher sampling rates as a waste of drive space.

If a new topic emerges from an old one here, but often leaps back and forth - carry on comments here. If the new question is totally new with no links, start a new one. That seems to work generally. I'm a bit intrigued by what you are doing and why you are asking some questions? I get the feeling that you've not quite asked what you wanted yet?

two mono vs one stereo has still left me a bit stuck as they're the same, bar the connector, so what is concerning you about this aspect?

I smiled at Dave's advice. I'm the worst case. I never read manuals. They are my absolute last resort. I think a product is badly designed if you need to read the manual for anything basic. Fair enough use them for a teeny feature buried away, but I learn by fiddling and winging it comes second nature. However, I accept I'm far from normal.
 
Shame on you Robert! Ha, my reading of manuals stems from my time as a service tech and one of my ever increasing tasks was demonstrating VCRs to customers.
Now, back in that day, owner's manuals were generally very good and since I was regularly hit with a new, to me, brand/model of VCR I HAD to read the manual so as not to look a total tit in front of the customer! Also, when customers phoned up with a problem the default advice was "had you looked that up in the book?" No, was the usual reply. I got so for any given problem I could quote them the page in the manual that dealt with it.

I don't mind you Rob or friend Jade floundering about, your life to waste! I just say AT LEAST get the drivers and hardware setup as the mnfctrs tell you in the first instance, otherwise all kind of annoying **it can ensue. For instance, for some AIs you have to install software with the AI disconnected. Others require one or more cons/diss'es during the procedure.

Mono/stereo? Did not take that much notice. AIs are at least two channel devices. "Stereo" is a way of recording and reproducing sound. Something many people get confused about.

Dave.
 
I'm what is educationally called a kinaesthetic learner - touchie-feely. Looks like we both did the same thing - Phillips 1500/1700 and the first VHS and beta, and alongside that U-matic to the commercial customers.

In Jade's case, she seems to want to do something quite specific but the different little bits of knowledge picked up are getting in the way rather than helping the flow. A reliable computer, with the drivers the manufacturers provide. Then, it sort of works. Normally, I have to say nowadays, first time with no setting up to do. The Tascam here has a control panel to adjust all sorts of things. I've never needed it - it just works.
 
"I'm what is educationally called a kinaesthetic learner - touchie-feely. Looks like we both did the same thing - Phillips 1500/1700 and the first VHS and beta, and alongside that U-matic to the commercial customers."

No, not at all the same thing Rob. My customers were Joe Public and I was handling Ferguson, Hitachi, Sony, Philips, Panasonic,Toshiba and a few other brands I have forgotten. That's VHS, Beta and phils 2000 system. Then each brand had a bog s model, better and best AND the fekkers changed every year! I would often borrow a top line model when it came out for a weekend.

As I said, you two can bugger and arse about with kit to your heart's I just say FFS get the initial installation done to the book!

And yes Rob I think we both suspect that this thread is the very essence of the phase "a little learning is a dangerous thing"!

Oh, and don't try "touchey-feeley" learning inside valve amps .

Dave.
 
There are plenty of good answers here Jude.
I dropped in before everyone got here, and was confused why you are outputing from the green socket, when you have a Presonus interface with stereo outputs.
Just read again that it doesn't always work, because of drivers.
You do not say what operating system version you are using. I might help to tell us all.
I had a similar problem with a Tascam interface. They stopped producing drivers for it after Windows XP.
So I bought a new interface from Focusrite, along with a new PC.
There's always some component turning its nose up at some other component.
I suggest you sell your grandmother, and buy a new interface, even if it is a modest one.
Oh, and Welcome!
 
rob aylestone said:
You mention ASIO reputation? Steinberg's design has in all the years I have used it been totally transparent and not even worthy of comment.
Click to expand...

There are definitely differences in driver quality. Take my recent experience with a Tascam Model 16 mixer/recorder/interface. I'm used to the computer being in control and having the ability to switch sample rates on the fly from the computer. The Tascam doesn't do this. You need to set the sample rate on the mixer separately from the sample rate on the computer and, if you do this in the wrong order, you are met with silence. Every other interface will switch sample rate automatically unless an external digital source is used.

My other recent experience is comparing the Zoom U-44 drivers and the RME Digiface/USB drivers. On my setup I use the Zoom drivers at high buffer sizes because I get the very occasional glitch if I go below about 1024 samples. The RME drivers seem to work perfectly well at 64 samples on the same machine. What is more, the RME has 32 inputs and 34 outputs whereas the Zoom only has 4 ins and 4 outs so the USB bus is much more heavily loaded when the RME is used.

Don't get me wrong, the Zoom is a great interface and well worth consideration if the OP is looking at the KA6 but the drivers aren't quite as good as RME's.
 
Hmm. Are we saying that drivers are doing sample rate conversion, and doing it badly? Surely the point of a driver is simply to get a data stream into a computer untouched? I happily accept that there could be a few badly written ones that are doing some rounding, or dropping data and this gets ‘fixed’ by error correction, but that is not their job is it? The actual output of a device should remain an absolute copy of the original. Never have I had your problems. If a driver is so poorly written that it doesn’t have the ability to autosense the received stream and make the necessary adjustments in the computer so it is correctly interpreted it is a very poorly written piece of software. Actually, I have had this, thinking back. Just once when I was using HHB portadiscs. Minidisc portables that had usb out, and not every computer understood them. Occasionally those that did misinterpreted the data as 32KHz sample rate, which they weren’t, and they made a shocking noise. Unplugging and re plugging usually fixed it. That’s the only driver issue that I have had that impacted quality. The rest were a go/no go combination, and I have had many of these. Not once have any of these changed audio quality. Maybe I have been lucky. Even the popular asio4all has always sounded absolutely fine. I have no reason to believe it changed audio quality, it just took time to process it. I’m guessing you are just talking about the drivers having communication issues, and not about audio quality, and this I can’t dispute. Some are just poorly written, and for me have never half worked, needing tweaks in buffer sizes.
 
I had a similar experience with digital cameras. They claimed high resolution pictures, but it turned out that they interpolated the raw
data, to artificially make the picture appear high res. You can't put data where it isn't.
 
ASIO drivers shouldn't be doing any sample rate conversion. All the data processing to or from analog would be done in the interface hardware itself. ASIO is just the protocol for transmitting multichannel data as opposed to normal Windows audio. It bypasses the normal data route that sound would take through Windows, such as Directsound or the sound mixers that are used with things like Realtec sound chips. Now, it's entirely possible that a driver is so poorly written such that it can't process the data quickly enough without a large buffer but that doesn't seem to be the normal situation.
 
TalismanRich said:
ASIO drivers shouldn't be doing any sample rate conversion. All the data processing to or from analog would be done in the interface hardware itself. ASIO is just the protocol for transmitting multichannel data as opposed to normal Windows audio. It bypasses the normal data route that sound would take through Windows, such as Directsound or the sound mixers that are used with things like Realtec sound chips. Now, it's entirely possible that a driver is so poorly written such that it can't process the data quickly enough without a large buffer but that doesn't seem to be the normal situation.
Click to expand...
This> https://www.soundonsound.com/forum/viewtopic.php?t=24765&start=60
Might give some insights to how AIs and their drivers differ in the ACTUAL latency they deliver. Yes the thread is an old one and the work does not seem to have been continued. The test guy had a lot of flack from AI manfctrs that did not like the exposure of hidden buffers etc. Maybe a contract was involved?!

Dave.
 
rob aylestone said:
Hmm. Are we saying that drivers are doing sample rate conversion, and doing it badly?
Click to expand...

No - nothing to do with any processing the driver may be doing
rob aylestone said:
If a driver is so poorly written that it doesn’t have the ability to autosense the received stream and make the necessary adjustments in the computer so it is correctly interpreted it is a very poorly written piece of software.
Click to expand...

Yes, that is pretty much the issue with the Tascam devices. Neither end has any control over the other end - so you have to set up everything correctly manually on both the computer and on the mixer.
 
This is great news to see so many people reply to what I was worried might just get burried among the unseen.

I got my interface and the install (by the instructions) was very easy actually. I havent used it long enough to meaningfully whore it out just yet its covering everything Ive wanted in this upgrade so far. I will say the ASIO control panel is fairly basic aesthetically speaking but thats par for the ASIO course in my expeirience.

I'm going through every response still but picking up on the keyword here and also having read around, I always find a lot of people seem to be at war with the devil they call latency. Part of me is curious why they have trouble to the degree that this bothers them but perhaps it is me not understanding why they have this issue, and I don't understand what they're referring to.

Correct me if I'm wrong but I find latency to be described by and hated for following reasons:

Most basically latency is lag in response time processing the sound through recording hardware to convert from literal analog waves of sonic pressure to digital data that can be written as binary data onto file as a record that represents the audio of the recording.

What I often assume they are troubled by or what it sounds like is an overloaded PC that freezes and causes mismatched data written into the file.
Perhaps there are more nuanced capabilities with the specific value scale of the AD-converter and wordclock in the interface.
I've heard the best home studio would buy these two units separately in sort of what looks like a rack or desk mounted form factor normally.
Is it just that some of these lesser quality cnonvertetr systems are not doing a good job of synchronizing the actual reallife atomic clock timeline to the momentary times in the reocrding timeline that the recorded samples are pinned onto? In a way thats irrelevant to whether the CPU resources are getting crushed or just fine.

Even at the lower end of the power scale, new generation high-count multicore and high-freq cpu's & enhanced mobo pairings offer the required capability. If it was borderline (i doubt it) at worst they would need to be disciplined and monitor playback tracking audio from second device for a track to play along to if needed. maybe this is inconvenient but heck what is a hobby for.

But this is maybe why Im thinking I dont know what they mean.
Besides reading everyone's take, and potentially responding here again, I'm also gonna be asking some other unrelated questions, maybe on different categories where theyre more applicable.

Thanks again
 
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