Jim Roseberry
-
Posts
1,178 -
Joined
-
Last visited
-
Days Won
9
Posts posted by Jim Roseberry
-
-
There are other builders who assemble custom laptops.
Some specialize doing so for DAW applications.
-
Not at all practical. The niche market would be so minute... there'd be no profit in the man-hours to create/produce/maintain it. Same goes for Linux based DAWs.
That OS would have to support 3rd-party plugins... or you'd be taking a 30-year step backward in time.
One benefit of Windows is the massive economy of scale. Remember back in the early days of PCs... where an IBM machine was literally $10k. Adjusted for today's economy, the cost would be almost $29k.
If you want the equivalent of a dedicated hardware recorder, Tascam makes a 24-track recording box. It's a completely "closed" system.
FWIW, The machines I'm running are in no way unstable, unpredictable, or poor-performers.
One can build a fast, robust Windows PC DAW. That has been my career for the past 30+ years.
-
6
-
-
23 hours ago, jesse g said:
Warm Audio fan here I love their WA 73-EQ, I have two of them and their WA Bus comp, however, no mics yet.
I've got a pair of WA273-EQ units.
Had to replace one of them (had a noise issue)... but I've not had a problem since.
They're not quite as nice as a real 1073, but they're certainly in that territory (fat/thick without sounding muddy/tubby).
-
1
-
-
I was just going to chime-in... 😉
Here's my response from the other thread... and I've added a bit more.
The 14900k is a fine performer.
You need large top-tier AIO cooling (forget air coolers).
I'd recommend top-tier AIO cooling for all three of the CPUs mentioned below.
The Core Ultra 9 285k is slightly faster... and at lower TDP (heat).
Using Cinebench R23 Multi-Core test for comparison:
- 14900k scores ~38712
- Core Ultra 9 285k scores ~42620
- 9950x scores ~42871
The AMD 9950x slightly outperforms the Core Ultra 9 285k.
14900k is a bit over $400
9950x is a bit over $500
Core Ultra 9 285k is ~$600
The Core Ultra 9 has a couple of advantages:
- Lower TDP (lower temps)
- Some reasonably priced Z890 motherboards come stock with a single Thunderbolt port. That's enough for most users... and it's considerably less expensive than a high-end board with TB (typically $600-$1000)... or having to use an AIC.
If you don't need the power of the Core Ultra 9 CPU, the Core Ultra 7 265k is about the same price as the 14700k.
-
2
-
1
-
I've got the 1273 and the 369 Compressor.
Both are very good for the cost.
-
1
-
-
The 14900k is a fine performer.
You need large top-tier AIO cooling (forget air coolers).
The Core Ultra 9 285k is slightly faster... and at lower TDP (heat).
Using Cinebench R23 Multi-Core test for comparison:
- 14900k scores ~38712
- Core Ultra 9 285k scores ~42620
- 9950x scores ~42871
The AMD 9950x slightly outperforms the Core Ultra 9 285k.
-
That sounds like USB related noise (from a poor quality USB cable).
Another potential source of noise... is a WiFi Router.
If you've ever played live using wireless mic or wireless in-ear-monitors, you've almost surely encountered Router noise.
It's very rhythmic/consistent.
I've had unshielded guitars (even from the likes of Gibson Custom) that picked up Router noise (my Router is close to my studio desk).
I now have a metal shield in front of my Router (half-baked Faraday Cage). It cuts down greatly on devices picking up the noise.
-
Ground related noise is super common in a home studio.
To minimize the potential for ground related issues:
- Start by powering the entire studio from a single outlet.
- Use gear that has balanced outputs/inputs... and always use balanced cables.
To trouble-shoot a ground related issue:
- Disconnect everything from the audio interface (just connected to the PC via USB)
- Connect a pair of headphones to the audio interface
- Turn up the headphone amplifier (be careful with your hearing)... and see if the ground noise is gone
If the ground noise is gone (most likely), the ground issue was with one of the devices that were originally connected to the audio interface.
If the noise is still present, it's almost surely a problem with the USB cable that connects the audio interface (swap out the USB cable).
In over 30 years of building DAWs professionally, the PC itself has literally *never* been the source of ground noise issues.
-
18 hours ago, Mr No Name said:
If I was to use a saffire pro 40 and send an output to a scarlett 2i4 connected to a laptop via usb, would this negate the need for firewire and saffire drivers? can the saffire pro 40 be used passively?
You can technically daisy-chain Firewire peripherals.
If you're using a PCIe Firewire controller, most have three Firewire outputs (often one Firewire-400 and two Firewire-800). You can adapt Firewire-800 to Firewire-400 peripherals.
Regarding Thunderbolt to Firewire adapter: Apple makes a Thunderbolt-2 to Firewire adapter. You'd need a Thunderbolt-3>Thunderbolt-2 adapter... and connect the Thunderbolt-2>Firewire adapter to that. The issue you're going to bump into with Thunderbolt-4 controllers; there's been a recent firmware update that breaks compatibility with Thunderbolt-2 peripherals.
If you just want to use the Pro 40 as an A/D converter (Analog to Lightpipe digital output), you could route the Pro 40's Lightpipe output (8 channels at 44.1k/48k) to a USB audio interface (assuming it has Lightpipe inputs).
-
1
-
1
-
-
Those cases are nice... but can be limiting with today's CPUs needing larger cooling solutions.
-
I've gotten more into outboard/hardware... and that's dramatically curbed my interest in new plugins.
Hasn't saved me any money... but that's another topic. 🤣
-
1
-
1
-
-
You have a ground-loop issue.
The laptop audio outputs are not balanced (meaning they're prone to hum/noise).
The easy solution is to run the laptop's audio output to a stereo DI box... and use its ground-lift switch to (safely) lift the ground.
A better long-term solution is to use gear with balanced inputs/outputs (using balanced cables to connect).
To reduce the odds of ground-loops:
- Power the entire studio via a single outlet
- Use balanced cables (TRS or XLR) for all connections
-
1
-
You need to disable CPU Core-Parking all together. If it's enabled and you're running at low-latency, it will cause audio glitches.
In my experience, the new Thread Director in Win11 (12th Gen CPUs and up) doesn't make a huge difference in performance.
I wouldn't migrate to Win11 expecting a major performance increase (or decrease for that matter).
If a machine is having DPC Latency issues, ASIO (Native or current) isn't going to resolve the issue.
Windows 11 is a fine DAW platform... but being a general-purpose OS, there's a fair bit that needs tweaked.
Native ASIO isn't a bad thing... but it alone isn't isn't going to change (negate) the need to tweak the OS.
-
1
-
-
1 hour ago, Starship Krupa said:
Did you read the article? Pete lays out very clearly why they are doing it.
The two biggies are first, that as we all know well if we've been trying to help people on the forum, not everyone is as savvy and squared away as we are about downloading ASIO drivers.
Take the example of someone completely new to computer recording. People have been working for many years with USB devices that just work when they plug them into their computer. So what is going to alert the brand new user that they should download and install this other custom driver? Paper manual? Good luck getting anyone to read one these days.
And anyone who's inclined to say "well, that's the price of admission, we all have to pay our dues," well, nope. What computer peripheral these days, other than an audio interface, requires jumping through all of those hoops to get working optimally?
ASIO can't perform as well as Core Audio if ASIO never gets installed. What was it that Byron Dickens used to say around here whenever someone was having trouble getting their audio to work? Something to the effect that 99% of those issues would be solved if the person would just use an interface with a good ASIO driver. I didn't and don't entirely agree with him on that, but he's not the only one who believes it.
Second, and this is also huge, Microsoft want to make system audio compatible with ASIO, as in, system sounds will still get through, and according to the article, this isn't practical if they're at the mercy of the interface manufacturers' driver. My system is a multi-use computer. I watch movies on it (and often wish to take dialog samples from them), listen to music recreationally, etc.
I'm picky enough about audio that I want to use my Focusrite interface to play this stuff back.
I tolerate a certain amount of the interface getting confused about sample rates when I start up the DAW while Apple TV is running, because I know the rain dance it takes to get it to settle down, and I know that Windows is at the mercy of Focusrite's driver.
If they do their own ASIO driver, they can give that driver the ability to communicate to the OS and tell it not to switch the interface's sample rate, to prioritize traffic from certain programs, etc. As it is now, people who use ASIO do it under the assumption that ASIO will take control of all audio, and anything other than one program at a time using the driver is far from guaranteed.
I've seen people struggle plenty with driver issues, and "YouTube doesn't work when I have the DAW running." I also know that for every person who takes the trouble to ask about it on an internet forum, there are probably at least 5 who gave up on it. And that's not good for pro audio on Windows.
Also, unlike Apple's Core Audio, ASIO as it is now can't aggregate devices, and that's a disadvantage. He mentioned that they're going to see if they can get device aggregation to work, too.
As for laptop users being able to use ASIO drivers, well, they can as long as the CPU in their laptop is an amd64 CPU. Correct me if I got the wrong impression, but I gathered from the article that it's currently uncommon for interface manufacturers to supply an arm64 ASIO driver.
BTW, is nobody intrigued by the other part of the news, Windows' USB getting optimized for audio? I think that is GREAT news. The reason I still use Firewire is that I've never taken to the idea of using an asynchronous bus that was designed to connect printers, keyboards, and mice. I know that it's time tested and zillions of happy users rely on their USB 2.0 audio interfaces (I even have a small one for portable use), but I don't think Jim's main interface is USB either. Quantum Thunderbolt, if memory serves?
A recompile followed by a LOT of testing.
Read the article... and it basically stated exactly what I thought it would.
- Class-compliant ASIO driver
- Arm CPU support
I'm coming from the perspective of looking for ultimate performance.
I'm using an Antelope Orion Studio Synergy Core (main DAW) and a Presonus Quantum (secondary machine)... which are both Thunderbolt audio interfaces.
I chose those specifically because they're pushing the limits of ultra low-latency performance (sub 1ms RTL).
The Orion Studio Synergy Core has great fidelity, onboard DSP, flexible routing, etc. IMO, All the best features rolled into a single audio interface.
I don't think native ASIO is a bad thing. I'm certainly not against it.
For me, it isn't bringing anything new/improved to the table. Same with ARM CPUs...
When it comes to CPUs, I want the fastest "workstation" CPU I can get.
Don't care if it's Intel, AMD, etc... (whatever offers highest performance at a given point in time).
Right now, I'm running AMD's 9950x.
Certainly Laptops have their purpose... and I see why some folks are interested in ARM CPU development (lower TDP, lower power demand, etc).
Power-management and performance-management typically don't equate to top-performance. Thus, my lack of enthusiasm.
-
2
-
FWIW, I don't get the excitement.
- Windows 11 x64 is already a great DAW platform.
- Laptop users can currently run ASIO drivers.
The only "advantage" of native ASIO is that the motherboard's onboard audio interface can be used via ASIO.
Most DAW users are running a dedicated audio interface... for a multitude of reasons (higher quality A/D D/A, lower noise-floor, balanced I/O, etc).
Some current audio interfaces can run sub 1ms total round-trip latency. Native ASIO isn't going to improve upon this.
Apple has had "Core Audio" for years (akin to Native ASIO).
Core Audio doesn't outperform current ASIO drivers on a PC.
-
2
-
Finding the right pointing device can really help those with RSI issues.
Many folks love a trackball. For whatever reason, they really inflame my Tendonitis.
Right now, I'm using a Logitech G703 Lightspeed.
Buttons and wheel don't have a lot of resistance... and that seems to keep things comfortable (for my situation).
Have never tried the Lift.
-
1
-
-
15 hours ago, J.War said:
If i understand correctly for a REAMP to be aligned with it's original PROCESSED take, you have to compensate for the reamp loop added latency AND for the tiny processing latency we just saw above (8-9 samples). Do you still agree with me ?
Yes, If there's any processing delay from the DI signal, that too would need to be compensated.
15 hours ago, J.War said:This is a zoom of the 2 reamps from the same DI, if i am not blind i think they are perfectly aligned.
But if you look carefuly at those 2 " peaks " at the left of the 2 grey markers, you can see there are slight differences in the waveforms. I think that explains why the phase cancelation is incomplete.
What do you think ?
That's exactly why I wanted you to record two separate passes of essentially the same reamp.
I was thinking they'd likely not be 100% identical (thus won't 100% phase-cancel).
The waveforms are aligned... and they're (obviously) pretty similar.
-
If I understand correctly, you're having issues with the reamped (originally DI) signal... and the original Kemper processed signals not aligning.
As a quick test, if you do two separate passes reamping the DI signal thru the Kemper, do those two tracks align perfectly (phase cancel)?
I would also measure the time offset between the Kemper's DI output and it's processed output.
Now that you know that the audio interface itself is recording properly time-aligned, what happens when you do a short test recording (both DI and Kemper processed)... then try reamping that DI signal thru the Kemper?
If the Kemper's DI is slightly latent, you just need to figure out the amount.
If that's the case, for the future... use a quality DI box prior to the Kemper. An all analog path would be zero latency.
If the DI is post A/D, it's definitely going to have some amount of latency.
-
First, you need to make sure your audio interface is perfectly aligning audio tracks (when recording).
Most ASIO drivers don't report the actual latency accurately... causing a "record offset".
Measure the record offset:
- Take a really short spike type signal (snare hit, impulse-response, or similar)... and re-record that to a second track via analog output to analog input (physically patch it).
- Now, zoom way in... and you're most likely going to see the tracks aren't perfectly aligned.
- Measure the time difference between the two spikes (measure this in samples. This is the actual record offset.
- In Cakewalk/Sonar: Preferences>Audio>Sync And Caching>Record Latency Adjustment (samples), enter the amount of of the record offset.
Verify the record offset is working:
- Record the original spike onto a third track (same procedure as above).
- Zoom way in... and you should see the first and third tracks (spikes) line up perfectly.
NOTE: You should do this for any audio interface (regardless of cost/quality). You only need do it once.
Once you've verified all tracks are now properly record aligned, recreate your Kemper External Insert effect (measure the latency).
Kemper re-amps should them line up precisely.
-
1
-
1
-
IIRC, Kemper has ~2-4ms of latency (input to output).
There's no way for any DAW software to automatically compensate.
However, if you take the time to configure an External Insert for the Kemper (within Cakewalk/Sonar), Cakewalk/Sonar can measure the latency and compensate for it. Note this only works if configured as an External Insert.
Otherwise, you have to manually align to the original track. This is true with any DAW application (not just Cakewalk/Sonar).
-
1
-
-
Core Ultra 9 285K (not currently in-stock)
- 8 Performance cores
- 16 Efficient cores
- 24 Processing Threads
- 5.7GHz Max Turbo
Core Ultra 7 265K
- 8 Performance cores
- 12 Efficient cores
- 20 Processing Threads
- 5.5GHz Max Turbo
Core Ultra 5 245K
- 6 Performance cores
- 8 Efficient cores
- 14 Processing Threads
- 5.2GHz Max Turbo
Performance gain vs 14th Gen is ~13%... but TDP is lower.
AMD's 9950x is ~4% faster than the Core Ultra 9 285K.
Some mid-tier Z890 motherboards come with useful additional features.
- Four M.2 slots
- Integrated Thunderbolt-4 port
Nothing mind-blowing... but definitely a step in the right direction.
With removal of Hyper-Threading, I was wondering if performance would decline (vs 14th Gen).
Glad to say that's not the case.
-
1
-
I've got a pair of WA273-EQ preamps (four channels).
One of the two developed an intermittent noise issue (early on... so Sweetwater swapped that out).
Haven't had a single issue since.
For the cost, they're hard to beat.
-
Heard about it yesterday. Insane world we live in.
One local guitar player friend plays the mini Friedman JEL head.
-
1
-
-
2 hours ago, Keni said:
Thanks Jim...
Is it relatively benign?
This is a software version of Codemeter. I'm guessing similar to iLok on the desktop?
Hi Keni,
I never had any issue with Codemeter.
MAJOR IMPROVEMENT
in Computer Systems
Posted
You're assuming all DAW users would embrace this concept.
Many folks are running Windows based DAWs just fine.
Many of my clients are using PC DAWs professionally to make records and compose music for TV/Film.
They're productive each and every working day.
Why would these folks willingly take a 30-year step back in time, going to a closed OS that lacks 3rd-party plugin support, etc... and be limited to what is essentially a closed-end hardware recorder?
Look at the Linux DAW market. It's miniscule... and (also) a solution to a problem that doesn't exist.
There's zero profit in Linux DAW development. That's an economic reality any fringe OS developer faces.