You guys might want to look at the New Intel Nuc i7 system
Here's some info and a review.
http://www.pro-tools-expert.com/vide...eid=28a2b0eb42
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You guys might want to look at the New Intel Nuc i7 system
Here's some info and a review.
http://www.pro-tools-expert.com/vide...eid=28a2b0eb42
KB you are ALIVE mate ! Happy new year buddy hope all is good !
I was about to send out for the KB search team !
I have to laugh at someone thinking that 2 x 2TB 'should be enough for even the most demanding music or post production'. I have several pro friends (generally making games music) with at least 6 x 2TB drives, and they still need more.
that's it Simon, it's a bit like DB's hated axe products encyclopedia comes on a 20TB hard drive as that's the size of the document haha
Only one screen is going to be too limiting for a lot of people. Not all, but when you're used to multi screens, it's hard to go back.
While I haven't used that particular model, an issue with the NUC series is they use laptop parts which are cut back to reduce the heat generated. You're not going to get the same grunt out of them as a regular desktop and even small form-factor desktops are a lot more flexible in configuration.
That particular NUC isn't cheap either - around £600 here in the UK (A$1000ish) - and you still need to fill it with RAM and add hard drives etc. My music PC has 3 drives, one for O/S and programs, one for projects and one for samples. I'd prefer to keep that configuration. I'd also like to see a couple more USB ports without having to use a hub. So many keyboards have USB connections these days, plus I already need two USB ports just for dongles. Although it has two internal USB ports - with that small size case I doubt you'd be able to fit a dongle inside the case. Also, with no Thunderbolt 3 to Firewire adaptors available, it would be new audio interface time and so even more cost, especially since my MOTU unit is still working perfectly.
Plus I've just remembered my two UAD PCIe Quad cards. Around £1400/A$2300 to replace with an external unit. No, I'll stick with a standard form factor PC.
Hey Woks how are you good sir. I hope you had a brilliant break and the new year is starting off brillantly . Guys have you seen the size of the Pro Tools project its massive and each track has plugins. It's super fast the latest i7 6th Gen mobile CPU and it's really quick 2.6 -3.5 ghz The memory is the latest Ddr4 and its fast too. I can make 3 of these for the price of one New Mac desktops and they perform faster than the macs too. It comes with the CPU you need to add memory and hardrives. The cost of base unit is $600 dollars for the average user 2 x 500 gb SSD is more than enough,and you can run ext hdd if you want too. Plus has Thunderbolt 3 40gbs which is blisteringly fast and USB 3. This isn't the old Intel Nucs but a completely new beast. Plus if you already have hardrives big bonus. I'm going to build one of these in the near future I'll let you know how it goes. I'll keep you posted. Maddog most all the new macs use notebook memory even the iMacs I have use notebook memory too. Strange I always technically think Apple are very behind on new hardware.
Thanks Guys
Kimball
My desktop DAW PC has been a work in progress ever since I first bought the hardware that's in it, I have just done the latest update of Windows 10 from Threshold 2 to the new Creator's Update on both my desktop PC and my Dell Inspiron 15 Laptop, so far it's been pretty good, only had one hiccup on the Laptop.
Funny thing is, I thought the new Creator's Update wasn't going to be available till later on this year, but I got a notification in Windows Update saying I could get it sooner rather than later by clicking on the displayed link, so I did and it took me to the Microsoft webpage where I could download the latest Windows Update Assistant, then I just ran it as administrator, the good thing is that it left all my files alone where they were and I didn't need to do any backing-up.
I think I may have figured out why my PC has been a bit temperamental lately, it's looking like the original 1Terabyte System HDD I had in it was in the process of failing, and it finally dropped it's bundle yesterday afternoon, been working on it ever since and I've just finished sorting it out.
last week i got hold of a 2nd hand Dell optiplex 980 for next to nuthin. its 4-5 yrs old but seems ok
now i can use the tascam 2x2 interface without lag or the general inferior pc pains
so far so good, trying out free software - presonus studio one prime, overloud TH2 and reaper...much learning i need.
great to toy around with late at night when real amp volume is of an issue at home.
wish i could spend more time with it but this earning a living (job) thing gets in the way...necessary evil for the fun tokens i guess
For $60.00 USD you can't go wrong with Reaper, it has a multitude of pretty good VST & JS plugins as standard more than enough to get you through almost any situation, these plugins are also pretty resource friendly.
Workflow varies from DAW to DAW but they all do the same thing so my recommendation would be to spend a bit of time learning one and stick to it.
Some time in the next day or two I'll put up something I have done bottom to top in Reaper, I've got about a dozen tracks that are pretty much in the bag but we added pedal steel over the weekend so I have to edit and remix them all.
There is blues, ballads and country all harmony heavy so something for everybody.
Most of the projects have at least 20 tracks and the 10 yr old Toshy laptop struggles a bit with the big ones but the i5 PC that i do all the mixing on sails through.
cheers Dave, reaper was the first one which i got TH2 vst to work so its a winner so far...and good to know of your positive feedback re usage.Quote:
Originally Posted by dave.king1
whilst i love nothing more than cranking up the old and new amp setup when permits, i also really like the ability to record and 'jam' along with drums bass etc as ive never really had the whole band experience, only primitive jams with mates and now son.
i guess it all takes the time spent on learning....best hobby ever!!!.....
love to hear some of your work too....mmmmm bluesy steel please
The mix is getting close, my reference track for the steel sound on this one is Marshall Tucker Band - Fire On The MountainQuote:
Originally Posted by Guvna19
https://www.youtube.com/watch?v=fjEMB4eK1Qs
Brown Bottle, Max Richards / Dave King composition ( 2017 )
Max Richards - lead vocal
Bob Browne - backing vocals
Wes Grady - pedal steel
Dave King - guitars & bass + drums / organ / strings VSTi programming ( guitar is Squier Thinline Tele with 52 Blackguard pups through Fender Mustang using AC30 emulation )
Recorded & mixed in Reaper using native VSTs except Waves SSL E Channel on vocals, 3rd party VSTi for organ, drums & strings
Mastered in AAMS
Still not completely happy but close enough to let friends have a listen, enjoy and any feedback more than welcome
https://soundcloud.com/suthol/brown-bottle
Respect !!!
my music taste is wide enough to appreciate that fer sure Dave.
and cause i got no band exposure, i make the mistake of being all about (my) guitar sound. and i know i need to learn to leave room for all the instruments to work together. like that song does
in saying that, im liking a bit more 'dirty rock aggression' for my blues ie George thorogoodness, black keys
love the flute too
Remember when you are mixing that a lot of the instruments share the same sonic space as the voice so you need to spread them to build a nest for the vocals.Quote:
Originally Posted by Guvna19
Typically I put the drums & bass down the middle along with the vocals, keys to one side and guitars to the other at about 60%, lead instrument moves to the middle of the mix in the solo.
Another trap is backing vocals, put all parts to one side by about 30% and the level will depend whether they are backing or harmony vocals.
In Brown Bottle there are 9 individual tracks dedicated to the drums, mic top & bottom / back & front etc and then mixed into a master, the snare has a smear of reverb and then the master has a bit more.
When we recorded the steel we did three takes end to end, plus another two endings which gave me plenty to cut and paste to get the final track, until a few months ago Wes hadn't played for 40 years so I reckon he did a rather good job with an unfamiliar style in a key that is quite left field for steel Dm.
Every track has a noise gate, EQ & Compression in that order and the EQ is High Passed at 40 Hz because anything below that is inaudible mud and will chew up you headroom.
Lead guitar & vocals all have delay slapback with the length set at 0.08, typically I use about 30mS of preverb on the reverb to allow it to settle.
Reverb as well as adding colour is also used to move parts back in the mix
All vocals have pitch correction because 70+ Yr old voices ain't what they used to be :D
After applying the noise gate the first step before any sound treatment is to align the parts to the grid, in Reaper the basic tools you have are nudge, cut and move or stretch / shrink and regardless how good a player you think you are a lot will be off the grid, once the alignment is done normalise the volume on each track which will simplify the rest of the processes.
EQ & Compression are your main production weapons but don't go heavy in one hit, don't be afraid to cascade them or do parallel compression.
Volume automation is the icing that finishes the cake.
All of the above is relevant to any DAW
There are others on here who also do the recording thing so maybe we should start another thread that is dedicated to recording, I spend a lot of time on TDPRI and there is a sub forum called Recording In Progress that has a number of recording professionals on it which is where I have learned a massive amount over the last 12 months
Dave, you can improve your sounds even more by not sticking to a fixed 40Hz low pass filter, but moving it up to the bottom of the audio range of the signal, especially on miked recordings but even on pre-recorded samples. E.g. A lot of vocals really have nothing much below 200Hz, (obviously some low male voices will have stuff down to about 100Hz so you need to listen and move the frequency point to where there's no audible difference) whilst there can be a lot of extra noise picked up below that which can muddy the mix and also falsely trigger compressors. Do it on several tracks and it is amazing how much clearer your mix can be. You can also employ a low pass filter on the higher frequency sounds to help reduce extra noise.
It's also often useful to use an EQ that has adjustable HP and LP filter slopes. Often the built-in ones have a mild 12dB/octave roll-off, which can still leave audible noise below/above the cut-off point. Something like the Tokyo Dawn Labs free TDR Nova dynamic EQ http://www.tokyodawn.net/tdr-nova/ will give you the options of much steeper slopes. The paid-for Gentleman's Edition version (which I use a lot) goes even steeper. This can be really useful on some problem tracks with a lot of bleed from other instruments.
Also, don't forget that at the indicated filter cut-off point, the signal is already 3dB down, so for bass guitar (low E fundamental at 42Hz), with a high pass filter at 40Hz you are reducing that fundamental slightly. So you've either got to set the cut-off point lower at say 35Hz, or use a filter at 40Hz with a much steeper slope so that the affected area above the filter cut-off point is much smaller.
Bottom E on a normally tuned guitar is at 82Hz, so you can set a filter at say 75Hz to cut off all extraneous noise below the guitar. Especially with gentler filter slopes, the higher up you can put the filter, the quieter any sub-sonic noises will be. E.g. with a 12 dB/octave slope set at 40Hz, any noise at 20Hz will be reduced by 15dB (because the filter is already at -3dB at 40Hz). But if you can set the filter at 80 Hz (an octave above 40Hz) then any noise at 20Hz will now be reduced by another 12dB, so will be 27dB quieter.
Or you can have the same effect by using a 24dB/octave filter set at 40Hz, which if then set to 80Hz, will reduce any 20Hz noise by 51dB, making it pretty much negligible.
So I tend to use this EQ as the first insert on most of my tracks (let alone using its other EQ and dynamic functions).
Good point on the LF hi pass Simon, I was generalizing a bit because I do look at the fundamental range of each track before setting the point.
Cymbals, guitar and other instruments that generate higher harmonics I hi shelf to keep the shimmer.
Basically trying to lay down some basics to get folks out of the blocks
Is there a link someone can put up for Reaper or any other half decent DAW software?
https://www.reaper.fm/download.php
Reaper
http://www.audacityteam.org/download/
Audacity is worth looking at too. Open source.
Definitely look into purchasing a $60.00 license for Reaper, I purchased one and it's been great so far, Reaper was created by some ex ProTools guys.
The videos on Reaper looks pretty awesome... Only hiccup I have is its not Linux based (Audacity is good, but not good enough) so I had to go out and buy a cheap new windoze based PC so that I could even try it for myself. Hopefully in a few days I'll be able to post some half decently produced audio clips...
As a side note. Has anyone any news on Soundcloud? Rumour on Facebook has it that the investors are likely to pull out thus causing it to possibly shut down or go off-line. Be a real buggar if it were true...
There should have been a decision made yesterday. Haven't heard any news about it yet though.
Reaper apparently loves WINE so Linux users rejoice, I have a Linux box but not tried it yet.Quote:
Originally Posted by Marcel
I believe that Soundcloud will continue with a slightly different business model
Just checked and it's been saved (for now) by emergency funding. The CEO has stepped aside.
https://techcrunch.com/2017/08/11/soundcloud-saved/
Spent most of today on the receiving end of that brutal reminder as why I have such a disdain for Windoze....
Brand new HP i5 PC..... over 5 hours from powerup to first login.... I should send a $1M invoice to Microsoft for spoiling my Sunday....
Latest x64 bit version of Reaper installed in under 5 minutes !!!!
ASIO Driver for my audio device took maybe 3 minutes.
From clicking "install" on Reaper to having a half dozen test recordings with Efx probably came in at just under 30 minutes....
Learning all the details of Reaper.... Ummmm....
Reaper is a lot more complex than it's given credit for, I've been using it for quite a few years so if you have any questions feel free to ask and I'll answer what I can
Dave, when i added a drum mp3 into reaper, it sounds crappy compared to just playin via media player, any idea why?Quote:
Originally Posted by dave.king1
In Reaper you are listening to quite a naked sound and a single sound file for a drum kit getting it sounding right is quite a challenge because each drum has different EQ & Compression requirements to sound good.
I use Drummica from Senheisser which is free, a bit of stuffing around to get running but it is recordings of a live drummer with a huge library of beats.
Drummica here http://de-de.sennheiser.com/drummica the site is in German so a bit of thought is required when working through the download but it is fairly intuitive
I sit down with my band mates and find the pattern and sound we want and then come home and create the drum tracks.
Typically kick ( front & back ), snare ( top & bottom ), hi hat, overhead & toms if applicable.
These are all individual tracks that need to be aligned, kick on 1&3 snare on 2&4 etc which takes only a couple of minutes , I then apply EQ & Comp to each track to get the sound I want, slight verb on snare, I balance them individually and then route all these through a drum master track so I can change the entire kit volume in the mix without trying to rebalance each item each time.
I also use automation to bring the snare and cymbals up where required.
Apart from the drum VSTi I don't use anything to treat the drums that isn't native in Reaper.
Kenny Gioa has a heap of terrific Reaper specific tutorial vids on YouTube.
Your main production tools are EQ and Compression and remember "less is more" with these tools, chaining a couple of EQs using small steps will give a better result than smashing it down unless you are looking for a specific sound
Keep asking and I'll do what I can to help.
If you are really keen you can also program your drum parts from scratch in MIDI and then load a VSTi such as GTG DPC3.
http://www.vst4free.com/free_vst.php?id=385
I've done it, bloody long process but it does enable you to get what you want if you have odd time signatures
So far managed a basic drum beat today in reaper using MTpowerdrum, great fun to jam along with.
when i add guitar for recording, i get a little snap crackle pop? not sure if i can change setting to correct?
PC is good, interface is good, average monitors, - maybe its the monitors i dunno.
dont like using my amps for drum parts.
any advice? Guv
Normally it's because the audio processing stops for a short while as the processor is too busy doing other things. Increasing the buffer size on your interface will normally stop this - but at the cost of increased latency.
How much you can do on a DAW without having to increase the buffer size mainly depends on processor power and the drivers for your audio interface. More RAM can help, as can having an SSD as the system drive, but most of it is down to processing power and well written drivers.
However, there are other things that can also take up processor time - things like 3rd party virus checkers, wireless networks and windows tasks running in the background. There are also programs that when sequenced to run by Windows simply take a long time before they pass back control to Windows, during which time your audio buffer may have filled up without being processed, resulting in an audible glitch. There are various latency checkers you can download, I've used this one http://www.resplendence.com/latencymon which gets a good recommendation from a fairly large UK PC building company that I know the owner of. If it highlights a program with a high latency, then you either need to shut it down (if non-essential) or find an alternative.
This is one reason why DAWs are still best used as dedicated machines (or at least dual boot).
Also, if you don't have a separate video card but rely on the on-board display, then that can take up a fair bit of processing if the DAW windows are scrolling.
Also check any power saving options - especially if it's a laptop. You want everything running at full power all the time with a DAW. you don't want windows deciding to ramp your processing power or hard drive speed down.
So if the PC is multi-purpose, you may need to set up a dual-boot system with minimal programs loading for DAW use - or at least a separate profile. There are some programs, like Adobe's Acrobat Reader, which constantly contact the web (at least it did a few years ago when I last used it). Also beware of any updater programs that keep looking to see if there is a new version available.
Thanks for chiming in Simon I've been hoping you would.
One of the problems I've had with VSTi in Reaper is random peaks the one I use for Hammond B3 does it with peaks way over +20db and this auto mutes the track and another piano VSTi does what you describe and it's often not noticed until the project is rendered.
The solution is to go to the JS plugins and apply the Master Limiter to the track in question, set it to about -6db this will control transients on the virtual input but won't stop you from getting the output where you want it.
I don't know if this issue is unique to Reaper or more likely the VSTi that I am using.
Probably the VSTi. With the floating point maths used in DAWs, they've got a huge theoretical dynamic range (around 1680dB), so peaks of +20dB aren't a problem at all. But that's not to say that a VST/VSTi plug-in or Reaper itself doesn't have some default protection to save speakers etc. in the event of detecting a particularly loud transient. I'm a Cubase person so not really familiar with Reaper, though I have tried it a couple of times.
Many VSTs modelled on real FX units behave in the same way to incoming signal levels, so levels above the system nominal 0dB will often result in distortion or severe limiting/compression.
Obviously the trick here is to carefully set the gain levels of instruments so that they don't go above 0dB, and generally have a much lower average signal level than this.
I'm just trying to contemplate what 1680dB equates to in real life......
It's like going from way beyond the quiet silence of outer space to less than 1cm away from ground zero of a 100GT nuclear blast !!!
That IS a huge (mathematical) range...
Another good latency tester app is one called DPC Latency Checker, it's free and I sometimes use it to fix DAW latency problems after I've tried various buffer settings, here's a link to the website where you can download it from:
http://www.thesycon.de/eng/latency_check.shtml
Something to remember is to keep input levels below 0dB when tracking (peaking at -12 to -18 is considered fairly safe), and output levels below 0dB when exporting to a finished .wav or whatever. Once "inside" the DAW, while the audio is being handled by floating point math, it is safe to exceed 0dB.Quote:
Originally Posted by Marcel
You really need a 'true peak' meter to set the master output level when exporting to .wav files and the like, and even then set the maximum output using the true peak meter to -0.5dB or so. This is because inter-sample peaks can push the real output up by up to 3dB.
You don't need to worry whilst it's still it's in the DAW and Paul_H said (though as I've previously said, some plug-ins may give you a lot of distortion or saturation if the incoming signal is too hot). So it's still best to keep everything below an indicated 0dBFS and keeping individual bus output levels to between -12 and -18dBFS. If you're working at 24 bit depth, then at -18dB you've still got far more theoretical dynamic range available 21 bit's worth) than a 16-bit .wav file has, so you're not giving anything away by working with a -18dBFS signal.
Inter-sample peaks are often the reasons that MP3s sound bad, especially if they've been taken from a .wav file that has sample peaks at 0dbFS or just below.