51m0n

I voted - rather tricky to choose, very good stuff in there, well done all!

[quote name='BigRedX' timestamp='1353933797' post='1879824'] Sorry to go slightly off topic but can you explain why moving one of the audio tracks isn't the same as moving the microphone placement when it comes to phase alignment. AFAICS moving the mic changes the time at which the sound reaches it. Surely nudging the mic track forward by a few samples achieves the same thing? [/quote] Its not altogether obvious is it If the two sources were identical, except for a delay then time aligning would work exactly as you expect. Thing is they are radically different sources because of the radically different chains:- 1) the DI from the amp is the output of the amp's preamp to the interface via a DI box (transformer at the very least if its passive), then into the mic-pre and on to the DAC 2) the input from the mic is that signal after it has gone through a power amp, a speaker cab ( a massive filter if ever theere were one) , across air, hitting another transducer (the mic), through another set of electronics (the mic-pre) before getting to the ADC. Ther two signals are very very different, they have different frequncy spectrums, different envelopes, everything that there is about the two sources are differetn except the thing that generated the initial sound. How can you possbly time align two sounds with different envelopes anyway? WHat do you line up, the beggining of the attack, or the end of the transient, or the sustain phase? They can (will) all be different now. As soon as you move one against the other the affect is to filter them where the frequencies add up or subtract from each other at that point in time. But the two signals are now different in the time domain (due to all the filtering of the speaker not being accurate, the mic no tbeing accurate, the air is effectively a filter iun this case etc etc) so you can only fix them in one place Phase alignment can help you get a good overall best guess across all the frequencies - effectively using the two signals to make a third that is the best combination, time alignment can take the two signals and make them start at exactly the same time, which isnt necessarily the best sounding result either. Does that help at all?

[quote name='MiltyG565' timestamp='1353937853' post='1879917'] Seriously man, i can not afford the extra gear. My cab is really not worth micing, and i have not got the expertise, or the equipment to go to such great lengths. I'm on sick leave from work, and will be for a while, so buying stuff is absolutely out of the question at the minute. I want to record, i want to use the DI out on my bass head, i want it to sound good. Cheers, i'll give them a call during the week [/quote] In thatcase by all means DI from the amp. Then to get the cab sound back use a decent free Cab simulator (Le Poulin's LeCab2 is great! Google it) to do the 'micing' for you.

[quote name='BigRedX' timestamp='1353939588' post='1879945'] Not necessarily - One of the options for installing EMGs is for active pickups and standard passive volume and tone controls. Have a look at [url="http://www.emgpickups.com/content/wiringdiagrams/Tone_Instructions_0230-0162B.pdf"]this[/url] for more details. [/quote] Yup spot on! I have a set of EMGs retro fitted into an old bass with passive tone controlsm, and they sound absolutely killer, yet all you can do on the bass is dial out some treble with the tone control or change pickup volumes/balance. Works a treat...

There is nothing wrong with a straight DI:- Bass -> D.I -> mixer/interface. In fact if you have the 'right DI' it will sound absolutely stella. If you need some amp based dirt, rather than pedal based then you will absolutely have to mic the cab. Why? Because a cab, especially a cab designed for dirt (ie a sealed no tweeter type affair like an Ampeg 810) is a huge filter on the output. If you DI the sound of the amp (there are DI's that can handle speaker level output, but most cant - rtfm!) the result will include a huge amount of rather horrid top end fizzy noise, which you will have to deal with. The cab does exactly that for you. If you want to mix DI and mic'ed then phase is a huge issue, simply moving the sound files time aligns them, but doesnt necessarily phase align them (complex subject, phase is dependant on frequency etc etc). If you are going to timealing them, move the mic track to match the DI, it will usually sound better that way round. In order to get phase right in the first place run some sound through the amp (nice bit of pink noise will do), watching the master level move the mic until you get the highest output. Thats about as cientific as you will get with the gear you have. In your position use the DI for the bottom end and the mic for the mid range grwol. Use any cheap mic you can get your hands on, positioning is the key to a great reault. Red5Audio do a great cheap kick drum mic (c £35) that will do fine on bass in a pinch....

[quote name='BassBus' timestamp='1353875964' post='1879309'] Interesting stuff indeed, but how did that machine get to the point that it could play that line? If indeed it did play that line. People had to design that machine. People had to build it. People had to program it. I don't think we are in a lot of danger. Once upon a time someone said we would be a cashless society. Someone said we would not need paper. Someone said CB radio was the future. Maybe I'm just one of these real idiots though. [/quote] It plays MIDI files. You can easily turn audio into MIDI these days. In other words, find your favourite bass lines, convert to MIDI, let robot play them. Easy. That is the point of programming and robots, you design and build them so that they then just need different data to do different things. Scary piece of kit, fortunately it cant play fingerstyle yet

Get a green screen too, then peoplp can have any background on video taken of their recording/reheasal (like crowd scenes from Queen in Brazil) "We played in front of 250000 people darlin"

Yeah, +1 on a couple of lessons. Nothing will help more in the first instance than maybe 3 or 4 lessons just to tidy up what you're doing a bit and send you down a path to low end glory in years to come...

When I were a lot younger a mate of mine was pretty damned good at all that shredding nonsense, but he became so paranoid that he might damage his hands (he also enjoyed a lot of martial arts) that he learnt to do it all left handed as well, both on a left handed guitar and an upside down right handed guitar . That made for some very very cool showboating tricks. Then all of a sudden no one was interested in that sort of thing and he came to realise he'd lead a wasted youth couped up in a room learning something totally useless.

That Behringer device will give you the bare minimum of what you need. Fine fo rwhat you are currently doing since it has a direct monitoring solution Audacity will drastically limit what you can do, and tie you to a very specific, and not very helpful workflow since it can't process audio as it plays it. This, more than any of the other really poor things about audacity effectively renders it useless for any kind of multitrack mixing at all. I'd suggest a look into Reaper, its the best bang for the buck of the current range of DAWs that I have found...

Awwww shucks

Mine sound fab.... Decent bass, decent interface (RME UCX) decent monitors. Job done...

[url="http://www.heilsound.com/pro/microphones/pr-35"]Heil PR35[/url]

Yer pays yer money etc etc Audacity is first and foremost an Open Source effort at a very simplistic wave editing program. It doesnt function very well as a multitrack recording program IME and IMO, and it certainly isnt designed for close to real time monitoring. I've spent too long mucking about trying to get a cheap solution that works. The value of that time is greater than the value of the new soundcard I just got so that I no longer have to worry about this stuff anymore. Any USB soundcard with hardware direct monitoring will enable you to do what yo need to do really. Audacity will just about keep up too if you're lucky (sorry but I have never had a great experience with Audacity, and I'm very pro open software - it just is mroe hasstle than its worth IME on all systems I've tried to get it to work on) On top of this, Windows' audio system is not remotely capable of real time monitoring (on a very fast current machine you will be awesomely lucky to get a latency of less than 30ms which is like having your amp 10m away),and so shouldnt be used at all. You need ASIO drivers installed, and even then all ASIO drivers are far from equal, and USB chips are far from equal too. The very very best USB interfaces in terms of drivers and latency are made by RME, you will be spending proper money on this kit, but its utterly brilliant in use IME. I get 2.5ms of latency with my RME interface and I havent tried to make anything more optimised at all yet....

Well the Blog will one day get there I hope. Here's hoping it gets to the point where people search there first and ask on the forum second. Then I know I'm getting somewhere (long long way to go before then though ) Its good revision for me too CT, I have to think about what it is I actually do before answering most of the questions. Its all so natural for me, I dont think "Oh, right I need to use some slow attack medium ratio medium threshold compression to accentuate the transient on this guitar rather than just make it brighter, since the actual tone is right but its nto making the impact I want it to in the mix" I just make the guitar sound 'right' to me. Its quite without conscious thought when I'm mixing...

[quote name='lurksalot' timestamp='1353428051' post='1874845'] KB you are miles further up the path than I am . 51m0n , Quality post, thanks millions, though it is probably waaaay past my expertise at the moment . Ed to add I have been through a few You Tube tutorials , but I find it very frustrating that a) even for beginners they assume you know stuff and use techy words lots of stuff they try do doesnt work for them first time so they do it again [/quote] This needs to be open in your browser at all times... http://www.soundonsound.com/information/Glossary.php

[quote name='JayPH' timestamp='1353428375' post='1874848'] Thanks Si. I get this more now. I'm going to mix some music tonight and will try this technique. Do you add EQ after you have adjusted your levels? [/quote] Its more complex than that. The actual act of getting a mix is a very iterative process. I tend to try and through up something with the faders as roughly as possible just to get a clue. After that I 'build' the mix in terms of groups, aux sends, fx I think I'm going to need etc etc... Boring stuff that can take ages, but if you dont do it then the computer fights you. Then I'll mute areas I'm not interested in at all yet, and listen for anything dodgy, I can spend two or three hours just mooching around to be honest, a little tweak here or there, learning how the song moves me. At some point I sort of change mode and just hack all the bad stuff out of the various tracks with EQ, change envelopes with compressors, look for the right ambience, adding more tracks and hacking more stuff out, refining the general balance constantly. I also spend time balancing sections (groups) together so that I can then control the level of a section easily. Eventually I am left with something very close. Then I usually leave it, and come back in a few days. Time to this point on average (for a serious mix) is at least 5 to 8 hours. Revisions and polishing from there can take an age or just be bang on.

You used to have tons of Warwick in, but then it never seemed to move as I recall, and Fender require you to put one of everything on the wall if previous conversations are true. Not really great that though for the punters. I havent been in for ages as I suffer no GAS at all these days. Last thing I bought from you was my sa450, which has been brilliant. Being in Brighton I think it would be cool if you had one of each Barefaced Cab on display, since they are from Brighton too. Not a lot of people really know about BF cabs and they are good....

The human brain can detect significantly less than a 3ms delay, but only in certain specific ways. For instance your ears are 17cms apart (on average) and that equates to a delay left to right of a sound originating hard right of about 0.5ms. You detect it as the sound feeling like it comes from the right even if the level is identical. Its a great way to achieve panning in a track if you want the level to match - except it doesnt collapse into mono at all well! If you play the direct monitoroed sound and the delayed sound at the same time you will hear less than 3ms of delay, but as phasing and transient blurring. Try it...

KB, you are really asking "How do I mix a song". Its a huge topic. The goal is to cause the listener to engage emotionally in the result. Don't be tempted to think that is a trite or simplistic answer, it isn't I am deadly serious. As simple as that sounds it is one of the hardest things to do, especially if you have no grounding in the subject at all yet. So how do you go about doing this? You have to learn to listen from the perspective of a punter, whilst also listening like an analytical machine, at different times one or other view point is more imoprtant. In most songs the lead vocal is 'the song' from the point of view of the punter, and to connect with it the emotional content must pour out (even if ihat content is no more than "let's dance" the empotional impact must compel the listener to do just that). We can assume that that lead vocal must be front and centre of the soundstage then, its what we want the listener to focus on where ever they are standing in the stereo field. Apart from that music generally has a pulse, contemporary music almost always does, and that pulse is what causes peopple to move, and getting them to move is a very short step away from getting an emotional connection, so the key rhythmic foundations also need to be in the center of the soundstage. Like anything a mix is only as good as its fundamental foundations. The kick, snare and bass must be (in a contemporary mix) front and center. In a more vintage mix from the 60's stereo was first applied as literally a switch with either left right or centre, and thats why you sometimes hear old tracks, or tracks that want to sound old (Lenny Kravitz) having a different somewhat odd pan(orama), or stereo spread, with (say) drums hard left, bass and guitar hard right and vox dead centre. Unless you are after the goofy old vintage feel just avoid this, treat it as a rule that should be stuck to unless ther are utterly compelling reasons not to. We now get into a bit more of a technical area. That sound stage with two speakers generating faux stereo actually really has only three definite and fixed positions and they are Left, Right and Centre. Yes you can pan by graduations of this, but whenever you do the exact source of the panned instrument will be murky at best. This is so imortant that a very large number of major league mix engineers literally only ever pan LCR (and that style is so prevalent that it is called LCR mixing). Bearing in mind that they may well be doing a lot of tricks to make this jar less (such as panning the reverb or delay ambience from anything panned left over to the right and vice versa) it doesnt change the fact that in their mixes there is no 'just a bit more to the right', it's all the way to the rigth or left or dead centre. So why pan at all? Seperation.Seperation.Seperation..... In order to have a mix be emotive its important that it has clarity, and this is achieved by creating space for the components of the mix, and one important way to do that is panning. Anything that isnt in the holy trinity of the foundation (bass kick snare) or the lead vocal(s) needs to be moved out of the way (panned). Personally I dont stick to LCR rigidly, there are elements of any mix that I like to slip into thos elesser defined areas off centre or off hard panned (hi-hats for one, esp with other percussion to balance against them). But I do always pan anything that can compete with the vocal out of centre. Dont worry about a mix not balancing in the stereo field sometimes either, that is cool too. It can be part of the payoff of getting to the chorus for everything to be balanced then where it wasnt just before. Guitars, pan hard, double track (not more usually) and maybe send any ambience returns to the opposite side for a bigger sense of space. Try it, it sounds like every big rock mix in the last 15 to 20 years, because it works really really well. Backing vox need to be a little bit ethereal, so I like them not hard panned completely, but very nearly, then the reverb/eq/delay on them needs to place them further back behind the lead. Oh, EQ! See my recent blog on using it 'properly'. EQ is used for a couple of things. Firstly, when you bunch a load of instrumetns up together , left right or centre, they are going to competefor frequency. EQ is used to cut out holes in one instrument for another instrument to occupy. Done right you cant tell it has been done at all but suddenly you can hear the relevant parts of all the instruments involved. Win! It also helps place things front to back in the mix, closer sources are brighter, further away sources are darker, a gentle wide roll off of up to 20dB at 20KHz can really give a sense that BVs are behind the lead for instance. Then there is the balance itself, which is a matter of taste firtst, what 'feels' right to you, is right. As long as you are in a palce where what you hear you can trust so decent acoustics are mroe important than hugely expensive speakers. Learn your monitors, and you can trust your ears. Proper use of comrpession can help with some asects of the mix balance, not just levelling but shaping the envelope of the sound, you can make things easier to hear by electing which things get their transiet mroe prominent than others, and which things get their sustain more obvious, balancing intime as well as frequncy and over all level, and pan. If you do it all 'right' it sounds killer, if you aren;t used to it or you are learning a bit at a time it can sound odd in the first instance. Time and lots of proactice and questions, and reading will get you there. Watching someone who really knows their beans mix will get you there a lot quicker IME! I would really recommend [url="http://www.amazon.co.uk/ZEN-Art-Mixing-Mixerman/dp/1423491505"]Zen and the Art of Mixing[/url] as a superb book on the art of mixing a song well. Its well written and will provide you with insight after insight into why you should do what to help you achieve a great mix.

[quote name='bremen' timestamp='1353339989' post='1873840'] SM57 = SM58 without the 'ball' pop filter. [/quote] Not precisely. The innards are the same, but there is a head on the sm57 too, and that is what determines the polar pattern of the 57 and changes hw much proximity effect it displays. It sounds pretty gash if that head gets damaged, as does the 58 minus its spherical windshield (due to the loikelihood of it being affected badly by pops and wind noise etc. Unfortunately you cant just buy a 58 and unscrew the head to have a 57 whenever you want. Pity really!

It wont be 0ms latency, at the very least there is ADA latency and on top of that there is the (very fast) DSP handling the modelling. But probably sub 2ms on anything decent, I would guess...

I really like the Red5 Audio kick drum, terrific mic, I got mine when they were on offer, for about £40 IIRC. Unbelievable deal! You can also use a Senn e835 on snare/toms/horns etc (I've used mine on all of thos and more, and its been fine, as well as for live vox)

[quote name='Thurbs' timestamp='1353093069' post='1871656'] All this talk about latency, what is the usual latency through a standard digital pre-amp? [/quote] Depends on the drivers, the chips yada yada.... ASIO drivers you may be looking at more than 20ms(+) for a cheapo interface, down to the RME UCX which is going to get you less than 5ms maybe even as little as 3.5ms. Using any built in kernel style stuff in Windows and the minimum possible woudl be closer to 30ms. Using the standard it gets worse too, if you use the bogstandard MS audio driver. Nothing to do with the speed of the machine either. Macs can manage slightly shorter roundtrips as a rule. All due to the Mac kernel (a FreeBSD unix variant essentially) allowing for more rigorous timing and interrupts and so on than the MS kernel does.

[quote name='lowdown' timestamp='1353325657' post='1873565'] For any Reaper Fanboi's............... Very nice looking skin here. Enjoy. [url="http://www.houseofwhitetie.com/reaper/imperial/wt_imperial.html"]http://www.houseofwh...t_imperial.html[/url] Garry [/quote] Now that is really pretty.... Dunnio if I'd use it, I dont care if it copies a desk look and feel really, I just want it to be efficient. But still, that is really nice. May give it a whirl, ta!