51m0n

[quote name='xgsjx' timestamp='1354031063' post='1881121'] Very true, but most dedicated audiophiles have their optimal sitting position, therefore EQ allows tailoring for that given position. [/quote] No it doesn't. You cannot eq out ringing in the room at frequency n being twice as long as ringing in the room at frequency m So you turn down the preception of too much bass, but actually in any given instant there isnt too much bass, over time there is too much bass. Turning it down with an EQ, doesnt stop the ringing, it just puts less energy at that frequency into the room, so now your balance is wrong, and you still have the issue of the ringing smeering the bass. It is a different issue, that [i]cannot[/i] be fixed with EQ. The use of EQ to fix acoustic issues in a serious listening room should be entirely debunked. Yes we use some EQ to try and help at a gig, but to be honest as everyone who has played their bass and wandered round the room can tell you cant fix the tone everywhere with EQ, its a myth, you have to compromise (hugely). And for any audiophile compromise is not a word that exists in their dictionalry (or they aren't audiophiles). Yet there is a persistance in the myth of EQing to the room. Also, please show me an audiophile HI-Fi EQ capable of the kind of surgical cutting we are talking about here. I am definitely not talking about your average gentle +/-9dB shelving EQ, something more akin to a 64 band graphic, or 20 band fully parametric would really be what is required. This is not the same as EQing for taste, that you can do to your hearts content (assuming you have sorted the room out so its not wildly messing things up after the eq), but dont muddle that up with EQing for the room

Oh and yes, I definitely advocate no tone controls on an amp. Get the room right and you dont need them, as someone said everyone hears differently, but, their day to day perception of sounds is generally very similar to everyone else* and equally importantly most mixes are done on reasonable monitors in reasonable room and decent mastering is done on incredible gear in incredible purpose built rooms. Don't forget that the biggest thing you can do to improve your listening experience on [i]any reasonable stereo in any normal room[/i] is to go some way to fixing the acoustic. Far cheaper to do than a lot of** the so called audiophile stereo equiptment (with attendant snakeoil fantasy products to add extra bite, gnarl, whimper, and beloved-sploink to your listening pleasure, with free crystals and shakra-gasm inducing spleen resistant leatherette finish) and with definite, measurable, guaranteed results. I would rather listen to a £500 stereo in a room with (at least partically) sorted acoustics than a £500k stereo in an untreated room. *Unless they are going seriously deaf, or have blown huge chunks of their top end hearing through misuse. In either of these cases adding treble or bass will not in any way magically fix the issue. **Who do I think I'm kidding, cheaper than [i]all of[/i]

[quote name='thisnameistaken' timestamp='1354029243' post='1881085'] I'm not ignoring the wink at the end there but I'm sure you know that even a low-impedance signal can get obviously degraded after 20 or so feet of cable. [/quote] Yes, it can, however, IME its far more likely that pedal board signal degradation is due to poor quality interconnecting patch cables. I use 10 ft in to the board and 10 or 20ft out of it, I couldn't care less about true bypass vs buffered - actually in point of fact I'd far rather have a really great buffered signal than a true bypass signal, since it will (again IME) degrade significantly less with cable length. However a good active circuit goes a hell of a long way towards making that unimportant (again IME). Yes you can probalby measure some degradation, but in live use I haven't been able to really notice any degradation in my board with either active bass, but you cantell (albeit slightly) with the passive one.

<Oar-In> Now this whole using EQ to fix the room thing.... Its a bit of a myth actually You cant fix issues with room nodes with an eq, since regardless of the settings on the eq the issues prevail. The issues are more often time and frequency decay issue, whereby certain frequencies (due to ringing) may be louder for longer in one place and quieter in another than other frequencies. If you turn the eq down for one frequency you can fix the issue for one, tiny place, but will change the problem elsewhere (thats the nodes) but you wont fix the ringing (as seen on a waterfall plot), and you cant fix every multiple of the frequency (which you need to) without ruining the sound everywhere. And hi-fi has never, ever come with tone controls close to complex and comprehensive enough to even make a stab at fixing an acoustic. </Oar-in> As for True bypass vs Buffered? Use an active bass and stop worrying about it

Scariest thing I've heard of though is a bit of software that can extract MIDI from audio so well that it can be used in combination with a grand piano capable of being played by MIDI to recreate and re-record a performance of a long dead artist, to the point where a new duet can be created with a living artist. Which was done recently at Manifold Studios (IIRC). Bonkers!

Best of luck - fingers crossed, have a rgeat time!

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...