APZX

Well since I'm still working on the last part of a song I figured a quick break from that might help a bit. Already, submitted my entry, though like an idiot I left silence in the end because recording is weird. Still love Super Metroid with its just completely off the wall weird but fantastic soundtrack.

Here let me give an example of what I've been able to muster with this tonight. This Beach from the SNES game Plok. I've been doing covers like this for a bit now just to help wrap my head around what I can and can't do yet. In this case I'm using 9 of synths. Even for the SNES the intro for this song is pretty darn epic. The basic list breaks down like this: Main Organ: TX81z | Strings: TX81z | Big Bass Intro Hit: Mopho | Brass: D-110 (layered with the Mopho) | Bass: x0xb0x through a ProCo Rat | Lead: Radias | Guitar: TR Rack | Clavinet Arp: SH-201 | Drums: DRM1 There is a lot I can do, but still so much yet to learn. Like rather than rely on compression for the kick or bass I used EQ and drove the inputs of the mixer to get a compression like effect. I'm still learning this whole setup, and it is a total blast.

That is years and years of building up. I've had that SH-201 for like 7 years now. The Radias for like 6. The DRM1 is probably the newest addition at only like 4 or 5 months old. It takes time to get all the hardware. I mean I had that mixer sitting around for about 2 years before I was actually able to get the funds together to start hooking up everything, and I still need more. When I say you don't want to know the amount of cabling that went into this I'm not joking. It seriously took me a whole day just to wire this up without running everything into patchbays like I should. The synths are plugged straight into the line-ins on the mixer. However, I ultimately want to run a hybrid setup so that will necessitate a lot more cabling as I'll want to normal through DACs to the line-ins, which will mean that I'll need enough patchbays to cover all the line-ins (40 mono inputs, 4 for the stereo, and 8 more for the effects returns which means a total of 52 points on patchbays). That then means at a minimum for cabling just from the DACs to the patchbays to the mixer I'll need double that. This doesn't even get into the processors or synths and their associated cables. Patchbays themselves aren't terribly expensive ($60 - 70 for some Behringer ones, and the Samson S-Patch Plus is like $120, but it is really well built). Individual cables aren't too bad, if you build them yourself with Amphenol connectors that is about $4 for a standard 1/4" TRS and then depending about $0.19/ft for some Canare L-2B2AT. This does require you know how to solder. Having said that, if you buy the cables from Monoprice of the length you want you're only going to save about $0.25 to $0.30 a cable on average. Sure a synth may cost $700 - $1,500, but the cabling will easily start to run into that territory if you've got a lot of them. And I do want more synths lol. Need to get the desk built for the mixer though That is coming in the near future (hopefully).

Since I went to the effort for some other folks I thought why not share the love? Also, I have no idea where to post this and this subforum seems the most appropriate. To start Kurzweil Music Systems is a company that I'm sure quite a few of you have heard of, but I doubt all around here have. Well in the early 2000s the company released a couple of FX boxes, and one of them was Rumour, which is basically a reverb machine, but it does have your typical FX except that it is really good at reverb. Some folks showed some interest in getting a some impulses of the unit. So, I spent some time away from doing my remix for Odai (sorry ) and selected 35 reverbs from the unit and sampled them. Quite a few of these sound fantastic on drums, others are just nice big lush reverbs, others are strange, and one was selected purely for its name. Hopefully, they have some use for some folks around these parts. I know I'm always on the hunt for more reverbs. Just for the fun here are some my more liked ones and what I like to use them on. 005 - Small Wood Booth: Drums. It adds just a touch of air and space to them. 034 - In the Studio: If darken the reverb just a little bit, it really adds a wonderful sense of warm space to the sound regardless of the source. 043 - Classical Chamber: Strings, Piano, Pads, and Vocals. It just has this right amount of tail to it. 050 - Real Niceverb: Works on pretty much anything. 068 - My Dreamy 481!: This is a pretty dark reverb and works well on sources that you want to emphasize their midrange components. Really really good on pads. 093 - Dreamverb: This is like the last one except richer. 133 - Lazerwaves: This is part delay, part reverb, part crazy. Just try it. Just a small selection of the 35. Hopefully, someone finds a use for them because I love the way the Rumour sounds. Snag 'em here.

No problem! Also, you're the first person who has ever really acknowledged that these boards exist that I've talked to. I was genuinely impressed with just the amount of features on offer from this board when I snagged it. Sure it may be a "Live Sound" board, but it does a perfectly fine job with all my synths and even the preamps are pretty darn good! Not that I'm some super snob about pres or anything, but they do actually sound pretty good. Let me be the first to say that most of this is used. The mixer I nabbed from a company that was in the process of upgrading to digital boards. They had two when I went over, my Series Two and an Allen & Heath ML5000 56 Channel. They were both the same price, $800, but I couldn't fit the ML5000 in my Blazer so I got the Series Two instead. Plus the flight case for the mixer, two power supplies in a flight case, and well yeah pretty much everything for it lol. It'll be easier for me to say what I bought new rather than what I bought used. The Roland SH-201 was bought new, really good for pads and aggressive leads, but it can do a few other things too and is super fast to program. The Korg Radias-R because I didn't buy the version with a keyboard, and in essence it is like a poor man's Virus. Really versatile synth. Dave Smith Instruments Mopho. I LOVE THIS THING! You can still find them for really not a whole lot and they're stupidly powerful little synths. Tons of sonic capabilities in that little yellow box. Synthesizers.com modular. This is thing cost me more than my computer and after getting to grips with it the thing is insanely flexible. After that it is basically my two distortion pedals, a bunch of cables, and a patchbay. Pretty much everything else was purchased used. I mean you can find Alesis 3630s commonly for less than $100, sure they're not the best sounding but they don't sound that bad. A D-110 can be had for anywhere from $50 - $100 and is a super flexible 80s ROMpler with 8 individually addressable instruments PLUS a rhythm section. The old Yamaha FM synths can be had for about the same cost. Both my TX81z's cost me like $80 each. Search your local classifieds. You never know what will pop up. I bought the D-110 and TX81z's from Japan, which necessitates a transformer because they're 100v devices instead of 120v. For the record I still use FL for sequencing and controlling all the synths. This is just a completely different approach to making music. In some ways a bit easier, and it others a bit harder. If I use a TX81z for a bass then that is all it is doing. I can't just call up a second or third instance of it for some killer FM E-pianos. Having said that, at this point I'm still trying to wrap my around loading the mixer and what exactly I can and cannot do with it. I've been doing covers of older VGM just to help wrap my head around it.

My space after recently getting around to actually setting it up. Not complete, but it works for the moment. Starting off with the DAW itself. A couple of HP ZR22w monitors, Neumann KH120A for monitoring, a Drawmer MC2.1 monitor controller, Focusrite Saffire Pro 24, and a pair of VU Meters from Crookwood. The computer itself is a custom built rig (i7 4770k, 32GB RAM, GTX 780, 256GB SSD, 2TB HDD, 2x 1TB HDD, in a Fractal Design Define R4). You can see the side of it on the bottom left. It is a really plain looking computer. However, when I turn around is when things get fun. Two tier keyboard stand with a Roland SH-201 and a Korg DS-8 (this thing actually needs work: new battery, keybed cleaned due to some sticky keys, Sysex dump, etc . . .) Synth rack! From top to bottom: Korg Radias, Roland D-110, 2x Yamaha TX81z, Korg TR Rack, E-MU Pro/Cussion, and finally an Access Virus Rack at the very bottom. Other synth rack, well more like a single synth and a patchbay (currently unutilized as I need to think of the best way to use it). The synth for this one is a Vermona DRM1 mkIII. Synth table!!! So, at the bottom is a 22-space Synthesizers.com Modular. Pretty much the Studio 22 one except with the addition of a Q167 LFO++ Module for a bit more modulation options. The top left is a Ladyada x0xb0x (TB-303 clone). Currently attached to it is a ProCo Rat for some Acid. I also have a Boss MT-2 that isn't pictured, but it is chilling to the right of the Alesis QX25. In the middle is a Dave Smith Instruments Mopho. You could see just a little bit of this guy in the last picture, but this is my mixer a 40+4 Channel SoundCraft Series Two (nicknamed Priscilla by the person who helped me to lift it because it weighs about 150lbs or 68kg and is the better part of 5ft wide or 1.5m). To my ears, generally neutral, pretty quiet, and ample headroom (can handle more level than my converters). And behind me is basically the DAW. Though I forgot to snag a picture of it, the rack between the mixer and keyboard stand is a processing rack. From top to bottom: Alesis MIDIVerb IV (Multi-effects), Kurzweil Rumour (basically a reverb machine), 2x Symmetrix CL-150 (compressors), Alesis 3630 (compressor), Alesis MicroVerb III (multi-effects), ART Pro VLA II (compressor), and finally a patchbay that is being used . . , yay? Under the mixer are its power supplies. And no you don't want to know the amount of cabling that went into this. All synths are controlled via MIDI and to do that I'm utilizing an Alyseum AL-88c which is behind the modular. And this is also why I haven't been working on my track Odai (sorry ).

Pretty much what Neblix & Flexstyle said. Though I do feel the need to comment on something Neblix stated about matching cable lengths. That is a "technically" right answer, but in practice outside of a few situations is actually completely unnecessary to be concerned with. It is correct because it does indeed take longer for the electrons to move through a longer wire, but then again you're talking about speeds in the region of like half the speed of light (yes I know the actual electrons are only moving mm/sec, but you don't notice that). There are situations such as with computer memory where the wires have to be the same length otherwise you run into timing issues, but for audio purposes I'd say that as long as you're not comparing a 100ft cable to 3ft cable you're probably not going to notice any delay or phase issues between the cables. Or let alone the fact that channel 1 on a mixing console is very far from the Master section and therefore should cause it to be noticeably out of phase with all the others when that isn't the case.

Ah the fun of recording hardware and trying to get it into the computer without noise. So, for the Sound Canvas depending on which one you have (last one dates to 2001) would determine the noise floor. The original SC-55 had 16-bit converters giving it a "theoretical" noise floor of -96dBFS. In practice though it is more realistically from that time (1991) the converters were probably operating around 14 to 15-bit for pro audio gear (the PS1 from 1994 operated around 13 to 14-bits). So, that would raise the noise floor about 12dB or so. Newer versions did gain better converters which inevitably leads to less noise. Now, that isn't actually all that inherently bad when you consider that a mixing console from the 80s probably had a noise floor right around there too, and you don't really hear people complain too much about the noise. I thought at first it could have been a mismatch in what the Sound Canvas was outputting versus what the sound card was expecting in. But I don't think that is the case given that you really don't see +4dBu on Phono connectors so the Sound Canvas is probably outputting -10dBv which the sound card should have no problems with. I have three other guesses though. 1 - The jack is dirty which is leading to a shoddy connection. 2 - The sound card is picking up electrical noise from the computer. 3 - There is a gross impedance mismatch going on which is generating standing waves causing an increase in noise. I'd suspect number 2 and 1 to be the most likely going on. That adapter will work perfectly fine. The only real difference between 1/8" and 1/4" electrically is that 1/4" can handle more power and it is just a more robust connector. Otherwise it'll work. Though since nobody has mentioned I feel I should. If you do get something like the UR22 or similar then to record stereo you'll need to use two inputs as each input is inherently mono.

What exactly do you define as a wide mix? There are several definitions to what constitutes a wide mix, and everyone views it differently. I've been told I make wide mixes more than once, but I feel mine are actually fairly narrow. How I approach the entire situation of it is like this. I make my mono straight up wonderful. In fact I do the majority of my mixing in mono because I feel it ends up creating a better sounding mix on the whole as I spend more time actually focusing on the general vibe of the track and separation of the instruments. As an added plus when you've got a decent sounding mono mix doing the stereo aspect is super simple and easy. Really the big thing I think a lot of people assume when they're mixing is that they need to use all sorts of techniques to generate the width artificially rather than baking it into the mix from the get go of the mix. That is to say they're not really trying to figure out how to build a sound stage in which they can place the various instruments in relation to each not only in terms of distance but also pan. With that being said though there are certainly tricks you can play to generate more width than you'd normally be able to with just pan. You mentioned Haas delays which is one way. Another way is to use reverb. One thing I like to do from time to time is create the space with just normal reverb and then on certain instruments use a really effecty reverb with a stereo width plug set to the maximum then blend the reverbs together in stereo so they sound good. The cool thing is that when the track is collapsed back into mono often that really effecty reverb disappears or gets much quieter allowing the reverb you used to create to really come to life. In busier mixes something I like to do is actually put the effects on the opposite channel. So, for example say I've got a piano in the left channel. In the right channel I'll place the reverb with a short predelay. On its own it sounds weird, but in the context of the mix it actually sounds very good. It is kind of like a Haas delay except that when collapsed to mono puts the reverb on the instrument where you expect it to be. Another option is Automatic Double Tracking, which basically is as it sounds. You create a double track and hard pan things. It isn't perfect but it can work wonders. However, most importantly I feel is that you really have to utilize a combination of these if you want to get extreme width and have it sound good.

There are a lot of things you can try to do a call & response. But first off pretty much the advice of hard panning L&R can be very jarring if you're not careful with how you approach it. With that being said I'll give you a few ideas of how to get away with the hard panning at least kind of. So, the ear is a fickle thing that really pays attention to the differences in the L&R. So, a single instrument just chillin' out there will be picked up almost instantly and become extremely grating very quickly. Though not all hope is lost as there are some tricks you can play with the ear. Without going into the changing the composition one thing you can do is actually hard pan the reverb to the other side e.g. Guitar in the left reverb goes to the right. Now, it will sound a bit funky and strange on its own. However, in the context in the mix it'll work especially if you've got something else playing in the middle. I cannot tell you exactly why it works, it just does. The only real thing to be careful is with EQ. Another option you have is to actually alternate the call & response's sides. That is to say if you put the call into the left the first time then you put it into the right the next. Again be careful with EQ. But doing this helps create something more interesting sounding, but also prevents it from being too jarring. Personally, what I'd start with is hard panned ping pong delays because with the proper use of filters would first prevent the jarring aspect by blending the L&R more, but it would also blend the call & response a bit better because they'd be feeding into each other as it were. Like echoes in a canyon or something. This is what I'd personally start with if you want them hard panned.

I'd just like to chime in on this and I've mentioned it before in this thread. The reason for having a peak of say -0.8dBFS comes from the fact that when it is placed into a lossy codec the level can increase not only the RMS (though not really perceived) and peaks can be generated that will clip the playback system. All of these loudness acts are a way to not only ensure that everything has roughly the same consistency when it comes to loudness, but also so that during play you're playback system is going to be unlikely to clip. MFiT isn't necessarily about loudness, but more about not clipping the codec on playback, but that is the gist of things.

Sorry for the mix up Danthr & Strader. Danthr, When I read your post initially it looked like dB full scale. For that I apologize. Though the discussion has definitely turned quite interesting now talking about RMS (which honestly is a crap way to measure loudness because it isn't weighted), Crest Factor (quite geeky), and all that fun stuff. Though I think it should be noted of equal importance to this whole thing of limiting are the Equal Loudness Curves. AngelOwlCity, It is important to note the difference between the two especially for someone looking to use a limiter on the master otherwise when you convert to a lossy codec it is possible to clip the playback of the codec.

dBTP vs dBFS is not semantics. There is a very big difference between the two. dBFS is decibels referenced to full scale whereas dBTP is decibel True Peak. There is a fairly substantial difference in what that means. Not semantics and to say so is just wrong. Even in the picture of Ozone they say True Peak Limiter. I mean if you're going to reference something by name, the CALM Act and then state the maximum allowed peak is -1dBFS when it clearly says in the specification it says -1dBTP? That is something else entirely. The specification calls for dBTP with purpose as dBTP and dBFS can be completely different from one another, not necessarily that they will be because Whittaker-Shannon Interpolation Formula. Now, if you want to start talking about how to use limiters and their various functions and such then start talking with actual definitions of the various aspects. Like what is a ceiling? What is the threshold? What does an infinite:1 ratio mean? What does the release on a limiter do? Why do some limiters have attack times? That is just a start of the various things you can talk about with regards to limiters. From here it is a natural extension to start talking about, "Well why can't I get my mix louder? When I keep turning up the limiter it just starts to pump in a nasty way." From a question like this it comes fairly naturally that the discussion evolve into topics covering mixing, gain staging, headroom, & monitoring. Instead as Neblix pointed out you mostly rambled without really getting into the specifics of how a limiter works, why they work, and why they're used. Limiters have a very defined purpose and the approach on how to accomplish it is just as interesting as their uses. I'll admit my post was rambling a bit, but it was mainly focused on one thing dBTP. The beginning was mostly an addition to your original posts with some history and trying to further specify precisely what is a limiter.

I'm gonna agree with Neblix on this. You have some good points here or there, but ultimately mostly just rambling on because you don't fully understand what all the functions in a limiter do exactly. I mean you're right in that a limiter is essentially just a highly specialized compressor where in ideally it prevents every single peak from ever going above whatever you tell it. There is more than one way to do this. Back in the olden days of AM broadcast there was actually a clipper at the very end of chain. Why? Because of the limitations of analog limiters. Then things progressed with better circuits and the likes until finally digital limiters came about. These offer options not available in traditional analog limiters because digital ones can actually look at the signal before it reaches the processing and this can prevent or eliminate a lot of distortion. In fact it is almost better to describe modern brickwall limiters as distortion hiding devices because that is what they're really doing if you really start to think about it. So, yeah a limiter is ostensibly a compressor, but fundamentally it is a very different device with a very distinct and unique purpose. If you want some interesting things to think about look at the actual names of a few of the old and beloved compressors out there. The 1176 is a Peak Limiter. The LA2A is a Leveling Amplifier. The LA3A is an Audio Leveler. The Gates Sta Level is a Compressing Limiter. I just bring this up because there certainly commonalities between a compressor and a limiter and even some crossover, but they are actually quite different. Now, I have to say that you got the CALM act a bit wrong. It isn't a maximum peak of -1dBFS it is a maximum peak of -1dBTP or decibel True Peak. Without digging through the whitepaper to see precisely how dBTP is measured and calculated, it is basically a predictive peak of what the reconstructed waveform would be without actually reconstructing it. Of course the accuracy of this depends on quite a few factors (primarily interpolation but who wants to get into that whole nonsense? I personally don't want to). This is actually important because you mentioned Intersample Peaks and this is precisely why there is the hard limit of -1dBTP. See, Intersample peaks are 100% valid but they'll clip pretty much anything. It is actually not all that difficult to get peaks that are +6dBFS if measured in the digital domain. In order to prevent these huge spikes -1dBTP was chosen as it should only let a very a small amount go above 0dBFS (the clipping point of digital). This is also sound advice when you're converting to MP3 or another lossy codec as they'll often increase the level of the track, which is actually part of the MFiT certification. Remember that even though the DAW is reading -.1dBFS if you measure the signal in the analog domain it could very be above that, which is really what the whole -1dBTP is meant to cope with. Unfortunately, if you want to really dive into this topic then you have to start invoking how the analog domain behaves as well and that is where things start getting messy with dBV and dBU and calibrated converters and calibrated monitoring. It really starts to get a bit funky honestly. Not terribly complicated once you think about it, but definitely confusing initially.

As I think more about this particular question I feel that some more information should be on offer than there currently is in the thread. So, as I said I don't really notice much of a difference in the way digital EQs sound. That is pretty much universally true for me. That doesn't mean that there isn't a difference. I personally despise the way linear phase EQs sound and so I don't use them despite having Pro-Q and Equilibrium from DMG Audio. They both are capable of linear phase with the proper PDC reporting to the DAW. However, I don't like how the filter rings equally and what that does to the spectral balance of a signal. But that is just me personally, and that doesn't mean I haven't found a need or use of that in a mix from time to time. That is something that PEQ2 simply cannot do. So, for utility purposes with complex material having an extremely flexible, powerful, and clean sounding EQ is something everyone should have in their toolbox. Otherwise as long as the developer gave enough attention to HF filters in relation to Nyquist then pretty much any Parametric EQ will sound the same with the primary difference being functionality. As an example of functionality between PEQ2 & Pro-Q real quick. If I want to change the Left channel of a sound only using PEQ2 then I have to mult out the part into two separate entities and treat just that side with an EQ. This is something that I can do completely internally with Pro-Q. Though the need to do something like this is pretty rare, but what about M/S? Pro-Q can do that without needing additional plugins. That right there is extremely useful functionality. Now, you may perhaps being wondering about so called "modeled" EQs as you've probably read about them. Well, supposedly the EQ I use a lot, Nomad Factory's NEQ-1972, is a modeled EQ. In certain situation flipping the "vintage" switch does alter the sound a positive way, but more often then not I don't hear a difference. But I've got more than one modeled EQ. There is Pulse-Tec EQ which is a Pultec with the mid range EQ part as well. This is a digital EQ that just doesn't quite sound like a normal EQ, which it shouldn't given that it has a tube makeup stage in the real unit as the actual EQ section is passive (at least IIRC the basic topology). Most of the time with this just putting it on something makes it sound better. No EQ is even necessary. However, you can perform the classic Pultec trick with it for bass or kicks, and it sounds fantastic doing it. Though the reason for this is that this EQ models saturation. Yeah there are differences in digital EQs sure. But honestly for the most part if you're just using a parametric EQ you should be able to achieve most anything you want with it short of an EQ that models saturation. Then you're kind of SOL with just a parametric EQ as that isn't something they do. A quick tip on EQs to make them sound more natural or more like they're analog counterparts. Most of the time when it comes to boosting try wider bandwidths. For some reason our ears are extremely sensitive to even a minor increase. If you ever look at the transfer curves for passive EQs you'll find that the Q of the filters in them is proportional in that the more gain is applied the narrower the filter gets. Now, if you're cutting start with narrower cuts in the first place. A lot of old EQs actually have asymmetrical boost & cut transfer curves. We're not nearly as sensitive to cuts as boosts and as a result you can get away far more when cutting than boosting. Do you need a fancy, shiny, super featured Parametric EQ? Not really. Are they handy to have around? Very much so. Do you need a modeled EQ? Not really. Are they handy to have around? Very much so. I guess that is kind of the point of my post here. You don't really need a fancy EQ, but there can be advantages to working with an EQ that has a fixed set of bands & frequencies as they can be faster to work with (I'm in this camp). Certain kinds of modeled EQs do wonders for certain things. There is a reason the Pultec is famous and Pultec models will show you why. They're super smooth and big sounding. Seriously, give a try to some of the modeled EQs out there. OverTone DSP offers a free demo on their Pultec emulation, the PTC-2A. A very sweet sounding EQ in general to try is TDR VoS SlickEQ. If you want to try a great sounding free program EQ give BaxterEQ from VoS a try. Just start trying and see what you think.

Eh the differences between most any digital EQ simply comes down to functionality rather than sonic qualities. For the most part you can emulate any digital EQ with a fully parametric EQ. Just how it goes. Though there is something to be said for the speed at which you can use an EQ to do what you need. For the most part I don't use fully Parametric EQs when I mix, I typically use Nomad Factory's NEQ-1972 which is like a Neve 1081 EQ. I just like how fast I can dial in what I need with it rather than to get all fiddly with the various settings of a fully parametric EQ. Though I do use Parametric EQs when I need to get in and do something cleanly or surgically. For those situations I use ReaEQ for more general stuff and Pro-Q for the more advanced stuff. Just find something you're comfortable using and go from there. If you like to be speedy with EQ then perhaps looking at an EQ that isn't fully parametric may be beneficial. Something free would be BootEQ from VoS. It isn't a surgical EQ by any means, but it has a good overall sound and surprisingly flexible. If you like the way it flows over using a fully parametric EQ then you should try and explore different EQ plugins. But sonically? I've never noticed a big difference between EQs except in the bands and how they react to the sound. Compressors on the other hand I have a lot more to say, but that isn't the question at hand. Use the digital EQ that sounds good to you and jive with.

@Rozovian I never got to personally enjoy this wonderful game on the SNES back in the day. My first Metroid game was actually Metroid Prime. However, after playing that I went back and gave Super Metroid a go and frankly was blown away by just how good it was. Anyway, back on topic. Thank you for noticing that I deliberately put simpler sounds against more complex ones! Going slightly off topic-ish for a moment. Something I'm trying to do with this remix is not only a remix in my own flavor of strange hard-to-define-genre electronic music, but that in my sound palette try to play on the Metroid OSTs in general or at least the ones I've played. For example the Theme of Super Metroid is a fairly dark piece with some very nice orchestral elements that are far more traditional (well as good as you can get the SNES to do orchestral stuff anyway). But then you take what I'm remixing here, Upper Brinstar, and it is kind of got this nice Electronic Vibe to it. However, compare Upper Brinstar to say Arrival on Crateria and it is a fairly stark contrast in the difference between not only the sound palettes, but also voicing of the instruments themselves. In fact the entire OST of Super Metroid is like that. Some tracks are simply heavenly in just how traditional they are, but then others are very dark and foreboding at their very core and not traditional in any sense. So, in this remix I'm also trying to pay homage to that disparity in the OST. That entire breakdown at 3:23 is inspired by darker parts of the OST as an example. Another is in the sound choice I'm using. Simple leading sounds with more complex backing & textural elements. The percussion sounds while very simple are far more organic sounding than most anything else (though they are electronic) and I'm contrasting those against far more electronic sounding instruments. I'm just glad that someone noticed it. Seriously, thank you! I hear ya on that, but I'm gonna keep it at the length where it is. I know there are some moments where it the composition is stretched a bit thin. I really do and I'm trying to figure out ways to overcome that. Timeaus pointed out something I hadn't considered before yet, which is to use the bell texture more melodiously. Which at a few moments I think will help with filling it out some. Also, if it makes you feel any better I did a nearly 16min track before, not a remix though. I really started to get into music with Trance. As a result I'm not really against longer tracks that are more droning and repetitious. But it can't be without purpose you know? I've tried number 1 before and it ended disastrously for me. I just don't think like that when it comes to composition. I can hear a melody and how to build upon it in different ways, but I cannot compose like that. As I start a composition I'm immediately in the sound design as that is also a part of the composition itself to me. I cannot separate the two like I can composition & mixing. Composition & sound design to me are too integral to one another to be able to separate. At least in my brain. I've tried it more than a few times and each time I never liked the finished results because they were never what I was hearing in my head before hand. Number 2 is really me separating composition & mixing. It isn't common that my final mix sounds just like compositional mix. I know this thread is supposed to have a single track in it, but as an example of this is I did a remix of Blue Fields from FFVIII. Compisitional Mix versus the Final Mix. In the composition I'm much more focused on getting instrument voicing precisely where I want things to be. I'm not really concerned if they necessarily fit together in terms of spacing, spectrum or things like that. I already do that for pretty much every track I make these days. However, for me separating out arrangement and sound design isn't something I can really do. They're just too intertwined in my brain. I know a lot people work like that, but if I hear compositions that way I immediately start hearing instruments that will fill up the track. Wow, I'm getting really long winded about this lol. But yeah number 1 just doesn't work for me for reasons that are hard to explain. Sorry But it still is a great piece of advice for folks. Thank you for the listen and comments!

More expression and more subtle emotion in the opening instrument? I'm not really sure how I'm gonna do that one. I mean I can add some stronger vibrato to it, and make the filter open a bit more, but I'm kind of running out of modulation slots on it. Even with that being said I'm not sure what you mean when you say some more subtle emotion to be honest. To me it emotes more than enough. I can layer it with something else, but then I could possibly run into a problem I've had more than once when making this track, and that is it as I add more instruments I have to be careful with what I add as it may do something I don't intend for it to do. So, yeah you're gonna have to be a bit more descriptive in what you mean exactly. Now, I can try switching the instrument out at one point and what not. But I already tried that originally when I added it, and found it to take away what I want that lead to do. The reason I've kept that lead the same for so long is because it is meant to be singular and feel small & insignificant. I want it to be almost a beacon of venturing into the unknown. So, I can see adding some stronger vibrato to it to make it sound a bit shakier at certain points, but at others I want it to have almost an absolute resolve. I don't disagree about the bass either. However, the problem is I don't want it to drastically differ for the simple reason of changing it too much radically changes the way you hear the track and how the textures work. I've been working on varying it, but it isn't a fast process. The last one I see what you're saying and I tried a piano already at that part. I honestly didn't like it. I felt it took the track into a territory that wasn't what I wanted. I've tried other things, and I'm not actually against the choir idea. The biggest problem with that is I don't actually have any good choirs, and I'm not about to drop the money on some either because I simply don't use them often enough to warrant the money into a good library of them. If you have some suggestions of some freebies I'll gladly give them a try. However, I can experiment with some other options to give a similar kind of effect. Look I don't disagree with the notion of making the second half of the track different, and it something I'm still working on.

All right! - What if you added a slight filter envelope on the choir-like pad you start using in the beginning? I hadn't considered that, but at some different points I had considered doing something similar. I'll give it a go, actually what might work in a more interesting fashion is to use a shelving filter and slowly open up through the intro. I have a couple of other thoughts I could do to make it a little more interesting, but this just confirms a suspicion I had. - Is there any way you can make the bells near 1:27 more tonal? Or are they just something textural for the background? They're more textural in nature, but I can make them more melodious with some layering. At least little bits and pieces that way. I have to be careful with what I add or I might find things getting to melodically dense. Most of the things I've added have been more for texture rather than melody. However, there are certain moments that come & go that I feel need something more. Perhaps giving some melody here or there will really help. - The repeated bass pattern repeats a lot. That's really what I'm most concerned about. But you do have some variation on the bass in the second half which helped (for instance, near 5:10). You and me both. As you noticed I did do some switch up towards the end. I'll work on it and try and come up with some more variations. The thing is that I really don't want to overdo it either. In fact I'm finding that to be the really hard part about this. Even a small change typically has a very dramatic effect on something I didn't want it to. Another thing I'm really concerned about that you didn't mention is the perucssion. It is for the most part the exact same thing all through the track. - How about a space-like sweep at 3:24 if you have one? One example I can think of is the one used for this track at 0:21. I heard some sort of sweep there, but I think it could be more obvious. The later instances of that sweep you have now, though, I would keep, though maybe you can thicken the tone up a bit with some slight distortion? I could totally do something like that. In fact I even have an idea for it. Actually this is a legitimately good idea. Though I'm gonna try and make it a bit unsettling or creepy sounding because that is the whole point of that section. But even so you've given me an idea. - 3:59 can have a transition, so some sort of long reverse can work (like a cymbal, noise sweep, etc), and whatever else you may think fits if you add that reverse. I did something in another track that would probably work here too. In that it builds up to almost a cacophony of sound and then drops to silence essentially. I was already starting to kind of lean this way because I feel that whole section afterwards needs something to better transition it to the end. Pacing is something I'm concerned about. That is part of the reason for the entire build up to the breakdown. I mean honestly I think there is good flow up the breakdown honestly. There are other issues of course with that section, but I don't feel like it has anything to do with pacing. The real problem I'm having is trying to fill out that second half with something that not only separates it from the from the first half, but also stands perfectly fine on its own. And this is where I feel the majority of the pacing issues stem from. The front half of the track flows well, but that second half doesn't flow nearly as well. As a quick aside. I know the mix right now has a multitude of issues, but I actually compose & mix at different stages. The overall track will sound different tonally before I'm done. Right now I'm just focused on getting the flow, instrumentation, and voicing where I want it. Thank you kindly sir!

This one gets a little hard to properly describe but I'll point out what I can. In the intro listen to the drums and how they have a particular kind of ambiance to them. Then at :40 when the guitar comes in it has reverb and space, but it doesn't match the space that the drums have. The tonality of the space is quite different and stark. Now, the way the track is pieced together works for quite sometime, but it starts to show problems again around 1:39 with the winds coming in and really shows up again at 2:09. Then at 2:36 it comes to light again with the bells. 2:57 is another example of the space of the various instruments not matching particularly well. Though at 3:10 things match up wonderfully, which could be for numerous reasons which include masking the issues or it could be the conversion to MP3 is wreaking some havoc there, but well that is how things go sometimes. 3:41 when the guitar comes in it just doesn't match at all with the other instruments. Now, let me be very clear about something. What I'm pointing out is quite a nitpicky kind of thing. Most people probably won't even notice and heck this track is so darn fun that I'd be surprised if anyone really notices unless they're looking for it. Now, you can have instruments with different reverbs; however, the reverbs have to be able to blend together in a harmonious way without causing any issues of space. This requires a certain attention be paid attention to the space being given to the instruments. It is something that for whatever reason really bugs me when it doesn't sound right. Sorry So, the drums are multitracked. I don't need to know what they're recorded with just that they were multitracked. If that is the case then you have a lot of options to treat them. Personally, I've always found with live drums that time spent doing the balance of them in mono with only one speaker on makes it much easier to come up with the balance and to detect any phase issues (doesn't sound like there are any). Some more options when it comes to dealing drums when mixing is instead of relying on a gate is actually automate the volume of the track so that it only plays when you want it to. If you need a little more "power" from the drums then you can use parallel compression to help with that. If you need some more "punch" from the drums then consider what you want to have more "punch" and use EQ, Compression, and Transient Designers to help give that particular instrument a little bit more. Also, consider buss level processing. A simple program EQ can work wonders sometimes. If you're cymbals are coming through too hard or something like that and you're finding a compressor is taking too much life out of them then use a de-esser. Think of a de-esser more as a HF Limiter rather than a de-esser. You'd have to tell me more specifically what you feel you're missing and I can give some more advice on approaches. Now, if you've got issues with needing to really dry out instruments in order to make them fit then you've got some volume, pan, and EQ issues to work out. Typically the reason for not being able to put reverb on an instrument is in really fast paced tracks where the reverb will cloud things up. In that case delays are normally used. Another reason is that the instruments aren't being given their proper place in the overall sound stage. When I mix I spend most of my time balancing the mix and basic EQ to enhance clarity of the various instruments in mono. This is before I even get into stereo. Everyone works differently, but it is something to consider. Having said that there are some things you can try with reverb to help blend the instruments a bit better together. Try EQing into or out of the reverb. Now, I'm mainly talking about a HPF and/or LPF here. The idea here is that there is particular set of frequencies that our ears are really going to be sensitive to in the reverb that give the real sense of space given by the reverb. Basically, send the instrument through the reverb like normal and then adjust the filters until you find a certain point of HPF & LPF where the reverb just pops. I cannot describe it any other way. You'll have to experiment. This isn't a new technique, but a rather old one. It is a way to eek out a little more clarity in the mix. It can help with cloudiness of instruments by allowing them to pop through more readily while taking up less spectrum. Another thing to try is actually sidechaining the reverb. When this is done it downplays the reverb when the instrument is playing but allows it come out more when it isn't. Again giving you more control to manipulate the space at will. I cannot really think of anything else to add currently. Sorry, that the post is a bit scatter brained. Been a long day for me. Feel free to ask more questions or really much of anything. I'll be around trying to figure out how to finish my track heh.

Drummer/Sound Guy? Well that right there is yer problem *j/k*. Serious now, are the drums multitracked like overheads, kick, snare (over & under), room, hat, toms, etc . . ., or are they just a stereo track? This is important because it really limits what you can & cannot do to them. If you've only got a stereo track to work from then EQ is going to be your best friend to get the desired clarity. If it is a multitrack then the focus should really be on building the drums from the overheads. They're what give the drums most of their sound. The kick & snare mics are really only there to add to the overheads. The limiter Timeaus is talking about is something done on the Master Bus and not really on the drums in particular. Though this goes a bit beyond what you're talking about you can limit the drums for a little bit more "oomph" if you want. But that really starts going beyond the scope of this topic. I know I didn't mention this earlier but there are some strange choices when it comes to the space given to various instruments. I mean they work together, but it is just that they sound a bit disjointed from one another because some have reverb whereas others don't quite. It sounds like they were recorded in the same space which gives them some cohesiveness, but not enough for them to be considered in the same space. As I said earlier without the multitrack or an uncompressed version of the track I'm pretty hard pressed into giving you much more specific advice. As an aside I can get very technical when talking about mixing because it is something that I've decided to really focus on. I've mixed a few Rock tracks for practice before so I do know first hand some of the issues that are going on here. Feel free to PM me if you've got specific questions that you'd rather see answered in a more private manner or feel free to post them here and I'll do my best to answer them. I do know a thing or two about mixing.

Yet another revision. The biggest changes here deal with an alternate percussion pattern and a few new bass patterns. In the first part of the track they're mainly there for accenting transitions. In the second half they're there to help add some more movement. I'm just trying to decide if I've got enough in the second half or if it needs more. See, I'd like to keep the second half a bit more textural with a more pronounced & contrasted melodic nature to the first half which gets into a lot more layering and creating a denser soundscape. I want to contrast the two. The problem I'm trying to solve is whether I've accomplished this and put enough into the second half for it stand on its own. Listen or download Revision 8

Timaeus222, I got that you were simplifying, but when the way you simplified it doesn't make any sense because it doesn't have anything to do with that. In general the knee of compressor can really be thought of like a car accelerating. You start at zero and then reach your desired speed. The ratio starts off at 1:1 and then gradually ramps up whatever you specified for the ratio. Also, I wasn't saying you're were calling compression evil. It just seems that a lot of folks view compression as an evil thing, and that makes a lot of people hesitant to use it. I wasn't talking about you and the overcompression issue you pointed out (which I do agree on) or anyone else for that matter. It was merely a comment to try and encourage the use of compression in a responsible manner

I'm sorry, but that isn't what a knee on a compressor is at all. The knee on a compressor is directly related to the ratio and deals with how gradually the compressor applies gain reduction via increasing the ratio to the incoming signal in relation to where the peak is compared to the threshold. So, it would sound like it is applying less compression because in general it is applying less compression on incoming peaks until the signal is at the threshold where the compressor is applying full gain reduction, but it has nothing to do with the tendency to react to loud peaks. That is still strictly the attack, release, threshold, & ratio. What the knee does is help blend between full compression on peaks versus low level peaks being slightly compressed. LBoC, It is kind of hard to give you any advice on how to cope with this without actually having the multitrack or uncompressed (i.e. no master level processing) version of the track. What i can say is that you've got too much going on down 150-400Hz. There are just too many instruments down there eyeing for your attention. Additionally, the bass is actually quite soft and could really use with some EQ to really bring out some of the finer details. From what I can hear if you were to say let the bass take the low end (sub 60-80Hz) and if you clean up the mud some and then let the kick run the 200-400Hz area that will help give the track some low end punch. If you still find there isn't enough detail in either of the instruments then start looking quite a bit higher in frequency. For the kick try bringing out the beater at like 5-7KHz. For the bass try bringing out the strings in the 2-3KHz range. That is at least one option. Another option is to actually sidechain the bass to the kick. There are some other options if you want to explore them, but I'd really just try and start with the old classic of letting either the bass or kick control the low end and then selectively controlling where each has dominance in the spectrum to bring out the qualities you want. On a really serious note don't be afraid of getting really heavy handed with EQ when doing this either because that may be your best course of action. Another thing to look at is using compression to do some transient shaping. Listen to the kick in the beginning. It is played quite softly, but it still has a good amount of bite to it at the beginning. You can use a compressor to really control how hard the kick hits. The same applies to the bass with compression. You can gain a lot by just compressing the bass. Don't be afraid of compression; it isn't evil.

I gotta say this is quite a bit of fun. Great mood, energy, feeling, etc . . .. Instruments are well recorded, the spectrum is quite nicely filled out. I mean there really isn't a lot to complain about in those regards. Everything is discernible in the mix. Having said that I feel that mix falls a bit short. There are certain moments that are simply fantastic like 1:16 - 1:36, 1:39 - 2:04, 2:09 - 2:16 as just a few examples. Everything just gels and it just works fantastically. However, there are moments where the track just doesn't live up to those like 2:19 - 2:27 or 3:42 - 4:22. The mix falls in on itself at those moments. The drums lose all their power & impact. It is at those moments you want the track to really come out and be driving. It is clearly evident by how they're played. I'd be more forgiving if there was more going on that required more spectrum & power. But really that isn't the problem. The way this is composed & voiced tells me that this shouldn't be the case. Just my observations. But seriously this is a really fun cover. Lots of great moments and honestly I can mostly overlook the mixing bits (there is quite a bit more than what I commented on, but that is neither here nor there right now) because it is just that much fun. Edit - Seriously, this track is so good I can almost overlook the mixing issues. That is really saying something for me.