|
Keefaz posted:It seems I accidentally posted this in the old thread: I posted a pretty lengthy reply in the old thread in answer to this question. In adition to that post: if you are compressing it hard enough with a high ratio be aware that the compressor only compresses sound that peaks above the threshold. So if you are playing very quiet notes, and these peak underneath the threshold then the compressor isn't affecting these sounds in anyway and that may be why they are getting lost in the mix. However, if this problem is severe enough it isn't something you should really 'fix' with a compressor as it indicates that you have alot of instruments swinging around in your mix that share the same frequency ranges in the same phase. So if you repeatedly do not get satisfactory results with a compressor I would suggest remixing or rerecording the relevant parts. I would not advise fixing any severe problems in the mixdown with compressors as they will never really go away unless you address the root of the problem.
|
#
¿
Feb 2, 2007 13:38
|
|
|
|
| # ¿ Apr 27, 2024 03:30 |
|
|
Keefaz posted:Thanks for that reply in the other thread, I just read it eariler this evening it really helped get my head around it. The more I looked at the wave and listened, the more it became obvious that I had some peaks that were forcing me to keep the track volume down. I used a built-in preset that compressed at -6db with a quicker attack and that really help to level it out. Thanks, man. A really commonly used ratio for lead instruments is 4:1 but I never understood why this was a rule of thumb. Like any dynamics processor you really need to look at the raw waveform and determine the amount of compression that is required on a per sound basis. If I am finger picking and I wan't a nice pronounced attack phase, I'll usually drop the ratio quite low and bring the attack envelope up to anywhere between 20 to 50 ms on hard/medium knee. It depends on the length of decay on the note and how pronounced the attack is. Having a 20 ms attack envelope on a compressor means that it doesn't do anything for the first 20 ms then it kicks in, which means the very sharp plucking bit at the start isn't compressed but then the compressor kicks in for the decay phase of the sound which is compressed. The knee determines how quickly the compressor kicks back in (or rather the linearity/non linearity of the curve). The ratio I tend to think depends on the sound being compressed so a ratio of 4:1 at a threshold of -30dB would mean that for every 4 dB over the threshold, the compressor will apply 1dB of gain reduction. So the effect of the compression ratio is wholley dependant on what the compression threshold is. if you have a 6:1 ratio you need to specify what the threshold is to get an idea of what the end result on the sound is. Although if it sounds better straight off, you are definitely heading in the right direction. And ultimately, the end result is about how good it sounds, not the numbers you are crunching. Hope you get the compressor thing sorted and start banging out really great guitar lines that just sit exactly where you want them to in a mix.
|
|
#
¿
Feb 5, 2007 14:17
|
|
|
From what I can tell from people that have problems with compressors over at Tranceaddict, its that they are twiddling knobs but it doesn't help because they have the ratio and/or threshold set to 0dB and 1:1 respectively. Which means that the compressor is doing absolutely nothing. So a common confused question that seems to crop up is: I bought this compressor but I can't tell if its doing anything to my signal? And the answer is 9 times out of 10, that the compressor is in fact doing nothing to the signal because you haven't set a threshold or compression ratio. Another thing that often throws people is the post gain dial. In some compressors applying post gain really changes the signal. Off the top of my head, those Blockfish compressors. In most cases though its just there to compensate for really heavy gain reduction and transparency is sometimes (often?) a desirable thing. So you can get really weird situations where people stick compressors into the signal chain with a threshold of 0dB but masses of post gain (which makes it into some sort of...preamp...or something). Theres no actual compression going on but some think it improves their guitar tone no end. Presumably because the output signal is just louder. There is of course no substitute for fiddling around by yourself and working out what does what and how it sounds. But at the same time, you can save yourself alot of confusion if you start early on knowing that a compressor with a 0dB threshold doesn't do anything. Beyond that you can save yourself alot of overcompression problems and alot of confusion by knowing what each variable on your compressor does and how they all interact. If you confused, I can explain so its all cool  Compression is one of those things in music production which is really easy to visualise so you can easily think of it in terms of waveforms. If you have a wave editor and a standalone compressor you can actually see what its doing. And after a while you can look at other single shot samples of things like bass drums and snares from a certain drum machine and you can tell how and where its been compressed (and by how much) by comparing it to a straight sample from the same machine. If you really get it, you start to realise that compressors can be used as sound shaping tools and that you can use them in conjunction with equalisers (provided you have a mixer with decent routability) to achieve the same thing that transient compressors and multiband compressors do. So if you know what you are doing, you really don't need Waves C4 or anything like that. As for good compressors, if anyone is interested in picking up a nice sounding software compressor, Voxengo Crunchessor is by far and way the best of the sub 50 bucks one. Its no contest. Further up that Kjaerhus GCO-1 compressor is spectacular for the money and probably the best software compressor I have ever used that doesn't require a DSP farm. The Sonalkis one is just as good but costs more. After that, you are in DSP territory and most of these compressors sound great, like that UAD-1 comp and the Sony Oxford ones. The UAD-1 Fairchild is ok I guess. You get some people going as far as saying that its rubbish which is...hmmm. It sounds like a broken tube compressor. Which is a good or a bad thing depending on what you want. Its not an all purpose compressor like Crunchessor though. Crunchessor has such clean post gain that you can use it like a preamp almost. I love it. For 35 bucks it has to be one of the best bang/buck plugins money can buy at this point in time. WanderingKid fucked around with this message at 18:42 on Feb 5, 2007 |
|
#
¿
Feb 5, 2007 18:39
|
|
|
Yeah. When you think about it there are no post processing effects that necessarily make a sound better. Thats subjective and it depends on context. You can't for instance cut, compress and EQ a lead guitar track the same way in 2 different mixes and neither would be 'better' - its not a question of being better but more a question of appropriateness. As for dynamic range. Dynamic range is really important for harmonically rich instruments. A violin for instance when bowed is very rich in harmonics and the arrangement of them is really very complicated. If you run a spectrum of a violin (realtime graph of amplitude on the y axis, frequency on the x axis) then you can actually see where the harmonics are. They look like a series of very tall sharp hills and valleys. To hear what effect they have on a sound, open up a .wav of a violin or a cello or something. Now using a paragraphic EQ make a very sharp notch or load a notch preset. Turn the volume on your speakers down alot before you do this. Turn the notch upside down by bumping the gain fader up to maximum. This will accentuate a very narrow band of frequency and you will hear this above anything else. Now sweep this notch across the entire audible range. Do it slowly and you can hear the harmonics being picked out - Do not do this with your speakers on full blast because you will DIE. Also, if you have trouble with identifying harmonics in instruments, this trick is a really good way to pick out harmonics and other interfering frequency ranges. Once you identify them just turn that notch upside down again and adjust Q to taste. Compressors are just tools that reduce dynamic range. But the point isnt to absolutely squash as much level out of a sound as possible. I mean you cant really perceive depth very well based on sound if everything you hear is constant and all of it is about the same loudness. Extreme compression will tend towards that. You also lose a sense of the complex harmonics of an isntrument beyond a certain point and all you will hear is mush. So why would you ever want to use a compressor? Seems like more trouble than its worth? One reason is as Yoozer says - to get a sound to 'fit' in with other parts of a mix a little better. The human ear is really quite good at detecting 'loudness' and especially sudden changes in the loudness of a sound. So if you have a mix and there are parts of it where the amplitude spikes suddenly you tend to pick up on it as being discontinuous and its annoying. Compressors can be used to manage sudden spikes in volume (In some situations these are called limiters). Compressors are often used for this purpose in Mastering Studios where the aim is to make slight discontinuous elements in a mix work together better. Secondly, some compressors 'colour' the sound that they process. I mentioned that UAD Fairchild compressor. Well thats an emulation of the real Fairchild which provides its post gain from a tube amp. It costs a bomb to buy a real one now. Tubes distort when you overdrive them and alot of people find this distortion quite pleasant sounding. So another reason for using a compressor is the combination of gain reduction and a coloured output signal. I think this is where some people describe the effect of a good tube compressor as having a 'warm' pleasant sound. Thirdly, you can use compressors to shape transient sounds. Whats a transient sound? Its a very sudden sound with a very sharp attack phase. A quick snare hit would be a transient sound because theres a massive, almost immediate spike in amplitudeas the stick contacts the drum and then the amplitude diminishes rapidly as the membrane of the drum vibrates. The whole sound lasts for a fraction of a second. Dance music particularly has alot of stacatto, transient type sounds - large bassdrums, snares, closed hihats, claps etc etc. Well with a compressor you have a tool that can reduce amplitude selectively. You also have an envelope on your compressor which determines when the effect starts and stops. So if you have a bass drum and for some particular reason theres a huge horrible clicking sound at the start of the drum, you could use a compressor with a high threshold, massive ratio, zero attack and very short release (say 10ms). Lets say the drum itself lasts for 0.6 seconds. What this would do is squash down that initial *click* but the compressor will release before the drum has finished playing. So this will make the initial click seem quieter. In this situation lets say you want a tiny bit of the click to come through, but not all of it. No problem, you could increase the attack time ever so slightly to 2 or 3 ms. Short enough to let a tiny amount of click through before the compressor slams in and starts applying gain reduction. If you have a wave editor you can see a compressor literally changeing the shape of a wave when you fiddle around with these settings. Fourth. You can use compressors with an extremely high ratio (i.e. 30:1) and a very high threshold (i.e. -0.5dB) to prevent clipping distortion where it would otherwise occur. A compressor with these kinds of settings is more often referred to as a limiter. Club soundsystems often have a limiter at the end of the signal chain to prevent sudden spikes in amplitude from destroying their amps. Fifth. Simply, loudness. If you have a transient sound like mentioned above then the first part of the sound is very loud but the rest of it is necessarily quiet in comparison. The loudest part of a 909 bass drum is nearly always the initial click, then it gets much quieter before fading out quickly. This means that when you normalise it to 0dB, the loudest part, the initial click will be scaled up to 0dB and the rest of the waveform retains its proportion in relation to the initial transient - i.e. the rest of it is louder but still way quieter than the peak. Now if this sound was massively compressed by reducing the dynamic range of the initial click, you will actually end up with a much much quieter sound. I mean you have just squashed the amplitude of that click right down. But you can increase the post gain knob on your compressor and apply poo poo loads more gain than you could ever get without compressing and not clip. Uncompressed, applying this much gain would just cause the initial click to soar past 0dB and distort. So you can use a compressor to make a sound unnaturally loud. Depending on the compressor the amount of post gain can also change the output signal, sometimes noticeably. Those Blockfish compressors do this dramatically (they are free so check em out) and I think this is the big reason for why they are popular. Conclusion: There should never be a situation where you 100% should compress a sound. However you can get a variety of different effects out of them depending on how you use them. The key to using comps well is to know what effect is appropriate, where and to what extent. If you take a recording of Ralph Vaughan Williams' The Lark Ascending, import it into Cubase and slam a bloody great compressor on the master bus with a 2:1 ratio, -60dB threshold, 0 attack, 0 decay, Hard Knee I guarantee you that you will make the original recording sound shitter in every possible way. Even if you had one of those amazing Neve compressors. Because its just not appropriate, and compressing everything indescriminately rarely works well. Yoozer is right on the ball when he says you completely could screw up the most beautiful singing voice on the planet with inappropriate use of a compressor. Even a good one. compressors won't necessarily make your guitar sound flat out 'better' all the time - hell some poor choices can make it sound alot worse. In the right place at the right time though they can accent a sound subtley or make it sound like it blends in better with a mix. And those tiny little details can make the difference between an OK song and something really great. WanderingKid fucked around with this message at 18:39 on Feb 6, 2007 |
|
#
¿
Feb 6, 2007 18:23
|
|
|
RivensBitch posted:An often forgotten and unused, yet highly important feature of compression is the key input. You can actually trigger the threshold of your compressor with another instrument. Let's say you have a part in a song where the bass is playing long, deep and sustained notes that eat up a lot of low end. Often these type of lines make the kick drum dissapear, especially if your drummer is playing softer during these parts, as he may be naturally inclined to do. If you put a compressor on the bass and send the kick to the key input of the bass compressor, you will compress the bass ONLY when the kick drum hits. If you play around with the settings you can make this subtle, so that it doesn't sound wierd but instead naturally allows the kick to poke out only when it needs to. This technique can also work well to let a vocal sit on top of guitars that may be a little too muddy or thick, eating up the vocal range. What you are describing is a mixer routing trick or the use of a side chain input (if you mixer has them). You can side chain all kinds of things - like delay plugins when you want a singer's voice to echo when the vocals duck out but you don't want the slap back delay to make the words unintelligible. You probably already know that though. I use FL Studio which has a pretty basic mixer - not alot of routing and no side chain inputs  So I can only simulate this side chain effect with a gate and automating effect sends.But I guess alot of people use it with compressors - 'side chain compressor' must be one of the most popular search terms on the TA production forum so theres a big deal about it. That pumping effect you describe is also used in EDM with a side chained compressor. Benni Benassi really rinses this trick and the gain reduction is so dramatic that you can hear it fade in and out rapidly. Some people love this effect. Others hate it. Side chaining comps is used all the time when a bass note strikes at the same time as a bass drum, normally in 4 to the floor dance music. I should have included it really but I just sort of forgot about it. And I would agree that a number of drum machine sounds have really boring dynamics. Most synthesized bass drums are sine waves run through an LFO set to trigger in an envelope. LFO modulates pitch of sine wave using a sine wave at 180 degrees phase. Thats the basis of a very crude 909 type kick drum anyway. You could look at this in a wave editor and its basically just an exponentially diminishing squiggle. Its so uniform that in dance music, it is often common to layer several different types of sounds (including hihats and snares as well as other kick drums) with this boring 909 type drum then bounce the whole thing as a composite sound. Its more interesting dynamically and harmonically that way. I think this has given birth to the 'Alphazone' bass drum which for lack of a better description sounds a bit like a gunshot going off. But with more bass. These turn up alot in hard house and the like, where they are compressed even further. Christ... WanderingKid fucked around with this message at 10:43 on Feb 7, 2007 |
|
#
¿
Feb 7, 2007 10:31
|
|
|
Yeah thats what I am getting at. The thing is certain mixers won't be able to do this. Im thinking mostly in software. For instance I have FL Studio and Voxengo Crunchessor which has a sidechain input. However it is greyed out when you run it in FL Studio's mixer so you can't use it. The sine wave kick drum thing was just how to synthesize a simple bass drum. Theres no compression involved in the process. It was just to demonstrate how boring they are dynamically. Thats just the basic process of building one in, for instance soundforge. Another example would be the basics for synthesizing analogue string sections. Basically its 2 (or more) saw wave oscillators, LFO modulates pitch of both saws in opposite phase along a triangle wave. Modulation amount is low and fairly slow (to simulate the natural vibrato that a violinist would perform). Output is rectified. Beyond that you can do anything to make it more convincing or less as the case may be. The TR-909 I believe doesn't add any noise source. It does have a ROM bank of samples which Roland have never disclosed. Thats one reason why you rarely see TR-909 clones - without those ROM samples it is very difficult to emulate 909 sounds - the bass drum and snare are the most difficult.
|
|
#
¿
Feb 8, 2007 16:34
|
|
|
ParthenoNemesis posted:I've been reading through this forum and trying to find something that already covers my question, and this is as close as I've come to finding an answer. Even so, I still don't think it answers everything for me, so if what I'm about to ask has already been covered somewhere, I do apologize (I did look!). Thats not a compressor issue. I have no idea what would cause that. I use soundforge though and I know it outputs about 6 decibels louder than bouncing wavs and mp3s out of FL Studio's mixer. But then FL Studio outputs pretty cold so I would expect that. Also, what are you playing the mp3 in? Winamp? Because winamp not only has its own gain control but it is affected by Windows Sound and Audio devices volume. You can check this in control panel. for instance, I have my internal soundcard set to play back windows system sounds and things because these blow my head off if I use my Delta 1010 on +4dB. So everything in windows (winamp, windows media player etc) always outputs way quieter than in FL Studio and Soundforge which use my 1010 instead. Try checking in control panel and seeing what the gain faders are set to.
|
|
#
¿
Feb 8, 2007 18:58
|
|
|
AtomicManiac posted:All I've got is that Jam-man, and a Warez'D FL studio and Cool Edit Pro v2.1(?) I would like to start recording just guitar parts, but eventually getting into the whole production of music thing. Where do I go from here? Eh...Well you could start by actually buying the programs you are using. Then you get an instruction manual amongst other percs (such as access to Imagelines technical help forum).
|
|
#
¿
Feb 9, 2007 12:57
|
|
|
AtomicManiac posted:Any Recommendations for the two? And what sort of price range are we talking (for both) I got Ram at around 100 and an External around 100-150? I'll stick up a little audio interface list tommorow, arranged by pricepoint. As for monitors. This is really subjective and you can spend a couple hundred or a couple thousand. Nearfied monitors largely reduce the problem of room acoustics because you monitor with the sound source very close to your ear. But it doesn't eliminate room acoustic problems and they can still make expensive monitors sound terrible. Other things like monitor placement can really change the sound of them. I have a pair of Dynaudio BM5as which I think are quite good. I ended up auditioning a whole range of monitors from the Alesis M1 all the way up to the ADAM P11a. I really liked the P11a. I kind of hated the M1. But you mgiht think differently. They all sound quite different and the BM5as I bought sound quite different to the P11a even though they are in the same price bracket. I thought this was disconcerting but it basically means you shouldn't take any advice on buying monitors except go down to your local retailer and give them a listen with your own ears. A note on some of the more expensive monitors - they tend not to have gain controls on the monitor itself so volume is controlled entirely in your software mixer. So be careful because things like windows system sounds can blow your head off if you flip all the +4 switches and pay no attention to the fact the output gain on your software mixer is maxed out. For convenience sake I would tend to go with active nearfield monitors. My BM5as are 50watts per channel I think although on the unit it says max 90watts. Either way, they go freaking loud. So if you want loudness I wouldn't pay a whole lot of attention to wattage beyond a certain point. Those KRK V8 monitors for instance have 250watt amps and you probably won't drive them to capacity when they are sitting 2 metres away from your head. They also made my ears hurt but you might have different experiences. One of the things I liked about my BM5as though is that you get a really amazing sense of direction from the tweeters. Its hard to explain but if you set them up symmetrically and keep your head still, then play a continuous sound, try moving your head a couple of inches to the left or right and you can feel the change in direction. Because of this, I keep moving my monitors around because I never sit still. Also, panning and stereo widening effects sound horrible when you push them too far which I'm not sure is a good or a bad thing. Also, owing to really lovely problems I had with Digital Village I have experienced what these speakers sound like when the low frequency driver doesn't work in the other, and when the tweeter has blown. So I have heard the tweeter on its own and the woofer on its own. I cannot for the life of me figure out where the lower mids come from and have concluded that these speakers are top and bottom heavy. I still like them though.
|
|
#
¿
Feb 12, 2007 19:02
|
|
|
Well, you can't produce what you cant hear. During hte time my BM5as were back at base I produced for about 6 to 7 months on a pair of free headphones I got with the Guild Wars special edition. Its the only thing in the box I still use. Well theres absolutely no sub bass response so you can play a tune like Hybrid - If I Survive, which has a hefty sub bass but you just won't hear it. So when you are producing and you want some sub bass presence you are kind of flying blind. The tendancy I used to do was amp up the bass until I could hear it. But if you can hear bass on those guild wars headphones then you know you have done something horribly wrong - just play your mix on a soundsystem with a sub and you will realise you have hyped the bass over the top. Now. You definitely can produce using these headphones. I've done it for 7 months and I do believe that given more time to work with them I will get used to this. I have started to get used to them already. The key is to know what your speakers/headphones sound like - where they hype frequencies and where they are deficient. But theres no doubt - a good pair of monitors in a good room with decent acoustic treatment - it makes life a helluva lot easier. You won't hype bass accidentally for one. Also, you can mix louder without the sound turning to mush. Once you push the volume up on guild wars headphones, the bass and treble starts to break up horribly. You can get used to it but you need to constantly A/B against different sound systems to get it right. All told, mixing on poo poo headphones is doable but it takes alot longer and you need to change your working habits. If you are starting out, its always a good idea to get monitors you like listening to because they don't fatigue your ears anywhere near as quickly and you make less obvious mistakes. So you will end up spending more time producing because you enjoy the process more. What price can you put on that? I admit I used to get fecked off at having to bounce the mixer output every 30 minutes to A/B it in my car. Save yourself the trouble.
|
|
#
¿
Feb 13, 2007 12:12
|
|
|
nimper posted:I think it's my fault.. I did put his name in the thread title In defense of the bitch I have no idea what the whole Mac/Pro Tools thing is about because I've used Pro Tools in Win XP and it has never given me problems. It just seemed like a totally absurd thing to argue about.
|
|
#
¿
Feb 21, 2007 14:54
|
|
|
I've decided its time for me to lose my M-Audio Delta 1010 and move up in the world as far as convertors go. Also, its been bugging me that I don't have an instrument level input so I have to faff around with discrete pres and a DI when I want to record my Guitar. I neither have a good preamp nor a good DI I've played around with a Rosetta 200 for about 20 minutes and I loved it to peices but sadly its more than 3 times over my budget. I was looking for something in the 828MKII kind of price range but there are a lot of horror stories about that interface. I never experienced any problems with the ones I used but then they weren't running on my computer and on my firewire ports. So I am now considering alternatives and shortlisted: MOTU Ultralite, MOTU Traveller, RME Multiface, RME Fireface 400, Focusrite Saffire Pro 26 I/O. Of these, I have not personally used any of them. I have used the Focusrite Saffire (not the 26 I/O) and it was basically a good card but there are things about it which bug the living poo poo out of me. For instance, The outputs are freaking hot and theres no way to switch down to -10dBV so I simply cannot monitor on this thing quietly if I need to. Saffire Control (the software mixer) has this horrendously annoying bug which makes all the gain faders default back to their maximum values when you save a mixer state. I cannot deal with this on a day to day basis Finally, the card has mic inputs but no instrument level input where I can plug my guitar in without a DI and some sort of spliced TS to XLR cable. So I keep wondering if the 26 I/O has the same problems.Do the Ultralite and Traveller have a history of problems that are similar to the 828MKII? I ask because alot of 828 tech is in both of these boxes. If the Traveller has significantly better convertors and pres than the Ultralite then I would be up for spending the extra 150 quid or so. Then theres RME which has always been lurking around at the back of the music production bus. I have no idea whether the Multiface is just a Hammerfall DSP card in a breakout box, whether its a modular system like Hammerfall and what it sounds like in practice. The Fireface 400 looks good for the convertors and pres but I think its only a 2 channel convertor - a bit like a half price Rosetta 200. And as much as I liekd the 200, I really wanted 2 more ins. I only ever need 2 outputs so the 8 I've got on my Delta1010 are a total waste. I don't use a load of outboard but I wouldn't mind having 4+ inputs in case I need to juggle a few synths. I don't really need any more than 8. Dont care about ADAT or AES/EBU support and will rarely use S/PDIF although some of these features come into the bargain at this price point anyway. What would you do in a similar situation and do you have any stories to tell about the cards I listed? Do you have an alternative? Your opinions would be greatly appreciated. WanderingKid fucked around with this message at 14:23 on Feb 23, 2007 |
|
#
¿
Feb 23, 2007 14:16
|
|
|
mofolotopo posted:Edit: I guess you probably wouldn't be able to use it with a laptop, though, if that's a problem for you. I should have mentioned that I am using a Small Form Factor PC so I have 1 free PCI slot and thats it. If at all possible I would rather keep it as a spare but if the Ultralite/Traveller works out better, it seems like I get to save the PCI slot anyway. And yeah Swivel Master, I've been anxious to try out the RME stuff. I think the Fireface 400 has the same convertors as the Fireface 800 which would be brilliant as I've heard loads of good things about their convertors. 4 analogue inputs just sweetens the deal. I'm sort of used to controlling gain through software. The Delta 1010 I already have has no physical controls on the front of the rack. I always thought that it would make sense to at least have a monitor gain control on the front of the rack itself since in an emergency (feedback loop!) you can quickly turn down the level. As long as it doesn't have any bugs like SaffireControl I guess its ok. But man its shocking having the monitor level shoot up by 100dB when you save a mixer state. Some day I will get a Rosetta 800 I think but not any time soon. Man is that thing expensive :\ Guess I should be grateful its not Prism Sound ADA8XR expensive... WanderingKid fucked around with this message at 18:23 on Feb 23, 2007 |
|
#
¿
Feb 23, 2007 18:15
|
|
|
The oscillator in question would be generating a 50/60hz sine wave, which has no harmonics. You can do this process in pretty much any synth that can generate a sine wave and has at least 1 LFO that can modulate the pitch of the oscillator in an envelope. There is usually an envelope or LFO triggering in an envelope via a high speed saw wave which modulates the pitch of the oscillator negatively over time. This means that the pitch dives the longer you hold the note down. Actually, in hiphop and breaks this was commonly done by sampling an 808 bass drum (which is synthesized using similar means but with a noise generator and a resonant low pass filter stage afterwards). So I would try both ways and see what you can come up with. The 808 has a really nice saturated hum when the pitch dive levels out and its hard to replicate. analogue filter saturation is pretty hard to emulate convincingly using softsynths and I have never had much success. filter saturation is the distortion that occurs when you overload the input of a filter. Its quite important for beefing up the bass end tail of a bass drum or for any overdriven sound when you swing the filter cutoff low. Other than that, its not going to sound hiphop unless you send the result through some valves, so if you have the real thing and some kick rear end convertors then feel free to go outboard and overdrive the drum like a motherfucker. If you dont have kickass convertors or tube amps then you can achieve sort of similar results using a tube amp simulator like Voxengo Warmifier. The thing about hiphop BDs is their simplicity. I wouldn't faff around with layering kick samples to get a composite sound - you lose the purity by doing that. Keep it simple and trying mixing the sine wave with some more harmonically rich waves like a saw. Try a sine/saw mix of 95/5 respectively and see how it goes and work on the distortion/saturation. Nothing too overblown as you don't want to go the way of gabber or anything.
|
|
#
¿
Feb 27, 2007 14:15
|
|
|
RivensBitch posted:no pair of headphones will reproduce the sound of your mix in a room. Logic has sidechain I/Os all over the mixer. In FL Studio you can't sidechain as such but you can simulate a sidechain like effect by linking two channel inserts/sends, choosing a peak gain relationship and drawing the automation into an envelope window using the paintbrush tool. Its not fully automatic and its not particularly fast but it achieves sort of the same effect. You can sidechain in Cubase - people on TA.com have done it but having not used Cubase since the Score days I can't comment on that one. All told though I am pretty sure you can either use a third party plugin or simply improvise your way around the lack of sidechain capability in your mixer. The only thing you need to be able to do is be able to modulate the gain of channel inserts/sends in the mixer. I think most DAWs can do this. Its not as quick or elegant as doing it the pro tools way but I figure theres not much you can't do in any of these programs. In a roundabout way you can get to the same place in pretty much all of these applications. The only difference really is the workflow and some people really love Pro Tools workflow. As for headphones. Well. It depends on alot of things. Bear in mind that nearfield monitors are designed to be listened to at a distance of no more than 2 metres and the reason for this is that the sound they project takes less time to reach the ear so room acoustics have less of an impact on the sound you hear. It doesn't work perfectly by any means but the greater the distance between your ear and the monitor, the more the room will influence what you hear. The human ear also perceives direction partly on the difference in time it takes for sound to reach either ear. Headphones (well built, well isolated ones anyway) will almost completely do away with the former influence which in my experience is often a good thing if your room is less than acoustically perfect. On the downside you get a very different sensation of direction, panned instruments and positional audio using headphones because sound from either driver reaches your ear almost instantly and at the same time. Hard panned left/right sounds feel unusual to me but its not something you cannot adjust to or anything. The key is to learn how your monitors reproduce sound. One thing I noticed about my Dynaudio BM5as is that they are very sensitive to positional audio. You have to keep your head in the same position when mixing. If I sit up straight/slouch or move my head a few inches to the left or right, I start to hear and feel that phasey kind of sensation when you listen off axis as with a stereo sound you are now changing the distance of your ear relative to the transducer. Therefore you really feel positional sounds on these speakers in a way I don't believe is possible on headphones. There are good and bad things about this. The good: excessively widened sounds really sound terrible on these speakers. You get a really good sense of panning an instrument and the direction it comes from provided you sit still and don't move the monitors. I think those Dynaudio AIR systems would be killer for surround sound monitoring because of this. The bad: You really do have to sit still. :/ I often get up and move around and sit back down and my chair isn't really a chair - its more like a giant beanbag on an ottoman. So I sometimes mix at home partially lying on my side (this is terrible by the way - don't ever do this )Sometimes I swear those speakers sound different every time I sit down so I keep having to move around to get the right spot and in the event of no such success I start moving the monitors around. A good pair of headphones eliminates this issue. It also eliminates alot of ambient noise (PC fans, traffic etc) which will probably get in the way. You can also get an awesome pair of monitor headphones for the price of a middle of the road desktop monitor system. I firmly believe you can monitor on anything provided you constantly do A/B references against other soundsystems (cross reference your mixes in your car, on your ipod and take notes). You will work out where each soundsystem is deficient and where it is hyped. I monitored for about 6 months on the free pair of guild wars headphones I got with the special edition. This was when my Dynaudio BM5a tweeter exploded and I was awaiting replacements. I got used to those headphones so much that I'm willing to buy another pair from any other GW collector's edition owners. My pair is getting tatty now. It is however, easier to mix on a good all round monitor as you make less obvious mistakes and you aren't essentially mixing blind. I feel there are things that are easier to interpret about sound when listening on headphones and things that are easier to interpret on desktop speakers. Ideally I would like to have a kick rear end pair of headphones, my Dynaudio BM5as and various other crapper soundsystems around the house, and my the soundsystem in my car. Because then I have loads of reference points that I can A/B against and work out precisely whats wrong with my mixes. I still need the monitor headphones. WanderingKid fucked around with this message at 14:39 on Feb 28, 2007 |
|
#
¿
Feb 28, 2007 14:30
|
|
|
Is it true that the pres on the 828MKII/Ultralite/Traveller become really noisy when you use the interface in high humidity? Some people claim they are really sensitive to temperature swings.
|
|
#
¿
Mar 1, 2007 14:36
|
|
|
Hustle Dozens posted:I've recently been interested in starting music production on my computer. Since I'm a mere beginner in the field I was considering buying the E-MU 1212M PCI card because its at a fairly good value and has the ability to upgrade when I gain some experience with this kind of production hardware. For the money the 1212M has no equal on the budget end of the scale. You need 2 spare PCI slots to get it working with all the digital I/O but you can use just 1 PCI slot if all you want is the analogue I/O. Specs wise you won't find better for less than 120 quid. You can of course get poo poo loads better but it'll cost you alot more. And I don't really think its worth your time upgrading until you can afford an E-MU 1616M. If you have one of those, I generally think of an upgrade from that to be something like an RME Fireface 400 - it truly kicks arse. Thanks Swivel Master for the recommendation - after auditioning one I'm selling my 1010 to get one of these beauts. After the FF400, an upgrade from that would be Apogee Rosetta 200. Mostly thats based on convertors, clocking and quality preamps (where applicable). Haven't taken into account the number of I/Os as these vary massively at all pricepoints. But you get a card with as many I/Os as you need. If you only ever do stereo recording and playback, you will only ever need 2 outputs. You can never really have too many inputs I guess but if you got a 10 in soundcard and you have like 1 synth module, then you will be wasting 8 of those inputs until you get more gear. Or something. Conversely you could run into a brickwall later on when you find you want to mic up your drumkit, your bass guitar and all your synths but you only have 2 ins. Thats a problem. If you can't stretch to the next price point (1616M being around 260 quid - slightly more than double the price) then you should be fine with the 1212M provided you can live on a stereo I/O.
|
|
#
¿
Mar 13, 2007 15:09
|
|
|
wixard posted:What are you talking about, it's been proven in court that they keep up with the industry's latest technological advances. Man, Pioneer's prosecution team must have pissed themselves laughing for hours when they saw the publicity shots for the DJX-1000.
|
|
#
¿
Mar 26, 2007 14:00
|
|
|
RivensBitch posted:I still love the aphex lawsuit where they opened the behringer unit in court and printed on the circuitboard was "(C) APHEX TECHNOLOGIES". The chinese had copied the unit so well they even got all the labels down perfectly.... Pahah! Behringer should change their name to Deadringer. I think it would be totally appropriate. v:Dv EDIT: out of interest, I know Pioneer filed a lawsuit against Behringer over their clone of the DJM600. Aphex has taken them to court. Who else has sued Behringer? And more importantly - how is Behringer still trading? WanderingKid fucked around with this message at 13:32 on Mar 27, 2007 |
|
#
¿
Mar 27, 2007 13:27
|
|
|
Jobless Drunk posted:This was me six months ago. Then I enrolled in a continuing ed recording program at a local university, and it made all the difference in the world. However, before starting that I did a whole lot of self-study, and continue to do so. I'd recommend you pick up a copy of Modern Recording Techniques by David Huber and start lurking gearslutz. Also, there are a couple podcasts that are worth mentioning, Inside Home Recording and Project Studio Network. Out of curiosity can I ask about some of the details of that program? I ask because I am reaching the limits of what I believe I can achieve on my own and I've been saving up to do a sound engineering course at Pulse Recording studios in Dublin. But the course is really pro tools heavy and Pulse aren't very forthcoming about many of the specifics of the program. So what was your experience of your course?
|
|
#
¿
Apr 3, 2007 16:38
|
|
|
quote:relatively cheap studio monitors (any help on this one?) Same opinion as Wixard on this one. I basically hated everything that clocked in under about £700 sterling. Alesis M1s sound no better or worse to me than the cheap Sony hifi speakers I had at home. At the time I was monitor shopping though, the ADAM A7s hadn't been released. Now you can get them for about 400 sterling for the pair and if they are anything like the ADAM P11as (and they supposedly are) then they should be good. The P11a was give or take the best monitor I auditioned for under 850 sterling. I liked it the most anyhow. Its worth pointing out that all the monitors I auditioned sound massively different. I found this disconcerting but ultimately it doesn't matter what you use to monitor. The most important part is you get used to how they sound and where they are deficient. Then make up for that deficiency by referencing your mixes on loads of other soundsystems (your mp3 player, the speakers in your car etc). Even if your monitors are amazing I get the impression you would have to do this and I still second reference my mixes on the crap pair of headphones I got with the guild wars special edition. I ended up getting a pair of Dynaudio BM5as and there are good and bad things about them. Mostly it depends on the environment you monitor in. You will get flutter echoes in very small rooms that don't have solid walls. Also, the tweeters are unbelievably sensitive which I suppose is a good thing but you have to monitor with your head pretty still. If you move your head a couple of inches to the left or right you can hear/feel the phase changes as the sound takes longer to reach one of your ears. Its literally a matter of inches. At first I thought this is crazy. But now I have sort of gotten used to it but my back sometimes kills me from sitting upright for so long. Occassionally I slouch over or lie on my side but they sound fricking weird when you do that. You can get used to that of course - you just need to keep monitoring on your side. If I'm doing this for a long time though my side starts to go numb. Hah. Well below this price point I didn't notice this behaviour on any of the monitors I auditioned. The M1s for instance just don't exhibit this behaviour and I can listen off axis and it doesn't make a whole lot of difference to the sound. I sort of liked the Tannoy Reveal 6s in a hard to explain kind of way but I hated those KRK Rokit and V series monitors. Those just seemed like really really powerful hifi speakers that bust your ears up at high amplitude. Wasn't keen on them at all. The Event TR-6 was quite nice but the build quality wasn't great and Event's tech support and customer relations are poor. Dynaudio on the other hand have been pretty good to me and there are a couple of guys working at their UK distributors (TC Electronics) which know me on first name terms and they said they can sort me out for replacements if I get problems. And they came through for me on two occassions and replaced two fecked up speakers that Digital Village shipped to me. Overall, it tipped the scales for me and I'm glad I spent the extra cash for the peace of mind and the solid speakers. I would suggest you do the same and spend a little bit extra for something that will last. WanderingKid fucked around with this message at 14:06 on Apr 4, 2007 |
|
#
¿
Apr 4, 2007 13:59
|
|
|
Well I'm not sure it works like that. The TR-8 has an 8 inch woofer, so it can reproduce a lower frequency tone than the TR-6 (which has a 6 inch woofer). The Dynaudio BM5a has a 6.9inch woofer with a frequency response down to 50hz. Which doesn't seem low on paper but I used to monitor on these in a small room with wooden flooring. It had 2 solid walls and two hollowed out walls with a high ceiling. 1 single glazed window. The room dimensions were about 10 foot by 15 foot so it was really quite small. The Dyns produced terrible flutter echoes in this room. You would get a kind of reverberation by clapping or stamping your foot so that should give you some idea of the kind of effect that your monitoring environment has on the sound of your monitors. I moved to a larger room with carpetted floors. I was now 1 floor up and I had 4 solid walls. I no longer get flutter echoes as badly and I can move my speakers well away from walls to prevent that hyped bass effect. But I should mention that I've used more powerful, larger speakers in this room and it just doesn't work. The standing waves are worse, they can go louder than is comfortable when monitoring in that environment. If you are monitoring in a tiny room and theres an slight echo when you clap, then pumping out loads of sub bass through an 8 inch driver and a massive amp is just totally pointless. It will sound terrible and is extremely off putting at high levels. I think you really need to consider the acoustics of your room when choosing monitors. Even though I hated the Alesis M1s, I work with a couple of people that use them and I've heard them in the same room as by Dyns. The woofer is much smaller and theres way less sub but you don't get standing waves in the same room like you do with the Dyns. Also, I think the tweeter is really unsensitive compared to the Dyns but again, moving your head slightly left and right doesn't produce that phasey effect so much. So I can find that monitoring with other people - where you don't always get the centre spot, is more ideal. I personally hated the M1s but you might like them and they might be appropriate for the room you monitor in. I think the only thing you can really do is try them all out yourself and see how it goes. It does help to get the best you can afford though (After you have auditioned a load of them). Otherwise you will always be thinking - what if I bought that monitor instead? One note on the M1s though. They seem extremely popular but by god - compared to the Dyns (in every room I have used the two in) it really does sound like someone stuck a shallow low pass filter before the monitor inputs. Then a high pass filter that sharply kills everything under 50hz. It just sounds so mute and centred, But I suppose thats why they cost about a quarter of the price. WanderingKid fucked around with this message at 12:07 on Apr 5, 2007 |
|
#
¿
Apr 5, 2007 11:50
|
|
|
Swivel - you use a condensor on your snare drum? Have you ever err..accidentally twatted it with the stick? Sm57s can take all sorts of punishment so its like - go crazy. But condensors are supposed to be way mroe delicate than dynamics right? Shoop - I was looking at an audio interface around MOTU 828MKII kind of pricerange and the Traveller was on the list (in the over budget category but I'd buy it if I really liked it). In the end, with a little encouragement from some ML goers I impulse bought an RME Fireface 400 and it kicks all kinds of rear end. I was mostly driven off MOTU because: 1) That stingy 3 month warranty. Who the hell guarantees a product for only 3 months? Any less and you would have to sell it as being disposible. 2) The noisy preamps. They aren't always noisy apparently but some people report having problems when they operate MOTU units in high humidity environments. 3) The Texas Instruments Firewire thing. If you use the wrong Firewire chipset then its supposed to be unusable and I remember DirtyNBL buying an 896HD and experiencing exactly this problem. I think he sold up and bought something else. 4) Several years ago 828MKIIs used to have alot of reports of noisy and extremely hot power supplies and I've been skeptical ever since. Its worth noting that loads of people use MOTU products and never experience any problems at all and generally think they are freaking awesome. But I have poo poo luck so I'd probably get a duffer. The Fireface 400 has awesome convertors. The preamps are a bit naff but they are onboards so I guess its unfair to expect an Avalon or something. Also the driver is rock solid which is really important for me. I can deal with less features but poor buggy drivers just makes producing a nightmare. It also looks horrendously ugly in its publicity shots - it doesn't look much like it in real life. Its not purple anyway and you can remove those horrid looking rack ears. It looks much less offensive then. Overall, I think for once, my gut instinct steared me in the right direction. If you can audition one definitely check it out.
|
|
#
¿
Apr 19, 2007 17:28
|
|
|
Swivel Master posted:ALSO, I REALLY like the Fireface mixer program. It's really easy to use, especially for having different headphone/monitor mixes. The software mixer is kickass. The only beef I have is that I wish there was hands on monitor level controls on the unit. But I've sort of gotten used to being careful from using a Delta1010. The rackfront of the 1010 is so bare and so flat that you could race snails on it. Come to think of it, the only software mixer I used that I hated was SaffireControl (on the Focusrite Saffire). And it was mainly due to one problem - whenever you save a mixer state it defaults all the gain sliders back to 0dB. I found this was horrendous as theres nothing like powering it up for the first time, slamming the gain controls down in your DAW to 25%, lowering all the gain controls in SaffireControl, saving state then pressing play. Have you ever heard an overdriven 909 snare at 115dB a mere 2 feet away from your head??? gently caress that poo poo. I was expecting a whisper and I got a gun shot.
|
|
#
¿
Apr 20, 2007 17:07
|
|
|
You can do it with any analogue/subtractive synthesizer and bitreduce the output. I made a pretty good clone of a megaman V song using an Access Virus B and 2 square wave oscillators, no filtering and generous amounts of bitreduction and vibrato. The drums are harder to get. Theres an art to making snes percussion and sadly I haven't cracked that particular nut yet.
|
|
#
¿
Apr 30, 2007 17:22
|
|
|
RivensBitch posted:It depends on the sources obviously, but I find that "mids" covers a lot of territory and those three instruments have their different spots. You have to use your ear of course, if you're using a plugin the with your mouse drag a greatly boosted width of frequencies across the spectrum slowly and listen to what you hear. it should become clear which frequencies matter and which dont, if you hear frequencies that sound bad pull them out, if you hear frequencies that are good leave them alone. Never boost. Take notes of the frequency bands that you pull out of certain instruments. If you are having problems making two instruments fit together in a mix, try to coordinate which frequencies you pull and which frequencies you leave in. Aye - I found vocals much easier to work with in the mixdown than say analogue synths and it comes down to the harmonic intervals. Pretty much every sound I have ever mixed with few notable exceptions (most 808 bassdrums with the attack setting low) span pretty much every audible frequency. Male vocalists will have their lowest harmonic interval in the lower mid ranges but the rest go way up to 11,000hz. And in between the (somewhat complicated) series of peaks you have alot of troughs. Whereas analogue synths for example output variations on basic waveforms. Harmonic sounds are often built out of saw waves and you could think of those as giant blocks of sound that carry a fundamental and all odd and even harmonics in linearly descending intensity down the length of the scale. So you shave off the stuff you don't want using a filter. And chugging guitar power chords are often tough to work with since the amount of distortion on them is typically high adding to the wall of harmonics that many hard rock bands try to create. Vocals are really different because the harmonics aren't so regular. One trick I use with paragraphic EQs is to switch the middle point on the EQ to a bell curve and narrow the Q so you get a very thin bell curve. Now turn down the master volume for the next bit. Increase the amplitue of the bell curve so its as high as it will go on the EQ. Sweep the entire range of the EQ to find the loudest point. Now increase the master volume so that its tolerable. This is just health and safety stuff. Now sweep the entire range of the EQ again with the narrow bell curve and listen out. You will notice the amplitude will suddenly spike at specific and ordered intervals along the length of the EQ. These are the harmonic intervals of that particular sound and are represented on a spectrum as a series of tall thin peaks in relation to a fundamental frequency (the lowest harmonic). With few exceptions there is mostly nothing below the fundamental (since this is the lowest frequency pitch reference anyway) so feel free to automatically hipass everything below the fundamental. Mostly when you layer laods of audiotracks of harmonically rich intervals then some ofthese harmonic intervals really overlap and start causing problems. Varying degrees of destructive phasing when summed to mono but mostly amplitude spikes caused by the frequency ranges of 2 different sounds being in the same phase. So it causes problems with headroom management. Do the trick I mentioned to find where the worst offending harmonic is. Once you have identified it (it will be the loudest) then you simply turn the bell curve upsde by giving it a negative gain. Voila - you now have a notch filter at the offending frequency. widen the Q as necessary. I tend to find wide Qs are better for where you want to subtle changes. If you hard notch out alot of harmonics you will notice that it can make vocals and a number of instruments liked violins sound very unnatural. And if you notch out the fundamental you can lose its pitch reference which can ound quite freaky. I tend to prefer using linear phase EQs for very sharp, very narrow notch/bell filters. Linear phase EQs are dead perfect and sterileand some of them sound it (Waves LinEQ for example). I tend to prefer non linear phase (analogue) EQs for very broad, shallow notch/bell filters. Analogue EQs will introduce a phase shift to the output signal and have other imperfections which some people find desirable (that are also mimiced by digital EQs that attempt to recreate the sound of analogue EQs). So yeah - digital EQs for precise surgical cuts and boosts. Analogue EQs for broad sweeping cuts/boosts.
|
|
#
¿
Jun 7, 2007 17:36
|
|
|
No I didn't make it up. Linear phase EQs are linear phase. There really is no other way you need to interpret it. I already said there are digital EQs that attempt to mimic analogue EQ behaviour. Think of them as virtual analogue EQs. I specifically said to ignore those if you are looking for linear phase EQs. As for the difference. Its clearly obvious if you use something like Izotope Ozone and you switch back and forth between its linear and non linear phase paragraphic EQ which you can do with one click. As for the information, alot of it is based on what Izotope recommends you do with Ozone. Its pretty sound advice for general purposes. 'musicality' is a bullshit term when used to describe tools like EQs and compressors. It depends entirely on what application you use them for. Even the most 'musical' compressor for instance can be made to sound very 'unmusical' if you apply enough gain limitation. There are some instances where you don't want a phase shift or at the very least want to keep it minimal. I used to hate LinEQ until I 'got the point of it.' But once again we dive into the 'highend' in a home recording thread. Nobody here is going to have an Avalon. If you do thats great - use whichever tool you feel most comfortable with. Alot of people here are going to be using VST based effects for their post processing - the very idea of using outboard without having stupidly expensive convertors is just silly. All software effects are digital but many are designed to mimic analog filtering using digital algorithyms. Thats why the distinction is sometimes referred to (read: simplified to) analogue and digital EQs. This information is intended to make some sort of sense for people who are confronted with a billion VST EQs and don't know which one would be better suited for a particular application. In the end there are no hard and fast rules and even those can be broken if you know them well enough. But as a starting point, its a good system to get your bearing and I'd like to see you elaborate on how this is demonstrably flawed - its you against Izotope at least on this score. and who says you have to use a mouse to operate a software EQ? Jesus, just assign the relevant CC values to some rotaries on your midi controller and voila - all rotary operation. Besides, if you are designing an EQ to be used on a computer it makes sense to make it functional for use with a mouse and keyboard. Alot of people say that analogue synthesizers are not meant to be programmed with a mouse and keyboard but Native Instruments Absynth proved that idea was bullshit. The problem of course is with GUIs that look and are designed to be programmed exactly like rotary control surfaces yet have to be operated with a mouse. Anyone who has struggled to mouse over a rotary on a synth like pro53 will know that its GUI was designed to look exactly like the hardware - it was never designed to be easy to operate using a mouse. Which of course is a problem considering NI were selling that product to people who will always be operating it on a computer. However, there are many ways its GUI could be altered to make it a cinche for mouse and keyboard operation. Allowing you to input numbers using the numpad for instance and having all variables have a numerical range display. Even without it you can use it exactly like hardware by mapping all the variables to a midi control surface. So what exactly is the problem? WanderingKid fucked around with this message at 11:11 on Jun 8, 2007 |
|
#
¿
Jun 8, 2007 10:45
|
|
|
RivensBitch posted:I think you bring up some good points but my specific criticism was with this statement: You may notice that I prefixed pretty much everything in my post with 'I tend to...' because its a habit I've gotten into that seems to be logical and works for me. It might work for you. Try it out. For those without any sort of system I believe its a nice set of simple ideas to start off the subject before you go off and experiment. You are of course correct in saying that analogue EQs vary in the degree of phase smear. And that there are nice sounding analogue EQs and not so nice sounding analogue EQs (in my and your opinion). The idea of linear phase EQs being appropriate for certain basic situations is fairly solid. You would use a linear phase filter for when you want to minimise or eliminate phase smear/comb filtering in stereo sounds. You would use a non linear phase filter if you didn't care either way. Some people find that phase smearing sound quite pleasant in certain situations if slightly unpredictable. Usually if I am making broad and shallow boosts on a paragraphic EQ you preserve alot of the harmonic structure of the sound so you don't really change the relationship between harmonic intervals that much (unless you boost using loads of gain). So I find these broad shallow boosts to be good for emphasising a broad characteristic of a sound - more like an emphasis. I find using EQs with extremely non linear phase characteristics can often sound quite nice for situations like that. When I am making very sharp, narrow cuts using alot of gain I often find that there are several things to think about : 1) Notching one channel of a stereo pair can be a pretty big source of comb filtering if you want it. 2) You can eliminate certain harmonics altogether by setting the EQ point to one of the loud 'spikes' in amplitude (harmonic interval or partial) and notching it out completely using a narrow Q cut with alot of -gain. This is the trick I described above your reply. 3) If you do point 2 to harmonic sounds and you do it to the lowest harmonic (the fundamental) then you can change the pitch reference of a sound. The best way I can explain this is with the 303s resonant filter. If you turn the resonance all the way up and sweep the filter's cutoff on a single monotone note played on all 16 steps of the sequencer, the tone will subtely seem to rise and fall in pitch as you open/close the filter. Thats because you apply alot of gain to the frequency around the cutoff point. When you sweep it over a harmonic peak it accentuates it and can change the pitch reference of the entire sound if theres enough resonance. 4) Hard notching out alot of harmonics in a series can make a harmonic sound seem very very unnatural. Its the same with extreme high/low/band passing filters. For various reasons I seem to use this notch EQ trick to diminish harmonics or amplitude spikes which would otherwise eat headroom or cause a clip. Sometimes you can stack several sounds (say a clap and a snare drum) in the mix and bring out certain harmonics in one or the other to change things like the pitch reference of a sound or to eliminate harmonic peaks which sound unpleasant or painful at normal listening level. So I happen to use this trick alot for corrective EQ - where I don't want to cause comb filtering or phase smearing and I just want to nuke an amplitude spike because its annoying me. Linear Phase EQs are great for that. It seems to me that you have used Waves LinEQ and was probably left with the same impression I was - I hated it at first because it sounds DEADPAN. Even moderate degrees of subtractive EQ can totally kill an interesting sound stone dead. Then after a couple of months of using Izotope Ozone's linear phase EQ and comparing it with the analogue modelled EQ I just kind of got the point of it and where I could play to its strengths. If you wanted something that sounds like it exploded out of a vaccuum tube amp that costs 10 grand and sounds a million dollars - you aren't going to get it with LinEQ. But you would be missing the point of it if you dismissed it on this criteria. However, there are no rules and most of them were made to be broken anyway. Its just a useful starting point for people getting under the hood of EQ and want to try something out before they start bending/breaking rules.
|
|
#
¿
Jun 8, 2007 22:21
|
|
|
RivensBitch posted:Now as for mapping midi controllers, I'm going to call you out on this one. Midi controllers have a range of 0-127, would you like to explain to me how you're going to get a clean sweep of frequencies by dividing a 20khz range into 127 values? I'm sorry but even the digidesign controllers (I own a control 24 btw) don't give you adequate control for dialing in an EQ, and they have 16,129 values. EQ is where analog has a much greater advantage and I think even the most scientific minds can agree that right now hardware controllers for digital equipment don't let you both broadly and minutely tune in the way an analog knob does. When using our amek at my studio I can with a single knob run the entire frequency spectrum and at the same time barely nudge and squeeze until I'm dead on. With an EQ plugin at best I can activate a "minute" adjustement feature but I've never had that form of control be as intuitive as a knob. Because even though MIDI can only have 128 discrete levels of sensitivity, you can still rapidly interpolate between those values. The more times per second the smoother the result. This for instance is how the Access Virus has a digital filter which doesn't audibly step. In fact, most virtual analogue synthesizers built within the last 10 years has a filter which constantly interpolates between steps for smooth operation. If it weren't for the perculiar behaviour of its resonance I would defy anyone to call out the Virus filter amidst a crowd of analogue polysynth filters. That of course is a totally loaded statement since pretty much all of them have filters with different phase characteristics, different saturation properties etc etc making it pretty much guesswork unless you have precise knowledge of how some of these filters work. Hell, the only virtual analogue synth I have ever used which had a zipper filter was the Roland SH-32. I hear the Sequential Prophet 600 has a zipper filter too but I can't comment on that one - I thought it was analogue. Have you never noticed before that you can divide up 19,980hz (the entire sweep range of a software paragraphic EQ like Voxengo GlissEQ) into 128 bands which equates to each band being roughly 130hz in width. Have you not noticed that you can sweep the cutoff and it wont actually step in 130hz intervals? Either from mouse operation or when bound to a MIDI controller? Now on a software EQ you can overlay spectrums and zoom in to something ridiculous. And on some of them you can input frequency and amplitude values to several decimal places. I think that Native Instruments Absynth allows you to input filter cutoff and resonance settings to 3 decimal places? Using a computer savant's best friend - the numpad. Its not infinitely continuous like analogue is but thats a lorra discrete intervals. When you have this much resolution I honestly can't tell the difference. How many different discrete values can you get out of 64 bits? millions? billions? Will you ever use those numbers or be able to tell the difference between 200.001hz and 200.002hz? No but thats the point. - I know what you mean about musicality in essence because I've used a Distressor before. It sounds absolutely class and its a scream to use. But I can't describe what it does much better than saying I enjoyed using it and if I had the money I'd have one myself. Beyond that you make up your own mind about what that might mean. WanderingKid fucked around with this message at 23:00 on Jun 8, 2007 |
|
#
¿
Jun 8, 2007 22:23
|
|
|
wixard posted:I've never used a linear phase EQ. From looking around it seems to me like in any application where you don't have full-on delay compensation (and any application where a 30ms delay is unacceptable) linear phase EQ is more trouble than it's worth. It takes so long it could easily put the entire track out of phase, instead of just slightly smearing the contents of the track like a traditional EQ would do. Aye. The thing about alot of drums is that they are atonal. Most membranophones dont vibrate in the manner a string would for example and instead of creating an ordered series of harmonics you get a quite complex series of partials. Sound on Sound did an awesome guide which should be available on their site called 'Synth Secrets' - The author explores the nature of membranophonic instruments and attempts to recreate the partials of a Timpani (I think) using a DX7. He does a good job too. I have nowhere near that level of knowledge on the subject but I figure its worth learning since you can start developing EQ routines and shortcuts that work consistantly for atonal instruments too. Some of the engineers in this field are truly insane like that - they have this mathematical understanding of sound design which makes me look like a total amateur. With regards to the latency - its not exclusively related to linear phase EQs. Convolution reverbs tend to have alot of latency too. Thats just part and parcel of working with software to a CPU limit. There are ways of working around this. For example I have separate instruments in separate project files and mix using several instances of FL Studio open. And I have to bounce alot and chop off the dead silence at the start of each wav. For alot of people this is a pain in the arse and for people like you that have regular access to full on pro tools rigs theres absolutely no point or imperative in learning to deal with this kind of problem when you don't have to with pro tools (0 latency and great sounding plugins wahoo!). If you don't have a protools rig or an awesome studio with awesome converters and outboard then you gotta bounce alot. Sure its a pain but look on the bright side, it costs alot less. Think of it as the cost of convenience.
WanderingKid fucked around with this message at 13:31 on Jun 11, 2007 |
|
#
¿
Jun 11, 2007 12:57
|
|
|
RivensBitch posted:I understand the mechanics behind trying to work around digital controls for digital plugins, I think my point was that it's still trying to recreate what comes naturally with an analog knob, and the feel to me is very important in the same way that the fretboard on a guitar is important to a guitarist. The results I get when I EQ that way are just better, and I credit both the quality but also the control. Nah I get where you are coming from. In the end its best to use what you are comfortable using and you are in the enviable position of having regular access to gear which is so well engineered it has 'x' factor. I admit if I had a Distressor I wouldn't use any of the dynamic plugins I've bought to date. I'd use the Distressor instead. But seeing as I can't use one regularly and I can't afford to buy one, I'm content to squeeze the absolute most out of some humble VST effects instead. I just learned the ropes like a lot of people in this thread by starting with a computer and some freeware plugins. I can't explain why a Distressor sounds so good or why its so satisfying to use. But I suppose if you could do that easily Empirical Labs wouldn't charge an armful for it. The thing that software EQ and Compressor developers are wisening up to though is that modelling the GUI of these processors on hardware is not intuitive as turning a knob on a screen with a mouse does not have the tactile response of turning a real potentiameter with your hand. Hardware units also have a rackface/user control surface which is streamlined specifically to make that unit a breeze to use. Whereas MIDI controllers are generic rows of pots and sliders which you map controls to. The only way I can describe this is as follows: I have an Access Virus B hardware synthesizer and whoever designed the user interface of this synth did so with great care because everything is laid out in the best possible way to design sounds fast. There is a software version of the Virus B - Virus Powercore. But the UI isn't as good and it doesn't program as easily when you bind all the CC numbers to a midi control surface. Its not laid out as intuitively as the hardware synth. Even though they sound almost exactly the same, I'd rather have the hardware. When it comes to outboard though most of us poor bastards couldn't afford a neve comp let alone the convertors to let it shine. Absynth is great in terms of software GUI design because its a soft synth designed to be operated via a mouse and keyboard. Unlike something like ImpOSCar which is modelled to look exactly like a real OSCar on your screen. Fiddling with tiny pots on a screen is not cool and binding all those variables to a generic midi controller can get fecking confusing. I think software EQs would benefit alot more by having GUIs that can be operated quickly and easily using a mouse and keyboard. I'd like to see a hardware EQ though that can do the kind of crazy stuff that Ozone's paragraphic EQ can do - It has spectrum overlays, infinite peak hold and and option to infinitely average the input signal so you can overlay the famous 6 decibel slope and compare your mixes. You can zoom in on the display something chronic and its designed to be operated using a mouse and keyboard. I am aware of hardware EQs that have x factor that I'd love to own. But in terms of workflow Ozone's EQ is genius. It may not sound as good as an Avalon but day to day its just a joy to work with. WanderingKid fucked around with this message at 13:18 on Jun 11, 2007 |
|
#
¿
Jun 11, 2007 13:12
|
|
|
Yeah. Check out the Lexicon PCM91. There are convolution impulses derived from this unit so you can try them out for free to hear what it sounds like. The upside to using a real PCM91 is that you get that sound but with no latency. You can run certain convolution processors with 0 latency (such as Voxengo Pristine Space) but you can kiss your CPU goodbye. The downside to using a real PCM91 is that it costs a loving fortune whereas SIR 1.008 with PCM91 impulses is free. Also, Rivens is right about reverb. Its derived from delay. All phase modulation effects are. The delay is just small enough that it is not perceived as a repeat of the signal. You can simulate crude reverbs using most tape delay plugins. WanderingKid fucked around with this message at 15:30 on Jun 12, 2007 |
|
#
¿
Jun 12, 2007 15:26
|
|
|
wixard posted:I think I could do that if I were working on my own projects and making all the decisions and only trying to please myself, but I can't imagine running someone else's mix session like that. Thats true. When I'm working with others on joint projects this is the type of thing I tend to take home and sort out myself. I usually find it uncomfortable when theres 2 or 3 other people waiting around as I'm waiting for the render bar to fill up. It takes a while when you have loads of audiotracks and you are using the highest interpolation settings. So everyone is just standing there in uncomfortable silence with a lit cigarette burning its way down to the stub and theres a little voice in everyone's head going 'hurry up for fucks sake.' Invariably, someone has to leave because their girlfriend is giving them a lift to their folks or something. Whatever momentum we build up usually loses its energy when we have to wait around retreading old ground like that. But thats why bands go to your facility to get their songs recorded. Not my house. Its also why they would pay you money for the service whereas with me, I'd mix for nothing (except the experience) and they'd just pike my cigarettes and get annoyed waiting around. For various reasons, this one amongst them - I prefer to work alone.  You should see the look on people's faces when you say something like: 'give us a moment - dual mono convolution reverb sounds poo poo when panned so lets set up a proper convolution chain with 2+ pairs of impulse responses so we can pan the wet signal and not have the soundstage collapse into mono whenever we hard pan an audio track hard left or right. Then lets do a render test and apply it to our snare drum. I'll chop the excess off the beginning and we can go again from the top.' 'How long is that going to take?' 'An hour? Maybe another 15 minutes on top of that so we can do some equalisation on the wet signal and mess around with the decay time so it sits properly in our master project. Then lets do another render test and compare the two snares in winamp. So maybe another 15 minutes waiting for bars to fill up...' 'Do you mind if me and the guys go and get a kentucky whilst you do the stuff then? 'Uhh. Ok...' Then its usually just me after that. And they loving wonder why they get upset when I mix their tune and it turns into Techno. Hey mang - what do you expect for nuthin? WanderingKid fucked around with this message at 19:13 on Jun 12, 2007 |
|
#
¿
Jun 12, 2007 19:00
|
|
- "OK, just give me a couple minutes to roll back to the original track, figure out if it's a dynamics thing, an EQ thing or an effect thing, decide what settings I want, then process the track again"
|
No recommendations really. I have come to the realisation that monitors are important but what is more important is getting used to the sound of the ones you own. There were plenty of people that made amazing sounding mixes on NS10s and they suck. If you know where the speakers are deficient, where they are hyped and how they compare to various other sound systems in your house then you are on the ball. Beyond that its down to personal preference and my advice is that you should not listen to anyone who tells you that you should buy 'this or that monitor.' The only correct answer is this: You should burn yourself a disc full of test tones, mixes and professional songs (for reference purposes). Grab some buddies and go down to your nearest retailer. Audition every monitor under your stated budget. Pick the one you like the most.
|
|
#
¿
Jun 16, 2007 04:14
|
|
|
RivensBitch posted:this would be a penny-wise, pound foolish mistake to make. The nice thing about a 57 isn't just it's excellent off-axis rejection or it's universally renowned versatility on a guitar cabinet, it's also the fact that it "just works" on about anything you could put it in front of. 57s (and 58s) are also loving indestructible. loving indestructible. You have to deliberately try to break one to actually...well...break one. 
|
|
#
¿
Jun 25, 2007 22:20
|
|
|
We have a couple of 58s in the house and the punishment those things have taken - the grill is totally mashed out of shape and the bodywork is scraped to hell and back where it has on occasion been used to wipe the floor. I've stepped on it at least once and one of my housemates (who has used it for vocals) has beat it up too. It just...wont...die!
|
|
#
¿
Jun 26, 2007 12:27
|
|
|
You need to state a budget and what you intend to use it for otherwise the exercise is pretty pointless. Its safe to assume you won't be in the market for a Lynx Aurora or a Lavry Blue or a Prism Sound ADA8XR. But even the difference between a $300 and a $600 soundcard can net you a whole lot more awesome. A soundcard is basically an input stage -> AD convertor ->  Do crazy mad production stuff in Cubase here -> DA convertor -> output stage. Its basically a device that lets you process sound through your PC. Some soundcards have useful extras like preamp stages, massive arrays of inputs and outputs (for recording tonnes of instruments simultaneously and mixing in surround sound respectively). Some have billions of digital I/O via ADAT. Others have external clocking capabilities like wordclock I/Os. Some have pretty much all of these things rolled into one package. But you pay more for these things and you need to know whether you will use them or not. If you are 100% software for instance, you don't need any inputs or an AD convertor stage and you can sink all your cash into something like a Benchmark DAC-1. Typically I suppose you would rate a soundcard by how good its conversion and clocking is. If it has preamps then how good its preamps are. Once you get to a certain pricepoint - MOTU 828MKII and above, the difference becomes quite difficult to tell on the conversion front. Over at gearslutz they love doing convertor blindtests and seeing which one comes up the daddy. Taken as a whole the results are mostly inconclusive. Recently, more people voted that the RME Fireface 800 AD convertors sounded better than an Apogee Rosetta 800s. I couldn't tell any significant difference when the FF800 was slaved to the R800's clock. :conf: Then they swapped clocks and the test was flawed so I discount that one. Then they tested both convertors using their own clocks...and I couldn't tell the difference there either. Before that, someone did a blind shootout between a Fireface 800 and a MOTU 828MKII and most people said the 828 AD conversion sounded better. I couldn't tell the difference on that one either (but I was using pretty crappy headphones for that test). Then some guy compared a Lynx Aurora and said it was better than a Prism Sound ADA8XR but previously someone said the Lavry Blue was better than the Prism. But the Lynx was worse than the Lavry. So...  So yeah. It does make a difference to a point. After that the extra cost goes on R&D and you enter the land of diminishing returns. How much can you afford then?
WanderingKid fucked around with this message at 17:50 on Jun 26, 2007 |
|
#
¿
Jun 26, 2007 17:34
|
|
|
Lets start narrowing it down a bit... Are you mixing in stereo or do you plan on working in surround? If you wish to mix in 5.1 at some point (i.e. for film soundtracking and sound design) you will want 6 outputs (centre channel, front left, front right, rear left, rear right and sub). If only plan on working in stereo then you need only 2 outputs. I tend to be of the view that you can never have enough inputs but some people don't want billions of them because they won't use them. Currently I use 4 outputs max but if you are recording drums then you probably want more than that (2 overheads, close snare, close hihat, kick drum). On the conversion front I think anything above 828MKII is decent so consider that your bassline and look up. Onboard preamps are nice if you don't happen to have a discrete preamp which rocks (like an Avalon or a Neve or something). Necessary if you record things like electric guitars or any microphone level source. Not necessary for line level devices (most synthesizers). Are you planning on doing alot of digital signal transfer?
|
|
#
¿
Jun 26, 2007 19:19
|
|
|
If I were you I'd look at: 1) RME Fireface 400 2) Metric Halo Mobile I/O 3) MOTU Traveller 4) MOTU 896HD You could probably get a Lynx Two for that kind of money but its low on inputs and has no preamps plus its a PCI card. You could get a Mytek ADC for 995 bucks which is pretty spectacular but you would still need a DAC (Benchmark DAC-1 for 700 bucks has to be one of the best discrete DACs you can get for under a grand). I'd recommend giving each of the listed interfaces an audition and see what you think.
|
|
#
¿
Jun 26, 2007 23:35
|
|
|
|
| # ¿ Apr 27, 2024 03:30 |
|
|
Slowfuse posted:gently caress budget too, what works?[/i] Straight converters where money is no object: Apogee DA16/AD16 combo Apogee Rosetta 800 Lavry Blue Lynx Aurora 16 Prism Sound ADA8XR Prism Sound AD2/DA2 combo Benchmark ADC/DAC-1 combo Most of those wont have preamps or anything and Digital I/O modules cost extra. You can get preamps from: Lavry Audient Avalon Neve But really, if you throw budgetary concerns to the winds you need a serious serious amount of money. Tens of Thousands of dollars at the very least. And you should audition this stuff blind. As well engineered as it all is, the law of diminishing returns always applies. Most folks I imagine would make do with the MOTU/RME/Apogee cards representing the top end of their budget - and you can get something pretty nice in the $750 to $1500 price range. I'm pretty happy with the Fireface 400. Eventually I'd like to get an Avalon U5 since I'm not bowled over by the pres. And then a Benchmark DAC-1 as a substitute DA. But I need to earn more bucks before I can think of doing that. Actually, a Fireface 400 + Benchmark DAC-1 alone would be pretty nice. It still comes down to how you use your tools and how good a musician/producer/engineer you are though. wixard posted:There is almost no hardware that only works on a Mac, and only 2 pieces of popular software I can think of offhand (Logic and Digital Performer) that are Mac-exclusive. I think Apogee Ensemble is MAC only. If you really like Apogee stuff and like working in Logic then thats a really big lure. WanderingKid fucked around with this message at 16:10 on Jun 29, 2007 |
|
#
¿
Jun 29, 2007 16:05
|
|