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Down Is the New Up posted:PC->audio system has been beaten to death already, but I wonder how important sound cards are in this sort of setup? I have heard mixed things in real life discussions with people on the matter. I'll layout my setup (or setup my layout?): A computer soundcard is (essentially) a digital to analogue convertor and an analogue to digital convertor. It has an array of inputs and outputs which allow you to convert analogue signals to digital signals and vice versa. In a nutshell: 1) The Analogue to Digital convertor (ADC) stage is used for recording incoming analogue signals. For instance, you plug a microphone into a soundcard input and record in a digital audio workstation like Cubase which runs on your PC. You sing into the mic and the mic capsule detects the variation in air pressure caused by the expelled air from your lungs resonating with your vocal chords. It converts these variations in air pressure into variations in voltage. This is our analogue signal. It then goes to the Analogue to Digital Convertor (ADC) stage where it is 'digitised' or quantised to use its technical term. You now have a digital reproduction of that analogue signal. The process of quantisation involves sampling the amplitude of the signal a number of times per second. This is the samplerate. So a CD sample quality recording (44,100 hz) would involve the convertor sampling the analogue signal 44,100 times per second. The number of discrete levels of amplitude that it is sensitive to is a function of bitdepth. CD Quality is 16 bits. It works exactly like bits in pictures. 32 bit colour means that each pixel can be have 16.7 unique colour values. audio sampled with 32 bits of resolution means it can have millions of discrete variations in amplitude. The convertor interpolates everything between the samples - joins the dots basically and you have a digital reproduction of an analogue sound. The ADC and input stage is for recording. If you don't do any recording you don't care about any of this. 2) To hear the sound, the digital signal is sent to the Digital to Analogue convertor (DAC) stage. This takes a digital signal and converts it to variable current. This tiny electrical signal is sent to the output stage where it travels along a wire to your speakers. Your speaker amplifier stage takes this tiny electrical signal and 'amplifies' it using an external power supply. This larger current then hits the speaker tranducer (the cone bit) which literally punches out this variable current thus causing the air close to the tranducer to ripple. So this variable current is translated to variation in air pressure once again. This variation in air pressure reaches the ear where it resonates with the ear drum and our brain interprets this as sound. Just for  purposes a speaker is basically the same technology as a microphone only it works in reverse. In fact with a few tricks you can scream into the woofer of a speaker unit and do some very crude recording.  Essentially then, a soundcard is simply a box which allows you to get sound in and out of your PC. Beyond that it gets more complicated but if you do ZERO recording and don't use your PC for any musical purpose then the only thing you really care about is: 1) The maximum bitdepth and samplerate of the DAC. 2) The signal to noise ratio (SNR) on the output. 3) The number of outputs (you need 2 outputs for stereo: 1 Left; 2 Right. You need 6 outputs for true surround sound playback: 1 Centre; 2 Front Left; 3 Front Right; 4 Back Left; 5 Back Right; 6 Centre Sub) 4) How reliable and up to date the driver is. If you do recording or plan to do some, then it gets more complex as you now care about ADCs, pre amp stages, MIDI integration, wordclock integration etc etc. so I won't address that unless someone asks specifically. Why is a soundcard important? You may be comparing a soundcard with an onboard chipset. If you can, avoid getting soundcards that come on PCI boards. Soundcards were never meant to be put inside PCs because of the number of electrical sources nearby which can generate alot of RF interference. The ideal soundcard will be one that comes in an external breakout box (which you want to place far away from your PC and any other electrical power supply) and which interfaces via Firewire. However when you think about buying soundcards you have to think of it like buying parts for a computer. You can spend thousands of pounds on an incredible studio quality convertor but if your speakers suck, you are not going to hear the difference. Just like buying a 3.8ghz CPU for a computer with 256mb RAM - you are just bottlenecking the system. So what soundcard should you get? This depends largely on your needs. 1) If you plan on listening only in stereo then you only need a soundcard with 2 outputs with 24-bit/96khz DACs. 24/96 is the DVD-Audio standard. If you wish to listen in surround sound you must have 6 outputs. 2) For various reasons the convertor isn't everything. How well it keeps clock and the integrity of the signal path (galv isolation being the ideal) are as important if not more so. Really really loving expensive converters tend to be over engineered so they have pristine signal paths, awesome clocking and convertors which can sample up to the DXD rate (384,000hz). You will not notice the difference unless you have extremely sensitive speakers and this is unlikely in the home. If you have very large powerful speakers that can accurately reproduce wide band signals (we are talking 35hz to 20khz+) then you will want better DACs. At least, you will notice your purchase alot more than if you listening via a pair of cheapo headphones. If you have a pretty ordinary average hi fi type sound system then the convertor issue is not that important. Even the budget soundcard market has fairly decent boards out there. The Emu 1212M for example which is give or take the best budget soundcard you can get in my humble opinion for under £130 sterling. For both listening and Recording. What about Creative cards? Creative cards are all derived from Emu boards. Creative bought Emu out not too long ago and the latest generation board they have, the X-Fi is derived from the 1212 board. In fact the X-Fi platinum pro is essentially the same soundcard as an Emu 1212M except that: 1) the 1212M sports 1/4inch balanced jacks. The X-Fi sports unbalanced RCA phono jacks. 2) The 1212M has slightly better SNR on the input. 3) The 1212M DSP chip runs a suite of effects processors that can be used for music production with no hit to CPU load. The X-Fi DSP chip runs a suite of effects processors for use in the latest EAX supported games. 4) The X-Fi has a number of gimmicky technologies like the 24 bit crystalizer which is nothing more than a glorified transient compressor. Recommendations: 1) Stick with brands you know or ones which are reputed to have stable drivers. There are few things worse than glitchy unsupported drivers. 2) Creative gets alot of stick in the recording community but mainly because their marketting department really lives up to the company name. The X-Fi is basically a good card because its basically an electronically unbalanced version of the Emu 1212M with plugs that fit most home theatre soundsystems instead of studio monitors. 3) To run the latest EAX in games, you must have a Creative card. 4) Always check how many outputs a soundcard has. You need 2 for stereo. 6 for surround. Check its SNR although to be fair theres barely anything between the big brands. 5) Brands to check out: Emu, M-Audio, Creative. You can of course get alot better for listening and recording but they also cost alot more money and have more specialist music making applications. For most people in most homes with run of the mill sound systems there really is not much noticeable difference. The most important and noticeable increase in sound quality you will get is not through a soundcard. It is: 1) through a speaker upgrade. 2) in the acoustics of the room in which you listen to them. Therefore, if you simply want better sound quality you will most likely want to sort these out first. Hope this helps mang. As for cables. The only thing you care about is that it is well shielded. Most home soundsystems will be totally unbalanced. Most home soundcards (all the Creative cards) are unbalanced so this doesn't really matter. My advice is to not bother spending tonnes of money on expensive cable. Just get a tough durable, well shielded one and you are fine. As I said you will notice a much more dramatic increase in sound quality by purchasing better speakers and listen to those speakers in a nice sounding room at the appropriate distance. WanderingKid fucked around with this message at 22:47 on Jun 13, 2007 |
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Jun 13, 2007 22:36
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A Balanced cable has 3 lines - signal, anti phase signal and ground. An Unbalanced cable has 2 lines - signal and ground. The answer is: Most modern balanced gear can support any combination of balanced or unbalanced lines. You do not need an op amp or anything like that. The first thing I need to know is what is the input on the speaker? Is it a TRS Jack input? An XLR input (black circle with 3 holes in the middle)? Now go to https://www.hosa.com. This site has some illustrations which you will find useful. You can buy cables which are terminated on one end with a balanced jack and an unbalanced jack on the other end. These cables have 3 lines but on the unbalanced end, the anti phase signal carrying line is grounded. This means that no matter what, this cable will always be unbalanced. It exists simply for connectivity reasons. Peruse the Hosa website and find what jack type connections you need. Your DVD player will most likely have an RCA phono output. So you would need an RCA to TRS or TS or XLR jack to connect your monitors to your DVD player. It doesn't matter which one because the system is not going to balance anyway and theres no anti phase signal so it will be ignored. To convert unbalanced to balanced you need a Direct Injection Box. These vary in price from the ultra cheap Behringers and Millenium D.I.s to the quite expensive Radial Engineering D.I.s The difference in price is mainly down to the number of extra features. To connect this up you would run a stereo RCA to TS cable from your DVD player outputs to the DI box. This part of the signal chain will always be unbalanced. The DI will electronically balance the signal and you will need a TRS/XLR to TRS/XLR jack to connect the DI box to your monitor speakers. This part of the signal chain will electronically balance. WanderingKid fucked around with this message at 00:05 on Jun 15, 2007 |
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Jun 14, 2007 12:57
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All a D.I. box does is match impedance really and break ground loops if you had them before. The more expensive D.I.s do exactly the same thing except they come with other useful bits and bobs (if you happen to need them). Such as high and low pass filters, phase reversal switches and more channels. There are two things you can do: 1) Run an unbalanced RCA phono to unbalanced TS jack from the DVD player straight into the speaker. The signal chain is totally unbalanced. I'd probably do this because its cheaper and I assume your cable runs are short. 2) Run an unbalanced RCA phono to unbalanced TS jack from the DVD player into a D.I. Run a balanced TRS to TRS jack into your monitor inputs. The signal chain is unbalanced before the D.I. and balanced after. To be honest, unless you are getting ground loops with the player you don't really need a D.I. The D.I. will break a ground loop if you have one. The only other reason why you might want to have a D.I. in the middle is if you are running hundreds of feet of cable and signal attenuation and crosstalk become a problem. Then you can eliminate it by using a short unbalanced cable between the D.I. and the DVD player and a really gently caress off long balanced cable between the D.I. and your monitor inputs.
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Jun 14, 2007 23:35
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Dogkicker posted:Question: I'm moving to a dorm room this fall and along with a new PC with a soundblaster x-fi card I want to get new PC speakers (I dont want to bother with getting a reciever). What 2.1 speaker setup would you guys recomend that would fill a dorm-sized room with quality sound? Keep in my mind I intend to use my PC primarily for music and not movies or games. I dont really want to spend over $170 either. 'sound quality' with respect to speakers would be more or less subjective and it depends on what you are used to listening to. The best thing you can do is burn a disc full of your favourite songs and take it down to your local sound system retailer. Then audition everything that clocks in underneath your stated budget. I personally think think that Blue Sky make decent 2.1 monitoring systems but they are freaking expensive. I have a pair of Dynaudio BM5as and I love them but they don't sound especially pleasant. The tweeter is incredibly sensitive - almost annoyingly so. Move your head left or right a few inches from centre and you can hear the phase changes. Its brilliant and annoying at the same time if you are mixing/recording in stereo. I know a few folks who really don't like this at all and would rather not mix on my speakers. Its also worth mentioning that your room will have a huge impact on the way that sound is reproduced. Depending on how bad it is and how obsessive you are then you may want to consider treating your room with acoustic panelling and bass traps as well as moving a few things around. The bottomline is this: Don't listen to anyone who tells you to buy 'this speaker or that speaker. Only you will know exactly what you like the sound of so go and audition them for yourself. You can audition more expensive, techically better speakers but if they are well out of your budget then doing so is more likely to make you want something you can no way afford. Naming a price and auditioning only 2.1 systems you can afford means you will be more likely to leave satisfied with your purchase which is the most important thing in the end. You really have to hear what you are purchasing before you part with your cash though. You wouldn't buy a car without test driving it first. WanderingKid fucked around with this message at 22:11 on Jun 16, 2007 |
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Jun 16, 2007 21:58
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McCheese posted:I noticed I've got some lint/dust building up on my silk dome tweeters, I tried a gentle application of compressed air to get it off, but it didn't get all of it. Any suggestions for cleaning silk dome tweeters? It shouldn't be a problem even if it looks aesthetically unappealing. The drivers in my monitors have a thin layer of dust on them permanently but I'd rather not prod at soft dome tweeter if I don't have to. It would probably be a thick caked layer of dust if I hadn't been booming it out on a regular basis - vibrate as much of it off as you can through normal listening and leave it at that.  Unless someone knows of a proper way to clean them? I noticed from my old speakers that the tweeter is actually quite resiliant. You can collapse the dome and poke it and even take a cloth to it. That said, my old speakers are basically disposible now - I wouldn't do any of that on my current speakers because the manufacturer told me not to. 
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Jun 19, 2007 01:28
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Pibborando San posted:Any particular reason for wanting to go tubes? Do you want your sound to be accurate or sound a certain way, because tubes are not accurate. They simply distort the sound, but if that sound is what you're after... Thats not what tubes do. The principles of tube amplification and solid state amplification are the same - the difference is that in tube amplifiers, current flow between cathode and anode takes place across a vacuum. In solid state amplifiers it is through a solid semiconductor. Tubes like semiconductors are used in the process of amplifying an electronic signal - Tube sound becomes distinctive in overdrive because they generate a non linear series of low order harmonics. Overdrive tube amps enough and the output soft clips. By contrast, straight solid state amplifiers when driven close to and beyond saturation point generate a linear series of harmonics across the entire spectrum of sound. When the peak signal exceeds the saturation point, the output hard clips quite suddenly (squares off). So the onset of distortion is more sudden and is perceived to be harsher. Now you can get solid state and digital amps that recreate tube overdrive characteristics and simulate the effects of soft clipping. A combination of these properties has led to the popularity of tube amplification amongst electric guitarists and the increasing wealth of digitally modelled tube amps and solid state amps that recreate tube amp behaviour is testament to this (see Marshall's Valvestate technology) - the onset of distortion is less sudden and the emphasis on lower order harmonics contribute to the perceived 'warmth' of the resulting sound. When the output does distort, the soft clipping does not alter the proportion or emphasis of the harmonic series as dramatically as traditional solid state amplifiers. One is not better than the other - they simply have different distortion characteristics, although tube amp distortion is overwhelmingly more popular than solid state amp distortion or digital distortion. Tube amps tend to be bigger and heavier than solid state amplifiers. They are more labour intensive to produce and the innards are more fragile. Tube amps have a higher rate of failure than solid state amps because they have more servicable parts that can fail - cathode poisining is an issue, tubes themselves can be physically broken and they can burn out. Tube amps can short too. Tubes will also have increasingly non linear distortion characteristics as they age. Some people like this effect, others do not. A combination of these factors and their desirability amongst musicians means they are expensive to purchase and fairly expensive to maintain. Solid state amps are smaller, lighter and require less maintenance because they have less servicable and mechanical parts. More so for digitally modelled amps, which require no maintenance, have very low rate of failure and are cheap to mass produce. Different tubes have different distortion characteristics. If you buy a tube amplifier you can mix and match tubes for a different emphasis on the sound. Amps that have multiple tubes in series have capacitors/transformers between them because of the massive negative bias at the input stage of each tube in relation to the cathode. This introduces a phase shift into the signal which will alter it because of comb filtering which occurs across the signal. The extent of the comb filtering depends on the extent of the phase shift and the results are somewhat difficult to predict. This inherant part of tube arrays is partly responsible (together with the emphasis on lower order harmonics) for what many people describe as 'colouration' of the signal. This phase shift characteristic is found in varying degrees in all analogue signal processers - analogue equalisers and tube compressors for instance. Tube amps are high voltage, current devices. The tubes themselves generate alot of heat but they also radiate alot of it. In the event of critical failure, the tube can be replaced easily. Because tubes generate lower order harmonics in overdrive, overdriven sound is more tolerable at very high amplitudes although I should stress - low frequency sound at high amplitude is just as bad for ear health as high frequency sound at high amplitude. Tubes really come into their own when they are driven behind massive speakers and they tolerate overloads better making them good if you plan to pump out music at unholy volumes. Transistor based amps cannot tolerate overloads quite so well. Solid state amps are low voltage, high current devices. Solid state amp distortion, with its emphasis on the entire spectrum of harmonics have more perceived higher order harmonics than tube amp distortion. At very high amplitudes, this is less tolerable to listen to and is bad for your ears. Solid state amps can of course be used to drive massive soundsystems, but if you overload them to destruction, you will probably have to replace the entire amp. Consequently they are less suitable than tube amps for driving sounds very very loud. This should be stressed - you will not be driving your speakers or amps this hard in your home. Unless you have no neighbors and no concern for the health of your ears. My soundsystem is a pair of active monitors with solid state amps built into the speaker cabinet. Generally speaking I like them but I work with people who will not use my monitors to mix because they dislike the way they reproduce sound. Similarly, I find their speakers odd to mix on. Ultimately, you have to listen to the soundsystem yourself and formulate your own judgement. Especially if you are going to spend large sums of money on one. You will gradually get used to the way your soundsystem sounds and you can equalise the output if sound reproduction is not great in your room. WanderingKid fucked around with this message at 22:04 on Jun 22, 2007 |
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Jun 22, 2007 21:06
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It does look nice. You even get to see all your little bitches light up. 
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Jun 23, 2007 00:51
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You really have to listen to speakers before you buy them because they all sound different - even speakers in the same price bracket. I was amazed at how different my Dynaudio BM5as sound compared to ADAM P11as. Which sound way diferent to Event P6s. Some people really don't like listening to my speakers and I live with a person who will not monitor on them. Get your cash together and get yourself down to a retailer. This is your budget. Ask to audition all the speakers below your budget. Do not audition a speaker you definitely cannot afford. Before hand, burn a disc full of your favourite tunes. Now go listen and buy the speakers you like the most. This is the only good advice when buying speakers. Doing anything else is tantamount to buying blind and if its money you can't afford to lose, it makes it all the more important.
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Jun 25, 2007 20:21
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Having not heard one I really couldn't tell you. If built into a well designed cabinet and with the crossovers matched appropriately (plus both woofer and tweeter being phase alligned) I don't doubt that it could be a decent little speaker. The important points though: 1) never buy any speaker without listening to it first - if you buy blind like this, do not be surprised if you dislike the sound of it when you hook it up at home. I suppose if you are building your own, this can be difficult but if theres any way you can audition high/low frequency driver in a ready built speaker - do it. 2) take into account that room acoustics have a very dramatic effect on how speakers project sound. Other than that I use speakers for a somewhat different purpose to most people in this forum - sound design. However, for that purpose it really doesn't matter what speakers you use so long as you know where they are deficient and how to compensate accordingly using equalisation and sound absorption/diffusion material to treat your listening environment. As a testament to this, its worth noting that the Yamaha NS10 is probably the most well known and well used reference monitor in the world (no contest) and its absolutely rubbish. That telephonetastic +7dB spike over 1500hz has something to do with that.
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Jun 26, 2007 09:35
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Internet Explorer posted:I actually have a similar question. While I haven't gotten any noise complaints, I'd hate to be creating noise for my neighbors and I'm a bit paranoid to the point where I'll turn down the sound on my speakers to where I can barely hear it. You have a few options but please bear in mind that soundproofing a room in a home is not possible (without essentially rebuilding it). The best bet you have is rigid fibre glass insulation. If you are a US resident the best stuff is Owens Corning 705 and you can get it in packs of 4 foot panels. If you live outside of the US you won't be able to get this stuff so you should use mineral wool (rockwool) instead. It works out at about the same price. How you utilize them depends on how your walls are constructed but they were designed to be slipped under floorboards and in between the air gap in your walls. If this is not possible for you, you should mount them in plywood frames and fix them to your wall/ceiling using a thick layer of adhesive. Dont couple them to wall surfaces (i.e. by nailing them directly onto walls). To treat an entire room in this manner will be expensive. If you cover an entire surface or line an entire wall with this stuff (and you do it properly) it does act like an attenuator for upper mid and high frequency sound - it wont do poo poo for low frequency sound (bass). The primary purpose of rockwool/rigid fibreglass is home insulation and sound absorption - i.e. reducing flutter echoes within rooms. You will notice that if you had a flutter echo in your room or a pitched reverberation, it will be gone if you do this correctly. Bass is a whole different story. Soundproofing as mentioned before is impossible. It would involve decoupling your entire room from every other exterior surface - basically, you float it and suspend it in a 'slightly larger room' using a sytem of springs (literally, very heavy duty springs). Then line the air gap with rockwool/rigid fibreglass. Auralex (I think) sells stuff called Sheetblok. Which is this black matting which you are supposed to staple onto your walls and so forth. It is supposed to provide quick and dirty sound proofing. In my humble opinion this stuff doesn't work very well (if at all) but its cheap compared to DIYing your own rigid fibreglass absorbers. Note that you can buy them readymade but the cost will be higher. When I get home I'll update this post with a link containing some instructions on how to build rigid fibreglass panels. Beyond that the only real attenuator that works is big, thick, solid walls. Preferably 2 big thick solid walls running parallel and with a gap of air in between. Which you can line with rigid fibreglass! You may have heard some myths about hanging carpets on your walls or using egg cartons. Carpets don't do poo poo. Egg cartons act as diffusors (they break up reverberation from flat surfaces) if you have enough of them but they turn your room into a giant firehazard. Acoustic foam panels act as absorbers and for that purpose they are somewhat effective but they are not very good at attenuating anything but high frequency sound - which your walls will tend to do better. They also turn your room into a giant firehazard except this time, it burns slower so you have time to escape. - To stop your speakers vibrating you need to mechanically decouple them from your desk. So you need to put some sort of elastic material in between your speaker and your desk. It has to be elastic enough that it returns to its original shape without the mass of the speaker on it but it needs to compress to some degree under the weight of the speaker. Do not use plastic materials - materials that deform and stay that way until you remould them - i.e. blue tac. Sand however works quite well and you can sit your monitors in 2 sand pits (a couple of centimetres high, not packed rigid) to get the job done. Also, don't use something too springy - For about 5 minutes I had an idea to sit my monitors on cushions. However, they will probably wobble and fall over. Which is obviously bad. Instead I would recommend buying something like this because it does actually work. Auralex has a rather 'creative' marketing department and they wax alot of lyrical about how these pads 'improve the performance of your monitoring system' - whatever thats supposed to mean. Either way, ignore all of that guff. It basically works for mechanical decoupling. You can DIY these things (and probably make a better one if you have measured the resonant frequency which is causing your speaker to physically vibrate and build your pads out of appropriate materials). but I don't know how or what materials would be the best for you personally. There is a sound on sound forums post that details this but I don't know where it is and have forgotten all the specifics. I'll try to find it. Do NOT buy those node feet. They do almost the opposite of what you need to be doing. The Auralex pad is a spongy, compressible material that retains its shape. In case you are worried - it is fire retardent (which means it does burn but it does so slowly. It ain't fire proof). You can think of these pads as 'speaker suspension' (rather like car suspension). The idea of car suspension is to limit shocks and vibrations caused from contact between the tyre and the ground. The idea being so that those vibrations aren't felt so much by the driver/passenger. Car suspension is essentially a spring between the mass of the ground (in contact with the tyre which itself is not totally rigid) and the mass of the car frame (and driver/passenger). The spring mechanically decouples to some extent the driver from the uneveness of the road. So riding a modern car doesn't give you a sore arse. Whenever you go over a speed bump for example, the spring compresses and relaxes thus taking much of the jolt out of the ride. The idea with decoupling speakers works in the same way - its just not as dramatic. Edit: Here are some urls you may find interesting: http://www.speakerbuilding.com/content/1011/page_9.php http://www.ethanwiner.com/basstrap.html http://www.6moons.com/ramef/6.html http://www.realtraps.com/install_mt.htm WanderingKid fucked around with this message at 11:19 on Jul 2, 2007 |
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Jul 2, 2007 10:03
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noface posted:Is there a large difference between optical outputs versus an HDMI audio output? I can't imagine there being much difference provided the optical out outputs at 24/96. Which is pretty much the standard bitdepth/samplerate at the moment. I don't really see 192khz catching on anytime soon.
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Jul 2, 2007 19:18
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Kash posted:... Soundcard analogue output 1 & 2 ---> Power Amplifier ---> Speakers Its pretty simple. I don't remember what kind of outputs the audigy has but you need to have some sort of analogue out. Otherwise you would need to do this: Soundcard digital output 1 & 2 ---> Digital to Analogue Convertor ---> Power Amplifier ---> Speakers Auditioning speakers is easy. You state a budget. Go down to your nearest retailer and go and audition everything they have under that price point. You do this by listening to some programme material you are really familiar with so burn a CD with your absolute favourite songs on it - the ones you have listened to so much you know every tiny detail of the track. Then give it a listen on all of them. I brought a pencil and paper with me and just marked which ones I liked the sound of. Eventually you will narrow it down to a couple you like more than the rest. Then start A/B comparisons. I had a budget of 1200 euros though, so the early selection process was to ditch the speakers which lovely lovely build quality. In the end, you buy the speakers you like the most. Or if you aren't totally happy, you save up some more cash and increase your budget. WanderingKid fucked around with this message at 13:41 on Jul 13, 2007 |
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Jul 13, 2007 13:36
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Local Yokel posted:This has likely been answered, but I scanned the first couple pages of A/V Arena for it and didn't see anything. Its mostly subjective. However, it is a general rule of thumb to mount all speakers such that the high frequency driver is level with the ear - this is because bass is perceived as being omni directional. How you arrange your speakers depends entirely on the dimensions and shape of your room as well as the acoustics. I have the front left and right speakers angled inwards at 30 degrees, tweeter mounted at ear height. I have 2 rear channel speakers mirroring the front with the driver mounted at ear height. Sub can be placed anywhere on the floor and the centre channel mounted on top of the TV. I've used the same setup in several rooms because I'm renting. One of the observations I made is that the same orientation doesn't sound the same in different rooms so room acoustics are a massive factor in how your system will reproduce sound. My recommendation is to get down and dirty and simply move each speaker around until you feel it sounds the best. On the 2 channel front, I'm constantly moving my monitors around because they sound different every drat time i sit down.
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Jul 17, 2007 00:31
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Boner Slam posted:Mostly not. All speakers should have the same distance to you, ideally. I sort of agree with you but room acoustics, dimensions, size and contents vary so much that there is no all purpose solution that will work for everyone. You just have figure it out for yourself and use a bit of applied 
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Jul 20, 2007 18:12
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ail posted:Now, I do have a question: Is there really any compelling reason to spend an extra $60 on a Creative XGamer over, say, a Chaintech AV-710? Music is more important to me than videogame sound, EAX has never really blown me away. You would hear a much more dramatic difference by looking into buying a well engineered pair of speakers rather than a new audio interface (specifically for its DA convertor and output stage - from what I can tell, I doubt you would use anything else). There isn't a huge difference on the DA front that I can tell between my Delta 1010 (bought 3 years ago for £299.99 sterling) and my Realtek AC97 integrated sound board (came free with the motherboard). The main one is that the Delta absolutely does not emit permanent low level crackling/glitchy noise (exacerbated by hard drive activity and so on). Whereas the AC97 does. I assume thats mainly because the I/O array on the Deltas is housed in an external breakout box, and its a good distance away from my PC and computer monitor. Either way, most of the time I can live with it. If money is no object, and music is your life, be done with it and buy a Benchmark DAC-1 or an Apogee MiniDAC or one of those Lavry DACs. Or something of their calibre (which is to say: over engineered to the point where the cost just isn't justifiable unless you have clients that will use it regularly). Spending heaps on an audio interface only becomes justifiable if you record and want a really good AD convertor, really good clocking, great sounding preamps and tonnes of analogue and digital connectivity + ASIO compatibility so you can record loads of channels at the same time without the whole system going to poo poo. If you just want to listen in your home, a budget firewire interface from the likes of Emu or M-Audio will do remarkably well and they are marketed towards a demographic which is (annoyingly) called the 'prosumer' crowd. Whether you cringe at the term or not I suppose doesn't matter because they fill that serious hobbyist niche fairly well. About the only requisite I think the non recordist should look for in an audio interface is to make sure that it electronically balances. Balanced cables barely cost more than their unbalanced counterparts. All balanced soundcards made in the past 5 years can accept any combination of balanced or unbalanced cables and devices so there is complete backwards compatibility. Even if you never use it, its there and doesn't cost much extra. If you do use it, you can pretty much rule out ever getting a ground loop plus you can now get away with stupidly long cable runs and everything is louder. If you have no real complaint with your current audio interface though, it will be much more worth your while to buy pair of speakers you actually like the sound of. Reasonable complaints I could think of would be: 1) Not enough outputs. Cant do surround sound now that you have bought your shiny new 5.1 speakers. 2) Driver sucks and is not detectible in a number of applications that require it and/or you constantly get DMA buffer underruns when running at high samplerates (which manifests as glitchy pops/crackles/stutters on playback). 3) Permanent low level staticky noise that is exaccerbated by such things like hard drive activity and which drives you nuts. More than likely an issue of RF interference and is particularly a problem with audio interfaces designed to be housed inside your PC case. Otherwise, save your pennies and look into buying bigger, badder speakers instead. If you can afford it, get both though.
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Aug 13, 2007 16:56
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American Jello posted:I may be setting up a system for my fraternity house to play music with. Is there any way I could set something up that would be party-proof, but still easy to plug an iPod or put a CD into? We listen to a decent amount of rap, and I blew out an old Sony speaker last year. Eh, you need a limiter. You can use a compressor - you just set the compression ratio over 10:1 or so and keep the threshold fairly high. Just below 0dB and make sure you arent clipping your amp when the signal peak at that level. Depending on how much gain reduction you are applying (i.e. how much signal is peaking over the threshold you may need to lengthen or shorten the amplifer attack/release envelope times but having them both too long will turn most songs into mush. Which would indicate that you are driving it too loving loud. If you have already blown out a speaker you were driving them too loving loud. Solution: Bigger speakers designed to cope with the signal from a more powerful amp. Impedance matched. A limiter will prevent stray clips. Clipping your amp being the number 1 probable cause of speaker failure. A limiter will also prevent sudden, massive increases in gain (for example where songs suddenly get louder) but that depends on alot of variables: the limiter's threshold, compression ratio, envelope times and whether the limiter's peak detection is via peak signal or RMS (Root Mean Square) peak signal. All of these are adjustible so you need to work it out depending on what you expect from the source material you are playing. RMS is a method of peak detection which persistantly averages the peak level of a signal over a time frame (usually between 500 and 3000ms) and takes the average as the peak signal (its always lower than the absolute peak). In layman's terms use a peak level limiter to supress clipping transient sounds like gun shots and short, sharp rises in level. Use an RMS peak limiter to supress longer sustained periods of loudness that rise above 0dB. To be honest if alot of the signal is peaking above 0dB and your amp's clip indicators are lit, you are simply driving your amp far harder than it was designed to cope with. Think of a limiter as the last line of clip protection. Nothing beats turning down the volume unless you have a soundsystem and amplifier which is designed specifically to pump out sound at ungodly levels. Even then you want a limiter at the end of the signal chain because it would suck so badly if you destroyed them and everyone's ears because of a monster clip. WanderingKid fucked around with this message at 15:51 on Aug 24, 2007 |
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Aug 24, 2007 15:45
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Pibborando San posted:A lot of people do bi-wire. They're audiophiles and generally are taking advantage of the two posts on each speaker for "bi-amping" which can have benefits. Thats not entirely true. Pretty much all 2 way speakers now have crossover networks which at its simplist level is a capacitor that attenuates low frequency signal to the tweeter (usually starts attenuating at 3000hz but varies and can be changed manually with the appropriate knowledge of electronics) and an inductor that attenuates high frequency signal to the woofer (usually around 800hz but varies and can likewise be changed manually). This is as much to do with safe operation of the low and high frequency drivers as any other consideration since feeding too much low frequency signal to a tweeter is a pretty easy way of hitting it with way more current than it was designed to handle, thus resulting in a burnout. It largely has nothing to do with sound quality although you can completely gently caress up mid range sound reproduction by setting crossovers inappropriately so the overlap between low and high frequency drivers is large and you have several capacitors in series rolling off the tweeter very quickly (and if the filter cutoff point is low enough will probably destroy the tweeter at high amplitudes). Or setting the filter cutoffs too far apart can essentially make a speaker sound like it is playing with a notch filter before the amplifier stage. Which nobody really wants. Pretty much every active speaker made since whenever already does this, all in the cabinet. If you have a passive speaker setup (separate amp and speaker) you really should biwire since operating a tweeter without a crossover network is like...well...do you want it to burst when you turn up the volume?
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Aug 28, 2007 20:40
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pim01 posted:I don't get it. Are you saying that passive speakers don't have crossovers? Bi-amping is a good option if you want to splurge for separate amps, but bi-wiring to one amp seems to me to be quite useless. If you don't like the crossover in your speakers, replace it (or the crappy components) with a better one. No. Google 'Passive Active Crossover.'
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Aug 29, 2007 21:35
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pim01 posted:I can see that there's all sorts of problems inherent in passive crossover networks, but your post made it seem like passive speakers are entirely devoid of any form of bandfiltering. That was not my intention - pretty much all speakers, passive or active should be operated with crossover network. To not do so would just be detrimental to the drivers. With actives you don't have to worry about wiring stuff and setting crossovers and so forth. Which I imagine the average joe in A/V really doesn't want the hassle of doing. Actives are just an easy, all in the box solution and a cure all for a number of little sins that just go with passives. Plus actives have their amp matched to the driver so you don't need to go through internet faqs on impedance matching and worrying about messing it up and turning your amp into nuclear waste. They also save space. Sort of. WanderingKid fucked around with this message at 13:40 on Aug 30, 2007 |
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Aug 30, 2007 13:31
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pim01 posted:I'd imagine it'd get expensive (and quite a hassle with powercords) when you keep in mind everybody apparently wants to have a 5.1 or even 7.1 setup. Ehhhh, I forgot about that. An active 5.1 rig would cost a fat wad yeah. quote:That is technically the only difference between passive and actives. Whether they have a built in amp or not. I like passives because that allows a lot of upgrade and sound tweaking possibilities down the line. The signal chain is different. In a passive crossover network the high/low pass filtering occurs between the amp and the speaker. In an active system it occurs before the amp. For the technical differences and problems between these differences see here. The short of it is that in active 2 way systems you have 2 amplifiers for 2 drivers and the source is connected directly to the load. Changing the impedance of the load has no effect on the crossover. Some active crossover networks are crazy in the sense that they are basically paragraphic equalisers (without the realtime graph :z) meaning that you can move the filter cutoffs around, change their resonances and so forth. I wonder if you could actually use your own paragraphic EQs to filter a signal before the input stage of a power amp. Just skip the passive crossover network. Hrmmm... Anyone tried it? You'd still need more than 1 amp. And I would imagine it would be basically the same thing as an active speaker only with a load more wires and wasted space and the power to blow your speakers to kingdom come by setting the tweeter cutoff to like, 200hz at +18dB and huuuuuge Q. WanderingKid fucked around with this message at 15:17 on Aug 31, 2007 |
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Aug 31, 2007 15:13
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Reisen posted:My speakers are able to pick up radio signals, and then play them very weakly. Thankfully it's too low to hear when I have other music going, but when the music is off it's incredibly annoying. The way I have the speakers set up, the off button is kind of hard to get to, so I'd rather not have to worm my way back there every night. First hit on google using the terms 'Speakers picking up radio frequency. Solution linked from above.
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Sep 9, 2007 06:30
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Cock Soup posted:I recently bought the Audiophile USB from M-Audio. At the music shop, the guy told me that I would be able to plug in and record my guitar directly via USB. However, I've had quite a bit of difficulty with the product: it picks up my guitar at a very low volume, and I'm starting to think it wasn't really made for guitar recording. I don't know much about this; do I need a preamp? Looking at the specs on the website, it seems like there is no mention of "intrument", except the MIDI connectors. You can plug your guitar in and you will get a signal. However... 1) It will be extremely quiet because your guitar outputs a very very low level signal. You will need to apply on the order of 40 to 60dB worth of gain before its even audible. If you apply this gain after the soundcard input stage via a software gainstage this basically means that you will get an overbearing amount of hiss and buzz. It sounds pretty horrid. 2) There is a huge discrepancy in impedance between your guitar output and your soundcard input. It wont bridge. In order to bridge 2 circuits the input load impedance needs to be on the order of 10 times greater than the source. A guitar pickup generally has a very high impedance (typically greater than 200,000 ohms but varies alot depending on signal frequency). A line level mixer input (such as the ones on your soundcard) will have an input impedance on the order of 20,000 ohms. Your problem is that your soundcard input stage has an input impedance that is generally around 10 times less than that of your source when it should be the other way around. This basically means that you will get alot of high frequency loading. The bridge pickup in particular (which picks up more treble) will sound very weak and the overall effect is that the sound seems very bassy and muddy with very little top end. Solutions: 1) You need to run your guitar into a preamplifier to provide on the order of 40 to 60dB of gain before the signal hits your line level soundcard input. 2) You need to run the preamped guitar into a Direct Injection box to form an impedance bridge resulting in maximum voltage transfer. Addressing both of these issues will sort out all of your problems. The simplest solution would be to simply junk your lovely soundcard and get one with instrument level, preamped neutrik inputs. Then its a simply case of plugging your guitar straight into the input and you are ready to go. And yes, MobilePre fits this bill because you can plug straight into the input - it is already preamplified and forms an impedance bridge using a built in DI. You can however get alot better. If you want to stick with M-Audio, Fast Track Pro is loads better. WanderingKid fucked around with this message at 04:57 on Sep 10, 2007 |
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Sep 10, 2007 04:49
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mono posted:Can anyone here recommend a receiver with some kind of auto-volume leveling feature? I've looked around a bit but I have no idea what an official term for this would be for search purposes. You need a compressor. A compressor will prevent sudden, massive increases in gain (for example where songs suddenly get louder) but that depends on alot of variables: the compressor threshold, compression ratio, envelope times and whether the limiter's peak detection is via peak signal or RMS (Root Mean Square) peak signal. All of these are adjustible so you need to work it out depending on what you expect from the source material you are playing. RMS is a method of peak detection which persistantly averages the peak level of a signal over a time frame (usually between 500 and 3000ms) and takes the average as the peak signal (its always lower than the absolute peak). In layman's terms use a peak level compressor to supress sudden loud transient sounds like gun shots and short, sharp rises in level. Use an RMS peak compressor to supress longer sustained periods of increased loudness. Basically, a compressor is an automatic level reduction tool. You simply dial in what level it should start working (threshold) and the how much it should work (compression ratio). WanderingKid fucked around with this message at 01:36 on Sep 16, 2007 |
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Sep 15, 2007 03:45
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I bri I posted:Is there ANY way to make my logictech computer speakers stop picking up mexican radio stations? It doesn't matter how loud the volume is, or if i'm playing music. Sometimes the feedback is so loud, I can hear it over the music or game or whatever. Answered 1 page ago in this thread. Come on guys this stuff is google fodder. Just typing 'speaker, radio stations, interference' into google gets you a tonne of solutions. WanderingKid posted:First hit on google using the terms 'Speakers picking up radio frequency.
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Sep 15, 2007 03:49
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Axiem posted:Okay, so here's the deal. I have some of these. I don't want to seem mean or anything but this question has been answered 3 times in the past 3 pages. Your speaker cables are acting as an antenna. This is why you are picking up radio signals. No this is not serious. Yes, this is easy to fix by winding all the cables coming out of your speakers (including the kettle lead that supplies mains electricity) around a ferrite core (which costs practically nothing) thus making them rubbish aerials. You can use ferrite beads too (those little cylindrical things you see at the end of a gamecube control pad). Using shorter cables will also make them less effective antennas. First hit on google using the terms 'Speakers picking up radio frequency'. (3rd time lucky I guess) Solution linked from above. WanderingKid fucked around with this message at 01:36 on Oct 6, 2007 |
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Oct 5, 2007 16:25
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Ferrite beads will work for that purpose. They are sometimes called chokes and you see them on alot of electrical appliances. My gamecube controller has them, as does my LCD flatpanel monitor. Attach the choke close to the termination point to the next device in the signal chain. Other than that, avoid having large numbers of cables running parallel or coiled up into reams. As a test, move your speakers and amp faaar away from your computer/monitor/tv/whatever and see if you still get the same problem. If you have your cables choked or wrapped around ferrite cores and your speakers/amp situated in the middle of nowhere and you still get buzzing, you have a grounding problem. A simple, easy way of fixing a ground loop without getting your hands dirty in electronics poo poo is to buy a cheap DI box and stick that between the two devices that have different potentials. Probably your amp and your speakers. If it breaks a ground loop and fixes your problem, keep the DI. If it doesn't, send it back to the store you bought it from and say 'sorry, I needed a preamp not a DI for my microphone' or some other excuse. You might not even need to make an excuse if the store has a no quibbles return policy. By the way, you can get a cheap passive DI for about 10 bucks. Replace the kettle lead supplying mains power with another kettle lead supplying the same volts/amps. See if that works. Ground Loops are pretty much the number 1 cause of audible buzzing/humming in audio systems, so I wouldn't be surprised if thats the beef. You aren't stuck with any buzzing/humming forever because it shouldn't be happening and the fact it does means it is either an issue of interference (which is fixable for very little money) or a grounding issue (which is also fixable for very little money). WanderingKid fucked around with this message at 01:45 on Oct 6, 2007 |
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Oct 6, 2007 01:36
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A home theatre system for 30 bucks?! I'm sorry but is this a joke?  You could probably get a remote control for 30 bucks...
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Oct 7, 2007 21:21
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A shame that those Klipsch ProMedia 2.1 system only dives to 60 bucks and under when its used or b-stock. One of those sites even quotes shipping at 35 bucks. The shipping cost on its own is over budget. Yikes. 
WanderingKid fucked around with this message at 23:14 on Oct 7, 2007 |
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Oct 7, 2007 23:08
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It will work even better if you get a ferrite core as well and wrap all the cables coming out of your speaker around them. You shouldn't get any radio frequency interference after that. You really do want a grounded plug as well as theres an electrocution risk by not having one. If it introduces grounding problems those are fixable by safer means anyway. Other than that, glad your problems are sorted. WanderingKid fucked around with this message at 02:33 on Oct 8, 2007 |
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Oct 8, 2007 02:07
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Pibborando San posted:Unless one cable is 6 feet and the other is 600, it won't make a difference (even then you may not notice a difference). That said I like to have identical lengths for aesthetics and my OCD. Add another zero to those THD figure. Delta 1010 DA - 0.0015% THD+N at -93dB Fireface 400 DA - 0.0015% THD+N at -96dB E-MU claims these figures: 1212M DA - 0.0006% THD+N at -100dB But really on the DA front it sounds pretty much the same compared to the FF400 and D1010 which is a good deal since you can buy one for about 120 bucks and on paper it beats the shite out of pretty much every convertor/receiver within 5 times of its price point. Its an option if you use a PC as a media hub. Those 10% THD figures have to be some sort of misprint. I have to believe that because I can't believe a major player in the market would peddle a device that is so obviously flawed and overpriced to the extent that I am seeing on numbers alone.
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Oct 28, 2007 05:24
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Those cables don't actually specify anything meaningful about their construction or material but generally speaking, if its under 3 feet in length you could string coathangers or paper clips together and it wouldn't make any significant difference. For longer cables you want to know the length of the conductor, what it is made out of and its thickness (measured in awg). Thicker cables have lower resistance. Shorter cables have lower resistance. You also want to know the diameter and material of the dialectric (insulation) between the cente conductor and the shield. The frequency of the signal along the conductor affects capacitive resistance though so the higher the frequency of the signal the more of an issue this becomes. Between all of those variables you can work out its impedance. For audio frequency signals, less than 20khz, cable impedance and return loss is irrelevant unless you run thousands of feet of it. For NTSC broadcast signals (megaherts range) its irrelevant up to about 50 feet. For HD video stuff its more of a problem because of the much higher frequency signal transfer and you really want to be looking at the shortest cable runs you can (less than 6 feet). As far as I know, Belden are one of the few cable manufacturers that even give you meaningful information about their products so they are a good choice. A quick google search shows they have a range of coaxial cables for video signal transfer for about 38 to 50 cents per foot. If you wan't a more in depth response - read this. With that information you could make your own cables. Although (literally) any old junk will do for audio purposes, I would tend to prefer buying from companies that don't have an over zealous marketing department, mainly because I despise the very common propagation of bullshit throughout this industry. It should not cost you more than a couple of quid, so don't get this idea into your head you need to spend loads of money because you really don't. Thats the electrical side of things covered. Mechanically, annealed copper is a good material choice for the conductor because its flexible. Just make sure the soldering work is good and it is otherwise rugged enough to take a moderate degree of flexing/twisting and you are grand. WanderingKid fucked around with this message at 14:46 on Oct 30, 2007 |
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Oct 30, 2007 14:34
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HDMI audio is digital. It just supports up to 24 bit/192khz. Whether you notice any difference above and beyond 16 bit/44.1khz depends very much on how well the source is recorded and produced and how well the conversion was performed at the relevant stages (if applicable). The difference is generally not great. Its barely noticeable really. You won't really notice what those extra bits are doing unless you listen really really loud. I'm talking not safe for your ears for more than about 10 minutes kind of loud.
WanderingKid fucked around with this message at 22:21 on Oct 30, 2007 |
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Oct 30, 2007 22:19
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Walked posted:Can you hook up 4ohm speakers to replace 8ohm speakers without ill effects? I'm slowly replacing my HTIB setup, and just ordered some Dana 630s, trying to avoid buying a new receiver until next paycheck though. It depends on the amp. So go and look at it and see what kind of load it can take. 2x 4 ohm speakers in parallel will present a 2 ohm load to the amp. Some amps can't handle that kind of power draw and will overheat unless they are specifically designed to cope with a nominal 2 ohm load.
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Nov 9, 2007 20:02
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pim01 posted:M-audio makes a nice product, although some people seem to have problems with the drivers. It depends on what you do. M-Audio Delta drivers are very solid in Windows XP and their ASIO driver is brilliant. It works with everything and works with every producer's rig I've ever had to install it in. Never had any technical problems with it and it has worked with every game I've ever played without hitch. Vista is another story but then alot of soundcards have sketchy drivers in Vista.
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Feb 10, 2008 00:10
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Silas the Mariner posted:I want to start making my own music using my PC and have heard good things about Garageband for the Mac. I've Googled and there are so many similar applications for the PC... so can anyone actually recommend one to me please? There are loads of applications that you can use to make music but you probably want to look at Digital Audio Workstations (DAWs) first. The most popular ones on PC would be (in no particular order): Steinberg Cubase SX Steinberg Nuendo Propellorhead Reason Imageline FL Studio Ableton Live Cakewalk Sonar E-magic Logic (now owned by Apple and up to date on PC only as of version 5.5) Digidesign Pro Tools (requires specific hardware) There are more but those are the ones I see crop up alot in various music production forums. Price varies alot as does workflow but apart from that it doesn't matter what you use since in a round about sort of way you can get to the same place using any of them. Your preference becomes just that - whichever application you use is a choice based upon what you are familiar with and what seems most intuitive to you personally. You can improvise around technical limitations in the software like the lack of sidechain inputs if you are smart and you know your tools well. Most of these programs have some way of connecting instruments and signal processers (both hardware and software) to a software mixer (which is integrated into the DAW). The only exception I can think of is Reason which is a closed platform (you are only able to use instruments and signal processors exclusive to Reason). All of them are MIDI compatible and with the exception of Reason is applicable to hardware (if present and set to send and/or receive MIDI messages). They all have some sort of piano roll/scoresheet/sequencer which allows you to click notes into a grid and have the computer play them back in a sequence. Wave editting: A number of the above DAWs have some sort of waveform editting capability but in my experience it is rarely anything but a compromise and for sample accurate wave editting you may want a program that specialises in doing just this. Sony Soundforge Adobe Audition Goldwave Reaper are some popular ones. Instruments and Signal Processors: Most DAWs support an open standard protocol that allows you to make virtual connections between software sound generators and signal processors. By far the most ubiquitous is Steinberg's VST. Most DAWs support VST (with the notable exception of Reason) and that gives you a choice of literally thousands of plugin synthesizers, romplers, samplers, effects, you name it. Go to https://www.kvraudio.com and browse through their plugin library to get an idea of sheer number of plugins you can get. Note that there are many excellent freeware ones. Also note that VST is so well established that software instruments and effects in general are simply referred to as VSTs even if they don't use the VST protocol. Other common open standard protocols are AUs (Audio Units) which are supported by Logic and Garageband and DX (Direct X) which is natively supported by FL Studio. The other common one is RTAS which is used by Pro Tools hardware and you will need some sort of Pro Tools rig to use these. Pro Tools sort of requires a thread all to itself because it requires dedicated hardware, is incredibly popular and is a well established platform for session recording. Another important one is Rewire which was developed by Propellorhead to allow Reason to be run as a plugin within another software environment (like Pro Tools or Cubase SX). When you do this you do not need to create a mixer in Reason - you simply wire Reason's native instruments and signal processors directly to the host's interface. This allows you to work in Reason (if you happen to like Reason's workflow) whilst giving you functionality that it would not normally allow via another host (i.e. VST support) - The number of software/hardware instruments and signal processors on the market are so great in number now that I couldn't really make a specific list here. Your best bet is to scout out new stuff on KVR and play around with some free demos. If you wish to make a particular sound you may want to post soundclips in the recording megathread in ML and ask what instrument or signal processor is capable of producing it. Hope this helps.
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Mar 31, 2008 03:50
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Pibborando San posted:edit: Producing bass is all about cone size and power. "There's no replacement for displacement" as they say. The lower the frequency, the more air needs to be moved for the same perceived loudness. Little bookshelf woofers simply can't move that much air, although a range to 55Hz is quite good for speakers that size (mine also use 5.25" woofers but go down to 40Hz because of a much larger enclosure but that's another story). I was looking into buying a Future Retro XS recently and stumbled upon this video on YouTube. Shame it costs 1.2 grand and you have to go on a waiting list but holy cow - look at that ADAM A7 woofer gyrate!
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May 27, 2008 03:05
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A decibel is a logarithmic unit. Assuming frequency, distance, everything bar amplitude are kept constant then the difference in sound pressure level between sound x and sound y = 10log(X/Y)dB So lets say X has twice as much power behind it as Y. 10log(2/1) = 10log2 = 3.01dB difference If X has 10 times more power behind it than Y then: 10log10 = 10dB difference If X has 1000 times more power behind it than Y then 10log1000 = 30dB difference With regards to perceived loudness - this would depend on the frequency of the signal since the ear is non linear and does not perceive all frequencies at the same amplitude as being equally loud. It is generally more sensitive to the ranges corresponding to human speech. Moreover, the ear generally begins to perceive all frequencies as being equally loud the higher the amplitude is. There is no accurate way of expressing a difference in perceived loudness between 2 sounds in decibels. Instead, you have people doing psychoacoustic experiements on people where they play a fixed frequency sine wave and increase the amplitude until a listener thinks its twice as loud as it was before. Theres a scale derived from experiments like this called phons and sones but I've never used them and don't know how it works. I've heard some people say twice the perceived loudness = 10dB but I have no idea how you would arrive at that number.
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Sep 26, 2008 01:10
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That frequency response information is meaningless anyway because it doesn't give you any information about how those numbers are achieved. It doesn't say if its weighted and it doesn't specify tolerances (i.e. flat to within +/1 dB between x hz and y khz). Then there is the influence of the omni mic and the room, which are almost certainly not constants. If you play around with an FFT you will quickly realize that you can make almost anything look flat by changing FFT block size and weighting, then arbitrarily lopping off the ends where it goes non linear. You aren't given enough information about how sensitivity is measured either. What you have are a bunch of numbers, a $50 price difference and the implicit suggestion that the more expensive one is better.
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Jan 11, 2012 11:02
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molotoveverything posted:I heard equalizers can damage your speakers if you slide any of the bands over +Odb because that introduces clipping. Is this true? Isn't moving the sliders up just increasing the volume? so along as you don't play your stereo too loudly too begin with and you don't boost any frequencies, lets say over +3 decibels you should be okay right? Not exactly true. A band EQ lets you increase gain in certain frequency bands independent of the others. Yes it is just like turning up the volume but only in that frequency band. You can still have too much gain and drive your amp and speakers over capacity by doing this but this is true of any device with a gain stage (frequency dependent or otherwise). You can boost +20dB on an EQ as long as you turn the main volume so you don't clip and you are mindful of the permanent damage high sound pressure levels can do to your ears. If it sounds distorted and/or your ears are burning after a few minutes, turn the volume down. The amount of gain you can apply in each band before distortion varies depending on the sound thats being played. If you have a looping 808 bass drum then most of the acoustic energy is concentrated in the low and mid frequency bands and theres very little energy near the upper limit of human hearing. So you can boost the hell out of the treble and not hear a whole lot of difference. But if you boost the hell out of the low frequency bands then you'll need to turn the main volume down alot because 808 bass drums typically have big bad bass and you've just made it disproportionately bigger and badder. It will also probably sound like poo poo which is another telltale sign that you are misusing an EQ. The opposite is true if you are looping only hihats. If you have a reasonably powerful soundsystem and you keep piling on the gain where theres alot of acoustic energy already, then you will eventually destroy something, whether its your speakers, your amp or your ears but that goes without saying. Just be sensible about it. WanderingKid fucked around with this message at 17:24 on May 29, 2012 |
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May 29, 2012 17:21
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revmoo posted:Tell me about soundproofing please. I need to reduce the noise pollution from my beat laboratory down a few notches. I'm thinking I need to fully cover the door into the room and put some panels on two of the walls. I don't need 100% reduction, just a bit more quiet. What should I expect to spend? I'm thinking around a hundred bucks or so to get a 50% reduction in sound leakage. Does that sound realistic? Also I have one window that currently has a few blankets nailed to the edges, is there a window-specific product I can get that would work better? Without knowing the dimensions of your room its impossible to price since the materials are typically priced per square meter. Not to burst your bubble or anything but I wouldn't be surprised if you had to pay over a hundred bucks per square meter in materials alone. There comes a point where you exhaust all easy/cheap options like closing your door and all thats left is putting up extra layers of drywall and soundboard. Then there is the ceiling and floor to deal with. It will add a couple of inches to the thickness of your walls and make your room smaller. Unless you are a general contractor or a serious DIY head, its probably not something you want to attempt yourself? Probably not a realistic option if you are living in rented accommodation. Honestly, the best advice I can give is to get in touch with a licensed general contractor in your area. Tell them what you need to do and get a quote, but it won't be one hundred bucks cheap. WanderingKid fucked around with this message at 13:10 on Jul 31, 2013 |
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Jul 31, 2013 12:04
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