what do you mean by audio low pass filter??
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A low-pass filter is a filter that passes low frequencies well, but attenuates (or reduces) frequencies higher than the cutoff frequency. The actual amount of attenuation for each frequency varies from filter to filter. It is sometimes called a high-cut filter, or treble cut filter when used in audio applications.\n\nA high-pass filter is the opposite, and a bandpass filter is a combination of a high-pass and a low-pass.\n\nThe concept of a low-pass filter exists in many different forms, including electronic circuits (like a hiss filter used in audio), digital algorithms for smoothing sets of data, acoustic barriers, blurring of images, and so on. Low-pass filters play the same role in signal processing that moving averages do in some other fields, such as finance; both tools provide a smoother form of a signal which removes the short-term oscillations, leaving only the long-term trend.\n\nContents [hide]\n1 Examples of low-pass filters \n2 Ideal and real filters \n3 Electronic low-pass filters \n3.1 Passive electronic realization \n3.2 Active electronic realization \n4 See also \n5 External links \n \n\n\n[edit]\nExamples of low-pass filters\n \nA low-pass electronic filter realized by an RC circuitA solid barrier acts as a low-pass filter for sound waves. When music is playing in another room, the low notes are easily heard, while the high notes are largely filtered out. Similarly, very loud music played in one car is heard as a low throbbing by occupants of other cars, because the closed vehicles (and air gap) function as a very low-pass filter, attenuating all of the treble.\n\nElectronic low-pass filters are used to drive subwoofers and other types of loudspeakers, to block high pitches that they can't efficiently broadcast.\n\nRadio transmitters use low-pass filters to block harmonic emissions which might cause interference with other communications.\n\nDSL splitters use low-pass and high-pass filters to separate DSL and POTS signals sharing the same pair of wires.\n\nLow-pass filters also play a significant role in the sculpting of sound for electronic music as created by analogue synthesisers, for example the TB-303 created by the Roland corporation. See subtractive synthesis.\n\n[edit]\nIdeal and real filters\nAn ideal low-pass filter completely eliminates all frequencies above the cut-off frequency while passing those below unchanged. The transition region present in practical filters does not exist. An ideal low pass filter can be realized mathematically (theoretically) by multiplying a signal by the rectangular function in the frequency domain or, equivalently, convolution with a sinc function in the time domain.\n\nHowever, this filter is not realizable for practical, real signals because the sinc function extends to infinity. The filter would therefore need to predict the future and have infinite knowledge of the past in order to perform the convolution. It is effectively realizable for pre-recorded digital signals (by padding the ends of the signal with zeros to the point that the error after filtering is less than the quantization error), or perfectly cyclic signals that repeat for infinity.\n\nReal filters for real-time applications approximate the ideal filter by delaying the signal for a small period of time, allowing them to "see" a little bit into the future. This is manifested as phase shift. Greater accuracy in approximation requires a longer delay.\n\nThe Nyquist-Shannon sampling theorem describes how to use a perfect low-pass filter and the Nyquist-Shannon interpolation formula to reconstruct a continuous signal from a sampled digital signal. Real digital-to-analog converters use real filter approximations.\n\n[edit]\nElectronic low-pass filters\n \nThe frequency response of a first-order filterThere are a great many different types of filter circuits, with different responses to changing frequency. The frequency response of a filter is generally represented using a Bode plot.\n\nA first-
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Tone Controls\n\nThe most common equalization system is probably the tone controls that can be found on most stereo systems. They provide a quick and easy way to adjust the sound to suit your tastes and partially compensate for the room.\n\nYou will often find the controls labeled 'bass' and 'treble.' Each of those knobs controls a special type of filter called a shelving filter, or more precisely, a lowpass shelving filter and a highpass shelving filter respectively. A gain greater than one will boost the signal, and if the gain is less than one, it cuts the sound. \n\nhttp://www.harmony-central.com/Effects/Articles/Equalization/#ton
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Paul's quite right -3.01(something) dB is half the power. It's based on the Bel (after Alexander Graham ) which is log10(some power / some other power) but is a bit of a big change (3 Bel is a factor of 1000) so people use the decibel (dB) which is 10*log10(P2/P1)\n30 dB =1000  -30 dB = 1/1000\n\nIn terms of your low pass filter, if P2 is the output power you get if you put a clean narrowband signal of power P1 and center frequency w into your filter then when 10*log10(P2/P1)~=-3,  w is your 3dB frequency.  \n\n\nGood luck
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You need a decibel meter
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Traditionally in English, noise is an unwanted sound (slighly bad or neutral).\n\nIn dealing with audio, noise is random hissing, like the sound of a TV with no arial (the visual of this is also called noise). When talking about signal to noise ratio, a high ratio means the desired sounds are much louder than the random noise (can be caused by interference).
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AudioMulch is pretty cool and relatively simple to use.  It also supports VST plugins - software modules you can "bolt on" to AudioMulch to add delay, chorus, distortion, etc...
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its called "feedback", distance your microphone fom your speakers.\n"should solve the problem".
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check out Audio Cleaning Lab by Magix\n\nwww.magix.com
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The answer is probably a yes, but the exact situation will change the answer.\n\nIf you are talking about background noise between the voice phrases you want to hear (Called the Noise Floor), this can be fixed fairly easily if the noise floor isnt too high. What you would do is use a device called an Expander/Gate. A Gate basically mutes (True muting or reducing volume by -60db or more) the audio when the volume goes below a certain level (Basically when you're not talking). If the noise isnt too loud, it will usually be masked by the person talking.\n\nAn Expander is like a more subtle version of a gate. Rather than killing the audio completely, it increases the difference of volume (expands) between loud and soft parts of the audio.\n\nMost video recorders reduce the dynamic range of audio using a proccess called compression. This has the effect of making soft parts of the audio intelligable, while making the loud parts not be over bearing. The problem however, is that this proccess increases the noise floor. I would hazard to guess that this is a major part of the problem you are experiencing if the audio you are using is running through the video recorder.\n\nIf this is the case, a gate will have a hard time functioning because the difference between the loud and soft parts of the audio will be hard for the device to detect (as the difference has been reduced). To compensate, stick an expander before the gate. This will add some dynamic range back to the audio, making it easier for a gate to work. This entirely depends on how much the dynamic range has been compressed... the more it has, the less effective these devices will work.\n\nThe operation of expanders/gates can be tricky. Threshold, attack, release, sidechain/key inputs, and ratio control operation are not immediatly obvious to someone who hasnt used an audio device with these functions on it. However, there happens to be tons of info on the net about these devices.\n\nRegarding using equalizers... they are mostly useful for background hums rather than broadband noise. If the noise sounds like noise (Like a "shhhhh", people talking in the background, wind, etc), EQs will not be very effective. If the sound is more like a buzz or humm or whine (Like an electric motor), then EQ will do very well. Look for a "parateric" EQ when doing this. There are tons of software that has this. Parametric EQs will allow you to not only center on the exact frequency, but the bandwith of the EQ can be set to be very narrow. The more narrow the bandwidth is, the less you effect your original sound. Also, you can kill pretty much all frequencies below 60-80hz (depending on the voice) without have a noticeable impact on the the voice. This will reduce some of the low end rumble in the audio.\n\nIf neither of these options work to satisfaction, one last option is a sampling noise reduction system. Not many programs have this... a free one would be GoldWave (Not very high quality, but still works in moderation). What it does is it takes a sample of the noise floor, notes the frequency response of the noise (called a stamp or print), and then reduces these frequencies from the audio. Used to the extreme, it will produce VERY weird artifacts... used well, this kind of noise reduction is currently the most effective in the world. If you can find software that does this dynamically (doesnt take just one "stamp"), the results can seem like magic.\n\nAll of these options can be attained through hardware, stand alone hardware, or via plugins.\n\nHope that helps.
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I would look into fitted plugs - check your area for an audiologist who can make them for you.  They are comfortable since they are fitted to your ear, and they can give you filters with varying levels of protection.
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24bit aftermarket soundcard.\n\nWorked for me. My onboard ACL850 was crappers.\n\n:D
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Need to have common mode noise rejection on the front end. \n\nLeads should have a ferrite bead or torus around a cable to prevent radio frequency interference from being transmitted down the wire.\n\nAliasing:\nTo avoid aliasing, the input to an ADC must be low-pass filtered to remove frequencies above half the sampling rate. This filter is called an anti-aliasing filter, and is essential for a practical ADC system.\n\nDither:\nIn A to D converters, performance can be improved using dither. This is a very small amount of random noise (white noise) which is added to the input before conversion. Its amplitude is set to be about half of the least significant bit. Its effect is to cause the state of the LSB to randomly oscillate between 0 and 1 in the presence of very low levels of input, rather than sticking at a fixed value. Rather than the signal simply getting cut off altogether at this low level (which is only being quantized to a resolution of 1 bit), it extends the effective range of signals that the A to D converter can convert, at the expense of a slight increase in noise - effectively the quantization error is diffused across a series of noise values which is far less objectionable than a hard cutoff. The result is an accurate representation of the signal over time. A suitable filter at the output of the system can thus recover this small signal variation.\n\nSampling Rate:\nSince a practical ADC cannot make an instantaneous conversion, the input value must necessarily be held constant during the time that the converter performs a conversion (called the conversion time). An input circuit called a sample and hold performs this task—in most cases by using a capacitor to store the analogue voltage at the input, and using an electronic switch or gate to disconnect the capacitor from the input.
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http://www.cooltoolawards.com/software/multimedia/audiogenerator.htm
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I'm assuming you're doing computer audio out TO some sort of sound system: eg. cord is going from computer to speakers.\n\nThere are 3 possible sources of noise:\n\n1.  Your sound card sucks and is not shielded.  This is unlikely unless you spent less than $10 on it.\n\n2.  The speakers themselves have poor electronics.  My guess is that since you called it an 'audio receiver' you're savvy enough to not have crappy speakers.\n\n3.  The cables themselves.  This is the most common source of noise in audio systems.  If you get poor quality or non-shielded cables, you'll get noise on the line from ambient electromagnetic radiation.  Make sure you get really good cables, because they are often the weakest link in ANY electronics system.\n\nMonster makes good cables, but so do other companies and Monster tends to be a little over priced.  You can tell a good cable by how heavy it is though, and how flexible it is - poor quality cables will feel light and be very, very floppy, whereas high quality cables will feel heavier, thicker, and won't flex quite as much due to the thicker shielding.
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goto the view menu- enhancements- show enhancements - click on now playing- click the arrow until you see the graphic equalizer and adjust that until it sounds right. thats all i can think of. also check what you have windows media players volume set on and what it is set on your computer speakers. that might affect the sound too.
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Broadband spectrum of non- regular pulses from 10 Hertz to\n200 KiloHertz in bursts of alternating frequency ranges\nof high/low,  to cover the entire human ear range of\nsounds, in a manner that is not consistent with other\nsounds, such as the regular pulse of CD/radio/MP3 speakers within a passenger vehicle playing music, and as diverse\na range so that people with a gap in one or more frequency ranges will pick up the other frequencies, and...\nirregular enough that the pulsing does not combine in a\nheterodyne-type of mixture with road tire noise, heater/airconditioning blower fan pulses, and composite road /drive train/wind noise to creat a harmonic blend that is dampened. Irregularity is a key to increase the awarness time. Alternating high/low frequencies means that the low frequencies travel well through the heavy frame members, and \nhigh frequencies travel thorough lighter areas of vehicle construction. Expensive automobiles have heavy ( weighted) padding glued to many interior car sections to cut out road noises - and it works. They also have lighter styrofoam and composite materials filling air pockets to dampen the low frequency road and drive train "throbs" comming from the engine compartment. I have had the radio on an alternative music station with synthesizers blasting unusual\nfrequencies, and much to my horror, glanced in my mirror to find an ambulance behind me trying to get my attention. It was not until I turned off the radio that I heard the pathetic BEE BAW of the siren. ... Locomotives use two or three different frequencies, which are quite LOW on the human audio scale, and these low frequencies can be heard MILES away. Transports, again use two or three lower frequency blasts from air horns, and again, the lower frequencies can vibarate right through most well insulated vehicles. The "standard" pathetic BEE BAW two tone regular patterned, weak siren common today is too narrow a region of the human range, too regular, too weak, and easily blocked by any expensive vehicle's sound insulation. Turning on any music can totally obliterate the noise.... Much better technology can be employed, ... and should be!\n How come I can hear some kids punked out stereo system BOOMing 2 city blocks away, as the car races towards me, and I can't hear an ambulance that is sitting on my back bumper?\nGo figure.
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Try here at the bottom.\nhttp://www.musicfromouterspace.com/analogsynth/coolnewlfo.html