Question:
how can frequency affect on quality of a sound file?
BOBBY
2006-09-14 15:42:38 UTC
I want to know why when I record a wave file with higher frequency I have a better quality, for examples 22050 Mhz is better than 11000, but why? what does exactly frequency mean when we are talking about a music or wave? if I increase the frequency, should I always suppose a better quality?

Thanks for your help

Bobby
Three answers:
Derek Southerby
2006-09-14 15:54:13 UTC
Waves are actually used in sound files if you look REAL close at 'em, like I sometimes do in Audacity or something similar.



The frequency is actually a direct relation to the number of times the wave can go up and down in accordance with the sound. If it is unable to change with as much frequency, then it is unable to produce as precise a sound as one hears from the vibrations (frequencies) in the air that we hear. So with the lesser wave frequency, the lesser the quality of the file. Stick with 44100 or so, which is I think standard for high quality sound.



Also, there is a higher setting, something around 48k. You don't need to worry about this high of a setting. The human ear, after a point, can no longer tell the difference between sound qualities.



Bitrates are a little different; for example, I can't tell the difference between 192 and 320kpbs bitrate (CD quality) MP3s, but I can definitely tell a difference between 96 and 128 kbps. About 160 is a safe point, where quality is ensured for that.



Hope I could help,

Derek
Vince M
2006-09-14 15:56:26 UTC
Dont take this as a knowledgable answer, but as a suggestion.



Just as the quality of a subsequent image is better by capturing it at a higher mega pixel rate, something similar may occuby recording a sound file at a higher freqency. The higher frequency may allow you to capture more sound detail, such as harmonics that may be occuring, either, at a faster rate than the freqency records, or even occuring "between" the waves.



As in counting pixels, this would only apply to the original recording. One cant simply take a small digital photo file and increase the file size and expect an increase in quality, one should not expect a sound file, captured at a lower frequency to magically sound better if re recorded at a higher one.
hotsauce919rr
2006-09-14 15:52:04 UTC
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.



A high-pass filter is the opposite, and a bandpass filter is a combination of a high-pass and a low-pass.



The 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.



Contents [hide]

1 Examples of low-pass filters

2 Ideal and real filters

3 Electronic low-pass filters

3.1 Passive electronic realization

3.2 Passive digital realization

3.3 Active electronic realization

4 See also

5 External links







[edit]

Examples of low-pass filters



A 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.



Electronic low-pass filters are used to drive subwoofers and other types of loudspeakers, to block high pitches that they can't efficiently broadcast.



Radio transmitters use low-pass filters to block harmonic emissions which might cause interference with other communications.



An integrator is another example of a low-pass filter.



DSL splitters use low-pass and high-pass filters to separate DSL and POTS signals sharing the same pair of wires.



Low-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.



[edit]

Ideal and real filters

An 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.



However, 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.



Real 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.



The Nyquist–Shannon sampling theorem describes how to use a perfect low-pass filter (equivalent to the Whittaker–Shannon interpolation formula) to reconstruct a continuous signal from a sampled digital signal. Real digital-to-analog converters use real filter approximations.



[edit]

Electronic low-pass filters



The 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.



A first-order filter, for example, will reduce the signal strength by half (about −6 dB) every time the frequency doubles (goes up one octave). The magnitude Bode plot for a first-order filter looks like a horizontal line below the cutoff frequency, and a diagonal line above the cutoff frequency. There is also a "knee curve" at the boundary between the two, which smoothly transitions between the two straight line regions. See RC circuit.

A second-order filter does a better job of attenuating higher frequencies. The Bode plot for this type of filter resembles that of a first-order filter, except that it falls off more quickly. For example, a second-order Butterworth filter (which is a critically damped RLC circuit, with no peaking) will reduce the signal strength to one fourth its original level every time the frequency doubles (−12 dB per octave). Other second order filters may roll off at different rates initially depending on their Q factor, but approach the same final rate of −12 dB per octave. See RLC circuit.

Third and higher order filters are defined similarly. In general, the final rate of rolloff for an n-order filter is 6n dB per octave.

On any Butterworth filter, if one extends the horizontal line to the right and the diagonal line to the upper-left (the asymptotes of the function), they will intersect at exactly the "cutoff frequency". The frequency response at the cutoff frequency in a first-order filter is −3 dB below the horizontal line. The various types of filters — Butterworth filter, Chebyshev filter, Bessel filter, etc. — all have different-looking "knee curves". Many second-order filters are designed to have "peaking" or resonance, causing their frequency response at the cutoff frequency to be above the horizontal line. See electronic filter for other types.



The meanings of 'low' and 'high' — i.e. the cutoff frequency — depend on the characteristics of the filter. (The term "low-pass filter" merely refers to the shape of the filter's response. A high-pass filter could be built that cuts off at a lower frequency than any low-pass filter. It is their responses that set them apart.) Electronic circuits can be devised for any desired frequency range, right up through microwave frequencies (above 1000 MHz) and higher.



[edit]

Passive electronic realization



A passive low-pass filter showing impedances of the componentsOne simple electrical circuit that will serve as a low-pass filter consists of a resistor in series with a load, and a capacitor in parallel with the load. The capacitor exhibits reactance, and blocks low-frequency signals, causing them to go through the load instead. At higher frequencies the reactance drops, and the capacitor effectively functions as a short circuit. The combination of resistance and capacitance gives you the time constant of the filter τ = RC (represented by the greek letter tau). The break frequency, also called the turnover frequency or cutoff frequency (in hertz), is determined by the time constant:







or equivalently (in radians per second):







One way to understand this circuit is to focus on the time the capacitor takes to charge. It takes time to charge or discharge the capacitor through that resistor:



At low frequencies, there is plenty of time for the capacitor to charge up to practically the same voltage as the input voltage.

At high frequencies, the capacitor only has time to charge up a small amount before the input switches direction. The output goes up and down only a small fraction of the amount the input goes up and down. At double the frequency, there's only time for it to charge up half the amount.

Another way to understand this circuit is with the idea of reactance at a particular frequency:



Since DC cannot flow through the capacitor, DC input must "flow out" the path marked Vout (analogous to removing the capacitor).

Since AC flows very well through the capacitor — almost as well as it flows through solid wire — AC input "flows out" through the capacitor, effectively short circuiting to ground (analogous to replacing the capacitor with just a wire).

It should be noted that the capacitor is not an "on/off" object (like the block or pass fluidic explanation above). The capacitor will variably act between these two extremes. It is the Bode plot and frequency response that show this variability.



[edit]

Passive digital realization

A model of a simple digital implementation of a low-pass RC filter is:







yn = αx + (1 − α)yn − 1



where:



yn is the current output value

yn − 1 is the previous output value

x input value

Δt is the time interval between samples

RC is the time constant

[edit]

Active electronic realization



An active low-pass filterAnother type of electrical circuit is an active low-pass filter.



In this example, the cutoff frequency (in hertz) is defined as:







or equivalently (in radians per second):







The gain in the passband is , and the stopband drops off at −6 dB per octave, as it is a first-order filter.



Many times, a simple gain or attenuation amplifier (See operational amplifier) is turned into a lowpass filter by adding the capacitor C. This decreases the frequency response at high frequencies and helps to avoid oscillation in the amplifier. For example, an audio amplifier can be made into a lowpass filter with cutoff frequency 100 kHz to reduce gain at frequencies which would otherwise oscillate. Since the audio band (what we can hear) only goes up to 20 kHz or so, the frequencies of interest fall entirely in the passband, and the amplifier behaves the same way as far as audio is concerned.


This content was originally posted on Y! Answers, a Q&A website that shut down in 2021.
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