Random converter. Set the positive and non-zero impedance value before conversions between decibel-milliwatt, decibel-microvolt, and other values. Teaspoons, tablespoons, cups or milliliters? Which are more convenient? Click or tap to find out! A logarithmic scale is often used when there is a large range of quantities like sound pressure, earthquake strength, light intensity, various frequency-dependent values like musical intervals, in antenna engineering, electronics, acoustics, RF engineering.

Logarithmic units allow representing a very large range of ratios by a small convenient number similar to scientific notation. At the same time, the sound power of quiet conversation is 0. Looks convenient?

## mW to dBm Conversion

Yes, but not to everyone! Actually, all people who are not mathematically or technically inclined can be easily confused when dealing with quantities expressed in logarithmic units like decibels.

Some people even think that logarithmic values more related to the era of slide rules than to the modern digital world. Logarithms were invented because they allowed turning multiplication into addition, which can be made much faster than multiplication. Among the scientists who contributed to the understanding of logarithms was the Scottish mathematician, physicist, and astronomer John Napier who in published a book introducing natural logarithms, which made calculations by hand much quicker.

An alternative unit to the decibel, the neper is named after him. A key tool for the practical use of logarithms was the table of logarithms. The first such table was compiled by an English mathematician Henry Briggs in Based on the work of John Napier and other scientists, an English mathematician and Anglican minister William Oughtred invented the slide rule, which was used until the mids by engineers and scientists including the author of this article.

In other words, the logarithm is a quantity representing the power to which a fixed number called the base must be raised to produce a given number. The answer is 2 or. Logarithmic units are widely used in science, technology and even in everyday things like photography and music.

There are absolute and relative units. Absolute logarithmic units express a physical value referenced to some specific value, for example, dBm is an absolute logarithmic unit of power with reference to 1 mW. Absolute units are ideal for describing a single valuenot a ratio of two values.

Absolute logarithmic units can be converted into non-logarithmic units of the same physical values.By using our site, you acknowledge that you have read and understand our Cookie PolicyPrivacy Policyand our Terms of Service.

Signal Processing Stack Exchange is a question and answer site for practitioners of the art and science of signal, image and video processing. It only takes a minute to sign up. I want to do some stuff on signals in the data, but only if it exceeds a certain range. What is the general procedure to get dB dBm, or anything from this kind of data? When you see a logarithmic dB meter, it is usually measuring the log of the power, i.

Remember, dB is a relative figure. Any signals stronger than that level will be clippeddistorting the signal. You can convert to dBm just by adding or subtracting the proper calibration value from the dBFS value — but you need to know that calibration for your hardware at the frequency of interest, such as by measuring it using a signal source of known power output ; it is impossible to perform that calibration purely in software since the digital samples are just numbers with no inherent units.

This is unreasonably large for a radio receiver. If you are just looking to ignore signals that aren't sufficiently strong this is known as carrier squelch or power squelchit doesn't matter what units you use, or even if they're logarithmic or linear, because you're just doing a greater-than comparison. It may also be relevant to note that if you filter a signal, you are by definition removing some of the signal power, so the measurement will be smaller.

Accordingly, the power in each bin depends on the width of the bin. Sign up to join this community. The best answers are voted up and rise to the top. Home Questions Tags Users Unanswered.

Asked 5 years, 4 months ago. Active 8 months ago.

**Sound Intensity and Decibels Distinctly Defined, Dude - Doc Physics**

Viewed 12k times. Kevin Reid 3 3 silver badges 12 12 bronze badges. What units would I use? If that's in Volt the formula above gives you dBV.

Active Oldest Votes. Units Remember, dB is a relative figure. Practical application If you are just looking to ignore signals that aren't sufficiently strong this is known as carrier squelch or power squelchit doesn't matter what units you use, or even if they're logarithmic or linear, because you're just doing a greater-than comparison.

Bank holidaysNote on bandwidth you probably don't need to read It may also be relevant to note that if you filter a signal, you are by definition removing some of the signal power, so the measurement will be smaller.

Kevin Reid Kevin Reid 3 3 silver badges 12 12 bronze badges.I am relatively inexperienced with ADCs so my apologies if this question is a bit naive but I could not find any clear answer here. With this set-up the input impedance of the receiver is a nice and fla 50 ohm from khz to about 50 Mhz.

The trouble is that I am unable to explain this result by looking at the datasheet of the LTC!! There is a 6dB difference that I can not explain in anyway! I concluded it is the SDR software that is responsible for this difference but I am not too sure really.

Can someone better explain me where this 6dB difference comes from? Is it the SDR software that I use that explains it? Is the reasoning I am making erroneous and if yes, can someone explain me why? A lot of this will be common sense but we might as well cover these for completeness and as a sanity check.

Yes I have factored in all the points except maybe your last bullet point regarding the ADC adjustment gain. It maybe the problem in fact??? Maybe the device or the PCB is somehow damaged and I am actually unable to select the 1Vpp to range for some reasons. You still get improved SNR but not as much as you might hope.

I don't know if this comes into play with what you are looking at. In an ideal system you are correct, dropping to the one volt range you will loose 6dB of SNR, but you are likely dominating the noise of the ADC with other noise sources so this 6dB will be reduces.

Lol missionsWhat is the jitter on the clock, what is the input bandwidth you are sampling, what is the frequency of the input signal? If you have a noisy amplifier that isn't being filter you can degrade the noise floor to a point where the sense range won't matter.

If you send a schematic I will have a better picture of where your noise is coming from. Each have a different behavior when I select an input range of 1V peak to peak instead of 2V peak to peak.

I am trying to understand if this difference of behavior is normal or not and from your answer I think yes but I would be grateful to you to confirm or infirm. The two circuits of the analogue front-ends are based on the ones described in figure 3 and 7 of the datasheet.

In circuit based on figure 3 of the datasheet, the analogue front-end is a ballun designed to operate between 5 and 60 Mhz, so within the first nyquist zone the sampling rate is Mhz. There is no anti-alias filter connected to the ballun, I have just connected a 50 ohm load to it.Join us now!

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Essentials Only Full Version. I'm sure her teacher is more familiar with the positive dB scale. The decibel will tell you how much louder or quieter one sound is when compaired to another, it will NOT tell you the precise volume of a sound although you could work that out For the purpose of the school work, dBFS would give perfectly usefull results, is readily available in Sonar and can be calibrated easily.

Just start with the loudest sound you're expecting to test and set the sound card input gain, or portable recorder mic gain to show Full Scale. Remember that for any of your results to be valid the sounds must be recorded with the same mic at the same distance to the sounds.

EDIT : or note the distance to the sound in the results. Sounds like a fun experiment.

Selphy cp1300And when the measurement equals that standardized level you get 0 dB-in-whatever-scale. In the case of dB SPL, when the measurement equals. In the case of dBFS, the reference is 'full scale' which, in digital, is the clipping point of the converter.

So when you're at the clipping point you get 0 dBFS and you go down from there. That's why you sometimes get positive and negative values for dB--it all depends on the reference level. When you interface digital gear with analog, you want to calibrate levels. Since analog usually has 'headroom' you calibrate so that dBFS equals 0 dBu on your analog meter--depending on the headroom of course. With dB, it's all relative.

You have to know the reference level for dB to mean anything. In that case, you're talking about the change in an already assumed dB scale of some sort. And it's not a linear scale. Karyn Now record your own Vuvuzela you do have one don't you? Who here doesn't?Don't forget the minus sign, when entering the distortion attenuation in decibels. The scales of VU meters show decibel values and percentage values. Addition of harmonic distortion attenuation of multiple devices. Total harmonic distortion THD is defined as the ratio of the rms voltage of the harmonics to that of the fundamental component.

This is accomplished by using a spectrum analyzer to obtain the level of each harmonic and performing an rms summation. The level is divided by the fundamental level, and cited as the total harmonic distortion expressed in percent.

This is the level difference between Harmonic Distortion unwanted overtones and the total signal in dB; see distortion.

The distortion of an audio device indicates the extent to which a sinusoidal input signal test tone a mplitude by non-linear distortions unwanted overtones and harmonics are added. It is therefore a measure of the occurring harmonic distortion. The value is given in percent, relative to the total signal. Harmonic distortion can be found with a minus sign in front of the dB value, the harmonic distortion is also specified with a positive sign.

There are variables that have no unit. The used browser does not support JavaScript. You will see the program but the function will not work. Nonlinear distortions are given as distortion attenuation a k in dB or as damping factor k THD in percent. Distortion attenuation a. Distortion factor THD k. Fill out the gray box and click at the respective calculation bar under it. Don't forget to enter the minus sign for the distortion attenuation in decibels.

Input k in percent. Input a k in dB with a minus sign.

Uss goldsborough reunion 2019The distortion factor was historically introduced before the THD-value and the ratio of the effective value of all harmonics to the RMS value of all harmonics plus fundamental frequency, that is the total signal. We use amplitude or voltage.By using our site, you acknowledge that you have read and understand our Cookie PolicyPrivacy Policyand our Terms of Service.

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### dBm to watts conversion

The similar unit dBov is defined in relation to power ratios so it's also an RMS measurementsuch that full-scale DC or square wave is 0 dBov, so that calculation is:. Sign up to join this community. The best answers are voted up and rise to the top. Home Questions Tags Users Unanswered. How to compute dBFS? Ask Question.

Asked 6 years, 11 months ago. Active 1 year, 8 months ago. Viewed 14k times. JustGoscha JustGoscha 1 1 gold badge 4 4 silver badges 11 11 bronze badges. Active Oldest Votes. Sign up or log in Sign up using Google. Sign up using Facebook. Sign up using Email and Password. Post as a guest Name. Email Required, but never shown. The Overflow Blog.

Podcast Programming tutorials can be a real drag. Featured on Meta. Community and Moderator guidelines for escalating issues via new response…. Feedback on Q2 Community Roadmap. Related 5. Hot Network Questions. Question feed.Fill out a gray box above and click the respective 'calculate' bar under it.

The reference voltage for 0 dBu is 0.

Sci fi moviesScroll down to find the formulas for voltage and power and the calculation of the absolute level. Explanation: What is "dBFS"? Digital Audio. Never express analog signal levels in terms of dBFS.

Follow this and you will not confuse anyone. The formulas for voltage and power and the calculation of the absolute level. The parameters of the mains or "power" sine wave form are summarized at the table below:.

The crest factor means the ratio of the peak voltage to the RMS voltage.

If you need to calculate an attenuator attenuation calculation you calculate a voltage divider. You will see the program but the function will not work. The used browser does not support JavaScript. Level L u. Level L V. Voltage V. Some say: The "u" in dBu implies that the load impedance is un specified, un terminated, and is likely to be high. What is dBu? All field quantities, like voltage or sound pressure are always true RMS values, if not otherwise stated. In mathematics, the root mean square abbreviated RMS or rmsalso known as the quadratic mean, is a statistical measure of the magnitude of a varying quantity.

## Converting dBFS to dBm with an SDR receiver based on the LTC2145-14 clocked at 125 Mhz

Also you cannot convert "dBA to volts" and vice versa. Conversion is only possible for measuring one single frequency.

Pro audio equipment often lists an A-weighted noise spec — not because it correlates well with our hearing — but because it can "hide" nasty hum components that make for bad noise specs. We don't use the dBm in audio engineering.

That belongs to power, we don't need here.

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