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#41
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dBFS
In comp.dsp davew wrote:
(snip) The VU meter is basically a bridge rectifier followed by a low pass filter. So it's mean rectified, not mean squared. So a 1dB difference for pure tone. Well, you can just change the numbers on the scale to represent the squared value. It is logarithmic (dB) in either case. We don't tend to use rms or mean whatever when talking about audio levels though, we just say "level" and that seems to be good enough. It's understood that when you reach 0dBFS you're in trouble shortly thereafter. In EE, RMS can mean different things. In some cases, it is the value that a sine would give, with the specified RMS value. That is, for peak-to-peak reading VTVM or mean absolute value reading analog meters, the calibration is for a sine. The only reliable means to know when the limit has been reached is the peak indicator (and then only if it's true peak which was the subject of a recent thread or two). As far as a reasonable measure of loudness, neither VU or peak or PPM are good enough, just a guide. But even with true peak, for live recording it is too late by the time you find the peak. This has all changed no with loudness metering, but that's another subject. -- glen |
#42
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
Al Clark wrote:
0 dBFS is a digital specification that represents the maximum level that a data converter can convert. For example 0x7FFFFF... or 0x800000.. assuming twos complement. It follows that the level of all signals will be = 0 dBFS It has nothing to do with the rms level at all. I woulda said that a 0 dBFS signal has the RMS level of a sine wave that just barely doesn't clip a converter (or, a hardlimited channel; it does not need to be a converter). (This is an important concept, of sorts, in that is shows that an N bit converter has a full-scale-signal to quantization noise ratio of 6*N + 2 dB, not the 6*N + 5 dB that some texts claim.) One can debate these things. Most outcomes of such debates are equivalent within a factor of two. Steve |
#43
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
In article , Randy Yates wrote:
Why do you say the VU measurements aren't RMS? Because of the meter ballistics? That's what I would traditionally say. A steady-state input into a VU measurment gives a value that should correspond to RMS, or at least average absolute value if it's a cheap VU meter. But a dynamic input into a VU meter would have peaks attenuated from RMS. When I designed consoles, we chose a compromise --peakier than VU, not as peaky as instant RMS. Got no complaints. Don't make such meter decisions without consulting marketing, and figuring out what the customers want. There is no one true way. Steve |
#44
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
In comp.dsp Scott Dorsey wrote:
(snip) It does seem that the FS applies to measuring devices, either analog or digital meters. NO. FS applies ONLY to digital system. When all the bits are set to 1, the meter goes to FS. Well, in at least one place FS is used for analog meters, and that is the sensitivity of an analog voltmeter, commonly specified in ohms/volt FS. -- glen |
#46
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
In comp.dsp wrote: Now we know when calibrating with a tone, 0 VU needs to be something like -20dBFS to avoid clipping when audio at 0 VU replaces the tone. On 11/20/2010 2:54 AM, glen herrmannsfeldt wrote: I thought for CDs the number is supposed to be -12dB, but I am not sure what is supposed to be at -12dB. Some CD recorders have an arrow at -12dB. (And the meter isn't likely to have VU ballistics.) It's just a number. It isn't "supposed" to be anything. I suppose that 12 dB of headroom would be reasonable if you were compressing everything that you recorded, as is the case with a lot of pop music today ("the loudness war"). When you reduce the dynamic range, you reduce the need for a lot of headroom since you have less uncertainty about the peak level. I sometimes record live high-school orchestra concerts. Because it is hard to know the level, I record 24 bit, then find the peak and RMS of each track. Then I figure out how many bits to scale each track by so that peaks stay below FS, and they should sound about right together. I suppose that's one way of mixing when you aren't listening and don't have controls handy. From a computer standpoint, it makes a certain degree of sense, but from a musical and artistic standpoint, it's absurd. -- "Today's production equipment is IT based and cannot be operated without a passing knowledge of computing, although it seems that it can be operated without a passing knowledge of audio." - John Watkinson http://mikeriversaudio.wordpress.com - useful and interesting audio stuff |
#47
Posted to rec.audio.pro
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dBFS
On 11/19/2010 9:07 PM, Scott Dorsey wrote:
I don't know what a Fs/4 sine wave is. I assumed he was talking about the frequency of the test waveform, one fourth of the sample rate. Perhaps he's looking at something tricky here (remember, this thread has drifted over to comp.dsp where they know nothing about music or audio - note my tag line), sampling a single frequency at an integral ratio of the sample rate so that every sample occurs at the same point on the waveform. That could lead someone who doesn't believe in the sampling theorem to the conclusion that the reconstructed waveform would be at the wrong level. -- "Today's production equipment is IT based and cannot be operated without a passing knowledge of computing, although it seems that it can be operated without a passing knowledge of audio." - John Watkinson http://mikeriversaudio.wordpress.com - useful and interesting audio stuff |
#48
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
On 11/19/2010 10:44 PM, Randy Yates wrote:
What units would a typical professional digital audio system use to measure RMS values of digital signals? They wouldn't, because they don't care. a professional (or even amateur) digital audio system doesn't care what the level is until it reaches 0 dBFS. You couldn't really be sure you were calculating the RMS value correctly based on looking at individual samples since there's a good chance that none of the samples occurred at the peak of the waveform. You'd have to convert the digital samples back to analog in order to accurately reconstruct the waveform. I suppose there's an arithmetic way to do that, but I'll leave it to the computer guys to figure that out. If I wanted to know the RMS value of a digital signal, I'd play it back through a D/A converter with a known relationship between volts and bits, measure the voltage with an RMS voltmeter, and then convert that back to bits. But I still don't understand your real question. I can read the words you're writing, but I can't get the significance of either the question or the answer. -- "Today's production equipment is IT based and cannot be operated without a passing knowledge of computing, although it seems that it can be operated without a passing knowledge of audio." - John Watkinson http://mikeriversaudio.wordpress.com - useful and interesting audio stuff |
#49
Posted to rec.audio.pro
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dBFS
On Sat, 20 Nov 2010 06:50:53 -0500, Mike Rivers
wrote: On 11/19/2010 9:07 PM, Scott Dorsey wrote: I don't know what a Fs/4 sine wave is. I assumed he was talking about the frequency of the test waveform, one fourth of the sample rate. Perhaps he's looking at something tricky here (remember, this thread has drifted over to comp.dsp where they know nothing about music or audio - note my tag line), sampling a single frequency at an integral ratio of the sample rate so that every sample occurs at the same point on the waveform. That could lead someone who doesn't believe in the sampling theorem to the conclusion that the reconstructed waveform would be at the wrong level. Sampling at an integer multiple of the frequency just makes the process clean. There is no leakage between frequency bins, hence no need for a windowing function to get clean sidebands. As you say, it certainly doesn't result in wrong levels. In fact I would say that where you have the option, it is the right choice to make. I can see that to somebody unfamiliar with sampling theory, they could assume that because - say - the peaks of the waveform never get sampled, they will be missed. Not so, of course. d |
#50
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
On 11/20/2010 12:00 AM, Randy Yates wrote:
What units would a typical professional digital audio system use to measure RMS values of digital signals? OK, well here are three examples: http://www.digitalsignallabs.com/m-1.png You'll note that all of the meters in your examples have their scale calibrated in dBFS. Hence my answer several messages back that you dismissed as "no information" is exactly correct. Where you have a digital meter that indicates both peak and "RMS" they're both displayed on the same scale. The program that calculates the position of the RMS meter bar has plenty of time to look at samples and can make a reasonable estimate of the RMS value regardless of the wave shape. The idea behind having such a meter is not as a guide for recording level, available headroom, or dynamic range, but rather to give an indication of the perceived loudness at a given time. It's somewhat useful to mastering engineers because it gives them a sense of how heavily the program material has been compressed in an attempt to get the average/RMS/perceived volume level closer to the peak volume level. -- "Today's production equipment is IT based and cannot be operated without a passing knowledge of computing, although it seems that it can be operated without a passing knowledge of audio." - John Watkinson http://mikeriversaudio.wordpress.com - useful and interesting audio stuff |
#51
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
In article , Randy Yates wrote:
So to infer from your response, one definition of dBFS would be something like this: dBFS = 20 * log_10(XPEAK / FSPEAK), Yes, I have already said that three times now in three different ways. What units would a typical professional digital audio system use to measure RMS values of digital signals? I don't know, because I have never seen actual instantaneous RMS values ever displayed anywhere. I have seen dozens of different time-averaged levels displayed in the digital domain but none of them were really RMS. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#52
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
glen herrmannsfeldt wrote:
Now we know when calibrating with a tone, 0 VU needs to be something like -20dBFS to avoid clipping when audio at 0 VU replaces the tone. I thought for CDs the number is supposed to be -12dB, but I am not sure what is supposed to be at -12dB. Some CD recorders have an arrow at -12dB. (And the meter isn't likely to have VU ballistics.) That's something different. The CD recorder meter, like almost all digital meters, is peak-reading. If you set your average levels so that they peak at -12dBFS, that gives you some headroom for an unexpected crescendo. It is a good rule of thumb. It is, however, only a rule of thumb. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#53
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
Mike Rivers wrote:
On 11/20/2010 12:00 AM, Randy Yates wrote: What units would a typical professional digital audio system use to measure RMS values of digital signals? OK, well here are three examples: http://www.digitalsignallabs.com/m-1.png You'll note that all of the meters in your examples have their scale calibrated in dBFS. Hence my answer several messages back that you dismissed as "no information" is exactly correct. Where you have a digital meter that indicates both peak and "RMS" they're both displayed on the same scale. The program that calculates the position of the RMS meter bar has plenty of time to look at samples and can make a reasonable estimate of the RMS value regardless of the wave shape. Note that those "RMS" measurements aren't RMS at all but are instantaneous weighted averages of various sorts. And you'll note that they won't match one another at all on different software or devices (UNLESS they match the ITU standard, which lots of systems including Pro Tools do not). If two devices give different values, which is correct? --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#54
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
Why does this have different than analog just because it is digital?
dBFS defines the reference for calculating the dB's just like dBV or dBu or dBSPL or dBHL or any other dB measure. It does not define the measure - only the reference. 0dBV is defined as the reference being 1V. any other dBV is 20*log10(abs(xV/1V)) -3dBVrms, 0dBVpk, and +6dBVpkpk all describe the same sine wave - the dBV describes the reference value while the qualifier rms, pk, and pkpk describes the type of measure. The confusion may because dBFS defines the reference level as the maximum magnitude that can be digitally represented in a system, the point of digital clipping. But you would still have to use the qualifier to describe your measurement as pk, rms, or pkpk. The reference only tells you how you are dividing to get the dB scale - it does not tell you what your measurement type was. dBFS = 20*log10(abs(x#/max#)) -3dBFSrms, 0dBFSpk, and +6dBFSpkpk describe the same maximum digital sine wave - and if 0dBFS = 0dBV in the D2A/A2D calibration it has equivalent meaning in the analog world of sine wave at analog clipping Though not all D2A/A2D have that convenient of calibration, nor is it likely the analog trim matches the analog clip so you can certainly have more peak analog signal than 0dBV. And if you want to describe that signal you would give the measure relative to 0V and the qualifier for the type of measure. |
#55
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
On Sat, 20 Nov 2010 11:28:45 -0500, Dick Pierce
wrote: Scott Dorsey wrote: I don't know, because I have never seen actual instantaneous RMS values ever displayed anywhere. The term "instantaneous RMS value" is itself meaningless. But if you have two samples you can have an RMS value with a meaning, albeit a pretty meaningless meaning. Within the terms of the act, one would assume a complete cycle of waveform has been measured to reach a conclusion about its RMS level. d |
#56
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
"Randy Yates" wrote in message
... | Al Clark writes: | | 0 dBFS is a digital specification that represents the maximum level that a | data converter can convert. For example 0x7FFFFF... or 0x800000.. assuming | twos complement. | | It follows that the level of all signals will be = 0 dBFS | | It has nothing to do with the rms level at all. | | The relationship between nominal rms levels and dBFS is loose. | The more bits you assign for headroom, the less bits you have for low | levels. | | A common professional audio tradeoff is 4dBu = -18dBFS. This would mean | that a +22dBu sine wave would just fit into the converter range without | clipping. | | It is also common that 0dBu = -18dBFS. This means the maximum input level | is +18dBu. | | Hey Al, | | I'm trying hard to see an answer to my question in what you wrote and | failing. | | Let me respond to you with this question: If you had a meter that | -24 dBFS with a Fs/4 sine wave, what would the peak value of the | sine wave be? | | --Randy | | | | | Al Clark | www.danvillesignal.com | | | | | Rick | | | On Nov 19, 5:09 pm, Randy Yates wrote: | Cross-posting to comp.dsp. | | --RY | | | | Randy Yates writes: | Also, what reference level does an analog peak-reading meter | use? | | --Randy | | Randy Yates writes: | | Hi, | | Some had responded here to my recent inquiry on levels that dBFS is a | peak measurement. | | If an RMS measurement needs to be made for a digital signal (i.e., on | a | digital mixing console or a ProTools plugin), what units are | utilized? | I | thought they were dBFS, i.e., that dBFS was an RMS measurement. | Apparently I am incorrect. Somebody please set me straight. | | -- | Randy Yates % "Ticket to the m | oon, flight leaves here today | Digital Signal Labs % from Satellite 2" | % 'Ticket To The Moon'http://w | ww.digitalsignallabs.com% *Time*, Electric Light Orchestra | | | | -- | Randy Yates % "She has an IQ of 1001, she has a jumpsuit | Digital Signal Labs % on, and she's also a telephone." | % | http://www.digitalsignallabs.com % 'Yours Truly, 2095', *Time*, ELO Randy, I think you and Bill Graham ought to get together. There's nothing worse than a troll who doesn't think he's a troll. Steve King |
#57
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
Dick Pierce wrote:
Scott Dorsey wrote: I don't know, because I have never seen actual instantaneous RMS values ever displayed anywhere. The term "instantaneous RMS value" is itself meaningless. BINGO! Mr. Pierce wins the kewpie doll! --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#58
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
But, regardless, there aren't any "different versions" of RMS, just the one: * * sqrt(sum(X1^2 + X2^2 ... +Xn^2) / n) but there are different values of n... and for a sine tone, the value of n doesn't matter because a tone is the same all the time....., but for real audio the value of n does matter and that is the crux of the discussion.. Mark |
#59
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
On Nov 20, 12:02*pm, (Scott Dorsey) wrote:
Dick Pierce wrote: Scott Dorsey wrote: I don't know, because I have never seen actual instantaneous RMS values ever displayed anywhere. The term "instantaneous RMS value" is itself meaningless. BINGO! *Mr. Pierce wins the kewpie doll! --scott Except that it is wrong. What is the instantaneous RMS value of -1... +1. RMS doesn't have to be an integral or a sum. b Rick |
#60
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
On Sat, 20 Nov 2010 11:43:03 -0800 (PST), rickman
wrote: On Nov 20, 12:02*pm, (Scott Dorsey) wrote: Dick Pierce wrote: Scott Dorsey wrote: I don't know, because I have never seen actual instantaneous RMS values ever displayed anywhere. The term "instantaneous RMS value" is itself meaningless. BINGO! *Mr. Pierce wins the kewpie doll! --scott Except that it is wrong. What is the instantaneous RMS value of -1... +1. RMS doesn't have to be an integral or a sum. b Rick It isn't. It is the square root of the mean of the squares. In the case of -1, +1 it is 1. The inclusion of the term "mean" says that there must be at least two measurements - any fewer and you can't have a mean. d |
#62
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
On Nov 20, 2:52*pm, (Don Pearce) wrote:
On Sat, 20 Nov 2010 11:43:03 -0800 (PST), rickman wrote: On Nov 20, 12:02 pm, (Scott Dorsey) wrote: Dick Pierce wrote: Scott Dorsey wrote: I don't know, because I have never seen actual instantaneous RMS values ever displayed anywhere. The term "instantaneous RMS value" is itself meaningless. BINGO! Mr. Pierce wins the kewpie doll! --scott Except that it is wrong. *What is the instantaneous RMS value of -1... +1. *RMS doesn't have to be an integral or a sum. b Rick It isn't. It is the square root of the mean of the squares. In the case of -1, +1 it is 1. The inclusion of the term "mean" says that there must be at least two measurements - any fewer and you can't have a mean. d I don't recall any such restriction on N. Certainly the measurement has just as much meaning with N = 1 as any greater N. Think of it as a limit as N approaches 1 then. The point is that it is as valid a measurement for a single point as it is for many points. It represents the equivalent voltage that would produce the same power as DC of the same voltage. Rick |
#63
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
On Sat, 20 Nov 2010 12:53:16 -0800 (PST), rickman
wrote: On Nov 20, 2:52*pm, (Don Pearce) wrote: On Sat, 20 Nov 2010 11:43:03 -0800 (PST), rickman wrote: On Nov 20, 12:02 pm, (Scott Dorsey) wrote: Dick Pierce wrote: Scott Dorsey wrote: I don't know, because I have never seen actual instantaneous RMS values ever displayed anywhere. The term "instantaneous RMS value" is itself meaningless. BINGO! Mr. Pierce wins the kewpie doll! --scott Except that it is wrong. *What is the instantaneous RMS value of -1... +1. *RMS doesn't have to be an integral or a sum. b Rick It isn't. It is the square root of the mean of the squares. In the case of -1, +1 it is 1. The inclusion of the term "mean" says that there must be at least two measurements - any fewer and you can't have a mean. d I don't recall any such restriction on N. Certainly the measurement has just as much meaning with N = 1 as any greater N. Think of it as a limit as N approaches 1 then. The point is that it is as valid a measurement for a single point as it is for many points. It represents the equivalent voltage that would produce the same power as DC of the same voltage. When you do it for a single point, the term RMS ceases to have meaning. For a single point it is just the voltage. For two points and above, RMS volts times RMS current give average power. d |
#64
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
rickman wrote:
On Nov 20, 2:52 pm, (Don Pearce) wrote: On Sat, 20 Nov 2010 11:43:03 -0800 (PST), wrote: On Nov 20, 12:02 pm, (Scott Dorsey) wrote: Dick wrote: Scott Dorsey wrote: I don't know, because I have never seen actual instantaneous RMS values ever displayed anywhere. The term "instantaneous RMS value" is itself meaningless. BINGO! Mr. Pierce wins the kewpie doll! --scott Except that it is wrong. What is the instantaneous RMS value of -1... +1. RMS doesn't have to be an integral or a sum. b Rick It isn't. It is the square root of the mean of the squares. In the case of -1, +1 it is 1. The inclusion of the term "mean" says that there must be at least two measurements - any fewer and you can't have a mean. d I don't recall any such restriction on N. Certainly the measurement has just as much meaning with N = 1 as any greater N. Think of it as a limit as N approaches 1 then. N is discrete, not continuous. That's the problem. You can't have half a sample... there is no application of limits. N=1 is a degenerate case. The point is that it is as valid a measurement for a single point as it is for many points. It represents the equivalent voltage that would produce the same power as DC of the same voltage. Rick Right. It's proportional to the instantaneous voltage measure. But it's degenerate as a weighted vector measure. -- Les Cargill |
#65
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
(Don Pearce) writes:
[...] When you do it for a single point, the term RMS ceases to have meaning. For a single point it is just the voltage. Correction: it is the magnitude of the voltage. I won't even get into whether this is a deterministic or random signal and, if the latter, how the mean is analytically defined independent of time... -- Randy Yates % "The dreamer, the unwoken fool - Digital Signal Labs % in dreams, no pain will kiss the brow..." % http://www.digitalsignallabs.com % 'Eldorado Overture', *Eldorado*, ELO |
#66
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
(Don Pearce) wrote in
: On Sat, 20 Nov 2010 12:53:16 -0800 (PST), rickman wrote: On Nov 20, 2:52*pm, (Don Pearce) wrote: On Sat, 20 Nov 2010 11:43:03 -0800 (PST), rickman wrote: On Nov 20, 12:02 pm, (Scott Dorsey) wrote: Dick Pierce wrote: Scott Dorsey wrote: I don't know, because I have never seen actual instantaneous RMS values ever displayed anywhere. The term "instantaneous RMS value" is itself meaningless. BINGO! Mr. Pierce wins the kewpie doll! --scott Except that it is wrong. *What is the instantaneous RMS value of -1... +1. *RMS doesn't have to be an integral or a sum. b Rick It isn't. It is the square root of the mean of the squares. In the case of -1, +1 it is 1. The inclusion of the term "mean" says that there must be at least two measurements - any fewer and you can't have a mean. d I don't recall any such restriction on N. Certainly the measurement has just as much meaning with N = 1 as any greater N. Think of it as a limit as N approaches 1 then. The point is that it is as valid a measurement for a single point as it is for many points. It represents the equivalent voltage that would produce the same power as DC of the same voltage. When you do it for a single point, the term RMS ceases to have meaning. For a single point it is just the voltage. For two points and above, RMS volts times RMS current give average power. d A mean of 1 sample is valid. RMS of a one sample measurement is technically valid as well, but perhaps not particularly useful. OTOH, If you knew that the signal was DC, a single sample might not be meaningless at all. With an AC signal, even a small number of samples may not yield a particularly good rms value. This issue doesn't begin to tradeoff exponential versus linear averaging, which changes the result as well. A very fast exponential averaging time approaches the 1 sample case. If you have an issue with exponential averaging, you can send me all your multimeters Al Clark www.danvillsignal.com |
#67
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
glen herrmannsfeldt wrote:
It seems to me that there is still some uncertainty in the meaning of dBFS. No. There are 10 kind of readers here, those that understand binary and those that do not. Well, consider that CDs are considered to have 96dB (or some similar number) of dynamic range. That is comparing a full scale signal (just about impossible in a live recording) Oh no, what is difficult to some is to stay in the comfy -10 to -5 zone re. FS instead of being at 0 dB FS for a number of consecutive samples. -- glen' Kind regards Peter Larsen |
#68
Posted to rec.audio.tech,rec.audio.pro
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dBFS
Randy Yates wrote:
Hi, Some had responded here to my recent inquiry on levels that dBFS is a peak measurement. If an RMS measurement needs to be made for a digital signal (i.e., on a digital mixing console or a ProTools plugin), what units are utilized? I thought they were dBFS, i.e., that dBFS was an RMS measurement. Apparently I am incorrect. Somebody please set me straight. dBFS is NOT a measurement method (peak or rms) but a specification for a signal level. Unlike dBm, dBu and dBV is has NO SPECIFIC PHYSICAL VALUE - it is simply the largest value that a digital system can represent. Cheers ian |
#69
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
On Nov 20, 3:56*pm, (Don Pearce) wrote:
On Sat, 20 Nov 2010 12:53:16 -0800 (PST), rickman wrote: On Nov 20, 2:52 pm, (Don Pearce) wrote: On Sat, 20 Nov 2010 11:43:03 -0800 (PST), rickman wrote: On Nov 20, 12:02 pm, (Scott Dorsey) wrote: Dick Pierce wrote: Scott Dorsey wrote: I don't know, because I have never seen actual instantaneous RMS values ever displayed anywhere. The term "instantaneous RMS value" is itself meaningless. BINGO! Mr. Pierce wins the kewpie doll! --scott Except that it is wrong. What is the instantaneous RMS value of -1... +1. RMS doesn't have to be an integral or a sum. b Rick It isn't. It is the square root of the mean of the squares. In the case of -1, +1 it is 1. The inclusion of the term "mean" says that there must be at least two measurements - any fewer and you can't have a mean. d I don't recall any such restriction on N. *Certainly the measurement has just as much meaning with N = 1 as any greater N. *Think of it as a limit as N approaches 1 then. *The point is that it is as valid a measurement for a single point as it is for many points. *It represents the equivalent voltage that would produce the same power as DC of the same voltage. When you do it for a single point, the term RMS ceases to have meaning. For a single point it is just the voltage. For two points and above, RMS volts times RMS current give average power. d I disagree. In for a single point it is just a trivial case. Clearly the formula still has meaning and it still works perfectly. Saying it has no meaning, has no meaning... ;^) Rick |
#70
Posted to rec.audio.tech,rec.audio.pro,comp.dsp
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dBFS
On Nov 20, 3:57*pm, Les Cargill wrote:
rickman wrote: On Nov 20, 2:52 pm, (Don Pearce) wrote: On Sat, 20 Nov 2010 11:43:03 -0800 (PST), wrote: On Nov 20, 12:02 pm, (Scott Dorsey) wrote: Dick *wrote: Scott Dorsey wrote: I don't know, because I have never seen actual instantaneous RMS values ever displayed anywhere. The term "instantaneous RMS value" is itself meaningless. BINGO! Mr. Pierce wins the kewpie doll! --scott Except that it is wrong. *What is the instantaneous RMS value of -1.... +1. *RMS doesn't have to be an integral or a sum. b Rick It isn't. It is the square root of the mean of the squares. In the case of -1, +1 it is 1. The inclusion of the term "mean" says that there must be at least two measurements - any fewer and you can't have a mean. d I don't recall any such restriction on N. *Certainly the measurement has just as much meaning with N = 1 as any greater N. *Think of it as a limit as N approaches 1 then. N is discrete, not continuous. That's the problem. You can't have half a sample... there is no application of limits. N=1 is a degenerate case.. *The point is that it is as valid a measurement for a single point as it is for many points. *It represents the equivalent voltage that would produce the same power as DC of the same voltage. Rick Right. It's proportional to the instantaneous voltage measure. But it's degenerate as a weighted vector measure. -- Les Cargill Who are you calling a degenerate??? :-( Rick |
#71
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dBFS
In article , Randy Yates wrote:
(Scott Dorsey) writes: That depends entirely on which averaging standard you decide to use. Most common is LKFS according to ITU BS.1771 loudness standard. You will never, never see this in the US, but RTW standalone meters can display it. Scott, sorry but I didn't see this until just today. Thanks. In searching for info on BS.1771 I also found this paper from Grim Audio, which, at a cursory glance, looks like it touches on many of the same issues I've been asking about here. It seems like most of the messages I have sent, you haven't seen. Let me reiterate he If it says dBFS, it is a peak-reading meter that reads relative to the highest digital value on the system. If it is some kind of average reading meter, it is not reading dBFS, but is reading something else. Because there are so many different standards for average reading, precisely WHAT it is measuring can be hard to tell. For example, the average meters on Pro Tools don't seem to match anything else or meet any known standard. The ballistics are faster than VU. If you actually need to have consistent and accurate average metering on digital systems, you use BS.1771 metering. Most people don't, though. Calling something RMS when it produces a weighted average is not correct. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#72
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dBFS
rickman wrote:
On Nov 20, 3:57 pm, Les wrote: rickman wrote: On Nov 20, 2:52 pm, (Don Pearce) wrote: On Sat, 20 Nov 2010 11:43:03 -0800 (PST), wrote: On Nov 20, 12:02 pm, (Scott Dorsey) wrote: Dick wrote: Scott Dorsey wrote: I don't know, because I have never seen actual instantaneous RMS values ever displayed anywhere. The term "instantaneous RMS value" is itself meaningless. BINGO! Mr. Pierce wins the kewpie doll! --scott Except that it is wrong. What is the instantaneous RMS value of -1... +1. RMS doesn't have to be an integral or a sum. b Rick It isn't. It is the square root of the mean of the squares. In the case of -1, +1 it is 1. The inclusion of the term "mean" says that there must be at least two measurements - any fewer and you can't have a mean. d I don't recall any such restriction on N. Certainly the measurement has just as much meaning with N = 1 as any greater N. Think of it as a limit as N approaches 1 then. N is discrete, not continuous. That's the problem. You can't have half a sample... there is no application of limits. N=1 is a degenerate case. The point is that it is as valid a measurement for a single point as it is for many points. It represents the equivalent voltage that would produce the same power as DC of the same voltage. Rick Right. It's proportional to the instantaneous voltage measure. But it's degenerate as a weighted vector measure. -- Les Cargill Who are you calling a degenerate???:-( Rick Easy babe! -- Les Cargill |
#73
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dBFS
Don Pearce wrote:
On Sat, 20 Nov 2010 08:03:14 +0000 (UTC), I woulda said that a 0 dBFS signal has the RMS level of a sine wave that just barely doesn't clip a converter (or, a hardlimited channel; it does not need to be a converter). (This is an important concept, of sorts, in that is shows that an N bit converter has a full-scale-signal to quantization noise ratio of 6*N + 2 dB, not the 6*N + 5 dB that some texts claim.) One can debate these things. Most outcomes of such debates are equivalent within a factor of two. Here's how the maths works. Lets call the clipping point 0dB. The biggest sine wave it can hold is -3dB RMS. (peaks just clip). The lowest bit level is - 16 * 20log(2), or -96.3dB Because the converter is perfect, the noise is TPD, which has an RMS value 4.8dB below the 1 bit peak. So the noise level is -101.1dB So the dynamic range is 101.1dB -3dB +4.8 or 98.1dB. That is the range from a just-clipping sine wave, to a signal equal in RMS amplitude to the noise. I agree with the result, but what is "TPD"? (I'm thinking "triangular" something...) S. |
#74
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dBFS
On Nov 20, 11:26*pm, (Steve Pope) wrote:
Don Pearce wrote: On Sat, 20 Nov 2010 08:03:14 +0000 (UTC), I woulda said that a 0 dBFS signal has the RMS level of a sine wave that just barely doesn't clip *a converter (or, a hardlimited channel; it does not need to be a converter). (This is an important concept, of sorts, in that is shows that an N bit converter has a full-scale-signal to quantization noise ratio of 6*N + 2 dB, not the 6*N + 5 dB that some texts claim.) One can debate these things. *Most outcomes of such debates are equivalent within a factor of two. Here's how the maths works. Lets call the clipping point 0dB. The biggest sine wave it can hold is -3dB RMS. (peaks just clip). The lowest bit level is - 16 * 20log(2), or -96.3dB Because the converter is perfect, the noise is TPD, which has an RMS value 4.8dB below the 1 bit peak. So the noise level is -101.1dB So the dynamic range is 101.1dB -3dB +4.8 or 98.1dB. That is the range from a just-clipping sine wave, to a signal equal in RMS amplitude to the noise. I agree with the result, but what is "TPD"? *(I'm thinking "triangular" something...) S. TPD= triangular probability density,, typically referring to a description of the noise used for dither.. Mark |
#75
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dBFS
In comp.dsp Mike Rivers wrote:
(snip on dBFS and such) (then I wrote) I sometimes record live high-school orchestra concerts. Because it is hard to know the level, I record 24 bit, then find the peak and RMS of each track. Then I figure out how many bits to scale each track by so that peaks stay below FS, and they should sound about right together. I suppose that's one way of mixing when you aren't listening and don't have controls handy. From a computer standpoint, it makes a certain degree of sense, but from a musical and artistic standpoint, it's absurd. Well, the idea is that they should sound about the same level, such that one shouldn't want to run up and change the volume control for each track. I don't think I could do that very well just listening to them, trying to memorize the average level over a 15 minute track. If the peaks are about the same, and RMS about the same then I figure that they will sound about right. If both peak and RMS are different by about the same amount (I might have changed the record level), then I adjust by about that amount. Usually that works. For a recent recording, the four RMS values were -36, -39, -41, and -35, all dbFS, peaks were -10, -11, -14, -11 dbFS, all with the same record level. -- glen |
#76
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dBFS
In comp.dsp Dick Pierce wrote:
(snip, I wrote) In EE, RMS can mean different things. Not in any "EE" I ever encountered. It means one thing: sqrt(sum(X1^2 + X2^2 ... +Xn^2) / n) It used to be, for DVMs, that was called True RMS. When they figured out how to actually do that electrically. Maybe they all do now, so it isn't a problem anymore, but it didn't used by be that way. Also, it isn't for the peak-to-peak reading VTVM, normally calibrated with an RMS (assuming sine) scale. Now, what's partly at the root of this discussion is that if we were to scale our system such that we called max(abs(X1, X1 ...Xn)) "full scale," what is the ratio of the RMS value of that set to the full scale value of that set. -- glen |
#77
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dBFS
Randy Yates wrote:
Also, what reference level does an analog peak-reading meter use? http://www.klay.com/klay/world_audio_levels.jpg link supplied by Hank Alrich in some other context. Roger Orban made a "multi-standard loudness meter" program some time ago, perhaps someone can remember the download link, it is very illustrative. --Randy Kind regards Peter Larsen |
#78
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dBFS
On Nov 20, 1:43*pm, Mark wrote:
But, regardless, there aren't any "different versions" of RMS, just the one: * * sqrt(sum(X1^2 + X2^2 ... +Xn^2) / n) but there are different values of n... and for a sine tone, *the value of n doesn't matter because a tone is the same all the time....., but for real audio the value of n does matter *and that is the crux of the discussion.. and the mean might not be equally weighted. RMS{ x[n] } = sqrt( SUM{ h[k]*(x[n-k])^2 } ) k where the weighting coefficients are normalized so that SUM{ h[k] } = 1 k essentially, you square the signal, run it through a low-pass filter with DC gain of 1, then square root the output of the LPF. there are many different versions of RMS. an infinite number of ways to do it. r b-j |
#79
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dBFS
On Sun, 21 Nov 2010 08:36:13 +0100, "Peter Larsen"
wrote: Also, what reference level does an analog peak-reading meter use? http://www.klay.com/klay/world_audio_levels.jpg link supplied by Hank Alrich in some other context. Roger Orban made a "multi-standard loudness meter" program some time ago, perhaps someone can remember the download link, it is very illustrative. http://www.orban.com/meter/ |
#80
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dBFS
On Nov 21, 7:56*am, robert bristow-johnson
wrote: On Nov 20, 1:43*pm, Mark wrote: But, regardless, there aren't any "different versions" of RMS, just the one: * * sqrt(sum(X1^2 + X2^2 ... +Xn^2) / n) but there are different values of n... and for a sine tone, *the value of n doesn't matter because a tone is the same all the time....., but for real audio the value of n does matter *and that is the crux of the discussion.. and the mean might not be equally weighted. * * RMS{ x[n] } = sqrt( SUM{ h[k]*(x[n-k])^2 } ) * * * * * * * * * * * * *k where the weighting coefficients are normalized so that * * SUM{ h[k] } = 1 * * *k essentially, you square the signal, run it through a low-pass filter with DC gain of 1, then square root the output of the LPF. there are many different versions of RMS. *an infinite number of ways to do it. r b-j This thread is about dBFS. Strictly speaking, it has nothing to do with peak, VU, PPM, RMS etc. etc. but for audio, in the music and broadcast industry, it refers to peak magnitude or VU, or PPM or RMS in some cases and possibly other measures I haven't come across referred to full scale of the metered range. Americans generally use VU, the UK and Europe use PPM generally. 0dBFS has units of FS. 0dBFS = one FS. FS is full scale whatever, but commonly refers to the value 2^(number_of_bits-1). We also use dBFS24, dBFS32 when referring to 24 and 32 bit audio for example. |
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