how do YOU read the RS spl meter?

[Scp53]

do you look at that average number readout for the las second of measuring? or do you look at the bar moving at the bottom that moves quicker?which is more accurate or more meaningful? i wonder if i havent been measuring the "dynamic" peaks of the music but instead measuring the "average" spl or what ever. im talking about the digital one. tell me what ya think. thanks

scp53

[Duke Spinner]

"a" ..or "C" scale .??

[dragonfyr]

I would suggest the C weighting and slow scale - you are more concerned with average levels then a momentary transient. And use a pink noise source rather then test tones for general level matching.

[Scp53]

i have been using the fast response on the meter. I'll change and go to the slow as suggested(btw, i have been using the c weighting). thanks. any other comments or suggestions?

scp53

[scriven]

What is it you want to measure? Are you looking to find the peak output? Are you trying to find the average output? Are you just trying to learn what your nice shinny new meter is telling you? Inquiring minds want to know!

[dragonfyr]

The Radio Shack SPL meter has two adjustments that provide you with a bit if adjustability designed to make it more useable for different applications.

These two switches are the 'weighting' and the 'response speed'.

The weighting switch allows you to change the meter's frequency response or "weighting", and the speed switch allows you to select the speed of its response to sound pressure changes. The weighting switch allows for switching between the standard 'A' and 'C' weightings. Choosing the 'C' weighting will make the meter relatively uniform over the frequency range from 32 - 10,000 Hz, and the 'A' weighting will make the meter more sensitive to frequencies in the range from 500-10,000 Hz. Additionally, the response switch allows for changing the speed of the meter's response from 'SLOW' to 'FAST'. A slow response setting will make the meter less sensitive to rapid changes in sound level and can be used for measuring average pressure levels. The 'fast' setting is more useful when peak sound levels are being measured since in this mode, the meter will respond to very rapid changes in pressure level.

<?xml:namespace prefix = o ns = "urn:schemas-microsoft-com:office:office" />

Additionally, you are generally better off using weighted pink noise as a stimulus for general level measurements and balancing output levels.

Meter Compensation Values:

Add the following values to the meter reading:

10Hz add 20dBs

12Hz add 16.5dB

16Hz add 11.5dB

20Hz add 7.5dB

25Hz add 5dB

31.5Hz add 3dB

40Hz add 2.5dB

50Hz add 1.5dB

63Hz add 1.5dB

80Hz add 1.5dB

100Hz add 2dB

125Hz add .5dB

[DrWho]

btw, have you read the little manual that comes with the meter? I'm pretty sure it'll answer any question you might have about it. dragon has already pretty much covered all the cool stuff though

I just wanted to add that when measuring peaks, there is a limit to how fast the needle responds to very short sounds (same concept as trying to measure "slow peaks" on the slow setting, which is meant to record average sounds). Ya, I'm not wording this well, but you can expect very short sounds (say a snare drum hit) to be up to a good 10dB louder than what the meter measures. Just something to keep in mind.

[colterphoto1]

the manual also suggests putting the meter on a stand (threaded hole provided) and measuring with the METER AT RIGHT ANGLES TO THE SOUND SOURCE.

Michael

[sbryan]

----------------

On 6/12/2005 11:12:40 PM dragonfyr wrote:

Meter Compensation Values:

Add the following values to the meter reading:

10Hz add 20dBs

12Hz add 16.5dB

16Hz add 11.5dB

20Hz add 7.5dB

25Hz add 5dB

31.5Hz add 3dB

40Hz add 2.5dB

50Hz add 1.5dB

63Hz add 1.5dB

80Hz add 1.5dB

100Hz add 2dB

125Hz add .5dB

----------------

Please provide additional explaination as to when and how to use these compensation values. For instance, I have a Marantz SR9300 with an Audio Control Bijou 1/3 octave equalizer installed between the pre-outs and the amp-ins of the SR9300 to drive Klipsch Reference 3 Series speakers (front/center/rear) and Hsu Research TN-1220 sub. I used the Marantz built-in pink noise generator and on-screen set-up menu to set the volume of each speaker output to be the same with the Radio Shack analog meter (weighing=C/response=slow). Using "The Sheffield / Coustic Test and Demonstration Disc" (tracks 19-24: One third octave band test frequencies) results were as follows:

25 to 63 Hz = -10 to -4 @ 100 dB scale

80 to 160 Hz = -8 to +1 @ 90 dB scale

250 to 630 Hz = -1 to 0 @ 80 dB scale

800 to 2K Hz = -3 to -1 @ 80 dB scale

2.5 to 6.3K Hz = -1 to +3 @ 80 dB scale

8 to 20K Hz = -9 to +1 @ 80 dB scale

1) Should all frequencies be EQ adjusted to be flat @ the 80 dB scale?

2) What adjustments would you suggest?

3) How would your compensation values be utilized in these adjustments?

I ran the previous test with sound on all speakers. I plan to disable the output for all channels but one and adjust each (LF/C/RF/RB/LB) separately as required. All advice appreciated.

Thanks

[dragonfyr]

----------------

On 6/13/2005 4:03:45 PM sbryan wrote:

----------------

On 6/12/2005 11:12:40 PM

dragonfyr wrote:

Meter Compensation Values:

Add the following values to the meter reading:

10Hz add 20dBs

12Hz add 16.5dB

16Hz add 11.5dB

20Hz add 7.5dB

25Hz add 5dB

31.5Hz add 3dB

40Hz add 2.5dB

50Hz add 1.5dB

63Hz add 1.5dB

80Hz add 1.5dB

100Hz add 2dB

125Hz add .5dB

----------------

Please provide additional explaination as to when and how to use these compensation values. For instance, I have a Marantz SR9300 with an Audio Control Bijou 1/3 octave equalizer installed between the pre-outs and the amp-ins of the SR9300 to drive Klipsch Reference 3 Series speakers (front/center/rear) and Hsu Research TN-1220 sub. I used the Marantz built-in pink noise generator and on-screen set-up menu to set the volume of each speaker output to be the same with the Radio Shack analog meter (weighing=C/response=slow). Using "The Sheffield / Coustic Test and Demonstration Disc" (tracks 19-24: One third octave band test frequencies) results were as follows:

25 to 63 Hz = -10 to -4 @ 100 dB scale

80 to 160 Hz = -8 to +1 @ 90 dB scale

250 to 630 Hz = -1 to 0 @ 80 dB scale

800 to 2K Hz = -3 to -1 @ 80 dB scale

2.5 to 6.3K Hz = -1 to +3 @ 80 dB scale

8 to 20K Hz = -9 to +1 @ 80 dB scale

1) Should all frequencies be EQ adjusted to be flat @ the 80 dB scale?

2) What adjustments would you suggest?

3) How would your compensation values be utilized in these adjustments?

I ran the previous test with sound on all speakers. I plan to disable the output for all channels but one and adjust each (LF/C/RF/RB/LB) separately as required. All advice appreciated.

Thanks

----------------

This compensation is for use when measuring this frequencies with tones, or creating a contour response.

But here is where I must express concern.

I understand what you are trying to do, but there is a fundamental problem.

{And I would be ecstatic if we could simply make this point and it's ramifications fundamental on this BB!}

The SPL meter will measure the composite response of both the direct and reflected signals at a point in space. Therefore, unless you use a method to remove the reflected signals ( i.e.: Don Keele's wonderful method of employing a LARGE open space such as a parking lot or Large driveway with the mic laid on the ground) you will not be measuring the isolated direct sound.

(And be aware that if you hold the meter, that a large amount of energy will be reflected off you!)

The composite response is fine for identifying nulls and LF modal response! And I would address this issue by relocating your subwoofer or analyzing the dimensions of your room in order to make a few Helmholtz resonators. But I would NOT waste my time with the EQ. You cannot correct for time based anomalies with such a frequency based tool.

But, as soon as you bring the EQ into the equation we are moving into the very domain of what you cannot equalize. You CANNOT correct for a non-minimum phase signal - meaning that EQ can ONLY correct for the minimum phase direct signal! It CANNOT correct a composite signal comprised of multiple signals resulting from superposition.

As an EQ employs LRC filters, the LC components cause an shift in the direct signal phase response. And while you will feel that an EQ can change the apparent response at a point, it is simply due to the relative modification in the phase relationships of the component waveforms that comprise the composite signal. In other words, the comb filtering polar response shifts ever so slightly, simply moving the problem around and modifying the problem ever so slightly. It CANNOT fix the error!

Thus, the fundamental problem is that the ultimate solution of using the EQ to try to correct for anything other then the DIRECT signal is an attempt to try to do something that the tools cannot accomplish.

If they could, we could simply forego all the acoustic analysis and simply buy lots of EQs as have been done for so many years!

And, ironically, this is precisely what we are again starting to see with so many of the high (on drugs) end manufacturers who are beginning to incorporate this very technique into their units with lots of fancy names and claims regarding "room correction" And EVERY such claim ranks right up there with the best of the spurious claims that we so like to attribute to Monster and Bose! So get ready to see that rarified clique become significantly larger with more 'marketing over engineering' BS!

Now lots of people persist in trying to do exactly this with a myriad number of variations! But it is a fundamentally flawed method.

So to repeat, while the composite response is fine for identifying nulls and LF modal response, I would address this issue by relocating your subwoofer or analyzing the dimensions of your room in order to make a few Helmholtz resonators. But I would NOT waste my time with the EQ. You cannot correct for time based anomalies with such a frequency based tool.

But again, you are welcome to spend your time doing whatever you might chose assuming that you aren't hurting others! But I suggest that you can find many more productive things to do with your time then this!

[Parrot]

----------------

On 6/13/2005 4:40:29 PM dragonfyr wrote:

the high (on drugs) end manufacturers

----------------

LOL!

[Nklipsch]

----------------

----------------

This compensation is for use when measuring this frequencies with tones, or creating a contour response.

But here is where I must express concern.

...But, as soon as you bring the EQ into the equation we are moving into the very domain of what you cannot equalize. You CANNOT correct for a non-minimum phase signal - meaning that EQ can ONLY correct for the minimum phase direct signal! It CANNOT correct a composite signal comprised of multiple signals resulting from superposition.

As an EQ employs LRC filters, the LC components cause an shift in the signal phase response. And while you will feel that an EQ can change the apparent response at a point, it is simply due to the relative modification in the phase relationships of the component waveforms that comprise the composite signal. In other words, the comb filtering polar response shifts ever so slightly, simply moving the problem around and modifying the problem ever so slightly. It CANNOT fix the error!

Thus, the fundamental problem is that the ultimate solution of using the EQ to try to correct for anything other then the DIRECT signal is an attempt to try to do something that the tools cannot accomplish.

If they could, we could simply forego all the acoustic analysis and simply buy lots of EQs as have been done for so many years!

And, ironically, this is precisely what we are again starting to see with so many of the ‘high (on drugs) end’ manufacturers who are beginning to incorporate this very technique into their units with lots of fancy names and claims regarding "room correction" And EVERY such claim ranks right up there with the best of the spurious claims that we so like to attribute to Monster and Bose! So get ready to see that rarified clique become significantly larger with more 'marketing over engineering' BS!

Now lots of people persist in trying to do exactly this with a myriad number of variations! But it is a fundamentally flawed method.

So to repeat, while the composite response is fine for identifying nulls and LF modal response, I would address this issue by relocating your subwoofer or analyzing the dimensions of your room in order to make a few Helmholtz resonators. But I would NOT waste my time with the EQ. You cannot correct for time based anomalies with such a frequency based tool.

But again, you are welcome to spend your time doing whatever you might chose assuming that you aren't hurting others! But I suggest that you can find many more productive things to do with your time then this! http://forums.klipsch.com/idealbb/images/smilies/2.gif"> http://forums.klipsch.com/idealbb/images/smilies/9.gif"> http://forums.klipsch.com/idealbb/images/smilies/11.gif">

----------------

In other words, the comb filtering polar response shifts ever so slightly, simply moving the problem around and modifying the problem ever so slightly. It CANNOT fix the error!

What do you mean by 'comb filtering polar response'?Do mean acoustical axis shift do to phase differences?

-> http://www.linkwitzlab.com/x-sb80-3wy.htm

How does one acheive a minimum phase system with a loudspeaker system,then EQ fixes the FR abberation problem doesnt it.

-> http://www.harman.com/wp/index.jsp?articleId=121

The math sends me to sleep,as its not up to scratch!

->http://en.wikipedia.org/wiki/Nonminimum_phase#Inverse_system

Cheers

Mike.e

[dragonfyr]

I would be glad to try to explain this in much more detail - or as little - as you like - but the site is not letting me upload anything! It simply times out...

And I unfortunately don't know how to embed graphics in the post!

But if you PM me I will gladly try to explain, and I can send a few diagrams that will make the explanation much clearer over email or Yahoo chat!

[WMcD]

I recall dragon and I have been at odds on other matters.

Here, I think we're in agreement to some extent.

One thing you can't remove with an equalizer is a dip caused by destructive interference, or a.k.a. a node (zero).

If you play a pure, constant, sine, you can walk around the room and find spots where the sound drops out. Or, you can sit in your favorite spot and vary the pure tone until you find that certain frequencies drop out there.

The reason for the node arises from the combination of direct sound and the sum of reflections from the room. In a simple case, the direct sound is considered zero in phase. A single reflection is 180 degrees out of phase from that and near equal in magnitude (or enough). The sum is zero. (There are more complicated combinations which can do the same.)

Please observe here that all the signals orginate from a single source (assuming one speaker). So you can't equalize "up" the null. If you increase speaker output, the direct sound is increased, but so is the out of phase reflection (which is being fed from the same source).

One thing to do is to add some sort of room treatment to reduce or alter reflections. Moving the speaker and toe-ing in may keep some of the beaming off the wall, but that doesn't work very much at low frequencies.

This is not to say that equalization can't help some issues. There are limits though.

Best,

Gil

[dragonfyr]

If we can find a way to upload a file I will show you some actual representative measurements of such anomalies describing the interaction between 2 identical sources from the frequency, 2space polar, and 3space volumetric polar response points of view.

You cannot EQ (correct) non-minimum phase interference with an EQ. You can only EQ the minimum phase direct signal.

[Deang]

Great stuff guys, keep it coming.

[Spkrdctr]

I'll be the devils advocate here... all of the above posts are correct. But, I think many people tend to slam EQ use because they go the extreme for their examples. The above examples are trying to use an EQ to fix major room problems and I would consider that extreme. When using an EQ, you need a light touch on the controls most times. You are not trying to fix nulls but trying to fix frequencies that may be peaking compared to other parts of the spectrum, such as an overly bright speaker (Klipsch??? Yes, sometimes..) So, an EQ has its place, but it is not for fixing major room problems as has been stated. I like using them to fix aspects of a speaker system, where they tend to do a good job.

So, use an EQ, but always remember that you want to use it in specific circumstances. This is a good thread! Seems to be some smart fellas around here.......

[dragonfyr]

----------------

On 6/13/2005 10:22:31 PM Spkrdctr wrote:

I'll be the devils advocate here... all of the above posts are correct. But, I think many people tend to slam EQ use because they go the extreme for their examples. The above examples are trying to use an EQ to fix major room problems and I would consider that extreme. When using an EQ, you need a light touch on the controls most times. You are not trying to fix nulls but trying to fix frequencies that may be peaking compared to other parts of the spectrum, such as an overly bright speaker (Klipsch???

Yes, sometimes..) So, an EQ has its place, but it is not for fixing major room problems as has been stated. I like using them to fix aspects of a speaker system, where they tend to do a good job.

So, use an EQ, but always remember that you want to use it in specific circumstances. This is a good thread! Seems to be some smart fellas around here.......

----------------

Sorry if this post is long...but pictures would make this easier!

Perhaps it is a matter of semantics, but given your description, I would maintain the the MOST common use of EQ by most is the erroneous attempt to address signal anomalies consisting of some combination of the fiollowing: the combination of direct (single source), or two or more direct signals, or two non-minimum phase signals that resulting in comb filtering/polar anomalies.

And when you think of it, this is the norm! For how can one easily only measure the direct signal without the interference of a secondary or reflected signal? How many people are using TDS (which is unique in its ability to 'window'/filter out the reflected signal or are listening in an ideal anechoic chamber? Thus any 'normal' signal consisted of two or more superposed signals, be they from two real sources, or two or more apparent sources by virtue of their difference in phase!?

(Pardon me a moment as I digress here in lieu of question asked in an above post...comb filtering and increased frequency dependent polar lobing arise simultaneously from the same cause - They are two different perspectives for observing and characterizing aspects of the same phenomena. The frequency response looks only at frequency versus intensity while the polar response looks at the radiation pattern of acoustic energy relative to gain at a particular frequency - and then the responses for different frequencies are typically superimposed over one another.)

The problem is that EQ can only equalize the isolated direct signal. And when you listen you cannot isolate it! And using an SPL meter, or an RTA you cannot observe it! They are limited to measuing the total combined/composite signal!

This presents a real problem! While we will maintain that an EQ can be used efectively to shape a direct isolated electrical signal that feeds a particular transducer, how is the average person to observe this signal isolated from the acoustical environment that sabotages that very desire!? Without a TEF or other mechanism utilizing TDS or MLS or other related time domain based measuring toolks, you cannot do this.

Now, in a few circumstances that is fine! A classic example of this practical circumstance of observing the combined signal reponse being ok is when we choose to observe the low frequency modal response of a room. You can observe the summed result of the interaction of the LF in and with the room. And from this one can adjust the location of the subwoofer to 'move' the acoustical effects 'around' relative to a particular listening position (while this will NOT correct for them!) or one can then determine the need for Helholtz resonators in the room. But this really does not require a meter to do!

But you still cannot effectively EQ the sub and more then other drivers in the room - independent of reflected signals and room affects. Despite the low frequency comb filtering and polar response anomalies being less apparent and prone to issues then the kid and high frequency signals by the nature of the lower Q characterisitcs and longer wavelengths of the low frequencies..

But is this what most try to do with either the low frequencies or the higher frequencies?? I maintain that it is not! Individuals continue to try to use the results of the SPL/RTA measurements to try to EQ the subwoofer or the frquency balance of the total system within the room! And remember, that is all they can do! As they do not possess the tools to isolate the direct signal independently of the additional ambient sources and reflections. Thus they can neither focus on the isolated source, nor can they focus on the isolated signals after they make changes to the direct signal component with the equalizer. So, even if the desire is to only address the direct signal and response, they can neither see the source nor their effect independent of the combined room effect.

Pardon me for saying this several times, but I am simply trying to cover every combination and permutation!

So, yes, an EQ can be used to equalize the frequency component of a direct signal. And this is one reason mixing (in SOME cases) and mastering suites (generally) (and I focus on Georgetown Masters & Masterphonic where I have spent the most time in the past 25 years) go to such great lengths to create RFZ (Reflection Free Zones) for mastering. They wish to hear the effect of processing on the direct sound, isolated from ANY room effects over which they cannot control!

But, as so few have the tools to accurately observe the isolated direct signal in a mixed environment, nor to measure the adjusted direct signal independent of the composite 'summed' signals. I would maintain that there are VERY few examples where anyone here can effectively employ equalization.

Thus, folks have little opportunity to EQ peoperly because they have no way to observe the beginning 'before' isolated direct signal response, nor the isolated 'after' results of their fiddling!!

So we are left with a very rare circumstance where one satisfy the conditions of isolating the signals.

And there are only a few tools that practically allow for this and are designed to maximize the allowable minimum phase equalization of those components that satisfy the required conditions in the real environment and not simply in a lab. And the two most common are TEF and SMAART. SMAART being very common in live sound reinforcement environments where TEF is overkill for a limited range of measurements.

The problem remains that we exist 'in' the listening environment, and that necessarily includes the room and the multitude of 'ambient' signals! And we need to realize the limitations of the frequency domain based tools such as SPL meters, RTAs, and EQs! And they cannot isolate the direct signal from the complex sound field!!! By understanding the limitations of these tools, we can learn to effectively maximize their limited uses while minimizing unintended 'damage' by not trying to make them do things they do poorly or not at all! And we can do this by recognizing the inherent limitations of our tools and learn when we need to switch to the correct that which we might want to identify, measure, and correct.

---still can't attach diagrams...

[dragonfyr]

And to address this subject one more time from a more 'live sound'/SR (sound reinforcement) oriented perspective:

What can an Equalizer Equalize?

The question What can an equalizer equalize? needs to be asked. Some claim to equalize the room. Is this possible? I think not.

When an electronic or passive equalizer is installed in between a mixer and a power amplifier we need to know that all it can equalize is the electrical signal being sent to the loudspeaker.

What comes out of the loudspeaker?

What comes out of the loudspeaker is called direct sound level, Ld. Early reflections from the floor, walls, and ceiling are called the early reflected level, Lre, and late-in-time, high order homogenous mixed sound is called the reverberant sound level, Lr.

When an electronic equalizer is employed it not only alters the Ld at the listener position, but also the sound power level, Lw, of the loudspeaker.

This in turn affects Lre and Lr but has no effect on Lamb. The question over the years has been how much can I alter Ld without throwing the baby out with the wash? Experience has shown that the answer is not much. Certainly not enough to drop a specular Lre having a full frequency response.

{Note: Refer to Sam Berkows articles referring to the need to reduce a subwoofers crossover point with increases in gain! http://www.prosoundweb.com/install/transcripts/samchat.html }

The Audience Effect

The so-called audience effect came about by people looking at the sound fields with 1/3-octave real time analyzers. What an RTA sees is the total sound field level, Lt, which is the combined Ld, Lre, and Lr plus any ambient noise present, Ln. Adjusting a sound system to a uniform Lt may or may not result in a sound you would want to listen to. In many cases the floor reflection can cause an operator to misadjust the direct sound level, Ld. Then when the audience arrives and covers the floor the misadjusted Ld is more clearly perceived and we say the audience affected the system. Really? Think with me for a minute.

What can the audience do to affect Ld from a sound system? The answer is, of course, absolutely nothing. Therefore, it is clear that the audience can only alter Lre, Lr and Ln. Now, ask yourself the question how can an equalizer adjust Lre, Lr and Ln? The answer is that it cannot.

I would hesitate to mention such obvious facts except for the remarkable number of articles and opinions appearing to claim the contrary. The only way an equalizer can cause a change in Lre, Lr and Ln is to do damage to Ld in the process.

What Can Affect Lre

Loudspeaker directivity factor, Q, is the classic way to handle unwanted Lre. Using a loudspeakers directivity to stay off of surfaces producing unwanted reflective energy is also one of the most cost effective solutions. The second approach is to use either absorption or diffusion.

Electronic Directivity Control

The increasing use of precision digital delays (i.e., 10 usec per step in contrast to normal digital delays of 1msec per step) to correct mis-synchronized loudspeaker arrays, where the mis-synchronization has resulted in directional lobing of the loudspeaker, demonstrates the importance and validity of directional control. {Additionally, limiting overlap of adjacent sound fields with higher-Q signals (more tightly focused think spotlight as opposed to floodlight) also minimizes the super-positional effect of the sound fields as well.}

Good Engineering Practices

Today, thanks to advanced analysis in the hands of competent users, good engineering practice has become:<?xml:namespace prefix = o ns = "urn:schemas-microsoft-com:office:office" />

1. Adjust Ld by measuring it alone with a TEF analyzer.

2. Optimize the reduction of Lre, Lr, and Ln levels by means of controlled directivity (Q) and measured synchronization of arrays.

3. Fundamentally control Lre, Lr and Ln through traditional use of absorption, diffusion, and noise abatement techniques.

What Point Am I Trying To Make?

The point I hope I have made is that electronic equalization in the frequency domain cannot correct phenomenon in the time domain outside of the minimum phase period (i.e., a few hundred microseconds).

[dragonfyr]

Wow! It attached!

Now I wonder what size it will download at? Let's see! I hope its not 6' x 8'!

For those interested in comb filtering and polar response - you will find some excellent examples along with 3space volumetric examples as well. And some pretty good commentary on the correlation beween depictions and domains and their real versus apparent effects...