audio blog

Using Convolution Reverb on your home recordings

2011-12-28T21:16:00.000-08:00

A little background: Auditory Depth Perception has been discussed on the blog in a fair amount of detail. So, I'm just going to do a quick recap of that information. Major factors that effect that perception in no particular order: 1) amount of reverb 2) timing of early reflections and their volume/tone 3) volume 4) amount of high frequency content.

1) The greater the quantity of reflections, the further the object is relative to the other objects making noise.

2) The closer the earlier reflections are to the original sound, the further the object is making the sound. See this initial time delay gap demo for a clear picture: http://www.syntheticwave.de/ITDG.htm

3) Softer(quieter) sounds appear more distant.

4) High frequencies are the most easily absorbed. A more distant object will sound more muffled.

How to implement this:

Place the least amount of reverb on the nearest object and conversely the most on the most distant.
Adjusting the EQ for more high frequency(or filtering less high frequency out) may make it sound closer.
The more distant the object, the less 'pre delay' the reverb should have. Different reverbs may have a different name for that parameter, but I think that's pretty standard. That's just a measure, usually in milliseconds, to describe when in relation to the original sound the reverb will start. Sound travels at roughly one foot per millisecond.
More distant objects should be slightly quieter depending on the degree of desired effect.

I like to overdo the degree of these variations on the signal b/c the listener has no visual cues to help them out.

My best example for a window view on a pseudo recorded performance and space(Pardon the vocal):
Everything That I Am by dantheman-10
Obviously I'm no pro and this sounds pretty rough to me right now. None the less I think it gets my point across. I think it's fairly realistic, but not 'real.'

Product Review: K&M Tripod Microphone Stand with Telescopic Boom Arm

2011-12-24T20:39:00.001-08:00

- Sound quality: N/A

- Reliability / Durability: Built very well, but I haven't used it long term. I don't have the most faith in telescopic boom arms in general. It certainly doesn't feel as strong as a one piece arm when fully extended and it only extends as far as a one piece arm.

- Would you buy this again if it were stolen?: At this point I may still go with another Tama stand, but I do really love this stand! It's sort of a toss up. I do like the machining on these better, but I am developing a preference for one piece booms. the K&M 210/2 would be on my short list to look at! Might be the best of all to me.

- Features that stand out: Build quality, machining(best I've seen), fit and finish, and stability/boom stays in place very well.

- What I don't like: Not so sure I like the telescopic boom. It seems strong when tightened, but just doesn't give me the confidence that a one piece arm gives me. Reliability is of utmost concern to me.

- Similar products used: Several On Stage Stands, the Tama MS205, K&M 210/8.

Product Review: Dayton EMM-6 measurement microphone

2011-12-18T11:03:00.001-08:00

- Sound quality: Well, this is a mixed bag. Technically this is a measurement microphone and really wasn't designed to record with. However, the response is of course very flat and the mic sounds extremely natural. The catch is the signal to noise ratio(I think). There's a lot of noise in this mic and it requires you to bump the pre amp up to loud and thus it easily overloads and distorts with ease making recording very difficult. I have a theory though that all these SDCs will have a better transients and High Frequency response d/t this little diaphragm that's lightweight. The size is smaller than the wavelength of the highest frequency we hear and will cause less audible diffraction/reflection of the high frequency content that makes its way onto the recording. I think this all contributes to the naturalness of small diaphragm mics. Too bad they all seem to generate a lot of noise as well.

- Reliability / Durability: It seems well made and sturdy. Remarkably so for this money.

- Would you buy this again if it were stolen?: Probably not. All measurement microphones need calibrated anyway so I'd probably get a higher end SDC and calibrate it. That would be more useful for recording and could measure just as well or even better.

- Features that stand out: Price/performance. $40 is not a lot of money to pay for this mic. It will capture a very natural sound as well, but it's still not a very good recording mic.

- Similar products used: Sound Professionals binaural mics.

No real complaints about this for its cost.

Dan

Product Review: Tama MS205 Microphone Stand with Boom Arm

2011-12-17T20:17:00.000-08:00

Finally, a decent microphone stand for an affordable price. I got the chrome version of this one so maybe this won't apply to the black one, but all the moving parts on this are very smooth. The boom arm stays locked in place like no other I own--mostly junk stands and a K&M 21140 boom. It's just a one piece arm, but it's got plenty of length--approximately 1 meter which is more than you can really use with the base. You can really push down on this boom and no slippage. The tubing is all thick and heavy, and the base is weighted. It gives you the impression that it will last a lifetime.

If you're like me and sick of cheap stands, this one is not cheap--it's inexpensive. Worth every penny. Sorry for the brevity, but it's a mic stand(not sexy).

Product Review: K&M 210/8 Microphone Stand with Boom Arm

2011-12-17T20:14:00.000-08:00

Well after being fed up with cheap microphone stands made in China, I decided to give Germany a chance to prove itself as a better value. No doubt it is a different class of build than those Chinese stands. The machining is awesome! Unfortunately I'm actually a bit disappointed with this stand. The base is beautiful and sturdy, all parts fit like a glove and move smoothly, but the boom arm does not stay in place nearly as well as my Tama MS205 does or even as well as my last working On Stage boom. The base is slightly more stable than the Tama, but the boom is just not as well executed. The wing nut grip is too small and I thought that might be why--I just wasn't getting enough pressure on the crank to make it hold. I'm not a big guy--5'8", 160#s, so this may be a problem for me and not some hulk. Turns out to be the case! Fortunately I have several broken Chinese stands with extra parts laying around. Put a new crank on with a longer handle--and all is well. Now the boom is very stable and I am pleased, but the Tama still has a better design in this department and a sturdier crank handle. Of course the whole assembly on the Tama is plastic and might be more prone to breaking if dropped. It feels sturdy though and it is thick. At the moment, like an initial quality assessment, I'd prefer the Tama overall by a slight margin. Only time will tell which is my favorite and which is more durable. They both feel very well made, but the Tama's boom arm assembly is better from the start. The K&M does have a little nicer base IMO due to its lower center of gravity, and all the machining is superior to the Tama's, but that difference is smaller than the boom arm difference IMO. Putting the Tama boom on the K&M base is ideal. Too bad they don't sell the Tama boom arm by itself. I'd take another.

Product Review: Samson C01 microphone

2011-12-17T20:10:00.000-08:00

- Sound quality: This mic is not too bad sounding really. People describe as the poor man's U87, but I have my doubts! Of the 3 LDCs I've used, this is easily the worst but still a usable mic. It has the most noise of the three by quite a margin and the smallest frequency range to my ear with the raspiest treble end. It's not as bad as this review makes it sound though. Compared to my PR20, this seemed like a step towards higher fidelity.

- Reliability / Durability: It feels okay, but nothing to write home about. Not too much cheaper feeling than the NT1-A to be honest. It's cheap enough to be easily replaced.

- Would you buy this again if it were stolen?: No. Having the NT1-A and the Blue Spark make this mic useless to me.

- Features that stand out: Better fidelity than some well regarded dynamic mics at a very low cost.

- What I don't Like: Sound quality is less impressive when compared against other LDCs.

- Similar products used: Rode NT1a and Blue Spark.

Sold this mic after getting the Blue Spark/NT1-A.

Product Review: Heil PR20

2011-12-17T20:07:00.001-08:00

- Sound quality: I like this mic. It sounds darker than the Recording Hack's graph would suggest:

Compared to a LDC, you'll get less fidelity in general. It has too much handling noise for jostling around in general, but shock mounted on a stand is where this mic shines. The proximity effect seems less pronounced compared to the SM57. It allows you to get up close and personal while singing and a tight pattern compared to my LDCs seems to remove the room from the recording at a distance. However, the lowish sensitivity make recording quiet instruments from a distance difficult b/c of the generated noise of the mic and cranked preamp. This is not your HiFi recording vocal mic.

- Reliability / Durability: Built like a vault! Dropped several times, thrown in the back of a truck for months and drove off road while in its case... never a problem. It feels heavy and solid and been around for years.

- Would you buy this again if it were stolen?: Yep.

- Features that stand out: Build quality and less pronounced proximity effect than expected. Also a generally great bass response even from a distance. The case is nice and has several replacement grills with a well fitted clip.

- What I don't Like: Handling noise--for its intended use, that's its only downside as far as I can tell. It would be great if they'd fix that.

- Similar products used: Shure SM57.

Just a workhorse to keep around.

Product Review: Focusrite Saffire 6 USB

2011-12-17T19:24:00.000-08:00

- Sound quality: The preamps in this crush my previous device's. The headphone output is also very clean, but lacks the output to drive more difficult headphones. They'll get my Shure SRH840s as loud as I'll ever need. I am still impressed by the sound of this especially for the money! I can't imagine I'll ever NEED better.

- Reliability / Durability: This little guy is built solid and tough! The case is all metal and it's heavy. The jacks also feel solid and hold cables tight. Pots all feel like hey are high quality as well.

- Would you buy this again if it were stolen?: I use a Mac and there is no other USB interface I know of that sounds this good for anywhere near this money let alone as intelligently implemented. I would definitely replace this if it were stolen.

- Features that stand out: Build quality, driver stability, and sound quality. I have never had trouble with this thing working.

- What I don't like: Nothing for its intended purpose.

- Similar products used: EMU USB 0404

This box still has me enthusiastic about it a year later.

Product Review: Emu 0404 usb 2.0

2011-12-17T19:20:00.000-08:00

- Sound quality: Well, looking at the specs you might expect great sound, but in this category it is a let down. I searched long and hard to find a USB I/O, comparing specs and reviews I had high hopes for this one. When I got it I was a bit underwhelmed and figured that it was probably just as good as it gets. I couldn't here the difference between it and my MacBook's built-in ones. I thought it was a testament to the MacBook's SQ! The pre amps have lot of gain--IF you don't mind a lot of noise! Well, that's not HiFi. I'm sure they are doing some cherry picking on their specs. The headphone output is not the most powerful either.

- Reliability / Durability: It's all plastic, the pots feel cheap, and the jacks are all loose. It hasn't broken yet, but it's felt like it could go at any moment since I bought it.

- Would you buy this again if it were stolen?: No--I'll give it away.

- Features that stand out: It has a lot of great numbers posted on the website.

- What I don't like: The cheap feel and the poor sound quality. They also wouldn't update their drivers to work with the newer operating system. Hopefully they have by now.

- Similar products used: The Focusrite Sapphire 6 USB

I still have this hanging around. People who've heard it next to the Focusrite wouldn't take it for free. They'd rather spend $200 on the Focusrite.

Product Review: Blue Spark microphone

2011-12-17T19:18:00.000-08:00

- Sound quality: I really like this mic. It just has a balanced tone and fairly low noise. Compared to my Rode NT1a, it doesn't go as deep in the bass range or as high in the treble. Still its tone is balance and extended enough to cover an acoustic guitar and most people's vocal with a fair degree of accuracy. Its cardioid pattern is not as tight as my PR20, but has enough rejection for studio use. Attached below is its on axis graph with the focus button on and off provided by Recording Hacks. The Focus just seems to reduce the low end which can help reduce rumble. Notice it also is a low noise mic, though it doesn't seem to be as low as the NT1a. It's still very useful and sensitive enough for quiet instruments.

- Reliability / Durability: Very well made mic. It's heavy and solid with a decent shock mount and a beautiful wooden case. Seems to be a league ahead of the NT1a in this regard.

- Would you buy this again if it were stolen?: I would. In fact I have 2 just in case something happens to one in a clutch and so I can do some stereo recordings. I got them for $159 on sale. That seems to be what they are commonly available for at the time of this review.

- Features that stand out: The best part about this mic is that it seems ideal for the one person and his/her guitar. The response, sensitivity, and low noise make that its ideal use.

- What I don't Like: The pop filter looks nice, but isn't particularly useful. It certainly doesn't muffle the sound though. If you get the older version, the shock mount was poorly built. They sent me a new replacement and it is fine, but still not as nice as some spider mount style ones. I can't say it's given me any problems though.

- Similar products used: Rode NT1a and Samson C01.

This is my most used mic in my collection. That says it all.

Product Review: Røde NT1-A

2011-12-17T19:16:00.000-08:00

- Sound quality: This mic has some outstanding features--it has a wide frequency range and extremely low noise. Recording my vocals tend to sound too bright dead on. When turned 45 degrees or more, it takes the edge off an produces a more neutral tone. I tend to hear this same brightness on other's vocal tracks as well. There are plenty of mods available for this mic and that's likely d/t the fact that the circuitry produces so little noise. Here's the Graph from Recording Hacks:

To me it seems to sound brighter than the graph would indicate which has been suggested is d/t time domain response issues caused by the protective grill.

- Reliability / Durability: This mic is definitely lighter and doesn't feel as solid as the Blue Spark. I haven't had any troubles with it however. It's build quality is closer to my old Samson C01. It comes with a very good shock mount, pop filter, and mic socks that allow you to leave it on the stand and not get dusty. It's very well thought out. The carrying bag is not a rugged way to protect these for transport however.

- Would you buy this again if it were stolen?: Maybe... I like this mic b/c it's about as quiet as a good preamp and it does sound great many times. It just isn't my go to mic. I imagine I'd miss it if it were gone at times.

- Features that stand out: The low noise and the wide frequency response!

- What I don't like: The general brightness on axis.

- Similar products used: Blue Spark and Samson C01

This mic has some nice features and is definitely useful, but the sibilant brightness needs tamed for my vocals in general.

Pioneer SP-FS51-LR Floorstanding Loudspeakers

2011-11-11T17:34:00.000-08:00

A lot of hype around this speaker. A friend sent me his pair for testing brand new. Let's see what we get:

That's as close as any 2 speakers I've measured and pretty well behaved. 45 degrees is nearly flat on both speakers.

Impulses:

Looks like Mr. Jones deserves the hype! :)

Large Driver Cone Treatement

2011-08-26T19:48:00.000-07:00

the Eminence Delta ProA.

Yes, their response graph is optimistic.

Prior to any treatment, the one's polar response looked like this:

I didn't do the other one out of sheer laziness.
After one round of stiffening and surround damping treatment I got these:

after the next round those went to these respectively:

another round makes
this:

into this:

and this:

into this:

Time elapse on these showed similar results, but I didn't save the graphs or the drivers.

Cone Treatments Demonstrated To Work (Time Elapsed)

2011-08-26T18:50:00.000-07:00

one driver over the initial curing as of June 9,2010:

Same driver in slightly different measuring conditions today June 22, 2010:

Prior to any treatment:

The Mod Podge Hard Coat still has 2 more weeks prior to full cure as of this graph where the Aleen's is well beyond its stated curing time.

Cone Treatments Demonstrated To Work (Geometrical Stiffening)

2011-08-26T18:44:00.000-07:00

Polar after large dust cap added:

impulse:

AVG:

The other driver:

Cone Treatments Demonstrated To Work (Surround Damping)

2011-08-26T18:38:00.000-07:00

Here's that same driver after damping the rim 3 times with the Aleene’s® Flexible Stretchable™ :

Polar

Impulse

Average of Polar

Here's another driver start to finish I did all the same things to:
Polar:

Impulse:

AVG:

Next up, Geometrical stiffness.

Cone Treatments Demonstrated To Work (Stiffening)

2011-08-26T18:27:00.000-07:00

Here's the new driver prior to cone stiffening treatment with Mod Podge® Hard Coat:

and after:

The impulse before:

and after:

Average of polar response before:

and after:

So what do we get? A higher rim resonance, deeper notch(es), reduced efficiency, and a little better impulse response.

Cone Treatments Demonstrated To Work (Damping II)

2011-08-26T18:18:00.000-07:00

Check out this same driver's impulse responses from the exact same angles. Here's just the first and the last:
first:

Last:

Then the first and last polar response average:
First:

Last:

You can see that the weight has apparently reduced efficiency.

Cone Treatments Demonstrated To Work (Damping)

2011-08-26T18:10:00.000-07:00

OK, here's a super cheap FRer's polar response from 11.25 degrees to 90 degrees in 11.25 degree steps straight out of the box:

Then I teated the surround with Aleene’s® Flexible Stretchable™ :

That was more change than I had expected--and for the better! So I tried again:

Notice that the rim resonance is down 6dB from the original after just 2 treatments! I though it could get no better, but figured what the heck, throw another slab on. This time I also added some yellow glue around the rim of the cone:

Another improvement so I did the same again:

OK, so in many respects this one is starting to look worse. I figured there was still a change and I should now just see where it goes. I didn't want to see worse again, so I didn't add any more yellow glue to the rim and damped the whole cone.

and again:

and yet again:

That's where I stopped for a listen.

I'll follow this up.

Infinity Classia 336BK

2011-08-01T23:21:00.000-07:00

These are too near field to represent actual performance.

no reflections!?

At regular price, I'd have to choose the Primus over these. At the crazy sale price I got these, I'd give these the go ahead. They are just much better built than the Primus. The Primus doesn't feel cheap, but these are very stout and have a more ritzy appearance. Even the binding posts are upscale. If you're a pretentious snob, these are your speakers.(kidding) Can't go wrong with either IMO and if you can find these at near the price of the Primus, I'd say go with these.

Infinity Primus 363

2011-08-01T23:19:00.000-07:00

These are too near field to accurately represent performance.

I gotta say I love this speaker. As far as I'm concerned this is the all time budget king. It's clean, loud and tonally accurate with a spacious sound. I'm still trying to convince myself the Classia is better, but they each have their strengths. I'm betting the Classia would withstand more high volume abuse, but that's not anything I'll likely test adequately. The only other speakers I've measured/listened to that I like more than these is the Mackie HR624mk2 and the JBL LSR2325. Too bad the Mackie/JBL don't interact with the room nearly as well as these do in the low end. These things eat the Mackie/JBL alive below 300Hz in my 2 listening room as do the Classia. Maybe this is just my rooms and possible placements, but I think this is just where floor-standers have an advantage. Above 300Hz both the Mackie and JBL seem more neutral and less flattering than these Infinity, but any of the 4 of them I could be happy with for casual listening above the room controlled zone. Makes me want to stick a crossover and a stack of woofers under those stand-mounted, bookshelf speakers.

The Theater

2011-07-22T18:14:00.000-07:00

Well, here's the front end of my Home Theater(mostly absorptive barring the TV)

Zoomed in a bit:

Cleaned up some:

OK, some I'm getting some more insights as to what constitutes great home theater. I've gone through massive quantities of reading(for a hobbiest), measuring, EQing, placing, and calibration. My conclusion is this: no matter what the acoustician tells you, acoustics treatments in any reasonable quantity for the regular guy doing this themselves on a budget will not get you to nirvana. No matter what the room correction software guy tells you, it can only do so much and will not lead to nirvana. You cannot EQ a cancellation. A) it will never fix it, and B) you'll lose all headroom both amplifier and speaker.

My personal experience with this goes like this:
1) placement of speakers is the biggest challenge to get right.

The first time I measured my home theater(of which I was so proud of the sound) it looked like this:

I thought this sounded great! Please no comments about me being an idiot. I already know.

Bryan Pape (works for GIK) gave me a strategy for sorting out my biggest problems after I was frustrated with DIY treatments. Some of you will know him from the forums and he is an excellent guy and seems he almost see's sound. Never met a more helpful/knowledgeable person other than perhaps Dr. Earl Geddes(who seems to be a bit more polarizing on the forums). None the less, he is damn honest even if his opinions are not popular. Experimental data continues to support his statements. Anyway, enough about that. After listening to Brian I ended up getting this:

and the AVG:

Then smoothed:

I ended up bumping the bass up for my personal preference after that, but never graphed it. All that with just placement just using Bryan's advice on how to place appropriately! Can't believe he didn't just try to sell me treatments. I can't give a better endorsement than that and I will buy treatments from him when the time comes just so his honesty won't be to his detriment. Besides, they have intelligent products that are reasonably priced. In any case, optimal placement for acoustics, just looked stupid. Sounded great, but looked bad.

2) Treatment and EQ: So I just bought an old model HK receiver model 3600. Nice rig and it sounds great. I hate to say it b/c it will make me sound like a subjective cook, but I think it sounds better than my defunct Onkyo TX-NR1007 ever did. I thought so in the store as well, but I bought the Onkyo d/t all the DBTs that people couldn't tell a difference.

My first shot at using the EZ-EQ looked like this:

With EZ-EQ turned off it looked like this:

I moved the speakers just about six inches, and to eliminate the floor bounce issue, I used a couple 15" square open cell foam block I have hanging around placed on the floor midway between the speakers and the listening position.

The six position average looks like this:

Then I removed the foam and took the same measurements again:

Admittedly, the perceived sound didn't change much, the SBIR influence is not hard to see. So I did another calibration w/o the foam to see what would happen(I also fiddled with the manual bass EQ to get a smoother more extended bass and added 1/6 octave smoothing):

Smoothed with foam:

How that compares do what Dr. Olive shows on his blog with Harman room correction related to preference:

It looks pretty close to me and tells me that their room is wisely treated b/c w/o a good treatment scheme, I was unable to duplicate their results.

I would also agree that this is the best my HT has sounded, but no one method would have got me there. You need to use all three and placement is the most important IMO. Now I'm going to enjoy some movies and my iPod d/t the dock that came with the 3600. It also has a great tuner. Truth be told I couldn't be happier with this and I'm sort of glad the Onkyo suffered an early death.

The Dan"TheMan"Cave

2011-06-14T18:13:00.001-07:00

For rationale click here.

Sketch Up

2D Detail:

Measurements:

Mackie in stereo

listening position JBL mono

RT20

RT30

RT60

RT60 mono

From performance position

From Dr. Olive's Study
and my new six position AVG:

Old 6 Position AVG:

Pictures from the Cave:

Right Front

Left/Front

Left Side Hidden Hendrix

Rear Wall

Right Side Diagonal Wall

Doorway with Movable Treatments

Woofer(Iris)

Me, Over-Caffeinated, Overworked

Binaural Hearing and Early Reflection Perception

2011-06-04T12:18:00.000-07:00

One perhaps nonintuitive fact about reflection detection related to binaural hearing is the reflections are almost equally detectable from the side walls or floor and ceiling--the medial or lateral plane. (Olive and Toole)

From Olive/Toole

The audible effects of these reflections however are different.

From the sides(lateral plane), reflections add spaciousness and source broadening due to interaural differences. Theater and concert hall designers have utilized this for decades as Dr. Toole's studies found this is preferred by listeners. Borish (1986) found that “the only subjective effect of delayed sounds when laterally displaced is spatial impression”. However, this spaciousness/envelopment comes at a price. Click on the pics below.

From NASA

Implications are essentially why THX has their surround sound standards--narrow loudspeaker directivity and surround speaker to the rear of the listeners. The best of both worlds--precise imaging, envelopment, and spaciousness. This also partly accounts for why the origin of high frequencies are more easily heard--less early reflections d/t higher directivity and they are more difficult to reflect IOW readily absorbed.

Medial plane Reflections: Drs. Olive and Toole found in "The detection of Reflections in Typical Rooms", that vertical reflections essentially effect the timbre of the perceived sound. Not so different than what's been found to be in Binaural Hearing and Direction Perception. Read it here.

Excerpt from Olive/Toole

Check out NASA's "speaker cage" used to help it build virtual auditory environments to vastly expand on this post. Finally a great use of our tax money! :)

The “speaker cage” developed at Armstrong Laboratory, Wright-Patterson Air Force

Base for HRTF measurements. Note the sound absorbing foam around the struts. Courtesy of

Richard McKinley, Armstrong Laboratory, Wright-Patterson Air Force Base.

The book of their research and its implementation is downloadable for free here. Highly recommended!

Happy reading.

Binaural Hearing and Direction Perception

2011-05-31T12:45:00.000-07:00

Binaural Hearing: We have 2 ears located on each side of our head at roughly the same height. The implications are we hear different in the horizontal and medial planes d/t our one ear's position relative to the other.

Vertical Directional Cues: Head Related Transfer Function. Our head, shoulders and pinna modifies the sound which aids in locating its source. The resonances and reflections cause our head to believe it knows the vertical location of an object by how much these bands are present compared to the rest of the signal. IOW the vertical cues are essentially timbre modification. The illustration(s) below from Spatial Hearing by Jens Blauert demonstrates the principle:

And resonance caused by the pinna/ear canal

The first picture tells us the ear has trouble distinguishing what spectral information is coming from the source and what is caused by the Pinna and ear canal.

Horizontal Direction Cues: You have interaural time and level differences. At frequencies smaller than the head (approx 1500Hz), we mostly asses the level difference b/c of the head shadow. At frequencies lower than 800Hz, IOW significantly larger than the head, we assess the phase difference. These frequencies are not exactly set in stone and it is thought that there is a smooth/overlapping transition from one mechanism to the other.

A great article on the subject right here. Just click.
demo
further reading

Hearing Beyond Haas

2011-03-05T13:48:00.000-08:00

Below 300-500Hz in a listening room the frequency domain is the most relevant acoustic metric to our perception. Above there, it's the the loudspeaker's anechoic response. I'll do my best to offer an explanation.

Brush up on this first: Haas Effect
Haas Demonstrations

The cochlea's operation varies with frequency and has been mapped. This is part of the place theory of pitch perception.

from Encyclopedia Britannica. The place theory is not perfect b/c you can play a note, remove the fundamental, and still have the same pitch perception. (Schouten, 1940) Other issues that cannot be explained by place theory:

the fine degree of accuracy observed in human pitch perception,
pitch perception of sounds whose frequency components are not resolved by the place mechanism,
the pitch perceived for some sounds which have continuous (non-harmonic) spectra, or
pitch perception for sounds with an f₀ less than 50 Hz. . . .

Taken from Acoustics and Psychoacoustics by David Howard and Jamie Angus.

Then there is the temporal theory--the timing of neural firing. At low frequency, neurons can fire as fast as the frequency to around 300 Hz. Above there they cannot keep up so pitch detection becomes progressively more by location of the firing on the basal ganglia with it being essentially by "place" at 5 kHz(some experts say 4 kHz and some as low as 1kHz). Below 1-5 kHz and above 300 Hz, the nerve firings are phase locked but fire in an almost random fashion. All this combined with ear canal resonance and pinna modifications above 2 kHz mostly explains the equal loudness curves as number of nerve firings is approximately equal perceived volume. We become naturally less sensitive to low frequency sounds as less nerves are firing.

We have very good freq detection below 300-500Hz because the neurons can fire at similar speeds as the wave forming. The question is "Why poor time resolution?" I'd say it's because the wavelengths are huge and take a lot of time to form. Simple physics. This fortunately falls into the range of where the room goes modal for the very same reason and decay times likely rise. Above there our time resolution is comparatively good barring ear canal resonances, but pitch detection less so as we'll see.

Let's look at Haas for another second, but not as it relates to directional hearing. We don't hear reflections as a separate event until at least 30ms according to the Haas Effect(which only looked at speech signals and ultimately depends on level as seen below).

(Toole, Sound Reproduction)

It takes a 50 Hz wave about 20 ms to even form, but only 2 ms for 500 Hz, and 0.2 ms for 5,000. The number of listens or complete cycles it takes for our ear to distinctly perceive a pitch varies with frequency as graphed below.

Log Frequency (Hz)

(Rossing, 1989)

To recognize a definitive pitch at 50 Hz it takes approx 2.5 cycles or 50 ms. You can see Haas's effect cannot operate here as it does for smaller frequencies. You can't hear an echo or a smudging of something that you can not perceive as a distinct pitch. At 500 Hz it takes closer to 8 cycles or 16 ms which is just inside the Haas limit. So at 5,000 Hz, it takes approx 80 cycles to hear a definite distinct pitch or 1.6 ms.

So what does this mean for the studio engineer, audiophile or home theater enthusiast? Below 300-500Hz, an EQ can be used to largely correct audible errors at the listening position provided the null frequencies are not too deep. The deeper in frequency, the more you can be sure the measured response will be more audibly accurate. Bass measurements need to be taken at the listening position(s) as the bass builds up before we hear it clearly, the measurement is shorter than the integration time of the ear, and the visible graph is steady state in the room. You can think of it as existing around us instead of moving from front to back and is part of the reason we don't hear our subwoofer's location. Theoretically we may not hear 'fast' bass, though I'd bet we can still feel it. Think of it this way; straight EQ doesn't function in the time domain, only the frequency domain. Since we have poor time resolution in the bass but great frequency resolution, an EQ can trim the peaks(but boosting deep nulls is a bad idea in a moderately powerful sound system). Placement of the subwoofer(s) and listener(s) is of primary consideration to avoid nulls. A cancellation cannot be fixed by EQ. Bass damping can help all aspects of bass reproduction, but a couple bass traps are no where near as effective as good placement. Room correcting software should be calibrated after trapping is installed and placement is at its best to remove listening position nulls. This paper from Genelec seems to agree experimentally. Perhaps we can say that an EQ will absolutely work for your subwoofer.

Here's the graph of the data from that paper:

Above there, 300-500 Hz, the speaker needs to have a flat anechoic response with a smooth off axis to reproduce the recording with a smooth off axis response and cannot be EQed by the 'in room response' d/t the ear's time resolution, the Haas Effect, and all the reflections contaminating the "in room" graph. For more information, see this.

Tightening The Loudspeaker, Recording and Room Connection

2011-02-27T23:07:00.000-08:00

The obvious:
Since none of you reading here likely live directly under the stars, you probably listen to or record audio in a room on a regular basis. I'm going to avoid all discussion of cars and headphone listening though I do plan on going into the headphone experience before too long. I'll also avoid discussing the bathroom and kitchen as they have more important uses than music listening.
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Breakdown of related topics:
Perception and Loudspeaker patterns:
http://dtmblabber.blogspot.com/2010/12/psychoacoustics.html
http://dtmblabber.blogspot.com/2010/12/above-modal-dominated-frequencies-300hz.html
http://dtmblabber.blogspot.com/2011/01/psychoacoustics-continuedrelated-more.html
http://dtmblabber.blogspot.com/2011/06/how-binaural-hearing-relates-to.html
http://dtmblabber.blogspot.com/2011/05/binaural-hearing-and-direction.html
Room treatments:
http://dtmblabber.blogspot.com/2010/12/room-treatment-advice-quick-and-dirty.html
http://dtmblabber.blogspot.com/2010/12/little-more-into-boundary-conditions.html
http://dtmblabber.blogspot.com/2010/12/mars-bringer-of-pain.html
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Now the new:

ITDG-Initial Time Delay Gap is difference between the first arrival sound and the first reflected sound.

What I'm about to say assumes you are after Hi Fidelity. That doesn't mean sounding like the original event, it means reproducing the recording as that's all you have access to and many if not most of them could be said to have no original event(think movies and studio recordings). OK, enough on that.

So we've already seen that these reflections will effect the frequency domain, but we haven't looked into the time domain. Sound travels at roughly 1 foot per millisecond. Check here for more specifics in metric. So you can take the time of the reflected sound path and subtract the time of the direct sound path to get your ITDG. A smaller gap time will tell your ear that the source is further away, a larger one closer. This may seem counterintuitive but it's more easily understood when you look at the demonstration here. The reflected path will be nearer the same length as the direct. Of course the amplitude(volume), frequency response and relative reverberant decay are also used to aid in determining distance between you and the sound source. I'm sure you've all heard the big, bad car stereo as it drives down the road. When it's close you can hear the treble loud and not so clear. From a distance, it's just bass and quieter. This is d/t air absorbing treble more easily than bass, but again beyond what I'm getting at. These are just part of distance cues. The clever home recording enthusiast may want to think about that during their next session. Hopefully I'll get to HRTF before too long for those guys as it can be used for other effects. HRTF(direction) HRTF(reflection)

So replace the loudspeaker in the picture with the live performance of your choice and decide to record it. Your microphone will pick up all these same perceptual distance cues. So what happens when you bring that space into your own through your loudspeaker? They overlap and wreak havoc unless your listening space has a longer ITDG than what's on the recording. Davis and Davis found 2-5 ms longer is needed to avoid this auditory confusion. They also add that a diffuse sound field should be present by the initial onset of the Haas effect--approx. 20 ms. This means we need a Reflection Free Zone (RFZ) on the front side of our listening room, and a diffuse back side. Dr. Toole found that these early reflections need to be reduced by 15 dB to be perceptibly inaudible and more so with Mozart. Live End, Dead End (LEDE) anyone? Absorbing the front wall reflections is an absolute must and front section of the side walls, floor and even the ceiling that the front loudspeakers sit in. The first contralateral reflection points may well depend on the size of your room. So, assuming propagation loss of a few dB, you'll need to damp the reflection points by at least 10 dB to get to the target loss. That requires an absorption coefficient of at least 0.9. I would shoot for as low as possible in the frequency range and hopefully into the modal region. That requires heavy duty material. 2" foam need not apply. At least 4 inches of rigid fiberglass or rock wool, and probably 6 inches of acoustic foam.

This graph compiled by Drs. Toole and Olive will flesh out the reflection detection picture with different signals:

Seems the ITU knows this graph well.

Further reading:
An excellent read and easy to follow.
http://www.soundonsound.com/sos/1997_articles/jan97/controlrooms.html
http://www.aes.org/technical/documents/AESTD1001.pdf
http://tech.ebu.ch/docs/techreview/trev_274-hoeg.pdf
http://alexandria.tue.nl/extra2/afstversl/tm/munster2003.pdf
classical control rooms
reproducing recordings

Psychoacoustics Continued/related more to speakers

2011-01-28T01:22:00.000-08:00

So why don't I advocate an omnidirectional speaker? It would make sense--reflections add spaciousness and detail. The problem is--they also add to timbre/tonality/spectral balance. The engineer that set the levels, EQ, pan, compression, etc... on your recording was not listening to an absolutely omni speaker. Odds are it was omni in the bass and more narrow in the treble just like every speaker I have measured. Therefor an omni is going to automatically be too bright in most set ups. That not even getting into the early reflection problems (click for more info on SBIR and early reflection detection ) and greater RT60s most home listeners are likely to experience.

B&K's studies concluded that the ideal steady state listening room response looks like this:

I'd say that's probably about right for most rooms. More psychoacoustically correct methods for evaluating a loudspeaker and a room are in the works: http://www.aes.org/e-lib/browse.cfm?elib=11797

The film industry's "X Curve" shows a similar trend:

but for a much larger listening space and again this is steady state. Some of that was done to hide flaws in the recording technology of the day as well.

So what would I look for graphically in a speaker? A smooth, flat response across roughly the front 30 degrees horizontally with a smooth, falling treble response further off axis. I also look for the vertical polar response so I know how to best aim the speakers. Bass has been found to account for about 30% of listener preference--so get some subs. Resonances are important! They will color the sound you hear. If you can see them in the polar response, you will hear them. BTW, my polar response graphs posted on this site shouldn't really be trusted below 1kHz--they are low resolution and too near field for anything to be drawn from them. Wider dispersion speakers should probably be kept away from the boundaries (see references below) which means they will require a larger room (and they typically are incapable of high output to start with). My 2 favorite speakers I've had in the house and measured are the Mackie HR624 mk2 and the JBL 2325. They are somewhere in the middle dispersion wise and are free from resonances/diffraction artifacts from 1 kHz to 10 kHz. That's what I'd primarily look for, and then how much output is it capable of. Resonances become more audible as output is greater and motor nonlinearities will rear their ugly heads as heat builds up.
http://www.gedlee.com/downloads/AES06Gedlee_ll.pdf

Narrower dispersion speakers can probably be used in any room at a further than 2 meters/ 6 feet in a cross-fired manor. They are also typically large (Larger speakers naturally have less dispersion), efficient, and handle high volumes well. This makes them ideal for just about any situation except near field monitoring because every movement will result in a large difference in audible FR(don't tell recording engineers--many won't believe it). They are also the most difficult to build well d/t the large cones and waveguides involved. Just listen to PA speakers and ask yourself if you would want that in your living room. Dr. Geddes, Jean-Michel Le Cleac'h, and I'm sure many others have gone to great effort to improve this sort of speaker system and they are not cheap.

So my opinion would be to avoid the extremes b/c the middle of the road is affordable, easy to get a great spectral balance from, and capable of accomplishing the output required of home speakers. Broad dispersion will require better drivers to get great output and thus cost more for HT needs, but will give you more than enough for near field monitoring. Narrow generally will give you all the output you could ever want, but the less resonant/great sounding designs are costly.

Oh, read this as well:
http://seanolive.blogspot.com/2008/12/part-3-relationship-between-loudspeaker.html

Bass must ultimately be measured in room as the room becomes part of the source at low frequencies. The time domain is less important here and the frequency response is the dominant part of what we hear. That's not to say the time domain can be ignored, just that FR is the most important thing to pay attention to in the bass range. Good time performance/quick decay is the icing on the cake.

The in room, steady state midrange/treble response is not important either except that it should be rolled off to some degree. A smaller room should probably have less than a large room. We have great time domain hearing resolution and the anechoic response of the listening window will dominate our frequency perception and the reflections will add detail, space, source width, envelopment and timbre as well.

More recently Dr. Olive did a subjective evaluation of room correction software. The results seem much in line with older studies.

Dr. Olive's Study
This is where this post started:
http://dtmblabber.blogspot.com/2010/12/psychoacoustics.html
http://dtmblabber.blogspot.com/2010/12/above-modal-dominated-frequencies-300hz.html
Further reading that will cost some greenbacks on the subject:
http://www.aes.org/e-lib/browse.cfm?elib=6248
http://www.aes.org/e-lib/browse.cfm?elib=7673

JBL LSR2325

2011-01-09T17:24:00.000-08:00

This one was returned to the store previously. It had no manual, no wrapping, and a crinkle in the tweeter dome.

More of my standards. Self explanatory at this point. The second one of the pair looks better:

It was new.

Review of Polar graphs

2011-01-08T20:27:00.000-08:00

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B2031P with cotton

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B2031P

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B1030A

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HR624 mk2

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Homemade

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Rokit 6

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LSR 2325

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Infinity Classia 336

Multiway/near field--low end poorly represented

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Infinity Primus 363

Multiway/near field--low end poorly represented

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SOS on monitors

The KRK Rokit 6

2011-01-08T20:22:00.000-08:00

Standard measurements here on the popular Rokit 6. I was listening to its larger brother in the store today and heard an issue in the MR. Perhaps it's an inherited trait. It's not as noticeable in this version while listening in the living room. The response is extremely broad and it makes it tough to get a clean measurement at home so I won't go into the time domain stuff. The treble does sound very smooth and a lot like the Behringer 1030A from memory but a bit less revealing. Another thing of note is that the vertical polar lobe is aimed a few degrees toward the woofer and it is again very broad at over 30 degree of flat measurements. I'd like to stuff these ports as well. The impulse has 2 broad humps that makes me think that's what I'm hearing in the bass. None the less, I'm not going to mess with them as these aren't mine.

A little more into boundary conditions

2010-12-29T20:56:00.000-08:00

The Front wall, ipsilateral side wall, and floor will all cause big issues with your frequency response that will be audible d/t the ear's increased integration time as frequency goes lower. In other words, we don't hear with the greatest time resolution at low frequencies so we'll hear the reflections as part of the source. This is why I like wavelets--but that's been touched on earlier in the DTM Blog. The contralateral wall, ceiling, and possibly even the rear wall will also come into play and more so in a small room I imagine. As the distance of the reflecting surface gets further away from the source, the less pronounced the dip will be thanks to the Inverse Square Law--every time you double the distance, you loose 6dB.

The equation for the front wall interference:
Cancellation notch frequency = [(344m/s) divided by (4 times the distance from the speaker to the wall behind it in meters)] (344m/s)/4Dm=fHz
picture for reference:

A quick Table for reference:

This equation for other significant boundaries:
notch frequency = [(344m/s) / 2(the distance of speaker to the listening position via the reflecting surface - the direct path distance from the speaker to the listening position)] (344m/s)/2(Drm-Ddm)

This picture may help with visualization:

floor bounce

A couple more quick reference tables for this SBIR modifier:

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For an online calculator (metric) go here:
http://mehlau.net/audio/floorbounce/
For sidewall reflections:
http://mehlau.net/audio/reflection_sidewall/

So for a practical example:
We have our speaker on a 3 foot stand. Its Baffle is 4 feet from the front wall and we are sitting we are sitting 9 feet from it at a listening height of 3 feet. That gives us enough information to find 2 of our first notches. The front wall will cause an approximate 70 Hz notch while the floor bounce will be a little over 300Hz. Keep in mind that the ipsilateral wall will still have its say as will the modes and to a lesser degree the other boundaries. For the ipsilateral problem it will be necessary to go to mehlau.net for a decent calculation.
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The previous section was just the the start of

Comb Filtering

The comb filter starts at that first notch frequency related to the SBIR. If you double that frequency, the interference pattern from the reflected wave is actually additive due to its phase relationship with the original wave. It is now back in phase and thus you essentially have 2 sources reinforcing each other. That gives us 6 dB of gain. This kicks off a pattern of alternation cancellation and reinforcement for multiples of the original notch resembles the teeth of those cheap black plastic combs you all have with the teeth getting progressively closer together as frequency get higher. It will also become less pronounced with increased frequency d/t narrowing directivity in the MR/Treble.

Here's the expanded tables from the first section on the effects:

Floor Bounce

Room treatment advice--quick and dirty

2010-12-27T19:22:00.000-08:00

Controlling bass is difficult in a small room and likely the #1 priority of room treatment.

In general: for typical box speakers you want to absorb the whole front wall(the one behind the speakers or you depending on the situation) with something thick like 4" thick or so. The further away from the wall, the better it will absorb lower into the bass range. Bass trapping the front corners with broadband absorption would go right along with this. This will reduce Speaker Boundary Interference Response(SBIR), reduce coloration and improve imaging. The next absorption point would be your first ipsilateral reflection and can be done with 2 inches of absorption material. 4 inches is probably better and space it a few inches off the wall. If your speaker have good off axis performance, this is optional depending on your desired effect. That can help reduce high frequency hash and harshness caused by diffraction as well as squash polar anomalies. That said, unmitigated early horizontal reflections have been shown to be "preferred" in blind tests. They will increase your apparent source width and can arguably be detrimental to pinpoint imaging.

Then the rear wall with the thick stuff for more bass reduction, but diffusion can be used if your bass is well behaved. Some people will also recommend absorbing the first reflection from the ceiling--others would say diffuse it. Diffusion makes sense because the ceiling height is high enough to allow it to work. The side walls can benefit from diffusion to improve spaciousness and sense of envelopment. Diffusion has to be thicker than absorption to be effective to as low of a frequency and it can be expensive. I would generally recommend broadband absorption midway between you and the speaker(or microphone as the case may be) to absorb the floor reflection--like a think foam pseudo-coffee table I use with stand mounted speakers. Measurements of the coffee table found here.

If building new construction or if you are willing to modify the existing structure, making you walls into resonating diaphragm absorber makes better sense for controlling low frequencies because the areas of highest pressure are right next to the walls. Porous traps reduce particle velocity and best used away from walls for bass frequencies.

I will continue update this when I have time/desire.

Rationale

More Rationale
Further Reading
Boundary Issues

Mars, the bringer of pain

2010-12-26T14:15:00.000-08:00

The bass region is a pain in the butt. You can see in all my graphs that this is where things get hairy. The length of the sound waves are just so long that they don't get to stretch their legs in a typical room. There are many opinions about how to deal with them, and none are ideal. Many people will say 'absorb them'. But where absorption is most effecting is a quarter of the offending wavelength into the room with an thick absorber. The is cause all sorts of issues as it means you will have bass traps in the middle of the room. You can't really diffuse them b/c they are huge! Some will say multi sub arrangements, other will say keep subs to a minimum to reduces sources of interference..... The best way I have found is to build some huge absorbers that nearly cover the front wall, do a lot of experimental placement of your subwoofer(s) for smoothest response in the listening area, ajust phase and crossover points and then EQthe max points as best you can. The only way to get rid of the nulls is plenty of damping by whatever means and good positioning. This next graph will show the effects of 2 large absorbers added 1 at a time to the front wall:

Not very effective in the deep bass range. I haven't applied any EQ in this system yet. There are a few graphs on this site in my quest to perfect the bass range. My best listen space graph is probably this--10 overlays across, up and down the listening space in my home theater:

It started out looking like this:

It sounded thin like that. I bump up the bass knob to taste afterwards.

The best look at room modes that I know of is here: http://www.hunecke.de/en/calculators/room-eigenmodes.html

Above the modal dominated frequencies--300Hz and up

2010-12-26T13:12:00.000-08:00

There are basically three types of loudspeakers as far as I'm concerned: ones whose off axis behavior bears little resemblance to on axis, loudspeakers with Wide radiation patterns and ones with narrow radiation patterns. Most speakers are of the first type and of little use to me for building a listening room, home theater, or mixing/mastering studio since we hear the early reflections as part of the original. It would be hard to treat your room to correct for wild sound radiation patterns unless you were to make your room anechoic. There's a belief that you can EQ poor loudspeaker radiation patterns, but you really cannot EQ acoustic problems. This is an acoustics problem so every problem you "fix" will create another. Newer software corrections seems to be improving these erratic loudspeaker's perceived behavior. Let's assume we want to hear the recording and not get into the wild world of mini monitors and unruly loudspeaker radiation patterns and you don't have the latest experimental room correction software.

Ok, wide pattern loudspeakers: above is a Linkwitz Pluto
( http://www.linkwitzlab.com/Pluto/specs.htm ) and we have a graph of one right here on the Blog--the Behringer 1030A:

So what sound effects can we expect from this type of speaker in stereo?
Spaciousness, good envelopment (sometimes described as 'liquid sound' or a 'wash of music' on audiophile sites), an improved vertical dispersion compared to narrow directivity, and detail. The "detail" seems to be a result of the ear getting 'a second look' according to Dr. Floyd Toole's proposal. The 1030A gave me so much detail, I heard things in recordings that really shouldn't be there and thought the speaker must be broken. I was wrong as I was later able to hear the same low level flaws on other speakers, just not to the same degree as they were exposed through the wide radiation 1030A. Some potential negatives from this system of loudspeakers has to do with early reflections that can be loud enough to cause image shift or poor imaging. I've never heard this problem, but I know others who claim it exists. Source broadening is real and a vague image could be described. Others might say it fills in spatial gaps... So is this a problem? That's up to you. The only time I've ever heard a serious image problem from reflection was with a Martin Logan panel speaker. Some sounds seemed to be coming from the front wall and the soundscape was essentially a mess. Imaging was out of the question. The guy next to me thought this effect was very cool. Panel speakers fall into my 'not to be discussed' category. These broad output patterned speakers tend to be small, inefficient, and require a lot of power for high output that they can't comfortably handle. They also tend to have a wide vertical window which makes them excellent near field monitors and a less abrupt change from omni radiation to wide in the bass to midrange transition compared to narrow pattern speakers. In theory this could give them an edge in tonal rendition. The Linkwitz link has a ton of info on setting these speakers up.
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Then narrow directivity made famous by Dr. Geddes and a version of his Summa The graph is displayed above and used to be posted here:
http://www.gedlee.com/summa_.htm
It was removed for more advanced displays and/or perhaps an upgraded design. I'll post this one for comparison sake and it's similar to my early attempt at loudspeaker design shown below.

Dr. Geddes advocates this type of system for many, pardon the pun, sound reasons--too many for me to go through them all. Essentially you can crossfire speakers with a tight toe in. The listening axis would then be off the center of the speaker by around 20 degrees. This is said and confirmed by my experience to have excellent spaciousness because of the high level of contralateral reflections, good interaural cross correlation, and precise imaging for many listeners. It reduces the influence of the room and very early reflections which are perceptually significant (these designs do not help most SBIR issues) and ITDG but suffers in near field performance due to large variations in FR with small movements and the narrow vertical and horizontal patterns--refer vertical polar response written earlier in the blog. THX recommends this type of speaker as does recent work of Princeton: http://www.princeton.edu/3D3A/Directivity.html though for use in a different way. The problem I had with type of speaker was caused by my particular design. You can see the ragged treble response and this resulted in some treble harshness. Graphically Dr. Geddes's speaker doesn't seem to posses this issue. It's just a better design well executed. Other things of note is that these speakers--narrow directivity--generally use a powerful, large diameter pro woofer and a waveguided compression tweeter. These tend to make a big, dynamic, and easy sound that pretty much any amp can drive them to THX reference levels and beyond. One particular advantage for the Summa is the fact that it maintains its pattern nearly to the region where a room goes modal and essentially becomes part of the source!

The other 2 speakers graphed in detail on this site are somewhere in between as far as directivity is concerned: the Behringer B2031P and the Mackie HR 624 mk2 with the Mackie leaning more toward the narrow. The B2031P displays some of the off axis nastiness that my design had but less pronounced. Reverberant rooms will expose this off axis nastiness more than than dead ones and reducing the very early reflections with these speakers is definitely a positive as is adding cotton balls in the ports beside the tweeter.

That's pretty much the basics of above the modal region and it should give you much of the info you need to make a reasoned decision about sound quality and buying a loudspeaker. I'm sure I will edit and update this in a few days or so. How much output and power handling will need to be addressed as well, but there is plenty on the web about that and little about this.

Next we go below 300Hz to Mars.
Study on optimum loudspeaker pattern for imaging.
How loudspeaker pattern effects perceived tonality.
Genelec Paper on Diffraction
Some Rationale
More Rationale

Venus and Mars

2010-12-24T20:37:00.000-08:00

Essentially somewhere around 200 and 300Hz, something funny happens; the response becomes dominated by the room below those frequencies but remains dominated by the loudspeakers polar response above it. This is called the Schroeder Frequency. There are several factors that cause this. 1 is that most loudspeakers are essentially omni at low frequencies and another is that these wavelength are similar in size to the boundaries of the room and the distance to the boundaries from the loudspeaker. There is no clear cut division d/t the difference in all these factors in each of our rooms.

Tonally what we hear above this region seems to be most dominated by the listening window of the polar response, and below this region it is the ungated measured response of the louspeakers in the room at the listening position.

More to come later,

Dan

Psychoacoustics

2010-12-23T19:48:00.000-08:00

I'm going to try and summarize some basic psychoacoustic phenomenon important for the audio, speaker, home recording or home theater enthusiast. It's hard for me to determine what is important to start with so I guess I'll start with explaining myself a bit. My understanding and writing of this is in no way comprehensive and it's not intended to be so. I just want to make this stuff accessible to those who don't want to read giant books.

So briefly: 1) The Haas Effect, often referred to as the precedence effect, is the principle that the first arriving sound tells the ear where the sound is coming from. Your head takes all those reflections inside roughly 30 milliseconds and integrates them into the original. Even if the delayed sound is up to 10 dB louder, the ear will still integrate it into the original. Reflections after this time period are heard as echoes and within this time period they contribute to the timbre, add spaciousness and a sense of detail to the recorded material.

That brings me to my next point, 2) a loudspeaker's polar response. If all those early reflections are integrated into the original, shouldn't they be spectrally similar to the original? Sound waves are coming off a loudspeaker in all directions, bouncing off your walls and into your listening space. It certainly seems reasonable for sound reproduction. This may sound like I'm advocating an omnidirectional speaker, but it's not so simple. I would think in the right room and positioning, I'd bet that Mr. Linkwitz's Pluto would be an outstanding speaker. Anyone in the Bay area has one and would let me listen, Just write me! :D

OK, I'll move onto just one more quick point today. I personally want to hear what's on the recording--not some radical distortion of it that may sound good to me on certain songs or whatever and terrible on others, so a flat frequency response is pretty much a must and a smooth off axis set of responses has to be there as well due to Mr. Haas's discovery. If the recording is really bad, your going to need an EQ--you'll need one anyway as I'll discuss later beneath 300 Hz.

Well, that's it for today,

Dan

topic continues here, in this post, and more recently here in detail.
Haas Demonstrations

Making a Polar Response graph

2010-12-22T10:19:00.000-08:00

OK, so how do I make all these polar graph? Good question. (ha ha) No seriously, I have a 4' by 4' piece of plywood that has a huge circle on it. That circle has a line that bisects it every 11.25 degrees. I place a speaker stand with an 2 line bisecting it at 90 degrees like an "X" overtop of the plywood so that one line integrates with the "0" and "180" degree mark and the other line integrates with the "90" and "270" degree mark. Line up the mic with the "0" to "180" degree line, set your level to a sensible range, and take an impulse or FR measurement using whatever method you have available. I use my MacBook, EMU 0404 USB, and a calibrated mic with Room EQ Wizard(A free program available at the Pro Audio Shack known as REW). After taking that "0" degree measurement, turn the speaker/stand to the next mark over and continue until you reach 90 degrees. Of course you can go further........ that's up to you and I encourage it. To manipulate the graphs into something more useful, see my previous post in this blog.

Info on Distance

Gating loudspeaker measurements

2010-09-18T22:02:00.000-07:00

So you want to get a nice clean measurement to design a speaker or measure a speaker you already have and you don't have an anechoic chamber at your disposal. Kind of a conundrum. You need a pseudo-anechoic measurement, but how does one make a pseudo-anechoic measurement? Eazy Peazy, you have to gate out the room reflections. IOW, remove the reflections out of the visible frequency response measurement. Since sound has a time element to it--it is a temporal thing, we'll have to look in the time domain to separate it out. In order to do that you need to see what is speaker generated and what is boundary/room reflection generated. The frequency responses at your left were achieved by various levels of gating using either the impulse or the spectrogram to discover where the room reflections enter the response. You can essentially close the gate on the reflection trying to get into the graph. In the impulse you can see where the major, high amplitude reflections come into play just after 6 ms, but the spectrogram makes things a bit clearer if a bit less absolute. In the spectrogram you can see not only the timing and frequency of the reflections, but their amplitude as well. Gating not only removes room reflections, but also the amount of data points in the measurement. So the longer the gate, the more detailed your gated measurement. As you look at the frequency response graphs and how they are gated, you can see the impact of the time and amplitude of the reflections shown in the spectrogram at that gate setting. You can try the same thing for any speakers you would like, but there are other ways as well. Read on.........

For woofers and mid/woofers, gated measurements can be less easy to be exact and the resulting frequency response graph becomes less accurate in the low end d/t reduced resolution effects of less data points. Good thing that there typically aren't huge problems in this frequency band d/t the relative rigidity of the cone and our perception of this range is dominated by the room anyway (barring baffle step).

The triangle in the top picture is the key to the math though the above(impulse) method is more useful--Pythagoras is your friend. 'A' squared + 'B' squared = 'C' squared. There are 2 right triangles in the pic above, just draw a straight from the midpoint to the floor and use it as 'A' or 'B'. The time difference between the arrival of the direct sound and first reflection is how you set your gate (reflected time-direct time=gate length). Sound travels at about 1ft per ms or 1 meter per 3 ms. Frequency Resolution of your gated frequency response = 1/Period(Gate). Let's say after you do your Pythagoras, a 6 ms gate is given(which would be excellent for in home). This limits your lowest frequency measured and resolution to 166.67Hz. Call it 170Hz. At 3 milliseconds your resolution becomes 333Hz or roughly 350Hz. You'll be able to get a longer time period/cleaner measurement from higher directivity speakers as the reflections won't be as powerful. The closer your mic is to the speaker and the higher the speaker is off the ground, the longer the gate you can have. However, the big chiefs in the industry say you need to get the mic away from the speaker for appropriate wavefront formation in the low end so it can be far field.

Them's the basics. Good luck.

Info on Distance

Forward Lobe

2010-09-06T12:08:00.000-07:00

Another important aspect of speakers is their vertical polar response. Every typical two-way speaker will have lobes and nulls in the response formed by the non coincident sources playing the same frequencies. In the near field--before the reflections become a major player--these lobes/nulls are audible with fairly small head movements if the nulls are not spaced very far apart. So ideally a "near field" monitor would have a broad forward lobe. The forward lobe is dictated by the distance between the drivers, the crossover frequency and slope, phase, and driver directivity at the crossover frequency, but that can be a heavy topic. Essentially you want to forward lobe to point at your ears. Check this out for a great visual on lobing: Falstad's wave interference applet.

Let's have a look at the studio monitors we have on hand.

You can see in the top graph that there are 3 lines that are basically flat. Since the speaker is rotated in 11.25 degree steps toward the tweeter, that corresponds to a 22.5 degree forward lobe that is angled 11.25 degrees toward the tweeter. Behringer lists the crossover frequency at 2 kHz for the B2031P(the speaker graphed). Seems pretty darned accurate. This is pretty good performance for the near field, but not ideal. The graph toward the woofer is 5 dB down at 11.25 degrees off axis. I got to do a little more precise measurements in smaller increments on this at a friends house and it's actually a bit better performance than what this indicates and the lobe is centered at 10 degrees toward the tweeter. I didn't include the graph toward the woofer

The next is the Behringer 1030A--graphs 2 and 3. The 2nd is toward the tweeter and the 3rd is toward the woofer. Combining those 2 you can see four lines that are basically flat. That is impressive performance--33.75 degrees of good behavior. The wide directivity of the drivers (small mid/woofer and shallow waveguide--see horizontal polar pattern for evidence), low crossover frequency, and close driver spacing contribute to this.

Now let's check out the Mackie HR 624 mkII, the 4th and 5th graphs.

The Mackie looks to have a bit more narrow lobe than the B2031P and it seems almost perpendicular to the baffle but slightly toward the tweeter. Not necessarily a bad thing, but good to know info. Placement is critical with these speakers. To be sure, these are my favorites out of the bunch, but they are the most difficult to get right. In the near filed, small head movements can definitely alter the sound. The 1030A seems immune to them. I listen to these Mackie monitors from 6-9 ft away. That gives you a vertical window of a little over 2-3 ft respectively. Of course you are also into the reverberant field at 9 ft so you can all but totally ignore the lobe. At 6 ft in my room I can still hear it with pretty dramatic head movements unless placement of me and the monitors is just right. At 3 ft away and a 1 ft vertical window, head movements can be an issue if monitor placement is not precise.

2010-09-05T21:37:00.000-07:00

OK, I did some more room tweaking with the Mackie monitors. I'm proud to say this is certainly among the best room responses I've ever seen and I did it with better placement rather than more treatment. Basically, that means I saved some dough. These are the averages across the room to make them simple. I threw the on axis vs. AVG response of the Mackie and the same with the center seat response and room average so the reader can make the correlation of how the polar response relates to what happens in the room above the modal region.

Updated Room Response

2010-08-21T23:40:00.001-07:00

So here's 10 frequency response measurements taken from the listening couch overlaid. Next is the average and followed by the decay time from each position and then the center of the couch--primary listening position. I gotta say I am pleased! This is the best sound I've ever had. Things are indeed looking up. The most audible defect is that 140Hz spike.

Mackie spectrograms

2010-08-05T21:38:00.000-07:00

Just another look at the Mackie HR624 mkII's diffraction.

Mackie HR624 mkII

2010-08-05T21:18:00.000-07:00

These are my standard measurements on this Mackie monitor. The nasty top octave isn't detrimental for me one bit. Also the vertical lobe is a bit narrow, but again doesn't seem to be a problem. Diffraction is excellent as is the off axis response. With the bass set to flat, it's a bit heavy, but stepped down a notch and it seems perfect. Overall I'm impressed with the sound.

Decay and Diffraction

2010-07-27T15:47:00.000-07:00

Gruesome title eh? Just a little side by side of the decay from 3 Behringer monitors.

Note the 1030A has the most reflection showing followed by the 2031P with cotton and the 2031P without cotton being the cleanest. So as you go lower in frequency and the dispersion gets more broad, the accuracy will be less precise for the graph with cotton and even more so for the 1030A. Also notice that the 1030A will get more reflections in the graph d/t its wider dispersion until the top octave.

Another Cheap Behringer Monitor

2010-07-27T15:14:00.000-07:00

I just did some routine measurements on the Behringer 1030A monitor. This thing looks pretty good on paper and the sound matches. This particular one unfortunately has a noticeable rattle while doing the frequency sweeps that wasn't heard with music and a sporadic treble distortion that is more frequent when the treble boost is engaged. t took me some time to figure out that the treble distortion was actually on the recordings I was listening to. I am just barely able to hear it on other speakers. This is not really a fault of the speaker except that it will expose problems on recordings that may have escaped the producers of the recording. The top is the vertical polar response toward the tweeter followed by the vertical polar toward the woofer. A fairly broad vertical lobe for such spacing. Next is the impulse response which is very clean and the horizontal shows a generally broad pattern with some degree of beaming in the top octave. Certainly no major diffraction issues here d/t the sculpted baffle.

My Room Response

2010-07-26T22:26:00.000-07:00

Here's the Frequency Response from three positions across my couch which is caused by the polar response from the B2031P and my 2 Allison subs. That bass end needs some work but that's typical.

Audiophile Buzzwords, a dicey subject

2010-07-26T22:04:00.000-07:00

How can loudspeaker measurements tell us anything about how a speaker sounds? Audiophiles generally don't think in terms of Polar Response, Impulse Response, Power Compression, etc...

They tend to look at things in subjective terms and often confuse musical terminology, Pace, Rhythm, Timing, Sound Stage, Dynamics, Spaciousness, etc.. for engineering possibilities and vernacular. This is a crude attempt by an amateur, myself, to bring these 2 worlds together. I won't go through all the buzzwords, just a couple to make my point.

Tonality can be looked at from many different metrics, but basically a polar response and impulse response will tell you what you need to know. Some people would argue for phase as well, but the science refutes that claim.

Look here for a good explanation what tonality is: http://en.wikipedia.org/wiki/Tonality

and you'll realize this is a music term, not a playback term. That said, wild polar responses will not have good tonality unless they somehow match the inaccuracies of the recording process--fat chance. If it does on one, it won't on any other. The recording process has as much to do with this as the playback. That's part of the reason why getting a polar response on a speaker is more useful than going and listening to your few favorite tracks. Rise time and Decay should also play a part and can be seen by looking at the impulse graph, CSD, wavelet, etc... With the impulse(s) and a polar plot, you'll have all that info.

What most people seem to describe as soundstage as far as I know mostly has to do with speaker placement and polar response. If you are shooting an even sound across your room and your room and your speakers are placed with the left on the left and the right on the right, away from the walls toed in, I can't see where you could go wrong. A narrow directivity should give you a better image where a wider, a better sense of space.

Spatiousness is another one of those touchy definitions. To me that's mostly reflection above the modal region (search for "Haas Effect" and the "psychoacoustics" post on this blog http://dtmblabber.blogspot.com/2010/12/psychoacoustics.html) and low level detail resolution if you're talking about what's actually contained in the recording. IOW if you want to hear the recording environment as picked up by the microphone and diluted or enhances through the process of production. Which means anything that interferes with that can have an impact. So from the loudspeaker standpoint, impulse, cabinet accelerometer CSD, now it even looks like capacitor vibration(so there may well be credence to more tweaks like God forbid, cables! Nothing has turned up there yet that I know of), and polar response will play into everything. In Dr. Toole's book there are studies that show wider dispersion adds to a sense of spaciousness. It seems rather intuitive. Also contralateral reflections play a role and subsequent elevations in IACC factor in. There are many things that can effect the low level resolution. I'd bet to some degree you can trace this all the way back to the source. This may be the most expensive, difficult and time consuming part to get to the "N"th degree. The room itself is also a large part of this.

Transient response is another one of those CSD, Wavelet, Impulse, polar. It's just rise time and decay. A CSD or Wavelet don't tell us a whole lot without the impulse response--ever really that I can think of.

Dynamics is another interesting topic that depends on wether you are talking physical or psychophysical. The best way to look at this graphically would go back to everything mentioned for transients, then also thermal capacity, power compression and efficiency.

I know, no publisher of specs is giving you this information, so knowing how to use it or think of it is of little use. In the end we are all left to guess. I wonder if informed guessing is better than uninformed. I bet anyone schooled in the issues at hand could do much better than I. Every time I read something new, I learn more and I'm betting any recording engineer, acoustician, transducer engineer, etc... could do a much better job than I just did.

Decay

2010-07-23T10:45:00.000-07:00

So The port stuffing made an obvious improvement in the response. Another view of the change is rate of decay. According to the psychoacoustic studies, the Cumulative Spectral Decay plot, or CSD, does not display the more audible delayed outputs. A Spectrogram would be more relevant. The top 2 graphs on the left are of the 90 degree off axis plots and the bottom 2 are on axis. You can see the cotton make the decay slightly faster but since the first reflection point of the graphs with cotton is sooner, there is a bit more noise late in the graph which contaminates the results. That can be reduced by graph manipulations, but left in for demonstration. Basically these don't tell us anything other than what the Frequency/Impulse Response tells us.

They just shed a different light on the situation.

Bass changes from port/enclosure stuffing

2010-07-22T22:30:00.000-07:00

OK, so Here's what happened on the low end of the spectrum coming from the B2031P speakers with the various modifications. The top graph is 2 inches from the speaker. When the ports are stuffed with cotton, you'll see the elevation of the low end. The next graph is 2 feet from the speaker. Here, the ported box actually has a bit more output.

More on the B2031P and diffraction

2010-07-22T19:35:00.000-07:00

I also wanted to test my diffraction theory with the ports next to the tweeter. So I got some cotton balls from the "stuff" drawer and filled the front of those ports with white cotton balls! Well, the results are something to see. The top polar and impulse graph are prior to port stuffing. The next are after the cotton balls were installed. Now I'd say it doesn't look like the woofer break up was any problem.

Inside the B2031P

2010-07-21T14:36:00.000-07:00

Here's a look at the Behringer monitor. I'll get some stuff to do modifications and see if it makes a measurable difference. I'm going to try and generate some plots using the new REW so we can get a better look at resonance issues. The crossover looks like a minimalist WGed 2-way with just a cap and resistor in the tweeter with a 2nd order low pass. The active version should do much better in this regard even though there's not a huge problem in the passive version. We see poly caps, an iron core inductor, and a wirewound resistor. The baffle is stout on the woofer half, but thinner on the tweeter. Enclosure damping is certainly less than I've ever used. That may be where we could best improve the design.

EDIT: The break up is benign on this cone! Response issues are caused by the enclosure/ports next to the tweeter.

A list of sites for the Home Theater/Audio enthusiast

2010-07-21T01:38:00.000-07:00

For the sake of saving money, it's probably a good idea to read some of the studies that have been done on the audibility of components and tweaks and how to set up your room to maximize your investments potential. If you have any interesting links, feel free to send them to me for posting. This is the short list of what I've found most insightful.

This site has the most DBTs in one site I know of:

http://home.provide.net/~djcarlst/abx_data.htm

Everything from capacitors and wire to amps, speakers, and cd players. Juicy stuff in the audio community.

Here's another dealing with amplifiers:

http://bruce.coppola.name/audio/Amp_Sound.pdf

SBT that agrees with previous findings:

http://www.matrixhifi.com/ENG_contenedor_ppec.htm

A video from Ethan Whiner that is very interesting:

http://www.youtube.com/watch?v=BYTlN6wjcvQ

Ethan's room advice:

http://www.youtube.com/watch?v=lbLVjHfHahg&feature=related

Dr. Olive's Blog is full of great information:

http://seanolive.blogspot.com/search?q=dishonesty+of+sighted+listening&updated-max=2009-04-09T15%3A24%3A00-07%3A00&max-results=20

Search around, it is one of the most useful sights on the web.

Everyone should really buy and read Dr. Toole's book: http://www.amazon.com/Sound-Reproduction-Acoustics-Psychoacoustics-Loudspeakers/dp/0240520092/ref=sr_1_1?ie=UTF8&s=books&qid=1239297857&sr=8-1

This will trump all the web searching you can do.

Surround Damping

2010-07-20T22:57:00.000-07:00

So then I treated the surrounds to damp the excess vibration. Looking at paper cones with accordion surrounds, the surrounds have a tacky, tar-like substance that stays sticky for a long time. I looked around and found Aleen's Stretchable-Flexible Fabric glue. It seems to retain it's tackiness for a few months now anyway--Ha ha. I also thought that since it was designed to remain flexible for a long, long time. Well check out the results.

Cone Stiffening

2010-07-20T22:39:00.000-07:00

So I looked into what was causing this poor cone behavior. As it turns out, it starts where the cone meets the surround--the rim resonance. That's where the cone starts to flip out and behave in erratically. So in order to raise the frequency when this starts to happen, the cone needs to be stronger. So I painted a few layers of Mod Podge Hard Coat to the cone and waited overnight to take the measurements shown to the left. You can see the rim resonance is higher and the impulse is more compact.

interesting references:
http://madspeaker.com/page2.html
http://voicecoilmagazine.com/media/klasco409.pdf
http://www.vibroacoustics.co.uk/audio/fsacbens.htm
http://www.nutshellhifi.com/library/speaker-design2.html
http://www.loudsoft.com/default.asp?site=FINECone.asp
http://www.klippel.de/measurements/sound-radiation-and-propagation.html

Cone Damping

2010-07-20T22:24:00.000-07:00

So, looking at the narrow directivity speaker below that just couldn't work out no matter what I tried electrically. Trying to make a notch filter too deep or too narrow will eventually backfire and it will become less deep. Remember the old adage, "you can't make a silk purse out of a sow's ear." Well I tried another route. Let's take a look at what I started with for experimentation. Shown are the average of measurements from the polar response, the impulse response, and the polar response.

The Listening Window

2010-07-20T18:41:00.000-07:00

The first three graphs are taken sequentially from 11.25 degrees to 33.75 degrees off axis.

Then the speaker was placed on its stand in its normal position.

The last 2 graphs were taken from the far ends of the listening couch and gated for distance and smoothed to hide the effects of comb filtering.

The spectral similarities are apparent.

Woofer Break Up Effects on Narrow Directivity Speakers

2010-07-20T13:36:00.000-07:00

To make a speaker with narrow directivity and a smooth transition in off axis response from woofer to tweeter, you generally have to crossover very near the woofer's break up region. The more narrow your waveguide's directivity, the nearer the woofer's break up will be to the crossover.

The top graph is polar response in 11.25 degree steps of the 10" Dayton waveguide, which the speaker is designed around, loaded by the Selenium D220t.

The initial "Rim Resonance" and how the woofer handles it is crucial in these types of designs. Finding the best woofer or even a good one is currently impossible using published graphs.

The second graph is of a woofer's polar response--not a particularly nice one. The next is after a make shift notch filter is added in line.

Then we have the respective responses after the low pass filter is added.

The final 2 graphs show what happens to those responses when the Waveguide is added. If the woofer's break up is as nasty as this one's, all is not lost as I'll demonstrate later.

You can see that a perfect off axis transition is impossible with this combination in its present state. The notch filter does improve it however.

Effects of Stuffing a Loudspeaker Enclosure

2010-07-20T13:18:00.000-07:00

Here's 2 graphs of woofers placed in a sealed enclosures. Can you guess which ones have fiberglass insulation stuffed inside and which do not?

Delayed Energy in the Behringer B2031P

2010-07-20T11:40:00.000-07:00

Here's the Impulse Response for the five B2031P studio monitors that I've measured. The graph shows how fast the output of the loudspeaker decays. Many factors effect this including resonance in the cone/surround and enclosure. A compact Impulse Response is then desired. You will also see at roughly 4 msecs the floor reflection. Notice the first graphs first reflection point is earlier than the others. It was measured under slightly different conditions. This notch, the first reflection, can aid you setting the gate on your time window when measuring a loudspeaker. You generally want to remove the first reflection from your measurement so it doesn't pollute your measured frequency response. The earlier your first reflection, the less resolution you'll be able to achieve.

Vertical Polar Response Behringer B2031P

2010-07-20T10:38:00.000-07:00

Another interesting aspect of loudspeaker output is the vertical polar response. Some experts will argue that it is unimportant, other will claim it's of utmost importance. I'll post two here so that your curiosity will be satisfied and see where the forward lobe is of this speaker. The top graph is on axis, 11.25 degrees and 22.5 degrees toward the woofer. The second graph is on axis to 45 degrees off axis toward the tweeter. According to the graphs, the optimal listening axis would be around the level of the tweeter.

Behringer B2031P Horizontal Polar Response

2010-07-20T10:02:00.000-07:00

I Just wanted to post these graphs for Home Theater, Home Audio, or Home Recording enthusiast. The Behringer B2031P is a cheap studio monitor that actually measures fairly well in the acoustic domain. Check it out. I've measured 5 of these and all are respectable.

To the left is the horizontal polar response graphs in 11.25 degree steps from the 0 axis to the 90 degree. None of the top five graphs have any smoothing applied and thus may look a little rough in the treble. So I added the 6th graph to the bottom of the page with 1/3 octave smoothing applied which is how many graphs for loudspeakers are shown. Of course you'll rarely see any more than one axis shown on a loudspeaker's published graph and as Dr. Earl Geddes would say, "there are reasons for that."

Look here for simple performance improvements.