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AES Paper on Emotional Impact of MP3 Compression

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Posted on December 9, 2016 at 09:55:52
Bromo33333
Audiophile

Posts: 3502
Location: Ipswich, MA
Joined: May 4, 2004
 AES published a paper called "The Effects of MP3 Compression on Perceived Emotional Characteristics in Musical Instruments"

TL;DR of it is: MP3 compression messes up the timbre of a lot of instruments, and may cause people to feel bad as they listen since the timbre and tone that's altered by the compression is part of what the study shows gives joy to a listener.

(You can get the paper by clicking through the summary article)

Interesting stuff!
====
"You are precisely as big as what you love and precisely as small as what you allow to annoy you." ~ R A Wilson

 

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Here's A More Direct Source, posted on December 9, 2016 at 10:33:05
Inmate51
 http://www.aes.org/e-lib/browse.cfm?elib=18523

Also, at the 2014 AES convention in Los Angeles, mastering engineer Bob Ludwig played several clips of audio compression. Wow. Once you hear it with "extreme" compression, and thereby know what to listen for, each successively less compression iteration still had audible artifacts.

:)

 

RE: Here's A More Direct Source, posted on December 9, 2016 at 12:31:18
Bromo33333
Audiophile

Posts: 3502
Location: Ipswich, MA
Joined: May 4, 2004
 Thanks for the link!!

I was an early adopter of MP3 for travel (I also used MiniDisc back in the day which was another similar issue) and could tell right away that the sound was a lot edgier and "dryer" sounding - paper like - than CD.

I also didn't listen to music as much when I compressed it. Though I was changing in my tastes. Then I got a SACD player and following that vinyl and ... wow!
====
"You are precisely as big as what you love and precisely as small as what you allow to annoy you." ~ R A Wilson

 

RE: Here's A More Direct Source, posted on December 9, 2016 at 13:28:58
Todd A.
 My first exposure to MP3s was at my son's house about 18 years ago or more. At the time, his stereo was completely comprised of equipment that I had owned - an Apt Holman preamp, Tandberg power amp, and Ohm Walsh 4 speakers, so I knew what this stuff sounded like. I remember thinking at the time how flat it sounded, and wondering how the hell he could stand to listen to it, although I didn't tell him that. That perception hasn't changed, BTW. I can understand how people might use MP3 for portable music, but not why it would be used in a home system. More and more, I'm leaning towards vinyl. Even with all of its faults, I still far prefer the sound over CDs or my music server (AIFF files), much less MP3. As always, YMMV.

 

RE: Here's A More Direct Source, posted on December 9, 2016 at 13:56:03
Bromo33333
Audiophile

Posts: 3502
Location: Ipswich, MA
Joined: May 4, 2004
 Totally agree. When I hooked my MP3 player to the stereo around 1999-2000 encoded at "near CD quality 128kbps" I was sad because it was so flat, no imaging, and was absolutely awful.

At higher bitrates it's OK for casual "radio" listening, but I still prefer FM radio to it.

Can't wait for hi rez streaming to be available someday (MQA and/or 24/96)!!
====
"You are precisely as big as what you love and precisely as small as what you allow to annoy you." ~ R A Wilson

 

RE: Here's A More Direct Source, posted on December 10, 2016 at 08:35:55
Inmate51
 With storage being dirt cheap these days (16 and 32 GB on a thin SDHC chip the size of a fingernail!), it's baffling why 24/96 isn't already the standard for hardware-based audio content.

:)

 

RE: Here's A More Direct Source, posted on December 10, 2016 at 09:49:58
JURB
Audiophile

Posts: 2056
Location: North Ohio
Joined: May 29, 2016
 "At higher bitrates it's OK for casual "radio" listening, but I still prefer FM radio to it."

Well then you must be listening to classical on stations with analog equipment and no bunch of digital crap embedded.

I find the rock stations way too compressed, and I mean dynamic range compression. The idea is to sound louder, the idea of which should have been abandoned on the FM band. On AM they did it so when you were in your car out of town and didn't know where the stations were you could hear them better.

A digital FM stereoplexer quantizes at 19 KHz. I think, compared to vinyl it sounds like shit.

For the next generation of digital they need to make the pieces very small. Forget about the 20 KHz limit, make it 40 KHz. Make the lower limit like 4 Hz. Then it is easier to design output filters with less phase shift and not so steep a rolloff. Then the number of bits/bytes. Regular CDs were 16 bit. Try about 48 bit and the real golden ears might just like it, or at east not be able to tell.

With today's technology a format like that would yield such short playing times on a DVDR that it could go the way of the betamax. Beta was better but the shorter playing times is why many went with VHS. Looked like shit but you could get three full length movies on one T-120 tape. Beta had a better picture and actually came out with (AFM) hifi sound first, but IIRC in beta 3 you only got 5 hours on a standard tape. That is not enough for two full length movies.

However, I think we are going into a time when we are going to see HDD recorders, put a 1 TB in those and you can afford to use a much higher format. Karaoke machines already do that, as do some DJs. No more busting your back carrying all the CDs, the other equipment is damn heavy enough. They might already have a higher format HDD recorder that I just don't know about yet.

Anyway, about FM, maybe it is partly where you live or that you have a truly superior tuner, but in general I have found MP3s to sound better than most FM. Listening to rock that is. The hillbilly station I think does a little bit better but that might partly be because the music is slightly different.

I do think I got lucky with the sound system in my music PC. Not so with my laptop, it has no dynamic range at all unless you go to the player (VLC) and turn its volume almost all the way down, not the PC volume. An external DAC I am sure would fix that. Maybe one of these days.

 

RE: Here's A More Direct Source, posted on December 10, 2016 at 19:22:20
bwaslo
Manufacturer

Posts: 245
Location: Portland, OR USA
Joined: September 10, 2006
 Actually, plain old analog FM stereo actually is based on sampled data -- the way it works (in the time domain) is the signal does a bit of Left channel, followed by a bit of Right channel, then Left again, then Right again, swapping between them 38,000 times a second. Then a 19kHz sinewave tone is played right on top of it, which tells the stereo decoder when to shift the signal toward the left or toward the right to separate the channels again. Hence, the sharp cutoff it has at 15kHz, to keep the 19kHz tone out of the speakers and to avoid images from the sampled signal. I know this because my High School science project was to make an FM stereo encoder (which managed to get used on a pirate station or two!).

And, yes, that's ANALOG FM, as started back in the 1960's! Digital FM is of course even more convoluted.
_

Make easy high performance diffusors:-->http://www.libinst.com/diffusers/Depot_Diffuser.html

Horn Design Spreadsheet:--> http://libinst.com/SynergyCalc/

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RE: Here's A More Direct Source, posted on December 11, 2016 at 04:26:34
JURB
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Posts: 2056
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 "And, yes, that's ANALOG FM, as started back in the 1960's! Digital FM is of course even more convoluted. "

Zzactly. I have been in arguments about whether that is actually quantizing or not. It is in one dimension but not the other. It is quantized in time but not amplitude, there are still an infinite number of amplitudes. But there is still that time thing.

Now when we quantize the amplitude a few other things happen. First of all if the source is completely noise free they have to add noise to it, it is called dither. It is just enough to trigger the ADCs and it sort of biases them like an amplifier. In fact on the old AAD CDs made from the old master tapes they had enough noise on them from the tape to not need dither and in some cases the quantization process subdued the noise in a very unobtrusive way, no Dolby, DNR or DBX or anything of the sort.

I actually was thinking of building a device that would do that outboard, and user calibratable. Set up a nice high bitrate, but then the lower bits close to digital zero are expanded just a bit to ignore most (but not all) of the hiss from the tape or whatever. Turned out that really the demand would never be enough and I would need too much help on it to really make money. To do it right you have to modify an ADC and really, the AAD CD did it well enough anyway.

But when they got to the DDD CD they had to add he noise, which you never hear. People, I mean alot of people REALLY could hear the difference, not just the golden ears. The dither, which is basically white noise actually biases the system just like an audio amplifier. you don't hear it, but without it, it just sucks. In fact even though they are extremely rare, there have been designs of amps with AC bias or even noise bias. Long abandoned designs, and I am not sure why. It almost seems more logical because it would tend to linearize the outputs' gain at lower levels better than DC bias. Or maybe not. Whatever. Since we do not hear of this now it obviously failed. Maybe it was too hard to get the thermal tracking right, I dunno.

But the fact remains that regular people could hear the difference, not just audiophiles, so they put the dither in.

 

RE: Here's A More Direct Source, posted on December 11, 2016 at 05:38:15
Dave_K
 There is no sampling in FM radio.

When broadcasting a stereo (L and R) signal, first the signals are band limited to 15 KHz. Then the modulator forms sum (L+R) and difference (L-R) signals. The L+R signal occupies the lower 15 KHz of the channel. Above that is the pilot tone at 19 KHz. The pilot tone is used to generate a 38 KHz subcarrier, and the L-R signal is amplitude modulated onto the subcarrier, which is suppressed for transmission. No sampling, just AM.

0-15 KHz - Sum (L+R) signal, baseband
19 KHz - pilot tone
23-53 KHz - Difference (L-R) signal, amplitude modulated on 38 KHz carrier

Most FM stations add other services in the remaining channel space between 53 KHz and 100 KHz such as radio for the blind, muzak, RDBS. And then the whole thing is frequency modulated onto the channel's carrier.

This approach allowed for simple & cheap monaural FM radio reception. All you have to do is the one FM demodulation and then filter out anything above 15 KHz.

For stereo FM, you also have to lock on the 19 KHz pilot tone, use it to generate the 38 KHz subcarrier (since the subcarrier was suppressed), demodulate the AM signal, and do another sum & difference to get the left and right signals back.

 

RE: Here's A More Direct Source, posted on December 11, 2016 at 10:19:34
bwaslo
Manufacturer

Posts: 245
Location: Portland, OR USA
Joined: September 10, 2006
 Dave_K, you're not wrong, and that's the FREQUENCY DOMAIN explanation. In the time domain (where the actual electronic process is, or at least was, done in practice) it is as I described. Filter out everything above 15kHz, you're left with L+R, no matter what domain you describe it in.
_

Make easy high performance diffusors:-->http://www.libinst.com/diffusers/Depot_Diffuser.html

Horn Design Spreadsheet:--> http://libinst.com/SynergyCalc/

SmallSyns:-->http://www.diyaudio.com/forums/multi-way/292379-s

 

RE: Here's A More Direct Source, posted on December 11, 2016 at 15:22:28
JURB
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Posts: 2056
Location: North Ohio
Joined: May 29, 2016
 "There is no sampling in FM radio."

Technically debatable. I know how it works. I guess what people say is that if it isn't quantized it isn't sampled. I call that semantics because even the analog stereoplexers chopped up the signal into pieces 1/38,000 th of a second long.

The technique of the modulation BTW used to be called double sideband suppressed carrier.

It still has its limitations. With more modern techniques such as pilot cancelling rather than a notch filter they could do 17 KHz, maybe a tad more, but it simply is not transmitted.

And with all the digital junk they transmit with it they probably simply can't. In fact some older tuners have a problem with all that digital crap and it comes out as noise even on the strongest of signals with more than ample limiting. There are people out there who modify older FM tuners to fix that. I believe, IIRC, the filter goes in the IF strip. I'm thinking but not sure that if you have a wide/narrow IF bandwidth switch, using narrow should eliminate the noise but then you get the increased distortion. Like, take your pick.

In the old days the other subcarrier was SCA, for store cast allocation and FM tuners started having filters for that. It ran at 67 KHz and I think it was frequency modulated. But not by much, the quality was not all that great because it was used for things like Muzak and maybe voice.

Now of course they got more in there. Your car radio reads out the name of the song that is playing, that is something new. And of course that is a source of noise for older tuners. And to tell you what, the newer tuners I have heard do not sound as good. But then many of them are on a circuit board about the size of a USB thumbdrive. theyy run on a new process, I forgot what it is called but somehow one chip does iit all. there is no IF strip, some don't even have anything that looks like a detector alignment. And the same 18 pin chip also does AM.

Now compare that to an old Revox.

I maintain my position that in most cases FM is good for the car, but not for serious listening. If you get a station without all the digital garbage on it maybe, but where do you find that ? And still, the 15 KHz limitation is still there. Most of you wouldn't put up with that in a cassette deck. At least not for hifi use.

" Then the modulator forms sum (L+R) and difference (L-R) signals."

A comparator did that. In a digital stereoplexer it simply switches between the channels. Much cheaper, no phase problems generating the pilot because a simple flip flop (astable mulitvibrator) does that. The system did make it a bit simpler, but better ? I doubt it. Just like the old redbook CDs speced out better than vinyl, sometimes the vinyl just sounds better.

 

RE: Here's A More Direct Source, posted on December 11, 2016 at 15:32:57
JURB
Audiophile

Posts: 2056
Location: North Ohio
Joined: May 29, 2016
 Actually this talk of FM is getting out of the domain of thread drift and into a hijack, which I don't like. the similarity is there though, that like 128K MP3s are good for the car or portable use, while the higher bitrate ones should be used on your good system. And I consider anything less than 128 unlistenable.

But how much of this is attributable to dynamic range compression rather than the data compression ? In analog recording, say on tape, you could go beyond 0 dB. It would create some distortion. On tape there would be some third harmonic, on vinyl it risks fast groove wear or even overcutting, which makes the master useless.

But with digital it has to be hard limited because you cannot exceed a certain, absolutely defined number. It has to be clipped like an overdriven amp. If not, in most cases it will make a terrible noise if over"modulated".

 

RE: Here's A More Direct Source, posted on December 12, 2016 at 06:11:13
Dave_K
 What I objected to was calling it sampling.

You can use a switching modulator to perform amplitude modulation, but that's not the same thing as sampling.

Mathematically, a switching modulator can be shown to produce the desired AM wave plus the square wave and a bunch of harmonics. The square wave and the harmonics can be filtered out. When you do that, you get the proper, continuous-time amplitude modulated signal. It is not a discrete-time signal, it is not sampled.

 

Most CDs are super compressed anyway so what's the difference? , posted on December 12, 2016 at 06:51:40
geoffkait
 Not to mention vinyl, SACDs and downloads.

 

RE: Here's A More Direct Source, posted on December 12, 2016 at 11:49:23
Bromo33333
Audiophile

Posts: 3502
Location: Ipswich, MA
Joined: May 4, 2004
 "Dave_K, you're not wrong, and that's the FREQUENCY DOMAIN explanation. In the time domain (where the actual electronic process is, or at least was, done in practice) it is as I described. Filter out everything above 15kHz, you're left with L+R, no matter what domain you describe it in."

Once you put the FM signal in the discriminator, the decoding of the L+R and the L-R happens simultaneously (Also the recovery of the RDS info). It doesn't switch back and forth in a standard analog demodulation the way you describe.

====
"You are precisely as big as what you love and precisely as small as what you allow to annoy you." ~ R A Wilson

 

RE: Here's A More Direct Source, posted on December 12, 2016 at 12:17:41
bwaslo
Manufacturer

Posts: 245
Location: Portland, OR USA
Joined: September 10, 2006
 Sorry, it does. I've designed and built such encoders long before whatever internet article you maybe read existed! It is done with an analog multiplexer (switch), a 38kHz square wave source, and a div/2 ff and some phase shift networks to get the synchronized 19kHz pilot tone. If you can find a block diagram of the old National Semiconductor PLL FM stereo decoder chips you'll see the decoding done in the (reversed) same way.
_

Make easy high performance diffusors:-->http://www.libinst.com/diffusers/Depot_Diffuser.html

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RE: Here's A More Direct Source, posted on December 12, 2016 at 14:02:31
Bromo33333
Audiophile

Posts: 3502
Location: Ipswich, MA
Joined: May 4, 2004
 Be careful.

I have worked in RF and Microwave in TV, Radio, and Wireless for over 20 (Almost 30!) years. Mostly on infrastructure and transmitters.

I just posted up the block diagram for clarity since I did not know your background.

Peace. I'll assume you are correct as I have not got into the guts of chips, and have mostly been on the transmission side of it.


====
"You are precisely as big as what you love and precisely as small as what you allow to annoy you." ~ R A Wilson

 

RE: Here's A More Direct Source, posted on December 12, 2016 at 14:17:28
bwaslo
Manufacturer

Posts: 245
Location: Portland, OR USA
Joined: September 10, 2006
 Not to do a peeing contest, but was in RF design (primarily for military and govt agencies) 26years as an Engineer, about 6 before that as a tech. 7 ears of electronic repair before that. Not god's gift to electronics but I do know my way around a signal chain.

Here's a link to an (old) article from when this stuff was new, see figure 2 which shows the waveform (without pilot) when one channel is an audio sinewave and the other is silent.
_

Make easy high performance diffusors:-->http://www.libinst.com/diffusers/Depot_Diffuser.html

Horn Design Spreadsheet:--> http://libinst.com/SynergyCalc/

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RE: AES Paper on Emotional Impact of MP3 Compression, posted on December 12, 2016 at 14:43:01
fantja
Audiophile

Posts: 15524
Location: Alabama
Joined: September 11, 2010
 Thanks! for sharing- Bromo.

no doubt, that excess anything is unhealthy, to state the least.

 

RE: Here's A More Direct Source, posted on December 12, 2016 at 14:56:33
Bromo33333
Audiophile

Posts: 3502
Location: Ipswich, MA
Joined: May 4, 2004
 It reminds me a little of my early days working on PA's for TV transmitters - the modulation for color TV was a very complicated one with AM, FM and all of that mixed up at once.

I got into it when they were working on the HDTV standards, and back when they were thinking of doing it analog. Quickly they started using QAM signals in Europe (where I was working at the time) when the US went their own way with 8VSB (developed by Zenith before being consumed by, I think, LG or Samsung - can't remember).

Good times.

Glad to see an old RF guy like myself here. Too often people do "engineering by google" so you never know where the other fellow is standing.

Didn't know about the commutating time domain sampling method. SOunds like a clever way to re-use circuity or save some costs. Wouldn't occur to me to do it that way. But as a high power RF PA guy, I am all about KISS - and reliability. Since when you are making a 30kW RF system, you really don't want anything to fail!


====
"You are precisely as big as what you love and precisely as small as what you allow to annoy you." ~ R A Wilson

 

RE: Here's A More Direct Source, posted on December 12, 2016 at 15:01:48
Bromo33333
Audiophile

Posts: 3502
Location: Ipswich, MA
Joined: May 4, 2004
 "Well then you must be listening to classical on stations with analog equipment and no bunch of digital crap embedded. "

There are a couple of radio stations where I live (Rochester, NY) that have pretty good sound:

90.5 (Mostly alternative, a community college station)
88.5 (Uni radio with eclectic programming)
91.5 (Our NPR station that is classical in the afternoon)

I do like some of the HD stations in the car. There is one that was on for a couple years that was nonstop commercial free announcer free Blues. Went off the air then - though it was compressed (though not obviously so) it was terrific content.

The older I get, it tends to be more about content than sound.

Siruis/XM is atrocious for sound quality where I am. When we were on our family vacation driving through Ohio, the sound quality was noticeably better (though still not good). Weird. Wonder if it is terrestrial transmitters there, that we don't have in Western NY because of our proximity to Canada? No idea.

====
"You are precisely as big as what you love and precisely as small as what you allow to annoy you." ~ R A Wilson

 

RE: Here's A More Direct Source, posted on December 13, 2016 at 08:11:37
Dave_K
 A bit of background info...

One method of implementing amplitude modulation is to multiply the input signal by a square wave at the carrier frequency, which can be implemented using diodes as switches. Because a square wave is just the superposition of an infinite series of cosines at the fundamental frequency of the square wave and its harmonics, the result of multiplying the square wave by the input signal is an infinite series of amplitude modulated signals. The fundamental and all of its harmonics are amplitude modulated by the input signal. Then you use a bandpass filter to select the one that matches the desired carrier frequency, usually the fundamental or third harmonic. This "switching" method of amplitude modulation is cheap and easy to implement in low level solid state circuits and is more linear than a square law modulator. So it's well suited for implementation on a chip and thus common.

When the FCC was soliciting proposals for FM stereo, two of the companies (sorry, I don't remember which) proposed transmitting sum (L+R) and difference (L-R) signals, with the sum transmitted in the 15 KHz baseband just like FM mono and the difference signal AM modulated into the channel space above that. If I remember correctly, one of the proposed implementations involved explicitly forming the difference signal and then using amplitude modulation, and the other proposal involved using a square wave to switch between the L signal and an inverted R signal. These two approaches are mathematically equivalent.

By the way, my background is originally RF as well (and computational electromagnetics). But I'm far from that now.

 

RE: Here's A More Direct Source, posted on December 13, 2016 at 08:36:53
Dave_K
 At first glance it seems like a semantic argument, but to me there is a key difference between square wave modulation and sampling. The former is continuous time process and the latter is discrete time. I suppose I've become a bit sensitive over this because so many people keep calling Class D "digital".

One reason why it's important is that discrete time information theory principles such as the Nyquist limit and Shannon-Hartley law apply to sampling but not square wave modulation. For example, you can theoretically use a switching modulator to amplitude modulate a carrier whose frequency is less than twice the maximum frequency of the input signal. Supposing the input signal bandwidth is 15 KHz, you can produce an AM signal whose carrier is 10 KHz. The bottom 5 KHz of the lower sideband will wrap onto itself, so the lower sideband needs to be suppressed, but it is possible to demodulate the AM signal and fully reconstruct the original input signal. If the input signal was sampled instead at 10 KHz, this would not be possible. Per Nyquist, the sampling frequency would have to be >= 30 KHz.

 

RE: Here's A More Direct Source, posted on December 13, 2016 at 18:23:05
JURB
Audiophile

Posts: 2056
Location: North Ohio
Joined: May 29, 2016
 "the modulation for color TV was a very complicated one with AM, FM and all of that mixed up at once."

Not exactly, unless you are including the sound carrier.

The chroma signal is like the FM L-R channel in that it is carrier suppressed. Also, the detectors in an FM tuner for L-R are actually synchronous detectors. The detectors in the NTSC system had two synchronous detectors, one running 90 degrees out of phase thus yielding two discrete signals.

As with FM, it is not that different on the transmitting end. But it is a bit more complicated than that. The I signal had a bit more bandwidth but only a few TVs could use it. The RCA CTC111 I believe had what was called "wide I" demodulation but then they used a crap CRT. It was the early days of the inline gun CRTs and the picture would have been much better on a delta gun CRT. The pitch of those early inline gun jobs was so poor they shouldn't have bothered. Wide I demodulation also required another delay line. That chassis also had one of the earliest digital COMB filters on the market.

The I and Q signals were the difference signals for red and blue, I forget which is which. The green was derived from a matrix circuit, when the red and/or blue went down, it made the green go up and vice versa.

There is some complex math involved as they found that the green carried the most detail so it was fairly predominant in the monochrome signal, which underlies those difference signals.

But there was no FM in there, that was phase modulation and was not even intentional because when you mix two waveforms that are 90 degrees out of phase you get different phases which range from the instantaneous to the quadrature.

More useless knowledge, I have tons of it.

Getting back to the OP here, these ultra good digital formats will happen when not only the media is invented, but when the sources are available. I listen to some music from the 1950s and find a 128K MP3 to be quite adequate, though even on that material I have sometimes noticed a difference with a higher bitrate file. When I downloaded I would get several copies of everything and delete the inferior ones. It is actually somewhat surprising how good some of those old recordings are, considering the times. Others not so good.

There is now the holographic disk. It makes blu-ray obsolete. It has so much capacity nobody can use it. Perhaps that is the medium of choice. The ultra high quality formats would take downloading back to the 20th century, like it was on dialup. Imagine a five minute song being 800 MB.

Here's a wiki on those disks :

https://en.wikipedia.org/wiki/Holographic_Versatile_Disc

If you wanted to send someone a song, depending on your internet speed you might be better off just burning one of those and snail mailing it. However the technology has not been perfected. They've apparently done it but it is not quite market ready I guess. But they're talking capacities up in the terabytes.

They get that ready, THEN you can throw away the vinyl.

 

Music Is Music, and Motor Oil Is Motor Oil, posted on December 14, 2016 at 09:37:22
geezerrocket
 I guess if your deaf or have shitty gear.

When people (educated people) attempt to tell me that there is no audible difference, it reminds me of that old Havoline commercial.......Motor oil is motor oil.



 

The 'religion' of motor oil, oil change interval, and brand., posted on December 14, 2016 at 12:09:58
Smelly_Socks
 Go to ANY 'car' forum. Post a topic with oil change, or oil change interval, or oil brand.. No matter how many threads are already there about it. A hundred more will be added to the new posting.
Folks expressing their heartfelt ideas on 'oil'
Some say change it every 1,000 miles, every 3,000 miles, Never, until the oil testing place says your sample sucks.
Dino oil, synthetic oil, blended oil. Low NOACK!! (claiming all else is worthless info)
Specialty brands, That some synthetic (heresy!) is not even synthetic.

I do not know of any audiophile topic which can raise the roof like an oil thread. Except the somewhat banned subject of 'double blind testing' LOL

 

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