Sonic's System

Sonic,

Please post pictures of your subwoofer adjustement. I am having difficulty visualizing your description.

I have been playing with my MGD sub trying to attain what you have described recently after completely setting up my system on a different wall in my listening room. The sound is just starting to become enjoyable again, and I still need to spread my speakers further apart. My sub sits atop a diy birch plywood platform raised above the floor about 6 inches using steel rods and brass MGD MTDs feet with the sub suspended by harmonic feet with MW placed under the feet.

Quote :

“At first thought it would appear that the purpose of a volume control is solely to enable the listener to have his music loud or soft according to his whim. Whilst it, of course, fills this requirement, it has a further purpose – that of adjusting the intensity of sound so that it is correctly related to the perspective as recorded or broadcast. If a voice is picked up close to a microphone in a very absorbent studio, then on reproduction that voice will take up a position at the centre of, and in the plane of the loudspeaker. For natural sound, therefore, the loudspeaker should radiate similar power to that of the original voice. If on the other hand the voice is picked up some way from the microphone in a more live studio, then the voice on reproduction will take up a position through the loudspeaker and a considerable distance behind it. It is clear the power required for the loudspeaker for natural sound is now very much less than in the first case. The position or perspective of the reproduced sound is fixed at the studio end and there is little that can be done at the listening end to alter it.

It follows that the volume setting for natural sound is to a large extent fixed at the studio end. Studio monitoring is usually carried out at a reasonable level and the whole aim, is to produce listening as from a favourable seat in the Concert Hall. Adjusting the volume control to a level to give this correct acoustic perspective will produce the most natural reproduction. The level is usually such that it is quite possible to speak to a person sitting next to the listener without raising the voice or turning down the sound level – as indeed this is possible in the Concert Hall. Raising the level to “bring the orchestra into the room” or turning it down to a low background will both distort the perspective, although this may have to tolerated on certain occasions. It should be pointed out that no amount of tone control or loudness control can affect the perspective, although these effect can be used to produce a new sound which although quite unlike the original is sometimes found acceptable.

Popular music is often recorded or transmitted with close microphone technique and would therefore tend to require reproduction at higher levels. It is in fact generally monitored at a higher level. There are number of other factors which have a strong bearing on optimum listening levels but it is outside the scope of this handbook to deal with these adequately. It will be realised that the volume control setting should receive careful attention and it can be emphasised that much listening is spoilt due to insufficient care on this point.”

Hi Sonic

You know sometimes I don't know what to think when people start talking about "how it is". I think people can relate to only as far as they have gone and because they have not stepped beyond they think this is as far as it is.

When I learned to engineer and later worked with engineers the one thing that I saw as a constant was that there was no constant. For example a recording engineer is a different animal than a mastering engineer. If you have any music that is recorded after the late 60s early 70s chances are you had 2 different sets of ears and 2 different studios making decisions on a recording. After the digital age got here the gap between the 2 became even bigger along with their definitions. I can tell what kind of system and what type of room Peter listened in when he made these comments and must say that "I think" he is limiting the sound and what we are able to do with it. As an example there is nothing that says music has to come directly out of the speaker in a recording. In fact I have mostly witnessed far left or right pan being outside of the speaker. With this being the case the plane of listening can be where ever you put it (front to back) as this is a function of the room, sound pressure and mass and not the speakers solely. Usually engineers have sound running right into the speakers because they are in listening areas not setup like good listening rooms. Playback rooms give you a third (and very different) perspective than both the recording engineer's room and the mastering engineer's room. So much so that most engineers I have ever know take the (in process) recordings to a third party trusted room and system to listen to, then come back and make guess judgements in the mastering room till they get it to sound ok in their playback room. If you ever take an on-line tour or actual tour of studios and the engineering process from start to finish it is a little scary the amount of (what we would call) guess work that goes into the recording. What may be placed and spaced in one part of a recording may be different when taken to the next stage (recording room to master room etc).

I absolutely, without a doubt, know that for a fact a well tuned room is revealing more of the music than a playback room for a studio. Here's example: You remember Celine Dion's Titanic sound track right? Well this was getting remastered in one of our studios in Nashville in 2001. When we played it in the studio and in the studio's playback room it sounded pretty good but the usual in the monitor sound was happening and it bugged me so I told the guy to burn a copy and bring it up to my place for a listen. Blew his mind!!!!! Not only did the music leave the speakers but we heard much more of what was on the recording that you couldn't even hear in the studio. This was not distortion, this was not make believe, this was the real thing. I probably did this same exercise about 100 times while I was in Nashville with the same results. The Oak Ridge Boys and Alabama among others actually moved into my studio/playback facility for a few months to make listening decisions and every time we went from the engineering room to the playback room the same thing happened, the sound in the playback room left the speakers and the music became 3D.

Taking this a step further, when we took the same recordings to other listening rooms some of the time it was in the speakers and other times not. It all depended on how well the system played back the content. The nice thing about doing this was being there while the music was being played and through the rest of the chain. Nothing like being there and learning.


mastering studio Nashville

Here is another part to look at. When the recording is in the first studio (recording studio) compression has not been applied (at least not as much as in mastering). Chances of the music not going into the speakers is much greater during this time. It's kinda like when we tune and open things up. The more things are opened and not compressed the more they fill in the room and not collapse into the speakers.

Personally I have not found many recordings I can not get out of the speakers or have control over front to back planes. Actually in my listening if something has a plane it is usually needing more freedom and tuning to get the 3D back into the music.

Quote :

So the issue of soundstaging is complex. Do the Absolute Sound people get the wide stage they claim? Do Von Scheweikert loudspeaker owner achieve this? I don’t think they are telling fables. Their systems are high quality and if set up right can give some big sound especially when played loud (and some reviewers state their playback levels – levels that Sonic finds appalling -- 103dB peaks at the listening seat!)

I don't think that the fables are being told to be dishonest but they are still fables when compared to real soundstaging. I have only heard 2 reviewers systems truly sound "Stage-Worthy" in all the time I've been touring. And that was after much tuning. I wouldn't want to start wars but staging among reviewers and manufacturers is a stretched science. The extreme staging Tunees talk about and the audiophile staging we often hear about in the story books (high end mags) are different events. Audiophile soundstaging is kinda fun until you are opened up to the real deal then you hear a world that is much different and sometimes makes it hard to go back to the 2D world. For me it's like this. Seeing a print of a great piece of art, seeing the real art (a big step) and seeing the real thing. Audiophile audio is like seeing the real art, It's beautiful with many colors and shapes. Sometimes you even fall inside of the painting and linger for hours staring at the movement and meaning. It's a fun ride and one that you can get lost in. But, when you hear the music wrap around you and disappear into a world of no boundaries it's like stepping from the art gallery on to the real location (you are there). You can walk around and explore any part of the landscape you wish. You can see and hear each part of every instrument and how it stands on it's know or works together with the other parts of the piece. For myself it becomes much more than listening. It becomes a moment that I am involved with.

I have never listened with a reviewer or manufacturer and had them tweak a system and say "here it is" and it take me away. It's always been the other way around so this makes it very difficult for me to make judgement calls. At the same time I have heard great listeners tweak and sit me down and I've been taking away into their land and understood where they were going with the setup. For example: One time I got home from tour and one of my guys had me come to their place to listen. I was a little hesitant because I didn't want to have to tell him that it didn't sound so good. I had heard their variation of prototype speakers before and it was not so hot. When I got to his place I could not believe my ears. Amused To Death sounded awesome and I sat through the whole thing thinking we have the next Michael I can move on and do what old lions do. I was so proud of him and he was rightfully cocky about his achievement. Then we set the speaker up in one of our showrooms and it sounded terrible. We must have taken that speaker around to 10 different places and at each it sounded like garbage. He was so angry he quite RoomTune and went back to what ever it was he did before.

I just do not believe in fixed sound. The chances that someone did it right and sent the sound to our doorstep is not a reality and as fun as the hobby might be the day will come where we all need to make a recording sound better unless we want to listen to our 10 fav CDs and pretend the rest sound as good. I can't do it. I have so much fun uncover the truth that it is so hard for me to listen on a long term basis to something that is so shy of it's potential.

Do the other guys even know what we are talking about? I don't think so.

Hi Garp

Here's a diagram of what Sonic built. It is wedged between the bottom surface of the Janis W-1 subwoofer and the floor,









Interesting to see that you are using Harmonic Feet under your speakers instead of MTDs. I've sound them Harmonic Feet to be rather mixed devices and only use them if I got no alternative. I get better results with MTDs or one of Michael's springs. The handwound crazy looking ones sound the best even if they can't take much weight and gradually collapse under the weight of anything slightly substantial. Consider using MTDs under your sub too and spike it down directly into the surgace of the platform unless that wood is precioous for any reason. For an amp, I would use an MW piece between the metal casing and the top of the cone or spring as an interface layer and let the point or bottom of the support devioce couple directly to the surface of the shelf/platform.

Sonic

Sonic,


Thanks for the update. I must be getting old after reading your reply, because I went back to replace the harmonic feet, and discovered that the sub was supported by MTDs. Yes, I have had mixed results using Harmonic Feet as well. I have plenty of the older springs as well, and use them throughout my system components.

Since I have plenty of MW, I will experiment with placements, like you describe, on the platform support rods.

Hi Michael

Good to hear from you on how recording engineers are largely/partly flying blind. I have been wondering about Tunee exeperince where instruments and backing vocals can appear at the shoulders, how noises can appear in random places and so on when the sound I have heard in a few professional studios is anything but surround, 3D and dimensional. Often audiophile may be appalled how flat fatiguing sounds from a big pair of Westlakes nearfield can be. Your observation explains what you (and Ivor Tiefenbrun too) said that we are only extracting 10% of the sound on a disk/LP.

I found this incredulous but after your account of recording studios, this is possible and there is so much even the recording and mastering engineers are not even hearing.

Hi Garp

My grounding gadget is useful only because the Janis W-1 sits on Michael's Gen 1 cable grounds. There is no real grounding this way so the cone/MW block and rod device contributes the ground. If you have an MGD subwoofer and MTDs, then maybe let the subwoofer sit directly on the MTDs and let them spike into the platform and see if that works.

Otherwise you cound make a ground device like Sonic did and try it under the platform. If your sub is using MTDs you should be getting pretty adequate grounding.

Sonic

Hi Sonic

The scary part of things for me in recordings is when the recording goes from the recording engineer to the mastering engineer. If the mastering engineer has no prior reference to the dynamics inside of the studio he (or she) has no real take on what they are deleting when they compress and EQ the sound for burning or pressing. Compression and limiting is an art totally different from the recording (capturing) stage. There's up compression and down compression, there's hard knees and soft knees (the angle or curve which the compression is made).

The easy way to describe compression is picture water that has to go through a tube as efficiently as possible. If too big of a tube the water will come splashing through the tube and have no pressure on the other end, if too small the water builds up on the front end and takes too much time to get all the water from where it is building up to where it is going. The thought is to get all the water through quickly with out wasted space.

Why not let all the water (sound) through naturally?

The mastering world is based on dynamic range. Our components, speakers and parts are built with limited boundaries as compared to real life dynamic sounds. If you ran the full dynamic range of a big sounding recording through panels we would be wearing mylar the first time a big kettle drum hit and I would be wearing a cone and spider. We would all be listening to drivers instead of delicate speakers. The problem with this of course is the loss of low dynamic detail and the fact that not everyone would have room for 10' tall and wide horns in their rooms. Even if you had the big speakers you would have to deal with the distortion of cones not being able to handle the true dynamic range.

Lets go back to vinyl. If we would have put the full dynamic range on vinyl it would have been very difficult to find a needle that could grab the deepest valleys and the tallest mountains on the surface, plus a typical big sounding LP would have no longer been a LP but more like 5 to 7 minutes of music per side. Fast forward: When the digital age hit remember when they were labeling speakers digital ready? Well this was because they had not yet developed the modern compression system gauge and the industry was afraid they were going to be blowing up speakers. Many recordings still have warnings on them for dynamic range. These recordings run dangerously close to the limits of most speakers and component values. Still even these recordings are not close to real life dynamic range. For those who have my speakers you are listening to speakers with almost nothing in the way of limitations on the cone and tweeter movements. The dynamic range of these are huge compared to speakers with more parts. Story time. Well when I was recording at "The Sound Lounge" in Detroit we were doing an acoustical guitar piece for "Jaheim". When we played back even this simple guitar rif without compression it threw my cones all over the place. It sounded fantastic but my subs and 60s would not have held up long under real dynamic uncompressed music. When we put on the studio monitors the magic of the music was gone but we could turn it up as much (almost) as the guys wanted (it was loud). Why the recording engineers play things so loud is because many of them are either musicians or are playing for musicians that are coming directly in from the studio to the control room to listen. This is why there are 3 stages of speakers that get used on most recordings. Studio monitors, mastering monitors and playback speakers, the 3 levels of dynamics.

Studio monitors: designed to playback the full dynamic range straight from the studio. Pluses, very dynamic and loud, minuses lack of subtle detail.

Mastering monitors: designed to handle the power from the original studio on short burst but more relaxed. Pluses, good for finding and gauging the compression cut offs (high and low), minuses, still not as musical as playback speakers (a little sterile sounding)

Playback speakers: designed to reproduce music in the home environment. Pluses, more revealing and soft sounding, biggest soundstage of the 3, less fatigue, minuses not able to handle the dynamic range of non-compressed music.

There are some crossovers to this rule but when you listen closely and for a long time you feel compelled to stay within the proper categories for the most part. I love it when a recording studio uses my speakers but at the same time I know that it's only a matter of time before someone is going to blow a woofer or tweeter.

break time

Well shoot, it's already in the later PM so I should just keep going on this and get caught up on my listening later.

Not only is Ivor correct in saying this, but we are also listening to the music being bent on both ends of the music's dynamic range as well as being filled in the middle. Don't get depressed it's not all bad but it is different than the real thing. In the mastering stage we take the full dynamic range lets say 1 to 10 is full range and squeeze it into 2 to 8, maybe even 3 to 7 if we're being linear about this.

Here's a very basic picture, if your not a studio person this might not make any sense, if you are this may seem too simple and maybe even incorrect, but lets give it a try.


"Compression has to be one of the most confusing and elusive effects out there. It's easy to know you need it just by watching your meters, but what does each knob and button really do and how does it all work? This article should answer those questions, and will touch on the "whens" and "whys" of compression.

Let me first start by explaining the basics of dynamic ranges in recording. First, we have the noise floor. This is the lowest level, where tape hiss and electrical hum reside at. Next we have the nominal level, which is the level that is best for recording your incoming signal in order to minimize distortion and overcome the noise floor. The distance between the noise floor and the nominal level is called the signal-to-noise ratio. Next is the maximum level, which is where distortion occurs at when your incoming level reaches it. This is the highest level in the total dynamic range. Distortion is something that you definitely want to avoid unless you are versed in the skills of good tape saturation (sometimes engineers will try to slightly distort the signal by pushing it over the maximum level because this will give a stronger sound to an originally weak one. However, in digital recording, any distortion due to overpeaking is distasteful.). Now the difference between the nominal level and the maximum level is referred to as your headroom. This is your safety zone, and this is needed to account for some stray peaks here and there without hitting the maximum level. And to wrap this up...the whole thing, from noise floor to the maximum level is called the dynamic range.

Okay, lets cover how compressors work. Imagine a recording scenario where you are starting to record some tracks on your multitrack recorder. You have set a good recording level for your instrument which is at or near the nominal level, but you notice that the incoming signal occasionally jumps up into the red. That is typically going to be the nature of either the instrument, your playing, or both. So, you don't want those distortions going to tape and ruining an otherwise fine performance. This is where the compressor comes in handy.

The Alesis Company a while back issued a brochure on how compressors work, and it gives the analogy of the compressor acting like your own dedicated engineer for that one track. It will monitor all the incoming signals and then act like it is pulling down the fader the instant that high volume peak occurs. In a more technical explanation, what the compressor is actually doing is reading the incoming signals, and then according to the compression ratio that you set, it knocks the hot signal down by that ratio. This allows you to keep the level down to one that is manageable and recordable, without the wild peaks.

Compression ratio you ask? Well, let me explain the 5 main controls. First, we have the threshold. Think of this as the decibel level where the compression will start working. I visualize the threshold as a line that is lowered onto the offending noise peak, and the lower the threshold level, the more the incoming signal will be compressed. This is because more of the noise peak is now ABOVE the threshold level so there is more to squash.

Next we have the ratio settings. This knob has different ratios on it like 2:1, 3:1, 4:1, and usually any combination in between. Okay, assume you set your ratio to 3:1. What this does is that for every 3dB your incoming signal goes over your threshold line, the compressor will allow only 1dB to pass. The level still goes over the threshold, but assuming that you set the threshold low enough and used an appropriate ratio, the peak will never reach the maximum level and distort. This is also due to the amount of headroom you have. Typically, I set my ratio first, and then use the threshold knob to find the point that the incoming levels are being compressed. This is done while watching the meters on the mixer, and you will see the offending peaks all falling within the same lower range which is nearer to the nominal level. Keep in mind that if your incoming signal is lower than the threshold level, (or the threshold is set too high), then none of the signal will be affected.

Next we have the attack parameter. Think of this as how fast the compressor acts on the peaks once they pass the threshold. Some instruments have a really quick attack sound as soon as they are played, and most peaks arise from this attack. Therefore, on instruments like bass and kick drums, you would want to set a quick attack.

The release parameter works by setting how fast the compressor lets go of the incoming signal once it has gone below the threshold level (where the signal doesn't need to be compressed anymore.) You could set the release to fast and cut off a signal quickly, or set it to slow which results in a longer sustain. Many guitar players use this to sustain their notes.

The last main function is the output setting. Typically, when you lower the threshold and the compressor kicks in to squash the signal, your nominal level will be lowered slightly depending on the amount of compression being used. You can then use the output knob to bring the input level back up to nominal. Be careful though, because by raising your signal back to the nominal level, you are also increasing the noise floor due to added noise from within the compressor itself. You may want to increase the trim on your channel or master fader so more pure signal is getting to the compressor instead. Everytime you patch your signal through another pathway (such as another processor), you are also adding the inherent noise of that pathway.

There is one other feature that not all compressors have, and this is the option to compress with "hard knee" or "soft knee". Hard knee is where the signal is compressed the moment it goes above the threshold to the full extent of the ratio that is set. Soft knee is where the compression is applied more softly so as not to sound so abrupt. This is similar to using the attack knob, and I use hard knee compression on signals like bass and kickdrum.

Hooking up a compressor is a simple procedure involving an insert cable. This is a Y configuration cable with one 1/4" TRS connector that splits out to two 1/4" connectors. One of these connectors is an RS and the other the TS. (I should mention here that TRS stands for TIP -RING-SLEEVE, with the tip being the send and the ring being the return. This way, the TRS connector allows signals to go both ways, and the TS connector allows on signals to send FROM the compressor to the mixer while the RS connector returns the signal from the mixer to the compressor.) The TRS end is plugged into the insert jack on one channel of your mixer, the TS to the compressor send, and RS to the compressor return. This creates a loop where the original signal leaves the mixer, goes to the compressor, is then compressed, and finally returns to the mixer.

As for using compression, that is a matter of personal preference. I use it only when needed. Unless I am going for a certain type of effect by heavil y compressing the signal, then I use it only for stray peaks, since putting a signal that isn't peaking through a compressor will only introduce more noise. Some people think that even though the signal is peaking out during recording, they can compress the signal in the mix and it will be the same. I used to think that myself but I realize now that when you put a distorting signal to tape, the damage is already done to that signal's sound. The track is already saturated with distortion and no amount of compression during the mix will make it sound as if it were compressed during tracking. That is why you should definitely fix stray peaks with the compressor when recording. Also, final mixes may also need a little compression even if you used it on tracks during recording. This is due to the summation of all the track signals.

The following are just suggestions of where to start setting your parameters for certain instruments. As I mentioned earlier, how YOU want to use compression is your personal preference.

Bass: Try starting out with a ratio of 4:1, and a fast attack and release. I usually use the hard-knee type of compression here since bass is such an attack-oriented instrument. But if you were playing smooth jazz bass, then you may want to try soft-knee. It depends on the sound you are trying to get.

Guitar: This depends on the type of sound you are using, but a good general place to start is 2:1 on acoustic, and maybe 3:1 on distorted guitar (although you may need 4:1 here.) To get a good sustain, try a 4:1 ratio, use a fast attack and slow release. Then play the note you want to sustain, and raise the ratio until the sustain is as long as you want it.

Drums: Drum signals are often compressed due to their hard-hitting attack volumes. If nothing else, compress the snare drum, because each hit will likely peak higher than other hits. Try starting out with a ratio of 3:1, and use a fast attack and release. If the signal is still peaking, try using a ratio of 4:1. This method could also be applied to the toms. As for cymbal hits, try starting with a 2:1 ratio (moving to 3:1 if needed), using a fast attack and a slow release (to preserve the natural decay time of cymbals).

Vocals: As with drums, compression is also commonly used on vocals. The ratio to start at varies for each singer, since some may be very strong and loud singers, and others quieter, having a smaller dynamic range. Try starting out with a 2:1 ratio, with a fast attack, and medium to slow release. Keep increasing your ratio until you get your peaks under control."

If audiophiles spent much time in this world they would be very confused when people start talking about the definitions of the absolute sound. Some how (and I'm not sure how) the audiophile has made their own world (complete with lingo) up that is completely different from the recording world and some how try to call it the same. When the typical listener talks about the audio chain they usually start at the source of their components but most of the work has already been done way before that to the making of recorded sound. What I showed above was just the tip of the iceberg as far as what our music goes through before it gets to us. For me it's exciting cause I know we will never be done tweaking. The other part that is some what exciting again is what Ivor said "10%". When I say to people that I can make a system that is 10 times better than what they have started with I'm not kidding. There is so much on the recording that we may never get to but by opening things up we can live in the world of 3D huge soundstages that are full of goodies.

In my experiences I have seen the jump between the size of soundstage studio vs playback as the playback easily being ten times bigger (if you count forward and backward staging). This is remarkable when you think about it. You are hearing much more (with a well tuned system) than the mastering guys are. All you have to do to prove this is look at the mastering studios space and see how extremely un-tweaked they are.

Story time again

I was in another studio in Detroit where we were showing off the harmonic feet and the MTDs at the same time we were recording an interview for VH1 (wonder where that interview got to?). The mastering room was huge but had things all out of whack which made the soundstage tiny compare to the real life recording that was going on in the other rooms. I played around with DRT and the cones in and out of all the rooms and the differences you could hear were giant however later when we put on the music at my pad and listened to it there was so much more to the music that you almost couldn't tell it was the same recording. It was more enjoyable and real in my room vs the mastering room. People started coming by almost endlessly to hear the recordings on my home setup as a judge just like when I was in Nashville and even Ohio (with the folks that would fly in from NY). As I play this back in my head I can see where the cut off is between the different stages of recording and why the sound can in many ways not be called the same piece of music at different times along the "over all" audio chain.

In many ways we the extreme listeners are explorers of what really went on in the recording room before it got the the control room.

Hi Michael and fellow Zonees

Done a few things this week:

a. I listened offside from in front of one of the speakers as Michael suggested. With classical music I got a soundstage that stretched fairly evenly across to the other side with the images of instruments and voices in the right places. At worst, the soundstage angled back a little on the far side. But with pop music, the system failed. A lot of sound was bunched up in the near speaker with scattering of images near the opposite corner of the room. No difference when I listened close to the right or left speaker - the results were the same.

b. Sonic liked the effect of the Janis W-1 grounding rod so I made a better one. Took a small MTD and a Michael Green genuine mild steel rod cut to lenght. This rod and MTD is wedged under the Janis W-1 between MW pieces. Bass definition is slightly improved because there is no MW block to damp the vibrations and because of the use of a proper/genuine MTD.

c. I found that the early pre-MTDs have a sound of their own. I used 4 under the mini platform suporting the Rega's two toriodal transformers. As the assembly settled a particular sound emerged. Warm in the upper bass but a little out of time with the rest of the music and a slight loss of information. I replaced the 4 early pre-MTDs with Harmonic Feet and the mid bass tightened up, music got into time and the information became clearer. So I changed Janis W-1 grounding cone to a MTD too. Tunees have observed how the MTDs are better than the early pre-MTDs which had a coating of some sort. These early ones are still better than the dozens of other cones around made out of aluminium, concrete, plastic, stone, crystal and such.

d. Sonic is using the same size of MTDs under each supprted device where possible -- for instance 3 large or 3 small, rather than 2 small and 1 large even if they are all the same height. Not a big change of sound or none but since I got enough Harmonic Feet to do this, Sonic might as well get it done.

e. Placed a 1/8" x 1.25" x 1.25" MW square over the top bearing of the Sony CD transport then placed the Harmonic Spring on this piece of wood for top tuning. Much improved clarity of sound. Before this the Harmonic Spring rested on the top of the plastic bar holding the bearing and the Spring did not make full contact with the bar to ground the transport due to the stregthening bars.

The Tune -- so much is in grounding things right. But the listening test shows a blockage somewhere at least with rock music.

Sonic

Hi Michael

That Nashville mastering studio picture -- what are those white structures inboard of the loudspeakers?What are the monitor speakers and how are they placed in relagtion to the side and backwall? Certainly this isn't an extreme nearfield set up, in fact it looks fairly conventional to me, even with PZCs ahead of and to the sides of the loudspeakers.

Are there tuned frequency traps/random diffusors in this studio?

As Sonic understands it, loudspeakers in a tuned system are usually placed close to the side walls, anything between 12 to 18 inches. My set up is 18 inches from edge of the MG 1.5QRs to the side walls.

What is the thinking behind such a close to sidewall placement?

Would this not lead to a shrinking of the soundstage beyond the outside speaker edges due near reflections forward of the speakers? Do different types of speakers behave differently -- line sources may need to be placed further from the side walls since they do not get more directional with increasing frequency unlike pistonic drivers. So Quads and Maggies in Tuned systems need to be further from the sidewalls than say a Classic 60?

Hello Bill333 -- how far are the outer edges of your Quad ESL57s from your sidewalls?

Michael, you remarked that you have heard that pan potting to extreme right or left can push an image outside the speakers. Do you have an explanation for this? Sonic understands stereo much in the L+R, L-R and proportionate mixing between two channels. Which means a discrete image of say a violin or a voice cannot be pan potted beyond the extreme edges of a speaker pair unless the instrument carried significant amounts of non correlated ambient information, it can give a sense of outside speaker imaging due to phase effects.

Sonic

Hi fellow Zonees

Sonic attended a listening session at the end of last week and came away with a nugget about soundstaging. A bunch of us were listening to a good system using one of the large BBC monitors for speakers.

As we listened, one of the guests pointed out that the soundstage produced by this system was like a rugby ball -- the images were big and tall in the middle/centre but pinched and small at the edges, at the speakers.

His explanation was that this effect occurs when the speakers are too far apart and too close to the side walls. The pinched images can lock sounds to the speakers and prevent a wall to wall soundstage from developing.

We moved the speakers closer to each other and without changing the playback volume went through the CDs again with Sonic and the other guests taking turns in the centre seat.

To Sonic, this system is limited because there is absorption used in this room and there is a lot of mass like heavy speaker stands. The tube (pentode) amplification is nice and the sources are both analog and digital though.

With the speakers brought closer in, the images at the speakers did start sounding freer and more similar in tone with the centre. Some imaging outside the speakers appeared with rock music and jazz had a better sense of bigness. The host, initially peeved, now sounded almost happy at the improvement.

This made Sonic think....my Magnaplanar 1.5QRs are 18 inches from the side walls and nearly 9 feet apart from the inner edges of the panels. I have felt a lack of bigness in the soundstage that tells me the recorded acoustic and instrument spread is much larger than the width of my room.

What if I tried a narrower speaker positioning....?

Sonic

Hi fellow Zonees

Sonic made two discoveries:

1. Varying the spacing between the speakers is something like focussing a telescope. At a certain spacing, the images all get to the right size in relation to each other across the soundstage, the rear stage sqares up into or beyond the front corners. When this is done, the same degree of toe-in (if any is used) should be maintained at every tested position.

Speakers too close and there is a blob of sound, a narrowness and too much image height. Too far apart and the soundstage and images are stretched at the edges and image girth varies, bigger in the middle, small at edges and more obviously plastered onto the speaker baffles. At the right distance, it clicks.

2. I guess the sources of blockage in my system are the pre-amp and amplfier transformers. They are the heaviest things around and concentrated mass too. They also sit on the same shelf of my rack with the amp even though the transformers are separated from the amp chassis. It think the transformers should be placed off onto separate shelves and these mechanically grounded.

Michael and others, your views?

Sonic

Hi Sonic

5/28 post

a) This is very good, the off to the side test says much. It also tells you which CDs are closer in recording technique to each other.

b) nice

c) So, you got to hear the difference between the Audiopoint and MTD. Now you can see why I couldn't live with the cloudy collapse of the Audiopoints. It was like someone put a glove over the transfer process. That along with the slight phasing thing left me with the need to do better. Wait till you hear the new cones. A big jump.

d) The new series are interchangeable. You have 5 MGA Audio Alloy cones to switch in and out at the same height.

e) It's all about how we grab the energy, send it and the delivery to the next part in the transfer.

5/29 post

The white things are windows.

The speakers are Bryston studio monitors.

The monitors are setup on a typical on axis studio text book setting. For the fun of it I set them up my way before the techys got there so I could see how big of a change it would be. Huge!! When the boys set them up according to the studio rule book the stage shrunk to the size of toy solders.

When you take the PZCs and platforms out of the room the system sounds like a tin can stuffed with foam (hope they never read this). This was obviously one of those high end studio rooms gone wrong. All text book, no common sense. Even though this new setup gets much praise I shutter at what we started with and the way it is setup after we left. You think audiophiles are stuck, you haven't seen anything till you work with trained engineers.

5/29 part 2

Quote :

What is the thinking behind such a close to sidewall placement?

Would this not lead to a shrinking of the soundstage beyond the outside speaker edges due near reflections forward of the speakers? Do different types of speakers behave differently -- line sources may need to be placed further from the side walls since they do not get more directional with increasing frequency unlike pistonic drivers. So Quads and Maggies in Tuned systems need to be further from the sidewalls than say a Classic 60?

Hello Bill333 -- how far are the outer edges of your Quad ESL57s from your sidewalls?

Michael, you remarked that you have heard that pan potting to extreme right or left can push an image outside the speakers. Do you have an explanation for this? Sonic understands stereo much in the L+R, L-R and proportionate mixing between two channels. Which means a discrete image of say a violin or a voice cannot be pan potted beyond the extreme edges of a speaker pair unless the instrument carried significant amounts of non correlated ambient information, it can give a sense of outside speaker imaging due to phase effects.

Two wonderful Questions. The lord has blessed you with a brain. If only we could implant this line of thinking into the audiophile mindset. Boy would it make life easier.

1) Side wall placement is something that I come up with solely based on the sound of the room and how sound waves are responding to the surface of the boundaries of the room. For me, it's a hearing thing. The whole first wall reflection thing is so messed up and I'm embarrassed that I have to talk about it in promotional material. Ever sense people got out their flashlights and mirrors they have moved further away from how much we need the support of walls to be a part of the speaker/room development program. I have always used the walls as one of the most important parts to my sound regardless of speaker type. I've been in some rooms where the speaker wanted to be almost touching the wall and others where the speakers rejected the sound of the wall. There are some walls able to build a very nice full range support and others that don't develop this support until you get away from the wall. The key to this is learning the room and where the Pressure Zones build and decrease. Usually you will find a place on either the plus or minus side of this zone that is perfect for making the speaker engage with the rest of the room. So think again of a room that is building up with pressure when you are thinking about placement. People that don't setup their speakers and ears in like minded places in relationship to the zones in the room get a disconnected sound. You feel separate from your speakers instead of being part of them.

Again throw out the rule books cause the ones who wrote it ain't liven in your room.

2) The loudspeaker is only one part of speaker system. We should always think in terms of pressure and space as opposed to the physical speaker. This is hard sometimes cause our audiophile brain is looking at an object that is sitting there and we think it should be the end of the stage. How do we make something go past the end, right? Well the reality of it is there is no end but the end. Meaning when the energy ends or the effect ends (phasing effect) that's it. You have unlimited space options cause you are in a room that is building up pressure. Pressure that works with your ear's sense of timing and phasing. Stereo is a wonderful thing when you think about it. You have 2 microphones sitting on either side of your head. These microphones never hear the same signal. One is always slightly delayed over the other giving you the concept of space and balance. Here's a great way to get anyone past the speaker being the edge of the stage issue. Play a piece of music on your system and listen to the way it stages then put on a set of headphones playing the same piece of music while your sitting in your listening chair. Compare the two sound stages. Pretty scary isn't it . Your ears are built to take in the sound of space. This means pressure and delays. A microphone in a studio does the same thing. It gathers the space, the whole space. In a typical studio setup you might listen to something that sounds like it is right in the speaker but the reality of it is (even without doing any phase tricks) this sound may be way outside of the speakers collapse ambient field and you don't even know it until you play it back on a system able to produce more of the harmonics and ambient field. How much more outside of right and left or how much more 3D is what is shocking. I'm convinced that studio engineers have no true idea of how big of a sound stage they are truly making. Again follow an engineer around on a recording field trip and you will come to the same conclusion.

Here's another way of putting it. There is a big difference between the words 2 left and right channels and 2 left and right speakers when it comes to recording. Speakers are only a tool to let out the recorded sound they are not the recorded sound itself. The recorded sound is usually as big as the room it was recorded in and because studio engineers are listening to a closed in setup the music they are producing is much bigger than what they themselves are listening to once it is in a good playback mode. Here's an example: ever listen to a piano recorded in a killer studio? In the studio it might sound like the piano is about full size because of how close we are to the monitors, sound board and such. Well take that recording home with you and the stage is huge, like 5 times the size of what you just heard in the studio. Often times you have to go back in the studio and change your miking as well as doing other things to shrink it down. I myself prefer the big sound but if your hunting for real size real space this is what you have to do.

We often hear voices in the middle of our sound stages sounding much bigger than real life. That is because we are thinking of looking at their face while we are listening instead of hearing what the microphone sees as space which is usually a ton bigger than a mouth. when people talk about right and wrong to me it is obvious that they have not done the process start to finish. A ton if not most engineers themselves don't listen start to finish so I can understand where much confusion comes in. But we in this world of ours should get to the point where we can enjoy the art without trying to make something happen that is not going to all because a studio tech doesn't understand what mike with what acoustics with what studio equipment gives them the real size. Could you imagine if they actually caught onto reality.

For me I just enjoy it for what it is and explore the outer limits along with the harmonic magic.

Hi Michael

If Sonic has brains, it is received from the Lord. True it is.

Before this discussion came up about distance from side walls, my MG 1.5QRs were 18 inches from the side walls (edge of panel, not centre, to wall). 0ver time I noticed that the girth and images closer to the wall were smaller and less dimensional than those in the centre -- I posted a scan of this if you recall.

After the listening test I referred to, I moved the speakers step by step to see the effect of distance while maintaining a constant toe in. The best point was close to 2 ft from the wall. Sonic found that any more to the middle of the room and the images bunched up in the centre with no outside info and lousy focus.

Going closer to wall caused a progressive decline in image size and subjective loudness. Closer I got to the wall the worse it got. I dunno what happens if I got the panel really close to the wall like 3 inches. My speaker cables are too short and trying this will involve moving the amp, the rack, the FS DRTs. Too much work just to see if the closer-you-get-the-worse-it-becomes reverse.

At this 20+ inches from wall setting, the image size are even across the stage, there is significant outside images and ambience and the volume and dynamic range has come up (9 oclock on the amp control sounds like 10 or 11).

Not bad for a week of tuning and settling. Then Sonic had that brainwave....looking for blockages I thought 'even though I have taken the heavy transformers out of the amp and preamp chassis, they still sit on the same shelf as the amp and preamp.....this is still a concentration of mass on the shelf.....what if the transformers were placed on separate supports? Cdimi and Hiend1 did this and they must have done it for a reason...'

Sonic don't have miniclamp racks and MGD shelves spare...wish I did....so I improvised.

I took two wooden footstools made from solid wood (not too heavy, but not lightweight) and moved the transformers off the MGD Clamprack shelves onto the stools. One stool for the heavy Magnequest of the Quicksilver and the other stool for the two toroids of the main amp. The toroid from the amp driving the Janis W-1 subwoofer was not moved because no suitable furniture was available. MW slices separated the transformers from the surface of the furniture.

When I played music -- David Munrow's Henry VIII and his six wives (EMI), Ravel's Piano Works -- Miroirs, Sonatines and Valse Nobles et Sentimental (Crossley/CRD) and McCoy Tyner's Extensions (Blue Note) -- I was speechless....

Sonic

very happy


BTW the distance from the wall is usually further away with hard walls, but not always. Always use your ears.

Hi Zonees

Here's some history of the BBC studio monitors -- by Harbeth's Alan Shaw:

This is the story of how a handful of talented engineers at the British Broadcasting Corporation's Research Department undertook the world's most sustained and comprehensive loudspeaker research programme. As with all great engineering achievements, what lies underneath is the human story of dedicated and often very young men, and I hope I can capture that here.

In speaker development the names D.E.L. Shorter and H.D. Harwood (who upon his retirement from the BBC founded the Harbeth company) are pre-eminent. But in piecing together the who, when and why of the BBC's in-depth involvement with loudspeaker R&D, one man deserves especial respect. None of this would have happened without the commitment, perseverance and genius of Harold Lister Kirke [5], Head of Research Department at its inception for the first all-important twenty years. I'd not heard of this remarkable and largely self-taught man when I started this research but his influence is everywhere. He applied the 'scientific method' and then shared that knowledge first with the industry then with the public.

Kirke is the truly unsung hero; a senior manager holding the purse strings; the commitment of necessary resources backed by his enormous reputation. He was to lead the BBC towards quality speakers which industry alone was then incapable of delivering.

The elegant, deceptively simple writing style of these BBC engineers and their almost kitchen table pragmatism caught my imagination as a schoolboy; it was (and is) a branch of engineering that half a century later still relies on much critical listening combined with the judicious use of technical equipment. Behind the leading names were many technicians and support staff who with dogged determination worked on ever smaller technical matters. I salute them all [1]; their hard work has become a life-long career for me; to really understand Harbeth and what we represent, we need to start at the very beginning of the history of broadcasting, and to incremental progress spread over nearly one hundred years.

How and why did the semi-autonomous government-funded BBC get so deeply involved with loudspeakers?

It's one matter to design loudspeaker systems from supplied drive units, but to undertake basic blue-sky research into loudspeaker drive units is a wholly more complex and expensive matter. Basic research leading to new products is normally undertaken by industry with the usual profit motive. But a government department undertaking the work of private industry is unheard of.

As we will see, this situation crept up on Kirke who found his department uncomfortably promoting trials of the upcoming wide bandwidth 40Hz - 15kHz VHF/FM system [2] but with monitor speakers from a previous era with their high frequency response not even half the transmission bandwidth and all manner of coloration problems.

Kirke didn't mince his words. He was evidently fully aware of the implications of inadequate monitoring. A programme of investigative work was put in hand: in 1948 he wrote and signed the report [3]. Of one British speaker unit he commented "This loudspeaker is distinguished by its unusually ingenious design and by its exceptionally poor performance." and of another "The performance of this loudspeaker falls so far below our existing standard that it was not thought worth while to carry the investigation any further." and of another "In the form submitted, however, it is useless for our purpose." He concluded "... it is thought that some time may yet elapse before a completely satisfactory commercial model becomes available."

The tone of his report is of disappointed realism. Yet that very same year marked the turning point with the masterstroke recruitment of Dudley Harwood from the National Physical Laboratory to the Kirke, Shorter team that would in time transform the global loudspeaker history thanks to their application of material science.

It is an insight into Kirke's character that he was willing and able to establish at the BBC his own independent technical research capability at arms length from the fledgling industry. He evidently had a particular interest in loudspeakers with a patent filed in 1933 in the USA [6]. It's difficult to imagine how the industry could perceive this enthusiastically; the BBC would then be judge jury and executioner - all at taxpayers expense.

Kirke could, presumably, have embraced and encouraged the industry, and maybe he tried. GEC, a major audio equipment manufacturer as perhaps the best example had good technical facilities. Kirke, one of the most respected radio engineers in the world and BBC Chief Research Engineer through the design and construction of Broadcasting House itself [4] would certainly have been known to them. Presumably they would have been willing to cooperate as better products would lead to more profit, and positive publicity. For whatever reason Kirke chose the expensive, audacious and politically risky step of engaging public money in loudspeaker research that was rightly the duty of an outside industry: he must have been able to make a convincing case to the Board of Governors through the Chief Engineer. We must conclude that he was under real pressure to solve the BBC's loudspeaker requirements and utterly convinced in his team; and utterly unconvinced by the capabilities of the nascent loudspeaker industry.

The chronological sequence of events that I describe in the next pages leads directly to the monitor loudspeakers we today design and manufacture at Harbeth. (AAS Jan. 2008)


Continued in BBC 1922-1960
________________________________________

References:

[1] MESSENGER, P., 1983. "Sounds subjective" Hi-Fi News & Record Review magazine.

[2] BRAY, John, 2002. 'Innovation and the communication revolution. From Victorian pioneers to broadband internet'. IEE. Google extract.

[3] KIRKE, H.L., 2/1948. "The selection of a wide-range loudspeaker for monitoring purposes (First report)". BBC Research Department No. M.008

[4] BBC publication, 1932. "A technical description of Broadcasting House".

[5] Small framed cartoon of Kirke, H.L. noticed by chance and photographed by AAS hanging in the corridor of BBC Research Centre, 25.02.08. It depicts the relocation of a BBC dept. to the BBC's Kingswood Warren mansion in 1948. Top left, Kirke is shown holding Kingswood Warren in his hands as he would a child. The caption reads, H.L. Kirke (Head of Research Dept.) "The Master".

[6] KIRKE, H.L. et al, USA Patent 1933, No. 1930757 for loudspeaker


The British Broadcasting Company Ltd. was formed in 1922 by three hundred manufacturers and shareholders as a commercial venture. As Chief Engineer, Capt. P.P. Eckersley had been involved with broadcasting from the very earliest moments; in February 1924, H.L. Kirke was appointed as Senior Development Engineer. By 1927, the influence of broadcasting on British society was so marked that by Royal Charter, this private company was rolled into the British Broadcasting Corporation [1] for a renewable term of ten years from 1st January 1927 under a Director General as chief executive with a formal executive, control, administrative and engineering structure. Funding was (and is) substantially provided by the British Government via a levy, the TV (then radio) licencee fee and the Director general reports to the govenmnent ministers responisble for broadcasting. The entire staff of the old Company was transferred to the new BBC.

After the formation of the British Broadcasting Corporation, Development Section was renamed Research Department and H.L. Kirke C.B.E., M.I.E.E was appointed as Senior Research Engineer, later known as Head of Research Department. H.L. Kirke held this influential and pivotal post from 1930 for an astonishing twenty years. He was immersed in the format battles for the newly conceived television service [2] and was in-post during the design and construction of Broadcasting House, London, opened in 1932. In the BBC Yearbook of 1933 [3] Kirke wrote a detailed appraisal of current microphone technology for the public that would grace the pages of a modern journal.

Under his guidance the BBC's iconic AXB microphone [4] was developed, made at one-tenth the cost of a bought-in commercial unit in a culture of 'best performance, lowest possible cost, make in-house if necessary.' He took his profession seriously inside and outside the BBC; records show that in 1943 (and presumably long before) he was active in The Institute of Electrical Engineers [5], and by October 1948 he was Chairman, probably the first Chairman since Jan. 1948 of the Acoustics Group of the Physical Society of Great Britain [6]. Proceedings show that on 25 Nov. 1948 he was in the chair. "The following paper was read and discussed - Apparatus for Acoustical Measurements at Low Frequencies" presented by Dadson and Butcher

The earliest recognisable subjective evaluation of the loudspeaker was not by the BBC but by commercial company GEC's F.H. Brittain in 1932 [7], followed in 1936/7 [8] and in 1938 with a remarkable report which covered "Frequency response (Its effect on sound quality), Balance (Permissible variations and limits), The Listening Room (Its effect on reproduction), Harmonics (Their effect on sound quality), The Validity of Steady State Measurements and Summary of proposed method of Assessment of all available acoustical measurements." [8]. It is curious as to why GEC with its manufacturing knowledge of loudspeakers, its measurement and listening facilities and its own anechoic chamber [9] were not more closely involved with the BBC or indeed any other speaker parts suppliers.

There is no mention of GEC drive units being examined in either BBC report [10, 11, 12]. Had they done so, and had the results been acceptable to the BBC, or had GEC been able to adapt them then the BBC may never have become involved in loudspeaker research although the GEC Presence Unit became, in time, a standard within BBC monitors. A horn tweeter drive unit minus the horn.

It is very fortunate for the speaker industry that Kirke had such a great personal interest in acoustics "Mr. H. L. Kirke observed that are the subject of acoustics had received too little attention, except by a very few, and it was still understood only by the few." [13]. This comment was seemingly made by Kirke in 1945, some fifteen years after the design of Broadcasting House, several years after Brittain's published review of loudspeakers and about three years before his chairmanship [6]. It is impossible to know what branch of acoustics Kirke was commenting on: it could have been studio acoustics, microphones or loudspeakers; he had an interest in all fields. In the area of loudspeakers, GEC's Hugh Brittain demonstrably had a thorough grasp of the subject, but as an employee of a commercial company, he would have been under no obligation to share that knowledge. But it is curios that GEC's knowledge and facilities was not drawn on. It is possible that there was a conflict of interest over GEC and the politics of the selection of the new TV system but "Although from the late 1920s the (GEC) laboratories maintained a watching brief on technical and patent developments in television, there is no evidence that the Laboratories contributed to the fundamental concepts of what became the Marconi-EMI 405-line system which was adopted as the UK standard, nor the Baird 240-line system which, after experimental transmissions over a period, was rejected." [14]

Kirke must have had reason to believe that unlike the broadcast television system concept which was presented to the BBC by industry, in acoustics matters the BBC could not or should not draw on a pool of external knowledge. Or perhaps he had little confidence in that external pool. A secondary theme to Kirke's audio legacy was studio and building acoustics. A generation of engineers later during the 1970 and 80s, that work would become the main activity of BBC Research Dept.

From its conception and patenting in the mid 1920s the USA designed Rice-Kellogg (I.S.R.K.) unit USA was available to the BBC. This was mounted in a suitably large polished piano black floor standing cabinet coded LB/3 and the complete system named LSU/7 monitor speaker. The back was open. As for the listener at home in the 1920s and 30s they would have listened through headphones or simple valve receivers.

The standard of monitoring in the studio over the LSU/7 would have been far superior to even the best listeners equipment. As the quality standard of domestic receivers increased, the performance gap between the sound at home and in the control room started to close. In January 1944, towards the end of the war, Kirke sailed to New York on the Queen Mary [15] [15A] [15B] to see for himself first hand the new VHF/FM radio service in action and progress in television transmission. Its likely that he was made aware of the development work underway at CBS Labs. by Peter Goldmark [16] on the modern LP record and rightly anticipated that the British audience would shortly follow American technology and entertainment trends. All of this amounted to higher fidelity.

The limitations of transmitting (on AM) with its restricted low-fi audio bandwidth had been commented upon in 1934 "The interpretational qualities of speech and music reside in the upper frequencies and it is not possible to obtain naturalness unless the frequency range extends to 12kHz ... One of the reasons why the reproduced version of the human voice sounds unnatural ... is due to the absence of frequencies above 5kHz..." [17]. Upon his return from the USA later in 1944 it must have been evident to Kirke that the audio frequency response window was opening. Clearly, Kirke had a demonstrable interest in loudspeakers and microphones at this time [18]. The LP record and the wide-band VHF/FM radio system was in imminent prospect. As McLachlan wrote in 1934 "In broadcast programmes, induction, valve, and other noises which are imperceptible in a system inadequate to reproduce above 4.5kHz would assume undue proportions if the range were extended to 12kHz. Consequently,to realise the full benefits obtainable by extending the upper register, the noise-level of the input in the studio and in the transmitting and receiving apparatus ... must be very low indeed." [17].

Sometime between 1945 and 1947 Kirke authorised BBC research engineers to start work on a replacement for the LSU/7 monitor. In 1946 D.E.L. Shorter published "Loudspeaker Transient Response..." [19] which, when computerised measuring systems became available thirty years later could be traced as the origin of the waterfall measurement plot.

The BBC Research Department's involvement under Kirke's supervision started by canvassing the speaker industry for a suitable wide-band driver. The next step was to verify its sonic and technical potential and wide bandwidth. The final step would be to approve its fitting to the new monitor cabinet. The final assembly would eventually be known as the LSU/10. The speaker investigation project was to became a greater challenge than even the new television system, and it ran for over thirty years. The performance of loudspeaker units evaluated by his research engineers was dismal. Seemingly the nascent loudspeaker industry had only a rudimentary technical understanding of their products and was practically clueless about how to maintain and improve quality. The industry put a bold spin on what was evidently extremely poor product. In 1948 Biggs told the public "Speech is an essential test (for a loudspeaker) and most people judge its reproduction accurately. The first test imposed on a loudspeaker by the Research Dept. of the B.B.C. is to listen to the reproduction of speech in the middle of a field, away from reflecting surfaces, and compare it with the original voice. A good loudspeaker should give good reproduction of speech, but it does not of necessity follow that it will be equally good on music, which requires a much wider frequency range" [20].

This was true. But in actuality what the BBC was experiencing as the fruits of the speaker industry was of desperately poor quality. "The performance of sixteen "high-fidelity" loudspeakers submitted by British manufacturers has been investigated. The results have been generally disappointing. The cause for this state of affairs is though to lie, not so much in any lack of ingenuity in design on the part of the manufacturers concerned as in the scant attention paid by them to simple subjective testing. It is true that the opportunities of comparison between direct and reproduced orchestral music are denied to most designers.

On the other hand, the majority of the defects noted in our tests were clearly shown in the reproduction of speech, and in fact the results of the music tests, apart from yielding some information in the extreme low frequency range, and on certain non-linear distortion effects, did no more than confirm the conclusions arrived at in the speech tests. The technical equipment required to carry out a speech test under carefully controlled conditions is no more than any reputable loudspeaker manufacturer might reasonably expect to posses, and it would appear that either the designers have made too little use of this simple technique or that, in using it, they have neglected to make frequent and critical comparisons with the original sound."[10]. In addition, Kirke drew a damning conclusion about the speaker industries negligible progress towards greater fidelity in over twenty years 'The I.S.R.K. unit in itself (dating from the mid 1920s, used in the LSU/7) is remarkable for its relative freedom from low-damped resonances ... although a very old design, gives a general performance more pleasing than that of most of the modern loudspeakers tested [10].

As Kirke reported said in 1945 ... 'the subject of acoustics ... was still understood only by the few.' [13]. But the new (LSU/10) monitor speaker was a real necessity and high priority with high quality wide band broadcasting just around the corner - time was running against them. There was a growing sophistication amongst consumers, an embryonic hi-fi industry, magazines and the consumer was curious about the subject of reproduced sound. A contemporary example for 1948 said " ... bear in mind that the reproduction (at home) depends upon the quality of the input ... a better speaker will expose the distortion of sound worse than an inferior one. The B.B.C. are by no means blameless in this respect. B.B.C. quality when it is good, is very, very good but some transmissions are very bad ...". [20]

In 1947, H. D. (Dudley) Harwood joined the BBC Research Department from the National Physical Laboratory where he had worked on hearing aid design and loudspeaker calibration [21]. The same year, D.E.L. Shorter, with Harwood about ten years his junior had been mandated by Kirke to get to grips with the BBC speaker requirements. Either Harwood was specifically recruited by the BBC to bring his technical expertise in loudspeakers to the department or it was extremely fortuitous timing of the right man at exactly the right moment. No time was wasted [10]. BBC Research Dept. 1948 (second) report [11] written by D.E.L. Shorter, investigated by Harwood and signed-off by Kirke opens 'In view of the urgency of the situation, it was decided that the investigation (into the two-unit loudspeakers being made by Research Dept.) be temporarily shelved in order to concentrate on the immediate task of selecting a commercial product'. This implies that fearing the absence of a suitable commercial wide-range drive unit, Kirke must have previously authorised the engineers to investigate combining a low frequency unit with a high frequency unit each within their optimum operating range. We now take for granted.

In their final report, jointly investigated by Shorter and Harwood, written by Harwood and signed off by the new Head of Research, W. Proctor Wilson (following Kirke's retirement) they open with the chilling comment that 'It may be stated at the outset that none of the (commercial) loudspeakers (drive units) examined was found to meet our requirements ...' [12] and report subjective opinions across the range of speakers such as '... the middle and high frequencies giving the impression of being detached from one another; definition was poor; the bass response extend lower than usual but was boomy in character; the high frequency response was deficient and the whole reproduction was heavily coloured ...'.

After examining over forty wide-range loudspeakers they concluded that 'the results have yielded much valuable information on the correlation between subjective and objective assessments of performance, a subject which will be more fully discussed in a later report'.

They had at least been able to select a concentric multicellular 15" Parmeko drive unit - a copy of an American Altec Lancing design which had performed adequately for the LSU/10 monitor speaker but they noted with concern that "... five further LSU/10 loudspeaker units, nominally identical (showed) considerable differences between the units ... of twenty more manufacturers samples all but one or two were found to posses objectionable (sonic) characteristic ... after a change in the manufacturing technique, six further samples were submitted but the general performance of five out of the six still exhibited (sonic) defects..."[11].

That was far from the end of the matter. Problems soon arose. 'Difficulties arose in repeating, in production models of the Parmeko loudspeaker, the performance achieved with the sample (in the 1-2kHz band) on which the selection tests had been carried out, and the present report indicates the extent of the compromise which must be accepted if these loudspeakers are to be adopted in their present form' and '... it was not possible to reproduce the characteristics of the prototype (1-2kHz) band, and it is thought that these characteristics were the result of a fortuitous combination of circumstances.' [11].

The LSU10 [22, 23] was eventually introduced around 1950 and was still to be found in the BBC in the mid 1970s - I remember one in the control room of BBC radio London, Hannover Square with the Lorenz LPH65 tweeter strapped to the front grille to extend the high frequency response.

The introduction of the LSU/10 in 1948 with its enormous 280ltr. thick-wall cabinet had a workable frequency range from 40Hz to only about 6kHz. It practice, it proved difficult to control the frequency response at the top end of the LSU/10's HF horn to better than plus or minus 3dB.[10?]. The 6kHz upper limit was adequate for monitoring in the BBC in the AM era (medium wave) from recorded music played-out from 78 rpm shallac gramophone records with their own restricted frequency response. But the LSU/10 was only a stop-gap.

Although the 1950s did not see the introduction of any new BBC monitors, this was a time when the BBCs detailed knowledge of the technical characteristics and limitations of loudspeakers and microphones was growing. Shorter's masterpiece I.E.E. paper, received No. 1957, published Apr. 1958 [26] opens with "Moreover, even a broadcasting or recording organisation must limit the size and cost of its listening equipment ...In practice, a monitoring loudspeaker is intended to represent the best product of its kind which could be used by a member of the listening public ...". But all of the BBCs research focus to this point had been concentrated in improving and controlling the quality of the middle and higher frequencies. It was not until a year later that James F. Novak [27] applied scientific synthesis to the low frequency performance of speaker systems with engineering models to explain and predict how bass drivers and cabinets behaved together at low frequencies.

_______________________________________

References:


[1] BBC Broadcasting House, London. Historical perspective.

[2] BURNS R.W. 'John Logie Baird: Television Pioneer'. Baird naturally wished to experiment in broadcasting television signals from one of the BBC's transmitters (and) he therefore communicated with each H. L. Kirke a engineer of the BBC who later was later to become its Senior Research engineer. 'Kirke was very interested in helpful', wrote Baird. Several transmissions were arranged the television picture being sent by telephone line from Baird's laboratory to a BBC studio. Kirke then put it on the air through the BBC's radio transmitter and Baird received it again by wireless at his laboratory. 'Complete success was achieved by this method.'. Only three experimental transmissions were given, the three experimental transmissions were unsatisfactory and they came to an abrupt end. 'Someone "up above" at the BBC, Kirke would not say who, had ordered the transmissions to cease.' ISBN 0852967977

[3] BBC Yearbook, 1933

[4] BBC AXBT microphone. This led to the 1953 design BBC PGS/1 ribbon microphone (Coles/STC 4038) is still in production in 2008.

[5] I.E.E. 'Discussion before the Wireless Section, 5th May, 1943'. "Mr H.L. Kirke said the title of the paper is perhaps unfortunate as it does not appear to give a proper indication of the subject matter and it is not very descriptive of apparatus..."

[6] Physical Society (Great Britain), Physical Society of London, Proceedings 14 Oct 1948, 25 Nov 1948, Imperial College, London.

[7] BRITTAIN, F.H., 1932. "A device for rapidly plotting loud-speaker response curves". J. Sci. Instrum.

[8] GENERAL ELECTRIC COMPANY (of UK) Ltd., Research Labs.. "The interpretation of acoustical measurements. Part 1. Loud speakers" Report No. 6666

[9] BRITTAIN F.H., 1937. "The appraisement of loudspeakers" G.E.C. Journal mentioned in book 'The GEC Research Laboratories, 1019-1984' by Robert Clayton ISBN 0863411460 9780863411465. Google extract here

[10] KIRKE, H.L., 2/1948. "The selection of a wide-range loudspeaker for monitoring purposes (First report)". BBC Research Department No. M.008

[11] "The selection of a wide-range loudspeaker for monitoring purposes (Second report)", BBC Research Department No. M.008/2, 1949/3

[12] "The selection of a wide-range loudspeaker for monitoring purposes (Final report)". BBC Research Department No. M.008/3, 1952/5

[13] INST. OF CIVIL ENG. pub. 1945. 'Discussion on modern theory and practice in building acoustics'. Engineering Division Papers 1753-7797. Exact date of quote unknown but by implication, post war, 1945. "Mr. H. L. Kirke observed that are the subject of acoustics had received too little attention, except by a very few, and it was still understood only by the few. Before the war the Acoustics Department of the BBC had carried out some experiments into studios which had more or less at the same shape but had different wall surfaces, one being irregular and the other regular; and it had been found and that, although the two studios, of identical acoustic treatment, had exactly the same rate of decay or reverberation-time, there was a vast difference between the sound that came to the ear from the two studios."

[14] The intense rivalry between the competing Baird and EMI TV systems is described here. Kirke must have demonstrated remarkable political and technical ability to navigate the BBC through to selection of the best (i.e. highest performance) system. The commercial advantages and prestige to the winning format supplier were huge. Also here the BBC examines rival TV systems.

[15] Kirke's visit to USA - reference to follow

[15A] US Dept. of Justice, 31 Jan. 1944, 'List or manifest of alien passengers for the United States'. Kirke is detailed on List 4, as 'Occupation: radio engineer' amongst eighteen government officials and one marine engineer. At that time of WW2, the liner Queen Mary was commandeered by the British Government. One can only marvel at prospect of stimulating conversations between diplomats and the two engineers.

[15B] Sailing to the USA in late early February 1944 from the UK was not risk-free because althougth the U-boat threat had receded, it had not disappeared.

[16] Peter Goldmark, pioneer of the LP record launched 1948. He was turned down by Baird for a job after meeting Baird for lunch in London

[17] McLACHLAN N.W., 1934. 'Loud speakers - Theory, Performance, testing and Design'

[18] BBC/KIRKE, H.L., 1945 'Improvements in and relating to PA systems ...' UK patent GB588355 588,355

[19] SHORTER D.E.L., 1946. "Loudspeaker Transient Response: Its Measurement and Graphical Representation," BBC Quarterly, vol. 1, pp. 121-129. Waterfall plots.

[20] BRIGGS G.A., 1948. 'Loudspeakers. The Why and How of Good Reproduction' 1st Edn.

[21] HARWOOD H.D., 1968. "New B.B.C. Monitoring Loudspeaker" Wireless World magazine

[22] SHORTER D.E.L., 1950. "Sidelight on Loudspeaker Cabinet Design". Wireless World magazine

[23] UK Patent 696671 'Improvements in and relating to Loudspeakers'. Application date :23 Sept. 1949. Published: 9 Sept. 1953 BBC/Donovan Ernest Lea Shorter

[24] COOKE, R.E., 1968 "High Quality Monitoring Loudspeakers" British Kinematography Sound & Television Journal.

[25] OLSON

[26] SHORTER, D.E.L., 4/1958. "A survey of performance criteria and design considerations for high-quality monitoring loudspeakers" I.E.E. Paper

[27] NOVAK James F., 1958. "Performance of Enclosures for Low Resonance High Compliance Loudspeakers" Journal Audio Eng. Society, presented New York

Hi Michael and Zonees

This has been a good tuning week after the big changes Sonic got from moving the preamp and main amp transformers off their clamprack shelves onto low wooden stools. I am still surprised at the extent of the improvement this gave.

Good learnings and experiences followed:

1.Moving the transformer of the amp driving the Janis W-1 subwoofer didn't work. The stage shrank and I lost harmonics in the bass and mids. For this I used a mini platform made of geniune MW from Michael and 4 resitone 3/8" rods. Wonder why given that this support is made completely out of MGD material?

2. Top tuned the CD player with a small Harmonic Foot that contacts the transport and the top shelf with thin MW slices. Very good -- more richness through the whole musical range though more in the upper bass and mids.

3. The soundstage goes nicely outside my speaker panels now in the new, closer-in position. Sonic had to adjust the sidewall sound shutters to optimise the imaging Left and Right and this seems to be programme dependent -- that is, some CDs need shutters toed-out some need toe-in or vertical. While this shows the power of the tune, a development like this is not something Sonic looks for or prefers.

4. Each new level of the tune is like peeling an onion (apart from the tears of frustration often shed to get there). You hear new things that can take the Tune in your system further. With the transformer tuning, I heard there was more harmonics in the amp driving the Janis w-1 subwoofer waiting to get out. Three large Harmonic Feet did good, four are even better.

5. Also found that the Sony CD player preferred to be on three Harmonic Springs rather than four. Maybe Michael can explain why some of my gear tune better with three and some four supports?

6. Will be trying three small MTDs under the Sony this weekend to see the effect -- the great Jim Bookhard used a combination of MTDs and a Harmonic Foot to tune his CD player.

7. For sure MGD plastic didn't work on the front faces of the two FS-DRTs. The extra reflection in the middle of the room called attention to the devices.

8. With all this, on properly recorded musick eg: Bach's Dorian Toccata and Fugue for Organ (Richter/Archiv), Sonic can stand in front of one speaker and while I hear part of the signal from that speaker, the soundstage now spreads thru the room and along the opposite wall across its 20+ foot length. There is dense music thru the whole room. Never heard this before....if the noticeable sound from the near speaker vanished, it will be a textbook Tune experience. But not quite. This works only with classical CDs. With rock, I hear the near speaker dominating the centre images and far speaker.

Michael, what does this tell you and what can I do?

What Michael wrote about recording tells me a lot. I think engineers are scared of change because they know the recordings they make have to be played back on thousands of different systems and rooms, cars, boats, headphones, computers, so they don't want to chance it. The big fear is to do something to the studio system which can sound great but can cause hundreds customers complaining about how the recording sounded strange or failed in their set ups (even if these are boomboxes). So they stick to what appears safe.

Also it seems that the microphone is also part of the instrumental mix as much as the violins, guitars and basses, drums. Change mic and pattern and the music changes. The systems engineers listen on and make their decisions with are only playing back a fraction of the music that is flowing by them. Most of the information in the signal just flows by unheard.

I wonder how a Direct Injected instrument like a bass, a synthesizer plays back on a tuned system? It can't have ambience and bloom because there is no room and no microphone.

Maybe most speaker designers are still thinking of the behaviour of their products as if they are playing in anechoic chambers. I wonder how a tuneable speaker and set up will reproduce music in an anechoic chamber where the "your room is yhour speaker" analogy breaks down. Like headphones?

I guess then that a signal panpotted exrtreme Left or Right will place the image right on the speaker as theory predicts but with a room and with tuning, the ambience is unlocked and this will cause psychoacoustic effects and the new data heard will cause the localisation to appear outside the speaker edges -- is this what you mean Michael when you discussed a panpotted signal moving way outside the speaker edges?

Sonic

"Michael, what does this tell you and what can I do?"


In a nut shell it tells me this is not any of the recordings it is still a blockage issue. Keep looking at what is in the boat that can be thrown over board and still get the job done.

Hi Zonees

One central question about high-quality music reproduction is this: Does the audio system end at the cones of the loudspeakers or is the room part of the system?

There is a school of thinking that believes the audio system stops at the speakers and the room has no part in the reproduction process apart from being something that needs to be damped, absorbed and generally got out of the way.

The Tune takes the opposite view that the room is a major part of the system -- the last mile to our ears -- and we need to co-opt it into our audio systems. It is a component as much as a phono cartridge. Everything in the room counts towards the sound -- if we have lots of glass and metal objects, there is an unsetting coloration while rooms with wood, especially tonewoods from Michael like the cedar, pine, magic wood give music. That is why a room stuffed full of furniture sounds different and nearly always worse than a lively room tuned with Michael's products.

This path of the Tune requires perseverance from users and a lot of effort invested to understand the room and making it work with the audio equipment.

Sonic

Hi Zonees

Sonic is experimenting with differing amounts of absorbsion and reflectivity/diffusion in my room – trying to get the optimum in the front and rear half of the space.

It is an engaging tune with just one Echotune or a 4 ft x 2 ft piece of tuning plastic sheet from Michael making an audible difference. But not got to a satisfactory point yet.

Also as I am working on this, Sonic found that the hand wound Harmonic Springs are great under the main amp, the subwoofer amp and the preamp with 4 per piece of equipment because the Harmonic Springs are soft and they squash under the weight of the gear if only three are used – and these are not “audiophile heavy” gear. And these Springs work well in top tuning the X-30 crossover and the Quicksilver. Better than resitone rods or the machine wound Harmonic Springs. They are so very rich in harmonics and transients it is eye opening.

Anyway, while Sonic is letting things run in, I been reading up on audio. I like reading the writings of Robert Everist Green. He is a reviewer and commentator in the Absolute Sound of long standing and IMO Pearson’s (now Harley’s) most respected and qualified writers.

Known as REG, Green is a mathematician and academic as well as a trained violinist. Though Dr Green is of the room damping and digital room correction school, he is one writer not to be dismissed out of hand. He is not exactly a fan of the Tune and any acoustic thought that involves room liveliness.

I found some articles on his site www.regonaudio.com that make good reading to give me (and maybe other Tunees) perspective to our thinking. Here is one piece by REG.

If any Zonees would like the URL of this work, it is:

http://www.regonaudio.com/HighRomanticism.html

Have a read!

High Romanticism and the Sound of Recorded Music
SHORTCHANGING ORCHESTRAL SOUND

It all started with Wagner. Every revolutionary has predecessors as well as followers and some of the elements of the musical High Romanticism of the late nineteenth century were present in embryonic form in the music of Schubert and Liszt and especially Beethoven. But it was Wagner who made not just preliminary steps but a definitive leap into what amounted to a new musical world altogether, the musical world that would be the natural home of Richard Strauss, Bruckner, Franck, Korngold, Scriabin, and a host of others.

Almost every orchestral composition from, say 1860 to 1913, was written either under Wagnerian influence or in deliberate rejection of it.

This is all familiar, of course. No composer has ever been written about more than Wagner. Indeed, few people have altogether. In the midst of the vast literature on Wagner as a philosopher of music, revolutionizer of harmony, redefiner of opera, master of poetry, dramatic interpreter of mythology, and, incidentally, political revolutionary, it is easy to overlook a basic point: Wagner changed the literal sound of music.

Listen in your mind to the opening of Mozart's Eine Kleine Nachtmusik and then to the opening of Das Rheingold. The Mozart is light, staccato, divided into short phrases, treble-oriented (the melody is the highest part, and harmonized from the top down) and, philosophically, clear and rational. Moreover, the music asks for clarity in performance: clear playing, separated instrumental lines, and clear acoustics to help with the first two.

Das Rheingold is massive and so long-lined as to have lines that go on indefinitely ("endless melody"); it is built from the bottom up and it is philosophically mystical. The performance requirements are warmth, continuity, fullness, bass solidity and relatively reverberant acoustics to support the sound.

Of course, this is oversimplified: Mozart contains Wagnerian passages (the overture to Don Giovanni); Wagner contains few Mozartian ones, but Strauss contains a great many. Still, the essential point remains. The Liebestod from Tristan und Isolde, Bruckner's massive sound structures, and Mahler's adagios inhabit a sonic world different from that of the Classic period - Mozart and Haydn-and to a great extent, different from that of Beethoven, too.

The Beethoven Fifth has the philosophical character of Romanticism heroic passion, extreme intensity and extreme contrasts. But its sound remains based in staccato-accented rhythmic structures, far more so than, say, a Bruckner symphony. This shift in sonic emphasis has vast implications for how the music should be recorded and reproduced. These implications and how they have been largely ignored or misunderstood will be my subject here.

To a great extent, the musical changes I am talking about are written in the scores. The massive orchestrations, the abundance of bass instruments - even the invention of at least one (the "Wagner tuba") - and the long melodic lines propelled by complex chromatic harmony are all literal parts of the music as written. But, in addition, one can hear the changes in the acoustics of the places where the music was intended to be played.

THE RISE OF THE RINGING HALL

Exactly how music was performed in the centuries before recording existed is not direct1y knowable. But how the concert halls themselves sounded is. For one thing, many of them are still around. For instance, the hall still exists (in Schloßsha Esterhaza) where Haydn's symphonies were performed during his long employment as composer to the Esterhazy family. It seats 200 and has a reverberation time of 1.2 seconds.

Of course, that was a private hall, in a sense. But the Leipzig Altes Gewandhaus, built in 1781 for public concerts, also had the same short reverberation time of 1.2 seconds.(The hall no longer stands, but the reverberation time can be calculated from the plans.) It held 400. This hall was renowned for its acoustic excellence - from this and other examples, one can deduce that something like 1.2 seconds was considered ideal during the Classic period.

Contrast these with the (fortunately) still standing Grosser Musikvereinsaal in Vienna, then and now regarded as the ideal for the performance of the late Nineteenth Century symphonic repertoire. It holds 1,680 and has a reverberation time of 2 seconds. This change from 1.2 to 2 seconds represents a fundamental difference: the former is dry, clean, clear - almost dry enough to function well as a speaking theater; the latter sounds truly reverberant and supports a warm, full sound. That the reverberation time in the bass is greater than in the midrange and much greater than
in the highs emphasizes the natural warmth of the High Romantic orchestra's sound.
The hall functions in effect as a bass boost "tone control" and a high-frequency reduction filter.

In actual concert hall listening, it seems to the listener that most of what one hears is direct sound and that hall ambience is somewhat secondary. So strong is this illusion that some people believe that concert hall acoustics is an unimportant subject.1 The illusion that the direct sound predominates arises from the fact that the ear/brain determines location from first arrivals. Thus the apparent location of instruments tends to be where they actually are, and the sound seems to come directly from its source.

The truth is entirely different. The direct sound is less than half the total sound in any distance larger than around six meters in a typically reverberant hall of usual size (20,000 cubic meters in volume, two seconds reverberation; the exact distance varies with the volume of the hall and the reverberation time). And, at greater distances, the reverberation completely predominates over the direct sound.

Audio people have a blind faith that the "precedence effect" for location - that first arrivals determine location -- applies also to tonal balance. This enables them to pretend that speakers that are anechoic-flat will sound neutral in rooms. However, this is definitely not true, and it is especially not true in lower frequencies -- say, below 500 Hz -- where reflections arrive before even a complete cycle or two of the direct sound has arrived.

In concert halls, it takes longer for reflections to arrive, usually around 15 milliseconds. This longer interval has the effect of keeping transients clear. (Since the reverberation has almost no high-frequency content, transients would be greatly dulled if the steady-state frequency response of the hall determined the perceived character of transients.) A cymbal crash always sounds like a cymbal crash! But, for sustained sound, the hall considerably takes over. And sustained sound is what the music of High Romanticism is about.

What all this adds up to is that, at any plausible audience location, the hall is contributing a lot to the tonal character of what you hear.(In Wagner's own place, the Bayeuth Festspielhaus, one hears only indirect sound from the orchestra, no direct sound whatever, because the orchestra is shielded from the audience by a roof over the orchestra pit!). The sound up close, where microphones usually are, is thus very different from the sound where listeners usually are (and should be). The up-close, direct sound is much louder compared to hall sound than is the case farther back. The audience usually hears a sustained sound with more bass and always with much less high-frequency content than do close-up microphones. The roll-off starts around 4 kHz, sometimes as low as 2 kHz. And, by 8 kHz, the roll-off is severe, and increasingly so as frequency rises. Remember: beyond 20 feet from the instrument, most of what you hear is hall sound, with almost no highs and slightly --but only slightly -- reduced presence.

THE MUSICAL SIGNIFICANCE OF TONE

The rise of Germanic High Romanticism was associated with an unprecedented interest in the literal sound of music. Beautiful voices had always been admired and exquisite instruments cherished. But the musical message of High Romanticism was embodied, more than ever before, in the sound itself, especially in tonal colors and tonal beauty. Richard Strauss always commented on his own compositions in terms of how well he had scored or orchestrated them. Franck, asked how he felt his symphony had turned out, replied, "It sounded well." And the quotation from Rimsky- Korsakov that AQ cited in Issue 108 (p. 99) makes the point explicit that the orchestral sound is a major part of that type of music, not a laid-on afterthought.

There is a clear contrast with the well-known fact that Bach's Art of the Fugue has no specified instrumentation. And indeed, the Art of the Fugue is as magnificent for four saxophones as for the organ or a strong quartet. (Actually, the saxophone version is particularly compelling, even though it is of course not a possibility Bach could have envisioned.)

The works of the Classic period have a similar, if less universal, mutability. Play the famous slow movement of the Mozart G Major Violin Concerto on the piano, add an "Alberti bass" left hand, and you have a more than presentable Mozart piano concerto movement: the music is hardly tied to the specific instrument at all. Try this with, say, the Bruch G Minor Violin Concerto, and the result will be disaster -- a violin piece played on the piano.

For the purely orchestral works of High Romanticism, the situation is even more extreme. Bruckner symphonies are perhaps imaginable on the organ to some extent, since an organ can approximate sustained orchestral sound. But any other alternative would lose much of the music.

Ironically, it was during the mid- to late Nineteenth Century that piano transcription rose to new heights. Recordings were, of course, nonexistent and live performances were complex, involving large forces. Piano transcriptions were the only way to hear the music frequently or in a domestic setting. But enormous effort was made to transcribe the sonic effects as well as the melodic and harmonic content. It was always clear that the real sound should be realized in the transcription as much as possible.

The results may sound in most cases like curiosities to our ears, but they were serious efforts to reproduce the real sound! The defining character of the sound of German High Romanticism was sustained tone - transients and percussion played little role. Of course, the initial transients of sustaining instruments have considerable importance for their perceived tone quality, as everyone who has played a record backwards knows. But still the fundamental sound of Wagner and the rest is the sustained, flowing, evolving legato. And the beauty of this sound is the beauty of sustained tone.2

A number of interacting factors were involved here. First of all, High Romanticism is intensely serious. And seriousness has always been associated with sustained sound, in the Classic period, too, and earlier even. In Mozart's Magic Flute, the gravitas of Sarastro and the Speaker is conveyed by their singing slowly and legato. Bouncy, cheerful Papageno sings fast and stacatto. Similarly, intense seriousness tends to ask also for a deep bass foundation. The serious intentions fit together flawlessly with the ringing, sustaining and bass-emphasizing concert halls.

Secondly, the ear/brain mechanism is far more sensitive to tonal distinctions when notes are sustained. Tonal beauty is thus intrinsically a more natural part of legato music. Perhaps this thought did not occur to nineteenth century composers explicitly, but it is even so of vital importance in considering reproduction of their music.

THE REPRODUCTION OF TONE

The principle of sensitivity to steady-state tone can be made explicitly convincing by considering evaluation of speakers, which differ quite largely in tonal character. A hard transient -- say a snare drum "rim shot" -- will tell little about a speaker's tonal character.Unless the speaker has severely curtailed highs, the rim shot sounds like a rim shot. One hears the tonal character far better with sustained sound -- say, slow organ music or even pink noise.

One can demonstrate the point by experimenting with an equalizer. A small shift in the midrange -- say, a third-octave wide -- will have a dramatic effect on pink noise or organ or legato orchestral sound. It will be almost inaudible on percussion, by comparison.

Here we encounter one of the great myths of audio, that transient behavior is the best indicator of musical quality. Ironically, it is quite true mathematically that perfect reproduction of impulses requires perfect reproduction of everything else (in a zero-distortion system, free of non-linearities). But, in listening terms, the ear/brain is more sensitive to sustained sonic errors.

There is in fact evidence accumulating that this is true even of phase behavior, not just frequency response. In visual, graphical, mathematical terms, phase non-linearity has dramatic effects on the shapes of transients. But these effects have very limited audibility. What is more audibly obvious by far, according to recent evidence, is the effect of phase non-linearity on complex sustained music, e.g., choral singing (c£, the research of the audio group at Essex University).

THE PROBLEMS OF PLAYBACK

The steady-state, sustained nature of the orchestral sound of High Romanticism and the musical importance of getting the sound tonally right combine to make the reproduction of this music extraordinarily difficult. Of course, in the early days of audio, the relatively simple problems of large dynamic range and full frequency extension seemed themselves insurmountable. But nowadays, with high-powered amplifiers and extended frequency and dynamic ranges of speakers, these difficulties are essentially under control. And CD has enough dynamic range, at least in principle, although just barely.3

The most fundamental issue is that it is extremely difficult to get the tonal balance exactly correct. On broad-spectrum sustained sound, wide-band errors of as small as 0.5 dB make obvious changes. Even 0.1 dB can be audible under certain circumstances.

And now we have to face the music, as it were: an actual speaker in a typical listening room - even an excellent speaker in a good, well-treated room - is doing extremely well to attain, say, ±2 dB performance over 60 Hz to 8 kHz. If you have a look at honest in-room measurements (e.g., in Don Keele's admirable reviews in Audio), you will see that the actual in-room response of speakers is usually all over the place, with wild peaks and dips.[Note added April ,2006: see ,however, the steady state response of digitally corrected systems or in some cases sufficiently well set-up systems with at most analogue bass EQ shown in the "Speakers in Rooms" section of this website.-- REG]

When the chips are down, this is, I think, the basic reason that musicians largely do not believe in audio much: in practice, the better, the more recent and the more strongly reinforced your memory of the literal sound of music, the less accurate audio is likely to sound to you. In my experience, if one plays a truly accurate audio system (and truly accurate recordings) for musicians, they are fascinated and deeply impressed. It is just that most audio systems and recordings are not good enough. Like the Scotsman who would laugh at jokes if they were funny enough (but none of them was), musicians seldom bear an audio system that seems good enough, because few of them are.

Another irony along the same lines as the increased audibility of phase in steady state arises here. The audio world is fascinated with "waterfall" plots as indicators of good transient response. Of course, mathematically, a "waterfall" is a representation of transient behavior. But in audible terms, it is the sustained sound that is most affected by problems visible in such plots. The transient behavior in the technical sense is most easily heard by listening to completely non-transient music!

The resonant "ridges" in waterfalls are heard as colorations. The low-level chaos of energy stored over whole bands of frequencies, not attached to discrete resonances, is heard as fuzz or lack of clarity. It is easy for a speaker to have transient "snap" -- all it needs is extended highs. But for clean, pure, uncolored sustained tone, a speaker needs to have truly clean transient behavior in a much stronger sense. (Just try to hear upper midrange resonances with, say, a snare drum rim shot - there is no real hope unless the resonance is enormous. Listen to a boy soprano, and the resonances will stick out like sore thumbs. Only in the bass are resonances strongly audible on transients.)

A certain amount of [perhaps unintentional] obfuscation exists here in the audio press in general. Those highly irregular "room response" curves are far better correlated with what one hears than the nicely flat anechoic curves that manufacturers and (most) techno-reviewers like to exhibit. This is especially so below, say, 1 kHz, where room effects often lead to wideband peaks and dips. This whole business is something that almost everyone knows about. But it is seldom discussed with the emphasis it deserves. (No boat-rocking allowed.)

A particularly conspicuous instance is the floor reflection. While carpeted floors can absorb high frequencies, all floors are strongly reflective below 500 Hz, say. Yet with a few exceptions (e.g., Gradient), almost no speaker designer builds into his speakers a response correction for this inevitable reflection. As a result, almost every audio system has large and broad-band irregularities in response between 150 and 500 Hz.

This ruined frequency range is, incidentally, the heart of orchestral music, being the octave above and the octave below middle C. No wonder orchestral music reproduced usually sounds very odd to anyone with a vivid memory of its real sound, even supposing you could find a half-decent recording of it to begin with. (Meanwhile, one keeps seeing reviews wherein the reviewer is trying to improve the sound of a speaker under review by, say, changing wires. No wonder people think High Enders are crazy.)

Incidentally, while DSP correction (digital signal processing, e.g., that of the Sigtech) is far superior to analogue EQ in that it allows very precise correction beyond the possibilities of analogue systems, even old-fashioned analogue EQ can be better than nothing if it is used judiciously. Most speaker systems in a room are so far out of balance in broadband terms that a little analogue EQ of octave bands can improve things a good deal.[cf. the "Speakers in Rooms" articles on this site, as noted]. What analogue EQ cannot deal with is abrupt peaks and dips (from reflection cancellations and reinforcements over narrow bands.). For this, one needs DSP. But the mania -- and the word applies -- for signal path "purity" led people to deprive themselves of a potentially helpful use of judicious analogue EQ in the pre-DSP days. Thus have we been treated to such unedifying spectacles as, say, people putting supports under the cables or gluing dots on the amplifiers of systems that have 5 dB errors an octave wide.

GETTING IT WRONG FROM THE START: THE AMERICAN WAY

The tonal nature of Romantic orchestral sound seems, at first, an easy thing to get right in recording, even if it is hard to get right in playback. A flat-response microphone set up at a plausible audience location should do the job.

And it actually will. One hears a good deal of platitude about how "microphones are not ears." While this is true at the level of subtle matters of space perception, it is also true that, for instance, a Blumlein mike set-up played back accurately is extremely close to getting tonal character exactly right.

So how did we end up with so many bad recordings and so few good ones? Why do almost all recordings produce over-close woodwinds, exaggeratedly loud and bright violins (usually pushed hard into the left channel), insufficiently full brass and excessive instrumental noise from all instruments? (When was the last time you heard a lot of bow noise and key clicking at a concert?)

The roots of the non-concert-sound, pseudo-high -fidelity recording lie far back in American musical and sonic history. It can be plausibly argued that Americans never really understood how Germanic Romantic music ought to sound, even though they embraced this music with enormous enthusiasm. The American rejection of Mahler and later Furtwangler in favor of Toscanini can be seen as symbolic of a technological approach to music completely at odds with the spiritual truth of Romantic tradition. And the American habit of building oversized concert halls has unquestionably led to a style of playing that emphasizes sheer volume of sound some sort of musical embodiment of the "bigger, stronger, more powerful is better" syndrome. (The very coining of the term "power music" as the theme of this issue would reflect a profound misapprehension unless it is interpreted to mean strictly emotional power.)

On the strictly sonic front, the mid-century American misapprehension of Romantic music was made explicit by two related developments in the 1930s: the appearance of the dry concert hall, emphasizing direct sound and subverting the ideal of the Nineteenth Century; and the development of recording techniques designed to reproduce direct sound only.

The first of these developments was the brainchild of acoustician F. R. Watson, who postulated that what was wrong with concert halls was that they had too much reflected sound and that the real sound of music was the direct radiation of the instruments. Idiotic though this idea is, it struck a chord in the American psyche beginning around 1925, and for several decades thereafter. The mid-century American sucker-dom for pseudo-science was aroused, and a surprising number of concert halls completely antithetical to the sonic ideals of Romanticism were built -- specifically as sites for playing Romantic music, as it happened, Romantic music being then as now the mainstay of American symphonic repertoire.

Around the same time, Bell Telephone Laboratories[now Bell Labs] developed an equally misguided idea about how to record music in stereo. On their own terms, the Bell Labs experiments made sense: the project was to reproduce music for playback in an auditorium through spaced, close-up microphones and, on playback, correspondingly spaced speakers fed by the corresponding microphone. This worked quite well. But when applied to reproducing music in a home environment, the method involved a fundamental fallacy.

The method worked for playback in a concert hall, because the hall where playback occurred supplied the needed ambience. But in a living room, which would not supply appropriate room sound, the result was and is a travesty, since Romantic orchestral music was written assuming that it would be augmented and sustained by concert hall ambience, and modified in tonal balance.

The connection between the two developments is clear: Watson convinced everyone that direct sound was what ought to be heard, and Bell Labs told everyone how to arrange to hear it at home.

Of course, these developments occurred before high-fidelity recording was possible. But when the LP recording and tape recording showed up - media with essentially full frequency range - the ideas were in place for a completely wrong-headed approach to the use of this full frequency range.

For decades, it never seemed to occur to recording engineers that perhaps they should try to reproduce what the audience would have heard. "High fidelity" became synonymous with "presence," "impact", "brilliance", "dynamics" - never mind that none of these was really a feature of orchestral music as it should be.4

One might have hoped that the rise of serious, High End audio would have reversed this trend. But, in the first two decades at least, High End remained obsessed with presence. Even now, close-miked female vocals are used to demonstrate supposed "realism" at audio shows - accompanied, of course, by drum sets and close-up instruments. And enthusiasm for the close-up recordings of the 1950s and 1960s has been sustained.

Sanity shows signs of prevailing. Things like, say, the Dorian Korngold recording [DOR 90216] or, especially, Fenby Conducts Delius on Unicorn [DKP 9008/9] have quite plausible (and consequently beautiful) symphonic sound. As the obsolete idea vanishes that the problems of audio center on getting enough high frequencies - an idea born decades ago and out-of-date for almost as long - hope is in sight.

THE PROBLEM OF SCALE

Orchestras are large. A full Romantic orchestra can easily occupy an area 30 by 60 feet. From a close-up position, the overwhelming impression is of large spatial extension. Without artificial signal processing, this impression cannot be reproduced in a home environment in detail. And, no matter how good its "soundstage", a home stereo system always sounds miniaturized, if the standard is the spatial impression of reality from, say, the conductor's perspective. From the audience's perspective, the situation is entirely different. From the twelfth row, if you close your eyes, the side-to-side extent of the orchestra is within an angle of 90 degrees or less, and literally perceived depth consists mostly of some general impression that things are some distance away.

This audience spatial impression is easily reproducible. In fact, the angular extent seems to happen almost automatically. (From close up, the angular extent of an orchestra is almost 180 degrees -- try to do that with your phase-driven, outside-the-speaker images.) But this spatial impression is utterly inconsistent with a close-up tonal perspective.

The point is that if instruments sound tonally close-up, then it is extremely unnatural for them not to seem very large too. If a concert grand piano is recorded so close that one hears hammer tone and action noise and the bright, aggressive tone they have close up, then it also needs to sound nine feet long. And if it is the solo portion of a piano concerto, then the orchestra needs to sound 50 or 60 feet wide.

Consider, for instance, the famous Byron Janis recording of Rachmaninoff's Piano Concerto No.3 on Mercury [SR 90283]. This is a wonderful performance and a musically moving experience. But as a recording, it is quite unnatural. The piano sounds very close and vivid, and quite large. But the whole first violin section is jammed almost into the left speaker, and whatever little extra extension one could get outside the speaker boundaries is far from the 50 or 60 feet of reality.

Ironically, considering how resolutely recording engineers used to refuse to back up, the same distance that makes instruments tonally natural solves the space problem. In a concert, the piano would be small, the orchestra larger but not too large (in angular extent) and a possible reproduced sound recorded at a distance resembling real music.

Yehudi Menuhin once described concert hall design as a "solved problem" - all one had to do was copy the old masterpieces. Recording is not exactly a solved problem to that extent. But a logical perspective, recording at an audience location, would certainly move things in a consistently promising direction.

Some things people do not do because they can't. Others, they do not seem to want to. The failures of recordings are in good part failures of intention, not of ability.

Another historical note is relevant: in an attempt to make their over-close spaced-microphone recordings not sound intolerably dry, recording engineers of decades past liked to record in environments that were too reverberant to be suitable for listening -e.g., halls that were meant to be full of people would be used empty or orchestras would be recorded in churches. This reinforced the too-close microphone techniques, because distant miking, especially with spaced microphones, would produce an intolerably "swimmy" sound in such an environment, or so people believed .(It was often not really true!)

For solo instruments suitable for close-up listening, this technique can work superbly (e.g., Water Lily's small ensemble recordings, and Dorian's piano recordings). But for orchestras, it is unsuitable.[Note added by REG, April,2006: In the meantime since this article was written, Waterlily has produced a series of superb orchestral recordings-with miking much more distant than the small ensemble recordings referred to].

Incidentally, many audiophiles would be shocked at how close, even today, the microphones usually are, if they were to attend some recording sessions, even "audiophile" ones. In trying to reproduce orchestral concert reality, we are usually starting with an unnaturally close perspective supplemented by an unnaturally reverberant environment, but with the reverberation reduced in relative level by the close-up microphone placement. (We won't even talk about the pop "audiophile" items, except to note that their vastly exaggerated "presence" tends to force people into playback systems that do not have intrinsically as much presence as they should, so that natural recordings sound too distant and "laid back.")

HOW THE FUTURE COULD BE

Audio has been something of a failure until recently at reproducing the sonic ideal of the music of High Romanticism. But it is at last beginning to offer the possibility of success.

The appearance on the scene of speakers that operate cleanly and free of resonance over the whole frequency range, and the arrival of DSP (it la the Sigtech AEC 1000)[and other such systems since: noted added April ,2006] to detect and correct room problems opens a new world, potentially. At the same time, recording engineers who are not mired in old ideas are beginning to produce recordings that are realistic in the true sense of offering something very like concert sound.[e.g., the Water Lily Russian recordings, note added April 2006]. The whole thing could start to work.

Of course, there are still some problems, ranging from the technical to the aesthetic. Conspicuous among the aesthetic problems is that the ubiquity of rock music, with its grotesquely exaggerated high frequencies, habituates people not only to some extremely bad music, but also to excessively bright sound, so that natural symphonic music can sound dull, to mention one depressing item. But at least the possibility exists that the future will be better than the past. As new generations come to appreciate the possibilities present in contemporary audio technology for reproducing music with true realism, and to see the obsolete and misguided "presence" concepts of 1950s "high fidelity" for the aberrations they were, progress will occur. The prospect lies before us of reproducing the true sonic glory of symphonic music.
________________________________________
REG

1 Otherwise sensible people have actually told me acoustic judgments are an illusion, and that substantial differences do not exist, on the grounds that controlled comparisons of halls are not possible. Fortunately, halls with adjustable acoustics have laid this particular piece of nonsense to rest.

2 Among the composers mentioned, Mahler is a partial exception, in his interest in transparent textures and detail. But in his adagios, he reverts to the legato completely.

3 In actual practice, recording digitally presents a problem with the 96 dB dynamic range, because it is necessary to avoid any overload in digital. One needs around 20 bits (about 120 dB) to be safe.

4 Orchestral music is moderately loud on occasion, although "moderately" is the operative word. (See HP in Issue Il, p. 301.) But it does not sound "dynamic" in the sense of audio because of its low levels of high-frequency content.

Hi Zonees

Been testing varying amounts of reflection and burn in my system to hear the effects and plan more tuning steps. Here's what I found:

a. At this stage my system is very sensitive to small changes in reflection or burn.

b. Increasing reflection or liveness in a room is not the route to build ambience recovery. The only ambience and reverb you should ever hear is whatever is recorded in the CD and LP. Anything else is distortion. Too much liveness can give a false echoeyness in the room and obscure the real ambience with a fake ringing. OTH a dead room swallows up low level details in the signal that includes the ambience. So it is a balancing act.

c. More liveness in the forward half of my room causes ringing and after testing, what I got now is about right.

d. The rear of my room is more complicated. Michael Green plastic on the rear of the FS-PZC behind the listening chair makes things too bright and ringy.

e. The best music to test the right amount of liveness seems to be bluegrass -- Martin guitars played fast. The notes must be clean and not smear. Mandolins too. These instruments with their high midrange content and transients right where our ears are most sensitive makes this a telling trial. Too much burn dulls the notes. Hurdy gurdyies work too and so does the barogue violin but a listener needs to know what these instruments sound like to make the test work -- the edge from the gut string needs to be "just so" and the jangle from the hurdy gurdy must not obscure the tune. Since I heard these instruments live so infrequently, Sonic will stick to the acoustic guitars.

f. In Sonic's system, I found that balsa wood isn't a good thing. It damps the sound in an odd way. The transient spike is OK but the note is followed with a dullness that is odd -- like a time delayed burn. Not pleasant. I discovered this with a test of supporting some transformers on balsa and ended up with all balsa in the room including two Shutters retired.

Sonic is still listening through the weekend. Let's see what else I learn. This is great fun....but methinks sometimes "will the day come when the listening is a stream of music for its own sake and not to seek this goal of tuning more and more...?"

Should I start collecting vintage tubes for a change like old GEs and Telefunkens, or is that another obsession?

Speaking of streaming, best I start planning for computer-based front ends. Michael, what is that about batteries?

The day of buying FLAC-formatted classical, jazz and specialties music via internet is closer than Sonic thinks. Record (LP) stores are growing. I won't touch tapes because many old formulations were made from whale oil. Sonic is not going down the tape route for this reason. And CD stores are going the way of dinosaurs.

Sonic

Hi Michael and Zonees

Sonic is still experimenting with different amounts of reflection in my room. More absorption is a no-no as you have seen from my earlier try outs. So it is reflection we try.....

One thing Sonic has not tried is to change the reflectivity of the window area. This is a 10ft x 6 ft picture window all glass, covered by a pair of cedar slatted blinds.

In this configuration, the area has never drawn attention acoustically and gave better centre imaging compared to a heavy curtain that hung from the top of the window (the curtain rail you see) to the floor. But this was at least 7 years ago.

I had enough plastic sheets from Michael to cover about 30% of the blinds. I used two pieces 4 ft long by just less than 2 ft wide. Clipped them to the wooden slats, each plastic sheet covering the mid point (width and height) of each of the two slats.

Played music.

The whole soundstage became like a fisheye lens.

Everything in the middle was pushed forward and enlarged. Yet the overall image height was about the same.

The tone went crazy -- there was an increase in volume but in an uncontrolled way. The midbass was uneven -- some instruments were distorted in tone and in a drum roll a tom at a certain frequency could sound explosive DOONG!!!

Male voices had too much grunt. The images at the speaker positions and beyond were shrunken in comparison to the centre.

Also a sense of exaggerated ambience -- some small scale violin works (barogue ensembles) sounded like they were recorded in a bathroom with the sound of a reverb machine mixed in.

"This is awful...all wrong" said Sonic....I probably switched some Pressure Zone on and it got out of hand. Of course energy like this can be tamed by Michael's gear and might even take my system a step higher.

But Sonic ain't got no more Tune gear. I also don't know where to start even if I did. So down came the plastic sheets and I get on with listening to what sounds nice again. Whew.

Something to think about but given the musick I now get there is less motivation for me to get into what could be starting over on a new path in tuning this room.

Sonic

Hi Zonees

Michael asked Sonic to describe the difference in sound between the mild steel anodised rods and small MTDs used for my X-30 crossover miniclamp rack and 1/4" and 3/8" resitone rods.

The miniclamp rack is 10" x 6" cherry finished MW. Very nice it is.

The resitone rods sound less metallic and more dimensional (like real musick) compared to the metal rods and MTDs. Now the mild steel rods and MTDs don't sound metallic on their own but against the resitone rods they sound less fluid, more mechanical.

Now my resitone rods were sharpened to a point. The 1/4" rods sounded better than the 3/8' rods, more free and relaxed and clear. But a sharpened 1/4" won't take any weight without the tip bending. Not even the weight of an X-30 with the cover removed will bend the tip and cause the clamprack to lean. The 3/8" rods do take the wieght nicely but they don't sound as airy and musical.

A pity since the X-30 is really lightweight. It is supported in the miniclamp by Michael's handwound harmonic springs with MW splices and bars. The resitone rod spikes rest on MW pieces 1" x 1" x 1/4".

Sonic has not tested resitone rods elsewhere in my system due to a lack of MW pieces and insufficient rods of the right length of weight-bearing capacity for other gear.

Sonic