[geoffmartin]

Cool! Thanks!

[geoffmartin]

Took me a while to find it, but here’s the article by Glenn Gould that I was thinking about earlier. I haven’t read it in at least 25 years…

The raw text version: The Prospects of Recording

A PDF copy of the original magazine article (starting on page 46)

It’s a long read, but notice that it’s from High Fidelity Magazine, vol. 16, no. 4, April 1966 – back in a time when such magazines contained more than just advertisements and reviews…

One fun quote (among the many in there) is the following:

I can’t believe that people really prefer to go to the concert hall under intellectually trying, socially trying, physically trying conditions, unable to repeat something they have missed, when they can sit home under the most comfortable and stimulating circumstances and hear it as they want to hear it. I can’t imagine what would happen to literature today if one were obliged to congregate in an unpleasant hall and read novels projected on a screen.

– BABBITT

[geoffmartin]

matador wrote:

Isn’t this part of the “full Unpluged Experience”? Like if you where there sitting on the carpet?

It’s not an acoustical sound, it’s a mechanical vibration. So, in order to hear it the same way it’s captured on the recording would be to lie on the stage floor with one ear mashed into the carpet.

Certainly an experience – but probably not desirable.

However, other noises like finger squeaks are definitely used in the mix occasionally. I once did a CD recording where (for one of the tracks) I put two Neumann mics near the neck of the guitar to get a reasonable amount of control not only of how much the left-hand squeaks got into the mix, but where they were placed in the stereo imaging. Listen carefully to that track in the “sweet spot” and you’ll hear the right hand plucking the strings to the left of the left-hand finger movements. (In case you’re wondering, the disc is called “Mona Lisa” recorded by Dennis Pinnock and Bill Coon. Good luck finding a copy… )

Cheers
-geoff

[geoffmartin]

I totally agree with Stan.

I find the opinion that a music reproduction system should sound like the musicians are in your listening room a little weird, since almost no recordings are made with that intention. We don’t watch Star Wars and expect to think that Luke and Leia are in your living room… Movies aren’t made to be like real life – they’re made to be better than (or at least enhanced versions of ) real life. It’s the same for music recordings: a typical classical music recording has the brightness and presence of being close to the instrument(s) AS WELL AS the reverberation and envelopment of sitting far back in the concert hall. In a typical pop recording (these days) the singers are in-tune, and make no mistakes – two attributes that are only possible with computer enhancement. The goal is to make the recording sound like the recording, since the recording doesn’t reflect reality – it IS the reality that your playback system should reproduce.

Glen Gould spoke and wrote about this decades ago – explaining that the reason he recorded instead of playing live concerts was because (in his opinion) the perfection that can be achieved through mic placement, mixing, and editing can never be experienced in real life.

So, in my opinion, Jack Dee can be generalised beyond the Pogues to include all musicians. 🙂 (Thanks for the quote AdamS!)

Of course, if we then take this argument to its logical conclusion, then you might arrive at the opinion that you can’t use modern loudspeakers to reproduce old recordings… I would generalise this though. There are many cases where playing a recording through a “high quality” loudspeaker + system actually makes the end experience worse. Two obvious examples of this are:

• I don’t need a system that has a frequency range that extends all the way into the stratosphere when I’m playing a 78 RPM shellac disc of Caruso. Better to roll off the top end with extreme prejudice to reduce the noise, since there’s no signal up there worth speaking of.
• Eric Clapton’s “Unplugged” album, played through a system with a very low low-end cutoff frequency will give you the ability to hear the sound of a foot tapping, coupled through a stage floor, mechanically shaking a vocal mic and resulting in a rhythmic rumble that probably wasn’t supposed to be there… Rolling off the bottom end cleans this up considerably.

Then again, if you want to hear the shellac crackle and the stage rumble – it’s not my place to tell you it’s bad. As the thread starts, this is preference we’re talking about…

Cheers
-geoff

[geoffmartin]

Stereomensch wrote:

I really would like to know if Geoff Marin uses an old Beovox speaker secretly at home

I don’t. Now you know.
🙂
-geoff marin

[geoffmartin]

Thanks!

[geoffmartin]

Tymphany.
-g

[geoffmartin]

Hi Trackbeo,

Just like the BV11/Eclipse/Harmony, the Theatre gives you the option to not use True Image if that’s what you’d prefer, and just like the Harmony, if you choose Downmix instead, then by using only a 5.x or a 7.x output, I think that you’ll get what you’re asking for.

If you choose 1:1 mode instead, then your output will be a direct reproduction of the input. So (as long as you have outputs assigned to the appropriate Speaker Roles) then the whole up- and down-mixing processing is bypassed – which may also be closer to your preference.

The goal (as always) is to make a system that has enough “handles” to be customisable for your preferences.

 

As for Floyd’s Circle of Confusion: as much as I like Floyd (I’ve known him for decades…) I disagree with his circle. In contradistinction to this concept, there is a reference: whatever the recording or mastering engineer heard in the studio. Take 100 recordings and count how many are made with the intention of making you feel like the musician is in your living room… The number will be very close to 0. (Just like there are no movies that are created to make you feel like you’re one of the cast, or sitting in a theatre watching a play.) So the circle becomes a line, where the point of “truth” is in the middle. Of course, achieving that target is impossible, since no two studios or studio monitors are identical… But this is a different discussion, best had over a pint.

Cheers
-geoff

 

[geoffmartin]

Not in Canada…. 8+8+11 = 25. We work in a hybrid system alternating between hexadecimal and Fahrenheit…

It’s fixed. Thanks for ripping the thin veil off my stupidity… 😉

-g

[geoffmartin]

This may help – at least as a start.

Cheers
– geoff

[geoffmartin]

Hi Sandyb,

Interesting comments. The thing that’s missing is the details of the configuration of the Theatre, which could have a noticeable impact on your experience.

I won’t get into the details of what, exactly, this means, since I’m working on a blog posting about it at the moment. I should be releasing it sometime this week, I hope… 🙂

Cheers
-g

[geoffmartin]

The cha cha is better than the waltz. Once in your head, it’s stuck there for days…

-g

[geoffmartin]

[geoffmartin]

Usually, the audio signal is represented as a change in voltage at the output terminals of the power amplifier. In theory, that voltage is identical at the input terminals of the loudspeaker.

The instantaneous current through the system is determined by the relationship between the instantaneous voltage and the impedance of the loudspeaker.

The problem is that this model is too simple. The power amplifier has an output impedance (which is typically independent of frequency, more or less…). The speaker wire has a measurable impedance (resistance and capacitance – therefore it’s frequency-dependent) that is a result of the wire’s construction and its length.

If the output impedance of the power amplifier is close to (or worse: larger than) the combined impedance of the speaker wire and the speaker (which is highly dependent on frequency), then the voltage at its output terminals is different from whatever it should be. This is because the amplifier’s output impedance is in series with the wire and the loudspeaker, and therefore the system acts like a voltage divider (measuring at the amplifier output terminals).

 

This means that the voltage at the power amplifier’s output will not be correct – but since the load impedance (the wire+loudspeaker) is frequency-dependent, then the error will be signal-dependent (since that’s what determines the frequency).

Therefore, if you have a loudspeaker with a low impedance at some frequency, then you need to use a power amplifier with a MUCH lower output impedance at that frequency in order for it to behave.

So, it’s not really the case that the loudspeaker is “difficult to drive” – it’s that the loudspeaker/wire/amplifer interaction makes the amplifer misbehave.

The simplest way to avoid this problem is to buy a power amplifier with an output impedance as close to 0 Ω as possible. This is a good solution if you have an amp in a rack, and you have no idea what will be connected to it tomorrow. However, if you are buying a “closed” system (an amp, some wire, and a loudspeaker) that won’t change, then you don’t need to spend quite so much money, since you can chose the amplifier helped by the knowledge of the wire+loudspeaker’s impedance.

Cheers
-g

[geoffmartin]

I’m a little confused about the initial problem.

Are you comparing the sound of a song

• played from Deezer on a Moment connected to the BL28s
  to
• played from vinyl played on a Beogram 9000 via a RIAA preamp through the Moment connected to the BL28s

?

If “yes”, then the difference in spectral balance is most likely to be caused by (in order of probability)

• different masterings of the track (Deezer vs. vinyl)
• wear on the stylus
• weirdness in the RIAA filter’s magnitude response curve

Of course, there’s nothing wrong with connecting the RIAA output directly to the BL28s Line Input, since the fewer devices in an audio chain, the better (generally speaking…), but I doubt that this will make a big change in overall timbre that you’re describing.

 

If “no” then never mind… 🙂

Cheers
-g

[geoffmartin]

It’s a good question – but I think that shellac (+ limestone) 78s are harder than PVC 33s. Also, the groove is MUCH wider, so there’s plenty of room for error.

Either way, it’s worth a try. Thanks for the ebay link you sent. This will give me a new project once the days start getting shorter here in Denmark.

Cheers
-geoff

[geoffmartin]

I forgot to add…

> you may not be aware of, but there is a 20 pages “technical sound guide” about beograms

I am aware of it… I wrote it. 🙂

 

EDIT: I didn’t realise it was a joke. I’m a bit thick this weekend, apparently. Sorry… 🙂

Cheers
-geoff

[geoffmartin]

Hi again,

No need to make excuses… I’m genuinely curious! I’m currently in the process of turning two sick 42VFs into one healthy one – but I don’t have many pickups with the cantilever still attached (the end result will have the 20º tonearm – not the later 15º version, and the SP1-2 pickups are hard to find in good condition…) Given the way I’ll be using this turntable (mostly with my 78 RPM disks) a DIY pickup rebuild would not be crazy.

I’ll have a look around for the sapphires – maybe I’ll have to call in a favour from an audio geek friend in Paris to look around the next time he’s at FNAC if I can’t find my own around here in the fields of Jutland… 😀

Thanks for the inspiration! Nice work, in my opinion!

Cheers

-geoff

 

[geoffmartin]

Too bad you’re in France and not next door. I’d be interested to measure this to see how good/bad it turned out!

On the other hand, I have some broken cartridges lying around. I might just try to duplicate your results by making some franken-pickups of my own… 🙂

I’m curious about the “chinese bulk saphirs”…

Cheers

-geoff

[geoffmartin]

I use my 42VF often – but that’s because it can rotate at 78 RPM…

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