The beatles please please me

Every Beatle album should be cut for full band width if it’s actually on the tape.

 

But that would mean there is a production master from which the UK was cut, wouldn't it? IIRC, they did not create production masters at the time, as mentioned in the other thread. They cut straight from the the "actual mix compilation" with the slipces. Is that right? @lukpac @slane ?
IF that is correct, Germany would receive a copy of the tape, then they would (or would not) make their adjustements. At the same time Abbey Road can cut from the original tape while making their own adjustements on the fly.
German cutting can very well turn out to be more clean/hi-fi even if it's 2nd generation compared to the UK. Cleaner cutting chain, less twiddling with knobs etc.

Ondra

 

Correct.

Of course, as noted, once you match EQ, Die Beatles isn’t any more dynamic.

 

Although I know you prefer CD or digital...but for great sound I'd go with an analog stereo cassette on a good cassette deck. You can always transfer to digital and then edit out the songs you don't like:

 

No argument there. But you would be surprised at how many cutting engineers filter out the frequencies below 30hz....40hz....and even 50hz.
Why? They think it will make it sound better. Secondly, it is common mastering practice to high pass at 50hz. Go on Utube and you will see "experts" advising the amateurs to cut everything below 50hz from their masters. No, I don't believe it either but it's true. There is even the real stupid secret mastering practice of "Overs."
This mastering engineer admitted that if the client asks for it (talked into it.) the odd very brief digital over is permitted - if it briefly goes over 0dbfs. Digitall distortion does not sound nice and should be avoided at all costs. But they figure if it's real quick and you don't hear then it's O.K.

This nonsense has even entered the recording industry. In a Utube video a "professional" recording engineer in a big major studio was saying how digital distortion on drums sounds great. Anyone hear how heavy metal sounds more distorted in the last decade? That's not not because the amps are real loud. Nope! That's the engineer purposely setting the record level on the drums to distort. No, i am not making it up.

 

The odd brief digital over won’t hurt a thing - it’s inaudible.

 

I have a NM copy of the West German Apple pressing, the same mastering as on the second Hörzu pressing. I did a needledrop of it some years ago.

That this mastering is less compressed is a myth that unfortunately is still being repeated today. It has exactly the same dynamic range as the regular UK pressing. I measured it.

The difference is the EQ.

 

Seems to be confusion about those dates.

The first German Beatles studio LP was With The Beatles, both stereo and mono versions came out in November 1963.
"The German tape dated November something of 1963" would be for WTB, not Die Beatles, as WTB was their only studio LP release that year.
Obviously sales were good enough to order the then back catalogue PPM album be produced.

The German Die Beatles stereo and mono versions first appeared in February 1964, some 10-11 months after their UK counterparts.
The stereo PPM master tape box notes that it was first sealed on 1st January 1964.
It was then reopened a short while later between 9th-17th January which is likely when the tape copy to Germany was made after the initial success of WTB weeks earlier.
That "tape copy stamp" in January 1964 fits in perfectly with the subsequent February release date.

Incidentally, the first pressing of Die Beatles is totally underwhelming despite the fabled uncompressed mix that was supposedly used, it's only the later 1960s 2nd cutting that has the sound.

 

Except that there are no signs that this special tape was used in 1963 (1964?) for the first pressing. The original Hörzu pressing is nothing spectacular. They apparently only started using it in the late 60s (?) for the second and subsequent pressings (the legendary A2/B2 cut). So that makes it even more mysterious.

But to expand on what you were saying, I do believe the Germans at that point had some leverage as to what tapes they were getting from the British. The German Electrola arm of EMI was quite influential—let’s not forget that it was an Electrola engineer, Peter Burkowitz, who designed the REDD mixing console that the Beatles started using during sessions for the 2nd album. And yes, stereo was quite big in Germany at the time, and mono to be phased out entirely within a year, ahead of anybody else.

 

Another problem is that the stereo and mono mixes that were done simultaneously (9 of the 14 tracks) match in terms of compression. If the stereos had then received extra work they would not match the monos as closely as they do.

Those 9 songs were one mix going to two machines (one in wide stereo, the other a collapsed mono version of that mix). If the stereo mix was originally 'uncompressed', then the collapsed mono mix would be too.

 

it depends on who you talk to. Some audiophiles are hearing it and have complained. But this practice is secret. well, not now. As for the "casual" over I guarantee your definition and theirs is very different. In recording you shouldn't be anywhere near 0dbfs. Yes with digital you can go right up to zero and level won't overload. Well, the digital level won't but the analog section will. Every A/D converter has an analog preamp in it and that overloads WAY BEFORE THE SACRED DIGITAL ZERO. The analog preamps start overloading over - 6dbfs. People who work in the industry know that 0VU does not equal 0dbfs. This is a fact. When professionals record they average out the signal at -18 dbfs (RMS) average level not peak. In Europe it's -20 dbfs (RMS). This where 24 bit professional converters are designed to sound their best. Easy to remember for 24 bit recording, 0VU = -18 dbfs (average not peak). This is A NORTH AMERICAN STANDARD. Some want in lower.

Now, once you get the signal into digital you can normalize the volume up. And you can raise it right up to 0dbfs since you are not dealing with an analog section anymore. In professional mastering studios they have very expensive digital meters that are calibrated (designed) much better than what the average person would have at home. But, you say, "0dbfs is 0dbfs." NOPE. Most cheap consumer converters take a reading every THREE SAMPLES and not every sample. Three consecutive readings of 0dbfs will equal an over which would be nine samples. Professional Calibrated meters take a reading every sample.

These overs are for clients that want loud disks. If you are just normalizing the volume, why would ask for overs? I suppose if you wanted an extra db of gain without limiting. Except I guarantee you a quick shot of 2db you will hear. No amount of casual overs will give a client even a db of extra volume. Oh, I may have been underplaying, the casual overs. Check the original mix of Vapour Trails - that's CASUAL OVERS. You can hear bass distorting on that.

We will have to agree to disagree gentlemen. There is no excuse for overs. Not in mastering and certainly not in digital recording.

 

Uh, what?

As far as overs go, in the early days of CD, converters actually had switchable clip indicators, so for example they could be set to not go off unless there were 16 clipped samples in a row. That was apparently the case for the version of Who’s Next mastered by our host, and nobody seems to mind.

 

If i went off an tangent (which I do sometimes) there was a reason. I thought it might help everyone with the issue.

Just a question because mood and intent don't carry on the web very well in text...I can't tell - Are you confused or irritated?

Let me explain because I don't think I made myself clear. Recording as high you can in digital just as long as you don't go over zero is what amateurs do. It is what the layman does. Professionals don't max their signal out to zero. Neither does Steve. Again you would be overloading your analog section. Just because there is a clip indicator there doesn't mean Steve is constantly maxing out his digital level. It seems to me you were implying that. If all that mattered was a overload then all we would have is 3 LEDS. One for signal detected, one for zero and one for overload. That's what all the numbers are for. (sorry, not meant to sound patronizing.) Heh, it took me year to find that out. Take a look at a lot of professional recorders. They go up to zero with no indicator for overload. Why? Because you should not be anywhere near there. Look at professional portable Wave recorders. The meter section will often only have a few important numbers on it: -18dbfs, -12 dbfs, -6dbfs, -3dbfs and of course 0dbfs. But no overload. If your recording in 24bit and your signal is maxing to often at zero or close to it your recording will sound harsh, cold, and generally unpleasant because the converter sounds it's best averaging around -18 dbfs and not zero.

Steve's converter is custom made. And I believe the input preamp is all tubes. And when tubes overload they produce even harmonic distortion as opposed to transistors that produce odd harmonic distortion. And I think we can all agree that tube distortion (real tubes not some plug in) sound kind of nice. Again sir, no mastering engineer recording in 24bit would max out his digital level to zero..A mastering engineer raises the level later in a process called normalization. I don't known if Steve does this..Why would he need to? Audiophiles buy his CD's because they sound great not for loudness.

Now on the other hand some Pro converters have a lot of headroom. I mean more than usual. If Steve overloads his analog section he gets more tube sound. If that's what is needed for the project. The fact that Steve has tubes in the analog section of his converters is a major game changer. The figures I give you still apply. However, if he decides (on occasion) to push his A/D converter passed the sweet spots Steve will get nice gentle tube even harmonic distortion and not the nasties everone else will get. Apples and oranges here.

A small note:
Mr. Steve Hoffman is a maverick. (I mean that in a good way guys) The way Steve masters is not the way the majority (probably all of them) of engineers master. He only uses software to edit. I understand he does fade ins and fade outs by hand. And he eq's going in (thank God) instead of the wierd way they do it now. Which is: Transfer the analog tape to digital....Do all the edits, compression, fade in and fade outs, other crap you guys don't want to know about....and then converted to analog again to use the MANLEY MASSIVE PASSIVE all tube passive mastering eq. Every mastering studio worth their salt has to have one. And any other analog stuff they wish to add....And then BACK TO DIGITAL AGAIN. That's 3 conversions! No wonder Mr. Hoffman's compact disks sound better.

I thought the 'analog section of the A/D converter' part was clear. You know where you plug the XLR cables from the quarter inch reel to reel into the converter. Before it gets converted to PCM it is an analog voltage. This section of the converter is called the input preamp. It overloads long before 0dbfs. Professional converters are meant to model analog equipment. A 24 bit converter's sweet spot is - 18 dbfs (RMS, not peak.) Why would that warrant a "Uh??!!" or is this too much detail? Am I going into to much ahhh...Audio engineering tecnobabble?

My apologies. I have been working in the recording business with my Uncle for the last 15 years. I unfortunately take it for granted that most of you will know what I am referring to. You misunderstood which is my fault. Old digital meters were often driven by analog circuits and were not very accurate. Yes a proper mastering meter will have adjustments like that. But just because it's there doesn't mean he used it. Oh by the way for 16 bit you don't max out to zero either. The layman does this. For 16 bit, 0VU = -12 dbfs ( average level, not peak) and you generally not peak over - 3 dbfs. This is the reason why a compact disk that is a flat transfer (No compression or eq.) in 2009 will sound louder than the same album released on CD in 1987. They didn't normalize in those days.

 

Neither.

As far as making ourselves clear goes, apparently it’s I that needs to do so. The version of Who’s Next mastered by Steve in the ‘80s has significant digital clipping. 15 clipped samples in a row is not uncommon on that disc. The point is only that few, if any, people seem to have a problem with what is actually significant clipping. On the contrary, that particular disc is well liked. Mastered on a Sony 1610 at Bruce Botnick’s studio, IIRC.

As far as your statement that the analog section of an A/D converter overloads before 0dBfs, please reference something that indicates that, as it suggests a completely incompetent design.

 

Well stated sir. And I will concede that yes, brief overs may not be heard and won't affect the quality of music. But I concede this point only for 16 bit recordings.

It's analog distortion. And how bad it will sound (if heard at all) will depend on weather it's a brief peak at say -2db or the meter is constantly hanging around -2db. i.e. a bass guitar playing several bars of quarter notes. Apples and oranges.

Do you have access to Pro Tools or a similar platform and U.S.B. audio interface? simple experiment. Set your recording bit rate at 24 bit. Sample rate does not mater. Run a steady signal through (keyboard middle C) the interface until it the reads 0dbu. You will see the recording level on your computer measuring -18 dbfs. (Averaging)
The first time I heard it that was my reaction. But digital and audio levels are not measured the same. Question? (Mine too) Why not retune or adjust these analog sections so that their peaks are the same as digital. That would work for peaks but not for the average level. Digital is perfect at until you get to 0dbfs and then no more. Analog isn't like that. With an analog recording level it's - ok a little unfocused....better...Good...perfect (0VU)...still pretty good but a bit harsh (3VU)....Too distorted (5VU) Needle now pinned most of the time. The analog section of a A/D converter works the same way. There is going to be a level where it is at it's best - 0VU. Music has dynamic range and when you record you need headroom. It would be impossibleto a make 0VU = 0dbfs. You would have no headroom left. The analog section of a professional 24 bit converter sounds it's best at - 18 dbfs RMS (Average level) . This is why they cannot make the converter sound it's best at the digital level of 0dbfs. No more headroom left. This idea about "using up all the bits" is ridiculous.

If Steve was going to peak at -3 dbfs he would have to set the average level at - 15 dbfs which he did not want to do. Better to keep it in the sweet spot and use the the extra 3db of headroom. Back in the day those converters did not do very well at low bit levels. Again, as I have said already - I have no problems with AF or DCC disks. Steve knows what he is doing. Most of these engineers don't. You want to hear an example of allowed "overs." Listen to the ORIGINAL MIX of Rush's, Vapour Trails. You can hear the bass distorting all over the whole album. It's horrible. Yes, the unmastered mix is bad but it doesn't have all that distortion over it. I know I have had the privilege of hearing a demo disk of the original (unmastered) mix a few years ago. Steve uses his ears when he masters. Apparently though the idiot that mastered the Vapour Trails cd did not. He allowed "overs" making a bad mastering job even worse. Most mastering engineers are not like Steve Hoffman. Don't fool yourself sir into thinking otherwise. Steve can do "overs" but you don't want anyone else doing it.

Here is what you have asked for.

PDF file. scan down to the VU meter part sir.
https://www.google.ca/url?q=https:/...BAtwQFggoMAY&usg=AOvVaw2aliZtSqJFX5elaKZjjPE4


A bit longer. A article on proper recording levels.
dBzee: Digital Recording Levels - a rule of thumb


And the video is not long..it's not short either. A well known mix engineer in Rap and Hip Hop.
KM MIX TALK BLOG 5: "STOP RECORDING SO HOT!" (www.KyrenMonteiro.com)

I freely admit when I am wrong. How can you learn if otherwise? I would hope that other members would do the same.

 

Digital clipping is digital clipping regardless of who is doing the mastering. Slight clipping will usually go unnoticed, while more severe clipping can cause very audible distortion. The clipping on Who's Next is very noticeable visually, but I don't think I've ever *heard* it. On the other hand, I have discs with severe clipping that is very audible.

As far as scales go, dBFS is the only absolute: 0dBFS is the highest you can go in digital, period. Trying to go higher results in clipping. The same isn't true for analog, however. 0dBVU means different things depending on if your standard is -10dBV or +4dBu, and not all digital interfaces have to be calibrated the same way. In addition, unlike digital, it's just a *reference level*, analog has headroom above "0". Analog equipment won't clip until around +24dBu, which is usually equivalent to 0dBFS.

Now, there's no point in recording at higher levels than you need to, especially with 24 bit where noise is less of an issue. But it isn't true that the analog portion of an ADC (and/or the gear in front of it) overloads well before 0dBFS, as is suggested above.

 

i would prefer the 2 box issues pre 1974, they are usually on better quality vinyl.

 

Sometimes on better vinyl. Mostly insignificant differences.

 

agreed mostly insignificant but the risk of surface noise is less on the pre 74 2 boxes so for those where surface noise is more an issue and hate any noise between tracks look for gramophone co rim text not emi records or all rights rim text copies.

 

Pre-74 can have noisy vinyl too. Not worth the chase IMO - all other things being equal.

 

there's not too much in it but i had a straight choice over a pre 74 and post 75 ( there was a hiatus in UK Beatles pressings ) in the same condition i'd go for the pre without hesitation. They are the closest to an original you can find on a budget.