Momentary Lapse of Reason Recording Question??

My best guess is that the original recording was analog and a digital recorder was used during mixing and mastering (ADD).

Analog recording is nonlinear enough that it functions something like a compressor. It compensates for musicians who can't modulate their playing effectively and gives notes a different shape.

Analog tape recording has built-in problems with bass. There is a high pass filter effect, and there are frequency response anomalies called "head bumps".

In contrast digital has no built in high pass filter effect and it has nothing that is even remotely similar to head bumps. In fact the low frequency limit of a digital recording is the inverse of its length which is a tiny fraction of a Hz.

Maybe they liked the artistic effect that the analog anomalies produced? Production is very different from accurate reproduction.

Add to this the fact that Pink Floyd has been recording since the mid 60's. They are probably quite comfortable using analog, and are well versed in analog effects.

How do we know it was a choice? Perhaps the recorder that was most available for the gig was analog.

What you are referring to is called SPARS code. They were always optional.

http://en.wikipedia.org/wiki/SPARS_code

"Several limitations of the code have led to it being largely abandoned."

The likelihood that someone with Waters budget recorded in a studio that only had analogue is about zero. Far more likely today is a studio that is totally digital due to the running costs and maintenance of analogue. Whether he could have done it directly into a DAW, or that was the only way to get that effect, it's a personal preference or he is being a Bono-esqe onanist is irrelevant because it's an artistic choice in production.

There are boutique studios that are proudly analog only.

http://www.emusician.com/features-interviews/0777/the-foo-fighters--grohls-garage/140095

http://www.soundonsound.com/sos/jun11/articles/foo-fighters.htm

There are boutique studios that are proudly analog only.

http://www.emusician.com/features-interviews/0777/the-foo-fighters--grohls-garage/140095

http://www.soundonsound.com/sos/jun11/articles/foo-fighters.htm

So? Doesn't change my point at all.

Ummm..., yeah, and Roger Waters had been out of the band for years by the time the remaining members, still engaged in a lawsuit with Roger over the right to use the name "Pink Floyd," recorded A Momentary Lapse of Reason. In fact, the acrimony with Roger was a factor in the band's deciding upon the title of the album. They had rejected a few other possible titles that they ultimately decided were too snarky. Therefore, Waters' budget has absolutely nothing to do with the recording of this particular album.

Waters has a huge personal wealth and no doubt his own very good studio and would have no difficulty getting funding from labels based upon his name and track record. He could probably fund most of the recording costs from what he dug out of the back of his couch, relatively speaking.

Hi A9X-308, But Will's point was that Roger Waters had nothing to do with this particular album.

Although I suspect that no Pink Floyd album has had a budgetary problem since the 60s. Any of the members probably had enough "couch change".

Correct. When the album was recorded and released in 1986-87, "Pink Floyd" was only David Gilmour and Nick Mason. Rick Wright was still with the band, but due to contractual prohibitions from a few years earlier, he was not a member and was in fact an employee. Roger Waters was neither a member or an employee and was not associated with the band or the album in any form or fashion. At that time, he and Gilmour and Mason were litigants in court fighting over the right to use the name "Pink Floyd." Roger had sued them seeking an injunction prohibiting them from using it without him. He lost.

^^ I didn't know the history behind that release, nor do I care, but it still does not change my point. Substitute Mason or Gilmour for Waters in my posts above - makes no difference.
They used analogue to record drums and bass for artistic or personal preference reasons, not because it would have been the only format available or budget etc.

IMO with this recorder thing, we're off in the land of almost zero useful information and infinite speculation over something that really doesn't matter in the big picture.

I've been involved with stuff like this and it all starts when someone says: "Let's use that one over there", and it seems like a good idea at the time, and that is what happens.

Analog tape isn't usually all that bad, and even when it is, you can hide a lot in a multitrack mix. Analog didn't die because it sounded bad every day, but because of the costs of making every day a good day.

A friend of mine is one of the very best analog tape machine techs in this region, and his workload went from more work than he could handle to needing to have a good day job in a few years.

there are many recording studio/engineer/etc stories often with detail on equipment, both current and historical in this
http://www.tapeop.com/issues/

The Video story on the front page sparked some questions about recording methods.

One point of view is that recording to tape and then digitizing will sound the best because the intrinsic jitter in the A to D process is minimized....

Another POV is the Direct to two track Sheffield guy in thestory who is using his customized and modded 24 bit 192 K A to D converter....for high resolution recording (24 192k) he does not comment on how he keeps the tracks in synch and not at all about jitter. He proudly notes that the musician can not distinguish their live performance from his recorded playback in his highly modded studio.

What are the tradeoffs in actually producing the final tracks..
with analog recording of tracks to to tape and then mixiing...
Digital recording of tracks and then using a digital mixer ...
How much of a concern is jitter in the recording process.. (an EU CD reocriding engineer noted that the red book standard was quite high)
Does software like Protools and autotune help or hurt the creatiion of good sound.
Edited by Tcatman - 5/9/13 at 7:50pm

Hi Tcatman,

I don't know what video story you're referring to, so my comments are out of context, and I apologize for that. I don't know what "intrinsic jitter" he is referring to. There is no "intrinsic" jitter in today's A-to-D converters - Jitter is caused by the clock that feeds the ADC. It is simple enough to derive a spectrally-pure (low-jitter) clock to feed to the ADC, so there is no excuse to have jitter in a consumer product, let alone a professional recording setup.

If your final output is digital, I don't see any point to including tape anywhere in the mix. Whether you consider dynamic range, channel separation, harmonic distortion, noise, number of channels, or almost any other specification, even a mediocre digital recording setup will out-perform tape. There is good reason to use tape, but if you start with tape, then stick with tape. If you intend to end with digital, then start with digital.

Again, If I missed the boat by not having seen the video, then I apologize.

This comment about jitter in the recording stream of analog to digital was made by a well known DAC designer. who always asserts that reducing jitter to as low as possible results in audible improvements in reproduction through the latest and greatest DACs. He never elaborated... I assume that by recording to tape and then converting to digital with a low jitter ADC was his gold standard because the sources of jitter were reduced and the trade off of limitations of tape were better ADD.

I would have thought that recording each track digitally was superior when going to a digital output as you note. However, I wonder how jitter associated with each track interacts as you go through a mixing process. When did recording set ups get this kind of gear? Does something like pro tools propogate error?

The Sheffield engineer of the video mag report was trying to capture a vital recording by going direct to disk without mixing (great attention to the recording setup) and that his high resolution ADC was the magic needed... Very different philosophy from how most music is put together. He is a huge proponent of the high resolution standard 24/192..

Considering both points of view... does the mixing process cost you something....(other then let the producer demand.. hotter... Hotter... HOTTER I SAY)

Hi Tcatman, And it is certainly true - clock jitter at the clock input of either a DAC or an ADC can cause audible distortion. When designing a DAC, clock-jitter used to be a serious headache - especially when the clock had to be recovered from the source material, such as with S/PDIF. Not anymore - it is now trivial to have a jitter-free clock. It seems clock-jitter has gone the way of the boogie-man (it doesn't really exist, but people are scared-to-death of it anyway).

There should be almost zero clock jitter in a recording signal chain, as the clock is derived from a quartz oscillator and is used to drive all elements of the digital chain. In my designs, I always start with a quartz oscillator for the master clock (MCLK), and derive the serial clock (SCLK) and the channel clock (LRCLK) from MCLK. I use those clocks to drive all ADCs and DACs, as well as the DSP, so all components operate in lock-step. It's quite simple, and the result is that jitter is in the parts-per-million range, orders of magnitude lower than what will cause distortion.
Good question. But first, let me explain that the audio tracks will not have jitter - jitter is only in the clock. If a clock with excessive jitter drives an ADC or DAC, then the audio will contain distortion as a result of the clock-jitter. The distortion will then propagate throughout the chain, as the tools would not know to remove it. There wouldn't be any interaction between channels, however.

Just to be anal-retentive, let me say now that jitter cannot be heard, as it is in the (inaudible) clock. It is the distortion that the jitter might cause that could be audible, and that distortion behaves as any other distortion, including the difficulty in removing it.
Although it is questionable whether we can hear the higher resolution that 24-bit/192kHz affords, there are very good reasons to record at high resolution, even if you later decimate down to a lower resolution. It is partially an issue of what the analog front-end can do, and partially an issue of what the DSP algorithms are capable of. For those reasons, many CDs are mastered at 352.8kHz (8x) and then decimated down to 16-bit/44.1kHz Redbook.

Yes, I would have thought that you would build a recording setup with a single clock… Now… how precise this clock must be could be a question. do I understand that when you mix several tracks on your way to a left and right track that the resampling and converson from 22 bits to 16 bits does not introduce more jitter and thus distortion to the wave form?
So, I am also trying to understand this comment.
An 8 percent standard is 8 parts in 100 which I think of as horrible. Your spec is in the u sec range with a parts per million spec. The DAC designer on this other thread notes
Which of course is 6 orders of magnitude lower then your standard.
So, I am confused…Clearly some recordings are superior to the vast majority. There is still real art involved in getting the sound onto the disc. So...
What is going on…. With a redbook production standard of < 8 % yet the Dac Designer is finding value in 50 psec.

Now, I suspect the majority of musicians are not techno geeks… and may not be at all worried about the sound quality of their music… It seems that those musicans who do care and demand vinyl prints or high resolution files are so rare that they become stories for being audiophiles. I am sure that they don't speak in terms of jitter and the resultant distortion Hell... Distortion is now simply what they want to generate for artistic effect.

Perhaps, a way to understand this issue is to ask... What features of a recording gets an engineer a Grammy Award these days?
Edited by Tcatman - 5/10/13 at 4:17pm

Hi Tcatman,

I'm going to work backwards - it's been that kind of a day . . .
First-off, I think that the DAC designer is suffering anticipation bias. His statement "Every time I achieve lower measured jitter, I can hear the improvement" seems to imply that he is trusting his ears and brain, which we know as unreliable. The fact that he states that he is trusting his own ears would have an adverse effect on his credibility.

By the way, 5-parts-per-million of a 12.288mHz oscillator would be approximately .4 picoseconds. Still, I've been told (so don't take it as gospel) that jitter less than tens of nanoseconds has been inaudible in double-blind tests. Hopefully, someone who knows of documentation will chime in.

With an MCLK of 12.288mHz, jitter of about .8 nanoseconds could skew a single bit-sample by 1%, which we could consider an audible amount. However, since each output sample is the decimated accumulation of 64 to 256 single bit-samples, some of which should be weighted in the opposite direction due to the jitter, much of that 1% should be canceled out. The issue would be that the total effect is going to be content-dependent, so the final result can't be directly computed.
I wonder what medium the master is on. Is it a master CD? From the context, I guess it would have to be.
This sentence makes no sense. BLER is the "block-error-rate", which is the ratio of blocks that contain errors to the total number of blocks. Since each block contains a Reed-Solomon error-correction code, the errors in the block should be corrected before being output. The 8% number (one in twelve) is to insure that the error-correcting hardware in the CD player does not get overwhelmed. What any of what he says has to do with jitter, I haven't a clue. Could he be one of those that like to invoke the boogie-man?
In the past, this was true, Theoretically, this may be still true. But in reality, this was a limitation that has since been solved.

MarkH,
Ok... I kind of like the notion of deconstructing what it takes to get a Grammy for recording engineering.

This iss the link to the Sheffeild engineer http://www.avsforum.com/t/1470742/the-new-fidelity-192-24

Based on MarkH's insight... My notion that the A to D step was the crucial development in the last 20 years seems to be off base by being greatly over hyped:)

So..
I imagine the key features are.
Recording studio has an acoustic signature that matches what the artists want musically...

Microphones come in all flavors and any electromechanical device will be unique... I am sure the engineer has his studio's carefully chosen microphones and they are unique...with various sound fields, dynamic ranges and quirks... and No off the shelf transducers for this recording need apply! The mic needs to match the dynamic range, frequency response and speed etc etc of the instrument....
Or you decide to record the sound field

the micro-phonic signal must be amplified... so you need something more then an op amp because you have to run in balanced config for long runs to the A to D converter.

At this point the channel clock and A to D derived from the master clock digitizes the wave form for that instrument and the microphone amp gain is adjusted.

The performance happens and tracks are written to disk in 24 bit resolution..... each channels wave form should have the same and very low distortion caused by jitter in the A to D conversion. Noise and distortion would be contributed mostly from the mics, amps, power supplies and cabling in that order.

So, now you play the tracks back through headsets or studio monitors.and mix.. so you can fiddle with intensity, phase (timing) re verb, equalization,compression ... all in the digital domain....
Am I correct that while you may or may not like the manipulation of the wave form... you don't add distortion in the mixing board?

and you will monitor the process through playback gear that you know every nuance of...... It doesn't matter what I hear on my stereo gear... just that the engineer knows that his fiddles on the board result in this specific sound and his mix will sound good on ear buds and top of the line Wilsons or Focals... .. Oh and also match what the artists want.

So... I have no clue what I don't know... (and could completely misunderstand the process)

So... where does the magic occur... What are the key steps where the specific gear makes the difference between OK recording and a Grammy recording to die for.

What are the key contributions of the mix engineer that moves the recording from OK to Grammy.

Thanks
Mark
Edited by Tcatman - 5/11/13 at 10:50am