Oh yep. That is blowing my mind already.
What the...? How the dynamic range expanded? Did you find the life in instruments again?
Source?
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Remastering (Improving) your own Audio Albums as Big F.Y. to corporate Loudness War trend.
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Hi Pal, DBX created a dynamic range expander and Type II noise reduction. It’s a filter on the machine.
I’m getting tired, so here’s a copy of how the tape manufacturer ATR explains how digital doesn’t compare in resolution.
NOTHING SOUNDS LIKE TAPE
WHY DOES MUSIC RECORDED ON ANALOG TAPE SOUND SO GOOD?
To understand why, a professional tape recorder provides the most lifelike reproduction revolves around a couple of important factors. The key lies in the inherent technology of the tape itself. Audio tape in use during the 1950s and ’60s provided approximately 65,000,000 magnetic particles per second of recording a quarter inch format at 15 inches per second (ips) tape speed. Each magnetic oxide particle or groups of particles takes on either a north or south orientation after exiting the recording head. Starting to sound like digital bit stream? Well yes and no. However there is one huge difference between analog tape recordings and even the best digital recordings.
RESOLUTION
The highest digital resolution today offers 4,608,000 bits switching per second. Not bad. Big improvement over the standard Red Book CD but it is not even close to sub-micron particle resolution of ATR Master Tape.
RANDOM PARTICLE STACKING
Quarter inch, two track ATR Master Tape running at 15 inches per second (ips) involves approximately 80,000,000 oriented and randomly stacked particles per track second. It’s not just the particle count but the random stacking that turns this super binary resolution into pure analog playback. This is why even a narrow track width recording still sounds so detailed despite the lower surface area.
Music is an intrinsic part of the human soul. It plays to our emotions, it talks to us, it calms us, it makes us rise to our greatest accomplishments and brings back our warmest memories. Why not record it on the best medium to achieve the best quality of sound?
This last paragraph is my favourite part.
May someone help me to find any scientific fact in there?
You can make fun of it, but it is the highest quality.
15 inches per second (ips) tape speed
Those particles get whipped around by the ultrasonic bias signal, then only the peaks of those waves (like "samples") are rectified(?), and low-passed into the listenable analog domain,
The tape signal is amplitude modulation of the bias signal, not just the mic feed.
The "signal" on the tape is not what comes out of the jacks.
Or am I terribly confused as sometimes occurs?
The tape signal is amplitude modulation of the bias signal, not just the mic feed.
The "signal" on the tape is not what comes out of the jacks.
Or am I terribly confused as sometimes occurs?
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I mean your talking about compression, right???
They had to compress the tapes 5 to 7 times to make an album.
I mean you do admit the vinyl is the superior source compared to the compressed digital loudness war file...
Well the master tapes, that’s the real deal of what it actually sounds like, before someone effed it up with their digital processing.
I’m sure some of you snowflakes are melting..... Just try to get a grip, because analog reel to reel tape, it’s still king.
They had to compress the tapes 5 to 7 times to make an album.
I mean you do admit the vinyl is the superior source compared to the compressed digital loudness war file...
Well the master tapes, that’s the real deal of what it actually sounds like, before someone effed it up with their digital processing.
I’m sure some of you snowflakes are melting..... Just try to get a grip, because analog reel to reel tape, it’s still king.
@Jim B, I'm going to be honest: You subscribed yesterday and your first posts are quite significant, already. If you intend to spread some subjective-audiophile BS here, you won't make a lot of friends on ASR...
I would believe reel to reel tape, it's still king once some (you?) will prove that claim. Not "because I say so".
I stand corrected:
Not rectified, and not AM.
https://hccc.org.uk/acbias.html
The basic scheme is to generate an AC signal at 5 or more times the maximum frequency that is to be recorded (e.g. 100kHz for the 20kHz audio range) and perhaps 10 times the strength of the signal to be recorded. The audio signal is then simply added to the bias signal. This is not amplitude modulation, but simple superposition.
So:
What happens to the precious source signal while the bias is swinging back and forth that makes it better than digital sample and hold?
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Mod note: Last few personal posts deleted. Guys, you really want me to babysit you? Ignore each other's posts if bothersome.
No worries, I’m out of here. This forum is filled with immature, ignorant people. Most of which couldn’t afford to buy decent stereo equipment and feel the need to discuss crap that isn’t even audible. Like it makes a difference. But what is really cliche is there inability to handle a difference of opinion so they resort to name calling. How very democratic.
I mean one guy doesn’t even know that changing your amp, changes the sound of your stereo. What is it that’s being discussed?? NADA
That one guy seemingly knows a lot more than you. In this forum, we go by what we can prove, not what we can shout as being right. If you have some evidence of changing an amp changes the sound based on controlled testing, then say it and everyone would appreciate it. If you are unwilling, then, don't let the door hit you on the way out.
Wherever you hang out, make sure you bring common sense with you that this is a hobby and not a reason for an outburst like I am responding to.
MRC01
Major Contributor
I struggled for years to understand what tape bias was all about: how it worked, why it was done, why normal, chrome & metal tapes used different bias frequencies (remember 70 vs 120?). That page is the best explanation that I've read. Now I realize I was confusing bias and EQ.
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Democratic? This isn't a let's vote on science site...no idea what that's all about...
The comment about what 'we' own or can afford, I find most amusing...
Anyway, there are places where your masterdebating style is likely much more appreciated.
Guermantes
Senior Member
I stand corrected:
Not rectified, and not AM.
https://hccc.org.uk/acbias.html
The basic scheme is to generate an AC signal at 5 or more times the maximum frequency that is to be recorded (e.g. 100kHz for the 20kHz audio range) and perhaps 10 times the strength of the signal to be recorded. The audio signal is then simply added to the bias signal. This is not amplitude modulation, but simple superposition.
So:
What happens to the precious source signal while the bias is swinging back and forth that makes it better than digital sample and hold?
If you wish to get nerdy (or nostalgic) about tape, Camras is a good place to start:
https://www.amazon.com/Magnetic-Recording-Handbook-Camras/dp/9401094705
digicidal
Major Contributor
Aww... I got here too late. Now it's starting to get all rational again... boooooring. 
Guermantes
Senior Member
Looking at the Adobe Audition screenshots, it appears the declipper simply applies a LPF to the clipped peaks. Or is there more to it? Any interpolation?
I frequently use de-clipping in both Cedar Cambridge and iZotope RX but usually there is noticeable artefacting on severely clipped material. No free lunch, I suppose.
Also: Does the DR Meter analysis in Foobar2000 apply any weighting? If it did, then I would have expected that the bass levels would have less impact on the DR score.
I frequently use de-clipping in both Cedar Cambridge and iZotope RX but usually there is noticeable artefacting on severely clipped material. No free lunch, I suppose.
Also: Does the DR Meter analysis in Foobar2000 apply any weighting? If it did, then I would have expected that the bass levels would have less impact on the DR score.