Redbook 44.1 kHz standard: theoretically sufficient timbral resolution?
- Thread starter Al M.
- Start date
Thank you kindly for re-posting this. I was just about to post something similar, as this is my standard answer to assertions that our audio systems do not need to reproduce anything above the basic range of human audible hearing.
The four air motion tweeters in each of my ribbon panels literally start at 18kHz.
Isn't the digital nerve racking for some people?
I spent some time, experimenting recording bands and artists. Even though I would get into that part of it full time. But after using my 'audio test generating subjects' I'm more decline to sell that all off after I give the next guy after I hot rod them into perfection. Once I recorded them they went on their way, what I found out is they thought they took advantage of me, however, they were just my organic signal generators. The only cool thing was using different mics and all of that in the process and internalizing how three dimensional sound waves are translated into the wire.
The digital clock signal has its little issues mucking up the signal on the way in and out due to the digital clock's harmonics bleeding into the band. So the engineers placed filters in that switch in and out at those sample frequencies. There are other frequency filtering afterwards that the other sample rates would use. All of these do add into the phenomenon called insertion loss. Which is a drop of signal power when a circuit like a filter is inserted. The perception of the higher sampling rates is the perception of change in signal to noise when the sampling rate changes, because of adding or subtracting those filters. I learned the 'point instant' instead of a 'stair step model' in regards to sampling. But quite frankly,its a mood point. Because that is not where the loss truly happens. It happens right before that when the signal begins its trip into a media change.
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I used different fancy boxes, and i/o and all kinds of these "recording interfaces" that consists of a signal path of microphone preamp, a line buffer into a a/d conversion chip. The strata of what you buy is what you pay for comes seemed like it it was true, but later I found, that is just on the surface. Those recording interface manufacturers purposely kept their schematics off the market. For I guess of being afraid of someone noticing, they are pretty much all the same. Another interesting point I cam across is they don't even create their own circuitry past what the chip manufacturers have in their "sample circuit" of their device. Going back further in school times and remembering a lecture in class about data sheets and their example circuits. I remember the professor saying, "The example circuit found in data sheets are simple examples to demonstrate how the device works. If you are truly designing something for a purpose, you should only use this as a reference guideline instead of the exact way it supposed to build for all applications". Remembering that, and seeing that yes indeed they didn't do anything but assemble the example circuit in their computer interface boxes over and over again. All those fancy mic preamps and all that going into a generic circuit not designed specifically for the task, but was designed to demonstrate the converter chip. I wonder why what I recorded didn't sound correct like too much of the wrong bass, or lack of mid punch or even the airy highs like I would hear on a 'mastered track' . After some time recording people at different levels, and notice differences in signal, it dawned on me that it has this insertion loss, and other parts compounded this. I stepped back and thought about how the older recording devices were made like the reel to reel so many admired more and even came up with emulators to add its not so good factors in to attempt to make it sound like a tape machine. So , instead of refining the interface, they market dsp to try to fix something it can't because it was lost at the time of conversion/capture. So I went and modified an ADAT interface first, I removed all of the generic circuit and replaced certain things to fix what was recorded. The mic preamp in front, when I jumpered it out of the unit into a power monitor for me to listen sounded pretty bad compared to the professional mic preamps that I had in a rack. Tinny, and the bass out of balanced. Interesting that their line in didn't bypass this stage, but attenuated the signal so it would go through all this. So I decided to bypass it. After that, I found that the unit recorded better than what I could buy. I found out later only a few high dollar converters were set up like this and they are still not available in the music stores online or physically. You had to contact them, then someone would quote you through a distributor they dealt with. I noticed that a lot of the professional recording gear of the past has low frequency roll offs built into them. So on another interface, I replaced the capacitor circuit on the microphone preamp with one of those transformers from a famous mic preamp, and lowered the capacitance values on their coupling caps in that unit so I can achieve that 80 hz roll off I measured on my reel to reel tape professional machine. The results in both modded units were spectacular and someone said "did you record this on your reel to reel?" Needless to say, he was shocked when I told him.
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Then the mixing game happened in post. There was all kinds of fun putting it together, good and bad. without modifications I had to do all this crazy processing stuff to get something sounding nice. After I modified it, it was esier to put together and also didn't have to do a lot of processing.
Then there was mixdown. I found it interesting that inside the multi track software, it sounded awesome, when I went to mix it down or 'bounce to disk' it sounded low and duller when I wrote it on a disk and played it. I know I did everything right, what was the deal? Later, I did a project and sent it into a mastering guy after auditioning several ones and looking at what they were doing. Some would eq it, other slammed it into a limiter or both but one thing I notice about all of them is they shortened the dynamic range and analyzed the effects, they what I called integrating soft clipping. So the format of CD and their loudness war ended up who can soft clip the best. When I analyzed the raw masters, I find that they were causing +3 to+5 dbfs of signal to exist. The digital scale is -150 to 0 dbfs, how were they doing it (soft clipping) I also notice they never let the signal fall below -3 dbfs which means to me, they are doing that to overcome the D/A insertion loss of the filters.
So the professionals ruined the digital format. That is why there very few dynamics. But on the other hand, too dynamic rendered a softer signal after conversion. Because we fell below a signal to noise before its fully converted back to analog. Some people swear by higher sampling rates, but I found that all it was a signal relationship of the insertion loss I discussed earlier. Later, I found if I mix really low and then master it up digitally, I gained more space and what was more what I mixed. It sounded great, until I burned it to CD and it was lower, softer than when I played it on a windows player.
I further analyzed the D/A process electrically and how circuits are built and noticed a few flaws inherited by the digital process. The sad thing about it is that one could make a wonderful D/A conversion circuit. But the commercial material was damaged to be mastered to go through the inefficient systems that were built to reproduce them.
Then there was mixdown. I found it interesting that inside the multi track software, it sounded awesome, when I went to mix it down or 'bounce to disk' it sounded low and duller when I wrote it on a disk and played it. I know I did everything right, what was the deal? Later, I did a project and sent it into a mastering guy after auditioning several ones and looking at what they were doing. Some would eq it, other slammed it into a limiter or both but one thing I notice about all of them is they shortened the dynamic range and analyzed the effects, they what I called integrating soft clipping. So the format of CD and their loudness war ended up who can soft clip the best. When I analyzed the raw masters, I find that they were causing +3 to+5 dbfs of signal to exist. The digital scale is -150 to 0 dbfs, how were they doing it (soft clipping) I also notice they never let the signal fall below -3 dbfs which means to me, they are doing that to overcome the D/A insertion loss of the filters.
So the professionals ruined the digital format. That is why there very few dynamics. But on the other hand, too dynamic rendered a softer signal after conversion. Because we fell below a signal to noise before its fully converted back to analog. Some people swear by higher sampling rates, but I found that all it was a signal relationship of the insertion loss I discussed earlier. Later, I found if I mix really low and then master it up digitally, I gained more space and what was more what I mixed. It sounded great, until I burned it to CD and it was lower, softer than when I played it on a windows player.
I further analyzed the D/A process electrically and how circuits are built and noticed a few flaws inherited by the digital process. The sad thing about it is that one could make a wonderful D/A conversion circuit. But the commercial material was damaged to be mastered to go through the inefficient systems that were built to reproduce them.
I was once a dealer for Golden Sound’s Ultra Tweeters, a quantum mechanical device with a claimed frequency of 2 or 3 GHz.
This is one of the most interesting posts I've read on WBF, thanks!
Are there recordings you've done that are available on Tidal etc?
Obviously overly compressing the sound is bad for dynamics, but one assumes business was falling rapidly so they had to do something. However, there are a lot of other perhaps less obvious reasons - hiding in plain sight! - why digital playback is rarely satisfactory to me here in the future.
this is my shortlist and yes I know what you’re thinking, “but isn’t Reed Solomon error detection/correction and the CD laser servo feedback system supposed to take care of all that?”
Isolate the CD player/transport from seismic type vibration.
Isolate the DAC from seismic type vibration.
Reduce scattered laser light inside the CD transport that would otherwise get into the photodetector.
Stabilize and damp the CD.
Eliminate the electric static charge on the CD.
Eliminate electric static charge on objects in the room.
Improve the transparency of the CD clear layer. Polycarbonate is only 90% transparent to the CD laser.
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Compression is a necessary evil in recording and production. Without it everything would be thin and out of scale with each other when the instruments are recorded individually.
converters made or modified for CD mastering changes the depth of the mix and like always the relative distance between loud and soft get shorter, the louder the mix is.
Here is someone covering one of the first converters available outside of having the custom made ones or an engineer soft clipping into a high end converter several times. Notice in the video the mixes changing depth in exchange for loudness.
I referring to aggressive over-compression, i.e., loudness wars, not to compression in general.
You can only compress a mix so much. But limiting which is a type of compression is the loudness war key. In the video above in post 129, the magic circuit that allows them to go above 0dbfs is a limiter.
A lot of CDs that have been coming out over the past ten years or so are flatlined. Apparently if there is some limit to the compression they utilize they haven’t reached it yet. Ever take a look at the dynamic range database?
There is limits to it depending on the frequency balance and relative power. But what really limits it is too short of a distance between load and soft. which sounds over cranked and distorted. The other extreme of having the distance great reveals a soft and weak recording. But normally an un-mastered track has 4-6 db differences between loud and soft. That normally gets mastered to around 2-3 db difference. But CD media is -3 to -5 difference which is less than 1db on the recorded material and signal +3db greater after the dac and about twice what 0dbfs can deliver.
In full scale terms (dbfs of hardware) yes. Even though you used to seeing the weighted level (LUFS) used to express dynamic range on a recording and using either DB or LU as the units
There is no simple formula that will give you x dBFS = y LUFS, because it depends on the frequency content of the audio signal being measured. So 1 dBFS does NOT equal 1 LUFS in general.
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