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Inside High-res audio: PCM vs MQA vs CD: 2L Sampler Comparison

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amirm

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Just watched the first part of the video and got to the part where @amirm explains 88kHz would be required to record the music.
Personally I found the microphones with the lowest noise (and best sensitivity) rolled off from 16kHz and I don't think I have ever owned speakers which went much above 20kHz and my ears went up to 16kHz last time I checked, a few years ago.
Since I couldn't hear it and my speakers couldn't reproduce it anyway is it still worthwhile to consider 88kHz as a worthwhile recording sampling rate?
Yes, philosophically. If not practically. :)

One can imagine that a master should record all the musical content. And once there, there will be a market for people who want to then acquire such a package as is, without a format conversion by the mastering engineer. Such format conversion to 16/44.1 no longer makes sense in download/streaming world because we no longer deal with the CD. The format conversion can have a subtle impact on fidelity both on bit depth and possibly the sample rate conversion.

I personally like to see a new high-res standard that embodies the above plus elimination of loudness compression (i.e. getting the content prior to "mastering" for the masses). Such content would sound better all around and would make a compelling offer then regardless of your hearing range.
 

Frank Dernie

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One can imagine that a master should record all the musical content.
I agree, but I would argue that the musical content is the audible part, not just all frequencies emitted by an instrument :)

I was an early adopter of file based music 20 years ago but have gone back to CD because I am not forever getting new firmware or other software and the file format changes. I have several file based devices from a Sooloos Control 15 to a PC based streamer from a small company near here, I also tried dedicating an iMac to file based music because the big screen was nice for artwork.
All of them are now either slow or no longer take updates. My 25 year old CD transport still works just as well as it ever did so is in every way less irritating for me than the file based stuff. I am a grumpy old bloke, I may not feel this way if I actually liked messing about with computers and all the time it takes rather than seeing as time wasted and didn't have 7000 CDs though.

The format conversion can have a subtle impact on fidelity both on bit depth and possibly the sample rate conversion.
Indeed, I have done blind tests on 3 different sample rate converters and only one was transparent to me.

I personally like to see a new high-res standard that embodies the above plus elimination of loudness compression (i.e. getting the content prior to "mastering" for the masses). Such content would sound better all around and would make a compelling offer then regardless of your hearing range.
I agree, less manipulation, rather as it often was 30 odd years ago is much better.
OTOH I still believe that CD is "high res enough" and that making the CDs not manipulated and streaming files compressed would make good sense, given the likely listening environments of the two.
 

RichB

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Yes, philosophically. If not practically. :)

One can imagine that a master should record all the musical content. And once there, there will be a market for people who want to then acquire such a package as is, without a format conversion by the mastering engineer. Such format conversion to 16/44.1 no longer makes sense in download/streaming world because we no longer deal with the CD. The format conversion can have a subtle impact on fidelity both on bit depth and possibly the sample rate conversion.

I personally like to see a new high-res standard that embodies the above plus elimination of loudness compression (i.e. getting the content prior to "mastering" for the masses). Such content would sound better all around and would make a compelling offer then regardless of your hearing range.

High Dynamic Range 44.1/16 music would make a far greater impact than inaudible frequencies ever will.
Next up is 3D audio, which is moving faster and will, it seems, finally kill MQA.

- Rich
 

Grooved

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Yes, I know.
Microphone position makes far more difference than the recorder sampling rate (or even what sort of recorder it is).
Having a higher bit-rate makes level setting easier (pretty well completeluy fool-proof in fact) and gives more latitude in subsequent manipulations, if you do them, so I do see a point in using 24-bit rather than 16-bit particularly for novices in level setting.
What I have still yet to see any convincing explanation of a way higher sampling rates could make an audible difference to music.
My StellaDAT won't boot any more. I suspect the EPROM was wiped due to the battery discharging whilst in storage (stupid of me to leave it connected, I know). Do you know anybody who could burn me a new EPROM or could tell me if that might be the problem? I did consider a StellaDAT II but went solid state instead, I don't need the StellaDAT but would like very much to get it working again.

From what I heard, they completely turn focus on the II version and never fix some bugs on the I (not about the sound quality), but they may have some parts left somewhere.
Depending on what kind of EPROM, it can be possible to duplicate one, but you will need a healthy one to do that. It was done on some other device, where generally it was not the battery itself but the EPROM having reached his limit. What kind of battery and EPROM is on yours ?
 

Grooved

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Yes, philosophically. If not practically. :)

One can imagine that a master should record all the musical content. And once there, there will be a market for people who want to then acquire such a package as is, without a format conversion by the mastering engineer. Such format conversion to 16/44.1 no longer makes sense in download/streaming world because we no longer deal with the CD. The format conversion can have a subtle impact on fidelity both on bit depth and possibly the sample rate conversion.

I personally like to see a new high-res standard that embodies the above plus elimination of loudness compression (i.e. getting the content prior to "mastering" for the masses). Such content would sound better all around and would make a compelling offer then regardless of your hearing range.

Right, be able to have choose between two versions, one "radio master" and one "less compressed" would be great.
Now, it will at least make two mastering process times, not sure they want to do this for all projects.
For years, the radio market was said t be responsible of higher compression than needed, but don't you think that now, the fact that a lot of people listen to music on small portable speakers gives them a reason to continue ?
Regarding pop hits, not strangely, the few tracks with less compression on an album are always the one that are not published as single and streamed on radio... and if you take the compressed ones, it's not always very compressed from the beginning, but crescendo, to keep attention and "hitting the brain" until the end of the song.
It's kind of a "magic trick" for creating a hit these days, even if it's not "musical magic" ;-)
 

Grooved

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I agree, but I would argue that the musical content is the audible part, not just all frequencies emitted by an instrument :)

Unless we want to use the sound of an elephant in your recording and play it during live show on a big system, you won't hear the sub-bass, but you will feel it :)
High Dynamic Range 44.1/16 music would make a far greater impact than inaudible frequencies ever will.
Next up is 3D audio, which is moving faster and will, it seems, finally kill MQA.
- Rich
If you want to record in stereo with a mics couple a live session, no problem if you're sure you got all from your setup and it doesn't need more editing.
As soon as you record and have to mixed a lot of tracks, you need at least 24bit, even if the final mix has a dynamic that can fit in 16bit. You can still convert the final file to 16/44.1 like it was/is done for CDs and lossless streaming, but if think most of the time, providing the 24bit version is just because of what was asked by a market, to get more sure you get something that didn't have been changed, maybe in the hope at first that you will get something with less compression, which can only be done if it's mixed/mastered differently.

In the end, our best chance to listen to an artist without too much compression is in concert, but I've seen a few one with a lot of compression too, and it's even more horrible than listening it in your room. Thanks, it was only a few times, mostly from band mixing electronic/acoustic sound and never on the hundreds of jazz concert I've watched, and on the opposite side, in the middle of some magical jazz concerts, I still put in my top5 a Radiohead concert in an auditorium that was just pure sound magic. Never too much loud and just feeling everything, just as beautiful as Wayne Shorter with Brian Blade, Dave Holland and Herbie Hancock in a small concert.
 
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Frank Dernie

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From what I heard, they completely turn focus on the II version and never fix some bugs on the I (not about the sound quality), but they may have some parts left somewhere.
Depending on what kind of EPROM, it can be possible to duplicate one, but you will need a healthy one to do that. It was done on some other device, where generally it was not the battery itself but the EPROM having reached his limit. What kind of battery and EPROM is on yours ?
I was in contact with Sonosax years ago about check and repair but they only offered a discount on a StellaDAT 2 and I had already started using a solid state recorder.
I am mainly interested to be able to get mine going to replay the master tapes from it.
I am not sure where the battery is, I have a mains power supply I mainly use. The removable chip has a label master V1.6b on it.
 

Frank Dernie

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Unless we want to use the sound of an elephant in your recording and play it during live show on a big system, you won't hear the sub-bass, but you will feel it :)
Yes but digital recorders record down to DC so that isn't at issue, it is the >22.05kHz frequencies I see no point in recording even when they exist.
 

voodooless

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If I'm not wrong, and even if you're talking about the "listening DAC" step, you're actually more confirming the answer I gave on "explains 88kHz would be required to record the music" more than confirming a comment stating the opposite of my answer, don't you think so ?

The question is not if you could, but if you should! You clearly missed the last line of my post post.
 
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Frgirard

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Where are the tracks with an amplitude of 80 dB?
Who are able to hear at - 80 or - 60 dB?
Who have a room with a NR of 15 dB?
Where are the speakers able to reproduce a dynamic of 60 dB?
Where are the domestic room can support a dynamic of 30 dB?
24bit for what? Demonstrate the existence of the multiverse

A record is made to be listen every where.
 

Grooved

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Yes but digital recorders record down to DC so that isn't at issue, it is the >22.05kHz frequencies I see no point in recording even when they exist.
Even if I don't see the need for that, if you that you may not hear but feel what is between 20 and 30Hz, you may think that you can feel the above of 22kHz. It will certainly not be a fundamental but harmonics, but it's possible you can feel it with your teeth, I don't know. But again, even if that was the case, I don't think it's necessary and I don't think it was the argument from people asking for masters. It's just that they asked for master and it created a market, but more to get the same thing than what was done in the studio.

The question is not if you could, but if you should! You clearly missed the last line of my post post.
In some cases it was needed, not because you want to record/listen above 22kHz, but because you can have a plugin or a device that works best at this rate, and just this thing that created the fact of using 88.2... when you have a plugin that works with it's own oversampling, it should work great without the need to record at 88.2. That's why I said it sometimes used not because you want your final master with this rate, but because creating was simpler with this rate. There are been several reasons.

Where are the tracks with an amplitude of 80 dB?
Who are able to hear at - 80 or - 60 dB?
Who have a room with a NR of 15 dB?
Where are the speakers able to reproduce a dynamic of 60 dB?
Where are the domestic room can support a dynamic of 30 dB?
24bit for what? Demonstrate the existence of the multiverse

A record is made to be listen every where.

I would say none of the reasons you gave, because they are at the listening step only.
The listening market has asked for getting, or listen to the marketing argument of providing, files coming from the studio, and these are 24bit for so much reasons other than the examples you gave, even if they are right.
And these other reasons are linked to : digital recording and mixing.
A snare drum can goes up to 120dB, cymbals more than that, nobody will take the risk of loosing time or use a mic placement to get less level at the cost of having a sound that is as good. Even if it will not reach it, you never record near the highest level accepted, you always keep security, so you don't have 96dB to work in 16bit.
You only use two mics away for your drums ? Easy to reach 105 dB around a drum kit.
These levels will be reduced in your mix, but you can take the risk to clip or having no headroom on any track while recording.
If you have a lot of tracks, there will a sum of their noise floor, with 24bit, you takes less risk that the sum comes up to a level it can be heard.
A guitar amp may sound different with higher volume than a low volume, and even if you will lower that once in the mix, the source while recording can be at more than 96dB to hit that right sound you want.
If you have a configuration where all sources are lower than 96dB (and far lower as you keep room and don't record near OdB), you could record each one with 16bit, but once you import it in your mixing session, you have to do it at 24bit or 32bit floating point. All processing will be better, you can avoid the possibility of digital clipping,...
Digital mixing and processing are certainly the main reason masters ended up being at 24bit more than anything else.
Masters were converted at 16bit in the CD area, but it changed after that, because of the market asking for studio files and/or the idea to re-sell some albums that were only on CD at 16bit before. Add that they got less money since CD have been shared via mp3 20 years ago, then streaming which doesn't bring you money that what CD did... Even with general increase of life cost, and so travel, hotel... the loss on CD sells is also a reason why concert tickets cost have increased.
Saying these ones are 24bit gives less doubt that it's the real file created by the artist team, but in most cases, it's not the artist that will decide which version will be delivered to providers because the master is the proprietary of the label, unless artist has it's own label.

In the end, studios need 24bit for several reasons (and other that I did not talk about), and the demand for studio files leads to 24bit files for listening.
And we are talking about 24bit, but we should not forget that most ADC or DAC will only gives you 21bit out of 24 in real use, maybe 22bit if put in a freezer :)
 
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voodooless

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@Grooved , audio production and playback are two different things. They have different needs, so should not be conflated. Specially the first has a lot of creative freedom. If they choose to use a bad filter that introduces aliasing, then that is fine, since the artist intended it this way. And you want some latitude in your files for filtering, processing, effects. Some effects even need a high sample rate to not introduce artifacts.

Playback is different. You want to playback the way it was intended. For that we don’t need “creative” filters. And if one thing is clear from these series of videos, it is that the artists had no intentions when mixing ultrasonic content. There is no time spend in shaping the ultrasonic part. It’s an afterthought, it’s just wherever it is.. and it’s obvious why: the people dealing with it can’t hear it just as anyone else.
 

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@voodooless , sorry but did you respond to me or to @Frgirard ?
because I agree on all of what you say, and if I'm not wrong, it's pretty much what I said in my answer to @Frgirard (two different things, high sample needed for some effects, not for providing ultrasonic content...) but you did it in a more concise way :)

@Frgirard last question was also asking why we ended up with 24bit files to listen and I tried to give some examples for why 24bit files exist and that this and high sample rate are not for offering higher frequencies than 20kHz or higher bit depth but because production was done with that, and since people want production files, they get 24bit and eventually higher SR. Am I wrong with that ?
 
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Who have a room with a NR of 15 dB?
NR ratings are not psychoacoustically valid. It is called *noise* rating for a reason. It is for determination of speech intelligibility and level of annoyance. Appropriate noise level needs to follow threshold of hearing with respect to computation of dynamic range. And there, research shows that plenty of rooms can approach or even exceed audibility thresholds. See: https://www.audiosciencereview.com/forum/index.php?threads/dynamic-range-how-quiet-is-quiet.14/

And my video on the same topic:

 
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24bit for what?
24 bit is not doable but 16 bit puts a limit on how loud you can listen before channel noise becomes audible. At 93 dB dynamic range for example, if you set your peak at that, then your noise floor would be at 0 dBSPL which can be audible. At higher levels, that noise proportionally gets louder.

Since the storage cost is trivial these days, I don't see a reason to limit ourselves to 16 bits if the content is produced at higher depth. Note also that even if the original live recording lacks such dynamic range, any editing in 24-bit format can create such low level detail.
 
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Frgirard

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24 bit is not doable but 16 bit puts a limit on how loud you can listen before channel noise becomes audible. At 93 dB dynamic range for example, if you set your peak at that, then your noise floor would be at 0 dBSPL which can be audible. At higher levels, that noise proportionally gets louder.

Since the storage cost is trivial these days, I don't see a reason to limit ourselves to 16 bits if the content is produced at higher depth. Note also that even if the original live recording lacks such dynamic range, any editing in 24-bit format can create such low level detail.
 

Frgirard

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Who produce record with a DR of 93 dB?
Who heard under - 60 dB?
Where are the record with a DR of upper 60 dB?
We work on sound we can hear.
No speakers are able to reproduce the Theorical dynamic of the 16 bit.
No domestic room are able to support the Theorical dynamic of the 16 bit.
No recording place, studio or venue has a NR less 30 dB....
 

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For a recording studio the value of NR20 (noise rating level) is recommended.
At 63Hz the value is 51 dB. The bass frequencies is the frequency zone with the greatest bearable dynamics.
120 dB - 51 =69dB.

A tenor can reach 130 dB out the mouth. NR20 at 250 Hz is 31Hz so we have 99 dB.

And all the records made in churchs, live concerts are far from the NR20.
 
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Who heard under - 60 dB?
What? Do that all the time, and every minute.

No speakers are able to reproduce the Theorical dynamic of the 16 bit.
You got that from where? Speakers have infinite dynamic range. Feed them silence and you get silence. Feed them any signal and by definition the ratio will be infinite. It is trivial to have a system that has peaks that hit 93 dBSPL and go to silence for lowest amplitude.

No recording place, studio or venue has a NR less 30 dB....
Did you not read the links I gave you? There was a reference to ex-president of Audio Engineering Society peer reviewed paper which included such surveys done properly:
index.php


Skywalker Scoring stage has noise level that is below threshold of hearing across the full frequency range. Davies Symphony Hall almost gets there as well. You seem to keep using the NR rating which as I explained, is useless in this regard. There is no such thing as a single number audibility floor in a room. It is a full frequency range as shown above.

There are also other factors here in that we hear below the noise floor. And that directional noise coming from a speaker is more audible than ambient noise.

You seem to have learned a few talking points which are rooted in complete misunderstanding of the topic.
 
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