High-res audio comparison: Linn Records Free High Res Samples

[dc655321]

The more slices you have, the more accurate the sound will be.

A thousand times, no.

 

[Frgirard]

Using ultra high frequency when recording doesn't only deal with how high of frequency sound you can hear but the resolution, the each Hz is a slice of that moment of recording. The more slices you have, the more accurate the sound will be. When playbacking a recording, the closer you get to the original recording frequency, the cleaner, more accurate more detail you will hear.

Proof?

 

[Newman]

I have a pair of Adam A7X with tweeters that go up top 50kHz. SOS review said this about that aspect:
Its frequency response now extends up to 50kHz, but the design is also a little more efficient than its predecessor and so can generate higher maximum SPLs. Now, I don’t think even your pet bat is going to get much benefit from the extra frequency range per se, but a wider frequency response generally means an even flatter phase response within the band you can hear, so it isn’t just a case of ‘more for the sake of the spec sheet’. After all, if everything in your audio system had a 20kHz bandwidth (‑3dB), by the time your signal had passed through all the different units in the chain, the final frequency response would be way down at 20kHz.Developing new extended‑range HF drivers is one thing, but they need to be driven by adequately powerful, low‑distortion amplifiers that have a greater audio bandwidth than the driver itself. Adam’s design team apparently explored existing Pulse Width Modulation amplifiers but found that the necessary low‑pass filter compromised the frequency response of the ribbon tweeter, so they developed their own Class A/B analogue circuit, which works up to 300kHz. Again, this may seem excessive but it helps to maintain phase linearity.
~ https://www.adam-audio.com/content/uploads/2016/11/review-adam-audio-a7x-soundonsound.pdf
Is this journo-hack hogwash or is this a valid point?

You can’t blame the journo for quoting Adam’s own ”hack hogwash” from their web page for the speaker. Well I suppose you could, if we assume it’s okay for manufacturers to write hogwash but not okay for journos to bathe in it.

But seriously, Adam get a demerit point from me for writing this reason for using AB amps. “PWM amplifiers need a low pass filter to separate the modulation frequency from the audio band, and this may limit the reproduction capabilities of the new tweeter.” Why write “may”, when they are building the amp into the speaker and can choose a PWM amp that definitely doesn’t? Disingenuous.

 

[dualazmak]

Hello friends,

Just for your reference, as my self-defence measures, I decided to routinely use High-Cut (Low-Pass) LR -48 dB/Oct digital filters at 25.000 kHz in crossover configurations in my multichannel multi-driver multi-amplifier system. Please visit here on my thread for the details.

 

[MWC]

You can’t blame the journo for quoting Adam’s own ”hack hogwash” from their web page for the speaker. Well I suppose you could, if we assume it’s okay for manufacturers to write hogwash but not okay for journos to bathe in it.

But seriously, Adam get a demerit point from me for writing this reason for using AB amps. “PWM amplifiers need a low pass filter to separate the modulation frequency from the audio band, and this may limit the reproduction capabilities of the new tweeter.” Why write “may”, when they are building the amp into the speaker and can choose a PWM amp that definitely doesn’t? Disingenuous.

? but surely that is exactly what they did and the next sentence following your quote from them says so, "So ADAM spared no efforts and decided to design its own high frequency power amp to allow the new high frequency unit to live up to its unique talents." Or do I misunderstand your explanation for your demerit?

 

[Newman]

Yes, you misunderstand.

But I don't know how to explain more clearly than I already did.

Since, like they say, they designed their own amp from scratch, then they could easily have made it a PWM design, and made sure it doesn't limit the tweeter's "unique talents". It wouldn't have been difficult.

So that's not the real reason they went AB. The real reason is marketing. Point of difference.

Disingenuous of them to throw us a BS spin like that.

 

[redshift]

My opinion, is I'd like the format in which something was originally recorded. If someone uses 24/96 I'd like to have it in 24/96. While much is made of possible ill effects of ultrasonic garbage it rarely is at a level to cause issues. Super careful recordings don't need to have that stuff anyway. I've done recordings and even young musicians have failed to find any advantage to 96 or 192 khz vs 48 khz I normally use. I might at most use 96 khz, and would see to it the ultrasonics are clean.

There are so many things an order of magnitude more important to the quality you hear upon playback than sample rate however. Maybe, maybe for some of the very finest young listeners 96 khz might be just the tiniest bit better like maybe, maybe 1%. If you recorded well at 48 or 44 khz it is 99% of the best possible for any listener which anyone would judge good, and 100% of what is possible for the overwhelming majority of all listeners.

People tend to do a lot of EQ/DSP these days on the playback side requiring interpolation and up sampling to maintain resolution, with it’s own slew of problems.

Why is it a problem distributing the material as 24bit/192kHz, it’s not as we are short on bandwidth and storage space for digitized audio signals?

 

[jaykay77]

People tend to do a lot of EQ/DSP these days on the playback side requiring interpolation and up sampling to maintain resolution, with it’s own slew of problems.

Why is it a problem distributing the material as 24bit/192kHz, it’s not as we are short on bandwidth and storage space for digitized audio signals?

Because "it's all a racket" as my grandfather would say. He was the wisest man i've ever known lol

 

[dc655321]

People tend to do a lot of EQ/DSP these days on the playback side requiring interpolation and up sampling to maintain resolution, with it’s own slew of problems.

How does using DSP require "up sampling to maintain resolution"?

 

[redshift]

How does using DSP require "up sampling to maintain resolution"?

Truncation.

 

[dc655321]

Truncation.

Uh huh...

 

[Tks]

Using ultra high frequency when recording doesn't only deal with how high of frequency sound you can hear but the resolution, the each Hz is a slice of that moment of recording. The more slices you have, the more accurate the sound will be. When playbacking a recording, the closer you get to the original recording frequency, the cleaner, more accurate more detail you will hear.

What if I use 8kHz sampling rate for my audio? Would playing it back resampled 192kHz allow me to hear "more accurate detail"?

 

[redshift]

What if I use 8kHz sampling rate for my audio? Would playing it back resampled 192kHz allow me to hear "more accurate detail"?

Yeah, you are effectively oversampling the signal. That way you could convert it to a higher res signal, perhaps go from a 16bit 8kHz to a 20bit 44.1kHz signal.

 

[Tks]

Yeah, you are effectively oversampling the signal. That way you could convert it to a higher res signal, perhaps go from a 16bit 8kHz to a 20bit 44.1kHz signal.

I know I am upsampling, but I am asking, will I hear more details because I went from 8kHz to 192kHz from a file that was originally 8kHz recording.

Also what would bit depth have anything to do with this question? Increasing bit depth of an original file's bit-depth does absolutely nothing in the slightest for the original sound.

 

[redshift]

I know I am upsampling, but I am asking, will I hear more details because I went from 8kHz to 192kHz from a file that was originally 8kHz recording.

Also what would bit depth have anything to do with this question? Increasing bit depth of an original file's bit-depth does absolutely nothing in the slightest for the original sound.

Says who?

Ever heard of 1 bit DAC’s? Funny how that could work?

https://en.m.wikipedia.org/wiki/1-bit_DAC

 

[Tks]

Says who?

Ever heard of 1 bit DAC’s? Funny how that could work?

https://en.m.wikipedia.org/wiki/1-bit_DAC

Says I. Are you going to answer the original question about sampling rate or are you going to derail by claiming upconverting a 16-bit song to 24-bit is going to improve the sound as well? (Making an even more dubious claim than the original upsampling one).

 

[redshift]

Says I. Are you going to answer the original question about sampling rate or are you going to derail by claiming upconverting a 16-bit song to 24-bit is going to improve the sound as well? (Making an even more dubious claim than the original upsampling one).

Yep it will; if you know a thing or two when you design the up sampler and interpolation filter.

 

[Tks]

Yep it will; if you know a thing or two when you design the up sampler and interpolation filter.

Okay, now - since you're I caught you in the mood for lecturing. Explain to the uninitiated, what exactly it is you think you're hearing (and how it will ALWAYS be an improvement, yet somehow never a detriment as is the usual case for people who take such claims to be true) taking a 8kHz recording, and upsampling to 192kHz.

So what I am asking for is, what has changed in the audio file itself that has now become audible from a metrics standpoint, and also I want to understand by what mechanism this is always going to be a better sound (we'll use SoX as an example, since I've not heard anyone make the claim that it cannot resample audio properly).

Lastly, and most importantly. Where is the testing that has demonstrated these claims to actually be the case, so something like proper blind tests (when upsampling native sample rates, to higher ones, produces audibly superior sound, and by what mechanism).

 

[redshift]

Okay, now - since you're I caught you in the mood for lecturing. Explain to the uninitiated, what exactly it is you think you're hearing (and how it will ALWAYS be an improvement, yet somehow never a detriment as is the usual case for people who take such claims to be true) taking a 8kHz recording, and upsampling to 192kHz.

So what I am asking for is, what has changed in the audio file itself that has now become audible from a metrics standpoint, and also I want to understand by what mechanism this is always going to be a better sound (we'll use SoX as an example, since I've not heard anyone make the claim that it cannot resample audio properly).

Lastly, and most importantly. Where is the testing that has demonstrated these claims to actually be the case, so something like proper blind tests (when upsampling native sample rates, to higher ones, produces audibly superior sound, and by what mechanism).

Classial signals and systems. Run of the mill DSP + ADC/DAC functionality.

 

[amirm]

Says who?

Ever heard of 1 bit DAC’s? Funny how that could work?

https://en.m.wikipedia.org/wiki/1-bit_DAC

That trades off sampling rate for bit depth. It gives no new information as there is none to give you.

If this were not so we would make all digital cameras 1 megapixel and then split out 100 megapixel images out of them. I hope you know that is an impossibility.