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Generic Budget USB to AES Converter Review

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Few members in Korea been using similar ones for Genelec 83x1s. (like 2~3 years ago?)

They came to conclusion 96kHz versions are faulty and only 48 or 192 ones should be used.

Not sure this is the same product (because I remember the ones used by members had fixed bandwidth). But 96kHz converters from Ali... I won't use any of them.
 
Few members in Korea been using similar ones for Genelec 83x1s. (like 2~3 years ago?)

They came to conclusion 96kHz versions are faulty and only 48 or 192 ones should be used.

Not sure this is the same product (because I remember the ones used by members had fixed bandwidth). But 96kHz converters from Ali... I won't use any of them.
yet I sent version 48 to ASR which seems faulty
 
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Why? If the stream is PCM (i.e. audio, not AC3/DTS etc.), what difference does it make if the volume operation - dividing each sample value by a fixed number - is performed before or after sending the samples over USB?

I believe it is what Genelec recommends:

In principle, the best result is achieved by adjusting the volume level with GLM.

Attenuating a signal digitally can lead to loss in resolution and that is why it is recommended to use GLM for level adjustment. GLM adjusts the signal level at a final stage before feeding the power amplifiers in the speaker. This ensures no loss in the signal quality.
 

In principle, the best result is achieved by adjusting the volume level with GLM.

Attenuating a signal digitally can lead to loss in resolution and that is why it is recommended to use GLM for level adjustment. GLM adjusts the signal level at a final stage before feeding the power amplifiers in the speaker. This ensures no loss in the signal quality.
 
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Too bad we can't have decent, cost effective usb to aes...
yes too bad

but for 100 dollars you can get a D10S with a 110 omhs SPDIF/AES cable and the problem is solved

when you have put 10k€ in your GENELEC + TRINNOV, you should not be afraid to add 100 € for a quality product
 
The "In principle" there is important - and a long way from "breaking everything"

In fact correctly implemented digital volume control upstream of the speakers is almost certainly inaudibly different from that implemented in GLM.
 
The "In principle" there is important - and a long way from "breaking everything"

In fact correctly implemented digital volume control upstream of the speakers is almost certainly inaudibly different from that implemented in GLM.
I don't have enough knowledge, I took the information from JK who knows very well about the matter
 
Because I am 10+ feet away and there is no reliable wireless way to control volume over GLM (their wireless remote sucks, nor can it work without direct line of sight) + needs to be accessible via the same 1 remote that is used for TV/Device control.
I was going to suggest the Gevol, but it seems to have disappeared off the web. I believe it was supposed to be controlled via IR or HTTP.

Previous discussion here on ASR.
 
It seems that if I use a Wiim Ultra as the source and then connect the Quloos QU02, it allows volume control of the AES output. My Bluesound node icon couldn't do that. I am going to try again later to rule out user error.

But...now I learnt that Genelec can't go into standby mode due to:

It’s quite common that the digital audio source keeps the digital bit clock active when the source is powered ON. That prevents the monitor or subwoofer to turn ISS sleep state.


It's like you get around one problem and another one shows up lol. Anyone else have had issues with ISS when using converters or even something that natively supports AES?
 
Interesting. Is there any study or literature that proves/supports the loss of phase at high frequencies, and/or showcases Shannon‐Nyquist's flaws?
No, because "sampling theory" "nyquist" and all the other "Monty" BS have nothing to do with the science of human hearing.

It would literally be like asking scientists who have actual science to do, to count angels on the heads of pins. Humans don't sample. So sampling theory is literally a bunch of BS when it comes to actual human sound perception -- it is an artifact of engineering, rather than some kind of universal law that applies to human hearing and sound perception.

There are no biological or neural structures that "sample" -- that's not how human hearing works. If you want to do "audio science" it is very curious to claim to be doing science when the actual subject of the science, who is doing the hearing of the audio, is a radically oversimplified "black box" with no formal theory that defines how it "perceives" . Or even worse, when one's artefacts and tools, combined with blindered-scientific-rationalist-thinking and total lack of knowledge of biology & cognition and/or reality, cause one to create a fake model of human perception that assumes "Nyquist" exists in the brain -- and then sanctimoniuously gaslight those who disagree and actually know something about the subject with insultingly stupid "Monty" videos. (Because "music" and the perception of 3D audio has absolutely nothing to do with sampling a continuous 20 khz sinewave -- rather, it has more to do with the overall system's impulse response to an infinitely short "wave", the exact opposite!).

This "Monty" BS is basically "safe and effective" science -- the science of not knowing, not asking, not finding out, not making a detailed model and theory of the subject being investigated. In short, that's not science -- maybe rename this site to "audio scientism review" and if you sign up now, you'll receive a free monty/redbook-is-safe-and-effective votive candle!

More precisely, Blauert found that the presence of frequency components from about 8,000–10,000 Hz is critical for accurate estimation of elevation. Langendijk and Bronkhorst showed that antero-posterior localization cues occur in the 8,000–16,000 Hz range (Blauert, 1969; Hebrank and Wright, 1974; Langendijk and Bronkhorst, 2002).

The bandwidth of a sound plays an important role in acoustic localization: the broader the bandwidth, the better the localization performance (Coleman, 1968; Yost and Zhong, 2014) under both open-field and reverberant-room conditions (Hartmann, 1983). Furthermore, the spectral content of the sound is an important cue for estimating the distance of the sound source. This type of cue works under two different conditions. Over long distances, high frequencies are more attenuated than low frequencies due to propagation through the air. As a result, sounds with reduced high-frequency content are perceived as being farther away (Coleman, 1968; Butler et al., 1980; Little et al., 1992). However, in order to obtain a noticeable effect, the distance between the source and the listener must be greater than 15 m (Blauert, 1997). For sound sources close to the listener’s head (about 1 m), by contrast, the spectral content is modified due to the diffraction of the sound around the listener’s head. For this reason, for sources in the proximal space (<1.7 m), sounds at lower frequencies (<3,000 Hz) actually result in more accurate distance estimation than sounds at higher frequencies (>5,000 Hz) (Brungart and Rabinowitz, 1999; Kopčo and Shinn-Cunningham, 2011).

Finally, sound frequency appears to play a role in front-back confusion errors. Both Stevens and Newman, and Withington, found that the number of confusion errors was much higher for sound sources below 2,500 Hz. Letowski and Letowski reported more frequent errors for sound sources located near the sagittal plane for narrow-band sounds and for a spectral band below 8,000 Hz. The number of confusion errors decreases rapidly as the energy of the high-frequency component increases (Stevens and Newman, 1936; Withington, 1999; Letowski and Letowski, 2012).
from https://www.frontiersin.org/journals/psychology/articles/10.3389/fpsyg.2024.1408073/full
"REVIEW article:

Auditory localization: a comprehensive practical review"​

Front. Psychol., 09 July 2024
Sec. Auditory Cognitive Neuroscience
Volume 15 - 2024 | https://doi.org/10.3389/fpsyg.2024.1408073

There should also be questions about 20-20Khz "hearing range" based on stuff like this:

According to the duplex theory of spatial hearing, the contribution of each binaural cue to sound localization is dependent on sound frequency. Specifically, ITDs are considered most relevant for localization of low frequency sounds (<1.5 kHz), and ILDs for localization of high frequency sounds (>1.5 kHz)10. Several psychoacoustic studies support this apparent cue-dichotomy for sound localization11-15; however, this theory has also been challenged by other studies showing that each type of binaural cue contributes to sound localization in a wide range of frequencies. For instance, ITDs conveyed by the envelope of high-frequency sounds, as well as the ILDs present in low-frequency sounds, can be used for localization16-18 (especially in reverberant listening settings19). In addition, no clear relationship exists between neural tuning to sound frequency and to ITDs or ILDs. That is, ITDs modulate not only the firing rate of neurons that are tuned to low frequencies, but also those that are tuned to high frequencies20. Neurons in the inferior colliculus of the guinea pig that are tuned to low frequencies even respond maximally to ITDs outside of the physiological range21. Comparably, neural encoding of ILDs in the chinchilla midbrain is frequency invariant22. Thus, binaural cues seem to be relevant for a wider range of frequencies than predicted by the duplex theory, and neurons encode both types of binaural cue irrespective of their frequency tuning.

Cortical mechanisms of spatial hearing​

 
No, because "sampling theory" "nyquist" and all the other "Monty" BS have nothing to do with the science of human hearing.

It would literally be like asking scientists who have actual science to do, to count angels on the heads of pins. Humans don't sample. So sampling theory is literally a bunch of BS when it comes to actual human sound perception -- it is an artifact of engineering, rather than some kind of universal law that applies to human hearing and sound perception.

There are no biological or neural structures that "sample" -- that's not how human hearing works. If you want to do "audio science" it is very curious to claim to be doing science when the actual subject of the science, who is doing the hearing of the audio, is a radically oversimplified "black box" with no formal theory that defines how it "perceives" . Or even worse, when one's artefacts and tools, combined with blindered-scientific-rationalist-thinking and total lack of knowledge of biology & cognition and/or reality, cause one to create a fake model of human perception that assumes "Nyquist" exists in the brain -- and then sanctimoniuously gaslight those who disagree and actually know something about the subject with insultingly stupid "Monty" videos. (Because "music" and the perception of 3D audio has absolutely nothing to do with sampling a continuous 20 khz sinewave -- rather, it has more to do with the overall system's impulse response to an infinitely short "wave", the exact opposite!).

This "Monty" BS is basically "safe and effective" science -- the science of not knowing, not asking, not finding out, not making a detailed model and theory of the subject being investigated. In short, that's not science -- maybe rename this site to "audio scientism review" and if you sign up now, you'll receive a free monty/redbook-is-safe-and-effective votive candle!


from https://www.frontiersin.org/journals/psychology/articles/10.3389/fpsyg.2024.1408073/full
"REVIEW article:

Auditory localization: a comprehensive practical review"​

Front. Psychol., 09 July 2024
Sec. Auditory Cognitive Neuroscience
Volume 15 - 2024 | https://doi.org/10.3389/fpsyg.2024.1408073

There should also be questions about 20-20Khz "hearing range" based on stuff like this:


Cortical mechanisms of spatial hearing​

Thanks for expanding on your stance.
 
It seems that if I use a Wiim Ultra as the source and then connect the Quloos QU02, it allows volume control of the AES output. My Bluesound node icon couldn't do that.
This AES USB audio device is like any other audio device to the host. The host outputs identical stream like it would to an audio device with DAC inside. It's just a decision of authors of the host-side software to avoid volume control operation to a stream which goes into a device which reports its output terminal as SPDIF. Sometimes its called a passthrough mode. Samples are not mangled by the volume operation to preserve bit-perfection which is necessary for non-audio streams like AC3/DTS. Players typically set a different sample type for these streams, the lower layer could base its decision about applying volume control on this. But sometimes the players do not set the non-audio flag (e.g. DTS can be played on a regular CD player with SPDIF output) and then volume control would corrupt the digital stream which the decoder before the actual D/A conversion would not recognize as non-audio (to correctly decode to PCM), but take it as PCM which would produce a very loud ugly noise. IMO the decision to avoid volume control also on devices which report the SPDIF terminal is just a precaution to lower the risk of this issue.

And the designers of Wiim Ultra apparently did no such decision (it's linux, not apple).
 
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No, because "sampling theory" "nyquist" and all the other "Monty" BS have nothing to do with the science of human hearing.
An entire post that demonstrates how little you understand digital audio AND how it relates to the science of human hearing.

Just one example:
More precisely, Blauert found that the presence of frequency components from about 8,000–10,000 Hz is critical for accurate estimation of elevation. Langendijk and Bronkhorst showed that antero-posterior localization cues occur in the 8,000–16,000 Hz range
Digital audio - with all it's sampling - perfectly reproduces those frequencies (back in the perfectly smooth analogue domain - as though sampling had never happened), both amplitude AND phase to levels irrelevant to human hearing. Yes, even redbook. The human ear literally has no way to tell that the signal has ever been digitised. This is because of ""nyquist" and all the other "Monty" BS" - and all the other engineering that has gone into making it work as it does.
 
Because I am 10+ feet away and there is no reliable wireless way to control volume over GLM (their wireless remote sucks, nor can it work without direct line of sight) + needs to be accessible via the same 1 remote that is used for TV/Device control.
Umm, what? The GLM remote is RF, not IR... It most certainly goes through a cabinet or two. If you want to control volume by some other remote, it's the main reason why that'll be a bigger challenge. IR control of GLM would require unsupported tools and hardware; The GLM protocol has been reverse-engineered enough to accomplish this and there are tools available (tested, working), and linux has a lot of IR control stuff available as well. Another route would be always-on GLM software and MIDI control - perhaps easier, perhaps not.
 
Umm, what? The GLM remote is RF, not IR... It most certainly goes through a cabinet or two. If you want to control volume by some other remote, it's the main reason why that'll be a bigger challenge. IR control of GLM would require unsupported tools and hardware; The GLM protocol has been reverse-engineered enough to accomplish this and there are tools available (tested, working), and linux has a lot of IR control stuff available as well. Another route would be always-on GLM software and MIDI control - perhaps easier, perhaps not.
The remote doesn't work unless line of sight for me. Not just line of sight, but has to be within a couple of feet at max. The connection to it also drops time to time. But most importantly is quality of life, if you have to hunt for another remote to control just the volume, that's a no go unfortunately.
 
Why? If the stream is PCM (i.e. audio, not AC3/DTS etc.), what difference does it make if the volume operation - dividing each sample value by a fixed number - is performed before or after sending the samples over USB?
Everything gets resampled and you lose resolution. Very simply put, the lower the output volume, the bigger the digital steps between levels in the signal become; Will one notice it or not is another question.

Re. Genelec SAM's, I prefer to alter the signal as little as possible before it enters my monitors, hence always -0 dB in, because there are no negative side effects in my use cases, even on the road. The RF remote works fine (I like the older model with less buttons) and the wired pot is very nice, if it's an option.
 
The remote doesn't work unless line of sight for me. Not just line of sight, but has to be within a couple of feet at max. The connection to it also drops time to time. But most importantly is quality of life, if you have to hunt for another remote to control just the volume, that's a no go unfortunately.
That's either a defective GLM brain, RF remote, remote battery or truly massive EM/RF interference going on... But yes, the QoL issue is understandable.

A hardware hack would work too, in addition to what I already mentioned. The GLM brain expects a logarithmic pot for the physical volume control (exact value depends on age of unit, but 10k ohm will work), so instead of the GLM protocol / MIDI, one could just figure out a way to feed it a suitable resistance. :)
 
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