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miniDSP Tide16 - Holy Grail with 16 Channel Atmos/DTS:X, high SINAD

peng said:
I know, I have quite a few of those, but now looking for movies, DTS: X or Atmos.
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There is next to no 96KHz Atmos as it halves the number of elements you can process in the system from 128 to 64, this actually becomes a serious limitation due to the way Dialogue, Music and Effects are separated and the fact multiple beds eat in to the number of possible objects.

I’d be happy with a processing bypass on the Tide16 so it can consume and reproduce PCM at high sample rates with all the processing bypassed but SRCs to 48KHz if you turn it on.
 
I'd be really happy if studios would produce more quality content at 48 than anything I could or could not hear from 96.

Unlikely that I will ever buy another disc, but I think my emphasis is clear - content, and more of it, not the sampling rate would be my preference.

Kind of funny to look for the discs that could prove very little - sorry to disagree with some of the gold members. We do have tendency on this forum to go behind blinds.
 
Curvature said:
Ever since I learned the mechanics of the sample theorem and audio compression, I have not given much thought to formats.
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Ever since I have fallen in love with titles that touch my heart, I have not given much thought if they came in HD or 4K, or just plain 5.1 that everybody gets nowadays as standard.
 
Oddball said:
Ever since I have fallen in love with titles that touch my heart, I have not given much thought if they came in HD or 4K, or just plain 5.1 that everybody gets nowadays as standard.
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Anyone who likes older media should appreciate its container and characteristic artefacts.
 
GXAlan said:
View attachment 506151

Single point measurements cannot distinguish these scenarios. As you add stuff like wides, heights, and various rears, being able to have the sensation of a bubble of sound benefits from this type of technology. Sony and Yamaha AVRs did a great job of creating a bubble which is why they were popular despite idiosyncrasies and “SINAD” issues. The less ideal your speaker placement, the more value something like Trinnov is.

When Trinnov first entered the home market, they thought of remapping as a good solution for surround sound setups with poor layout. They then decided to move upmarket where good layout plus remapping gives you amazing immersion.

This also provides improved precision in measurement since you get repeated data 4x as fast for amplitude.
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My comment that you're responding to was specifically in response to another comment about time alignment. Detecting where sound sources are in space for other reasons is something completely different.
 
Curvature said:
Signal delay (constant per frequency) shouldn't affect final alignment, regardless of the cause.
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It does where it is different for different sources, which is common.
 
Ultrasonic said:
It does where it is different for different sources, which is common.
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In what way does a upstream source outputting the signal cause a per channel difference? That would imply that the source is broken.

Turntable, N64, Apple TV, whatever. They'll each have their own signal delay. The only thing they will affect, potentially, is audio/video sync, which should be set per input. Unless I'm missing something, I don't see how sub vs. main vs. satellite alignment will be off once set at the AVP.
 
Can this prepro let a user do 3 way active crossovers on the fronts, for instance, similar to the one from Storm Audio, but at a reasonable cost? Or did it get dumbed down into another one of the regular prepros?
 
multichannel_guy said:
Can this prepro let a user do 3 way active crossovers on the fronts, for instance, similar to the one from Storm Audio, but at a reasonable cost? Or did it get dumbed down into another one of the regular prepros?
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Thats what minidsp has always been about, yes can run active 3 way speakers.
The only limitation currently is you cant liad custom FIR filters into each channel, but reading between the lines that functionality will probably come down the track as the firmware matures...
 
AudioJester said:
Thats what minidsp has always been about, yes can run active 3 way speakers.
The only limitation currently is you cant liad custom FIR filters into each channel, but reading between the lines that functionality will probably come down the track as the firmware matures...
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I am not sure obviously but when I "read between the lines" it looks to me like miniDSP had to agree to eliminate custom FIR filters for each channel in exchange for being able to include DIRAC Art.
 
Keith_W said:
Is there any way to achieve lip sync with streaming video?
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Any Kodi implementation can delay/advance audio and video independently up to several seconds. But you are of course limited to what Kodi can play. There are YT and Netflix plugins but being 'unofficial' are obviously limited in various ways. But for locally stored content it's fine.
 
chelgrian said:
There is next to no 96KHz Atmos as it halves the number of elements you can process in the system from 128 to 64, this actually becomes a serious limitation due to the way Dialogue, Music and Effects are separated and the fact multiple beds eat in to the number of possible objects.
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Thank you for the info, I thought it was just me not knowing how to search. Again, I don't think it matters, just being curious that's all, though I do think miniDSP should eventually up their game to 96 kHz as there obviously are many people who for whatever reasons, are convinced that they could hear better SQ, often cited more "air" if sampling rate is higher. To me, that's just typical internet hearsay, once someone found a seemingly good good and make some claims, over time, more people would believe it, actually heard(thought they heard) it and keep spreading it..
 
Curvature said:
In what way does a upstream source outputting the signal cause a per channel difference? That would imply that the source is broken.

Turntable, N64, Apple TV, whatever. They'll each have their own signal delay. The only thing they will affect, potentially, is audio/video sync, which should be set per input. Unless I'm missing something, I don't see how sub vs. main vs. satellite alignment will be off once set at the AVP.
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Sorry, I meant acoustic source. E.g. a subwoofer with built in DSP vs main speakers without.
 
peng said:
Thank you for the info, I thought it was just me not knowing how to search. Again, I don't think it matters, just being curious that's all, though I do think miniDSP should eventually up their game to 96 kHz as there obviously are many people who for whatever reasons, are convinced that they could hear better SQ, often cited more "air" if sampling rate is higher. To me, that's just typical internet hearsay, once someone found a seemingly good good and make some claims, over time, more people would believe it, actually heard(thought they heard) it and keep spreading it..
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My hearing has passed the point of being able to hear high frequencies, but just because I can't, doesn't mean that others can't.
While there's definitely a lot of evidence that CD is transparent, much of it is public domain and widely read and taken as gospel.
There is also evidence that high resolution audio does make an discernible difference , but it tends to sit in AES papers that cost money, and are not widely read.
Where you have contradictory evidence like that, my view is that you should take the result with the highest positive result, rather than the lowest negative result.
There's also evidence that just because you can't hear anything under 20Hz or over 20kHz, it doesn't mean that a 20 to 20k bandwidth is transparent.
Yes that sounds completely unintuitive, but so far we have all been making the assumption that audibility can be based on audibility of continuous tones.
There was some research at Kyoto University and published in the Journal of Neurophysiology Volume 83 Issue 6 that concluded that although listeners could NOT detect any sounds that were high-pass-filtered at 22kHz, they COULD tell the difference between full-range audio, and audio that had been low-pass filtered at 22kHz.
 
Sokel said:
Still, that's fixed, along with the distance.
Mic's don'tcare about it.
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Mics care in that they measure it, which is what we want. Note my point was accurate geometric positioning information of acoustic sources misses this aspect and so isn't the gold standard for time-alignment.
 
Ultrasonic said:
Mics care in that they measure it, which is what we want. Note my point was an accurate geometric positioning information of acoustic sources misses this aspect and so isn't the gold standard for time-alignment.
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Mics (and apps) care about capturing the signal the moment it emerges.
Smart mics (like arrays) also know from where this signal emerges.

Add the above, also add the levels info and you're there.

A little OT, but it's the same way law enforcement now uses arrays to pinpoint a gunshot.

Here's a relevant article:


(Edit: I see my crystal ball and wish for a sphere-like array, along with the app to visualize it, can you imagine that? )
 
welwynnick said:
My hearing has passed the point of being able to hear high frequencies, but just because I can't, doesn't mean that others can't.
While there's definitely a lot of evidence that CD is transparent, much of it is public domain and widely read and taken as gospel.
There is also evidence that high resolution audio does make an discernible difference , but it tends to sit in AES papers that cost money, and are not widely read.
Where you have contradictory evidence like that, my view is that you should take the result with the highest positive result, rather than the lowest negative result.
There's also evidence that just because you can't hear anything under 20Hz or over 20kHz, it doesn't mean that a 20 to 20k bandwidth is transparent.
Yes that sounds completely unintuitive, but so far we have all been making the assumption that audibility can be based on audibility of continuous tones.
There was some research at Kyoto University and published in the Journal of Neurophysiology Volume 83 Issue 6 that concluded that although listeners could NOT detect any sounds that were high-pass-filtered at 22kHz, they COULD tell the difference between full-range audio, and audio that had been low-pass filtered at 22kHz.
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If I remember right, you are also an EE, so you must have a good understand of Fourier, Nyquist etc, 48 kHz can cover high frequencies up to 24 kHz (yes I know there are other things like ringing, and other stuff to consider), that gives a margin of 4 kHz above the generally accepted human limit of 20 kHz. Transparency, in absolute sense it one thing, in practice, even if "others" can hear higher frequency, it doesn't mean (in fact highly unlikely) that they could discern the difference due to including those higher frequencies. That's not even for single pure tones, let alone in music/movie tracks when the waveforms are complex. Those research you cited, are just research, I know may PhDs, so I know how those things are, but wouldn't say any more about that. Anyway, I am all for devices capable of 96 kHz sampling rate regardless, so that there would be a huge margin for even super humans so no argument there. As to 192 kHz, good to have too, as long as price wouldn't be jacked so high lol..

By the way, thank you again for that link but I couldn't order those 96 kHz movies on Amazon.ca, it linked to Amazon.com only. Will have to try elsewhere..
 
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