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

dsd4life said:
I am waiting for SHARC and Onkyo or whomever to design the DSP in the following configuration and configruable:
1. Flatten
2. Multiple Sub woofer null and wave management
3. Steering like Sony does
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This is already handled by AVRs and processors with Dirac Live, including products like the miniDSP TIDE16.
Steering = upmixer. If you want exactly what Sony does, you’ll need to ask Sony to build such a device for you.
 
Yes, I also admit to drawing up notional plans to add four atmos channels and the required in wall wiring. I might even use a few ascilabs.
 
markus said:
To the best of my knowledge, they do not.
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either you are wrong or minidsp support are wrong since the direct quote from the reply from from minidsp was

‘higher sampling rates would increase the cost of the Dirac Live license‘
 
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chelgrian said:
either you are wrong or minidsp support are wrong since the direct quote from the reply from from minidsp was

‘higher sampling rates would increase the cost of the Dirac Live license‘
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Seems like your miniDSP agent was making a speculative statement. There are no cases where Dirac has charged more for a license for higher sampling rates, afaik.

I would never assume that a support agent for a different company is in-the-know about anything outside their direct purview (and even then sometimes they barely know their support script). Even if it was Dirac support, they wouldn't necessarily know something like that.

We can try to do conjure @Flak to see if they have any knowledge of this topic.
 
I asked them a similar question about my HT and HTx that I did not even purchase DL, the reason given was DSP, that was the limit when there were 8 channels, the Flex and even non Flex can do 96 kHz because the only have 2x4 channels.
 
chelgrian said:
either you are wrong or minidsp support are wrong since the direct quote from the reply from from minidsp was

‘higher sampling rates would increase the cost of the Dirac Live license‘
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The quote does come from https://support.minidsp.com/support...understanding-the-various-dirac-live-versions, but that article is from 2021. To my knowledge, the license is no longer dependent on sample rate.
Another possibility is that miniDSP got a better deal by agreeing to stick to 48kHz, but we’d never see those details. Either way, I know of processors that could run Dirac Live at 96kHz with the same license, but don’t because of performance concerns.
 
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chelgrian said:
The only 96KHz Dirac implementation I’m aware of is in the Theta Casablanca and you can go look up the price of that for your self.
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Theta do claim to run Dirac at 96kHz, and that's been repeated many times. However there was some contradictory evidence here:
I contacted Theta about this issue, and asked...
My understanding is that Dirac Live is strictly fixed at 48khz output by Dirac, yet the specifications for the Theta Casablanca Va indicates that it operates at 96khz.
Given Dirac’s 48khz fixed output, is the Casablanca Va’s ‘input’ processing up to 96khz rather than it ‘post-processing’ at up to 96khz?
If the Casablanca Ca does indeed achieve post-processing’ at up to 96khz, how is that possible given Dirac Live strictly operating at 48khz post-processing?
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It's taken >6 weeks for Theta to respond...
The Casablanca has a master internal clock which determines how the internal circuits communicate with each other.
The PR 5 board has the Dirac DSP on it. The DSP maker chose the Dirac software to work at 48 Khz to be more stable. The Dirac software is converted to a 96 Khz signal before leaving the PR-5 board.
The 96Khz signal is sent to the DAC where the signal is upsampled to a 384 Khz signal before becoming an analog signal. All the sample rates have to be converted to a 96 Khz signal before leaving to go to another part of the processor. One master clock working at 96 Khz determines at which frequency the Casablanca CBVa works at.
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So it sounds like the DSP board I/O interfaces run at 96kHz, but the processing on the board is done at 48khz. This contradicts Theta's own explicit claims:
Today’s upgrade to the Casablanca IVa brings full 4k HDMI HDR with HDMI 2.0 compatible inputs with HDCP 2.2 and includes full HDR capability using state-of-the-art audio bandwidth to 192 kHz, increases DSP processing power by a factor of 10, adds world-class digital room correction and optimization from Dirac Live operating at a world-leading 96 kHz .......
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To the best of my knowledge, the only AVP that processes Dirac at 96kHz is Datasat.
 
welwynnick said:
Theta do claim to run Dirac at 96kHz, and that's been repeated many times. However there was some contradictory evidence here:

So it sounds like the DSP board I/O interfaces run at 96kHz, but the processing on the board is done at 48khz. This contradicts Theta's own explicit claims:

To the best of my knowledge, the only AVP that processes Dirac at 96kHz is Datasat.
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I think Dirac itself can run at 96 kHz and possibly up to 192 kHz. If I remember right even Audyssey can do that.

The device itself is limiting it to lower sampling frequencies for various reasons, one being the DSP bottleneck.

I can use DLBC at 96 kHz sampling rate but it is the PC standalone version. That’s for playback, for calibration, I can only use 48 kHz because I have the Umik-1 that can only do 48 kHz.

Found the following, follow an interview with Chris K, by audioholics.com:

Audioholics: The top frequency for correction is 24kHz, implying that Audyssey is functioning at a 48kHz sample rate. Does this mean that high resolution content (for example 192kHz or 96kHz sample rate PCM) will be downmixed?

MultEQ is capable of applying correction from 10 Hz to 24 kHz.
Chris Kyriakakis: There are two parts to this answer. A loudspeaker does not reproduce acoustic energy above about 24-30 kHz even if it was in the content (with the exception of super-super tweeters), and a microphone cannot capture acoustic energy above that range. So if there is no information captured then, there is nothing for the filter to do up there.

Now, there is content encoded at higher sampling rates of course. We offer MultEQ at 96 kHz and even higher if needed so that the content can be processed without downsampling, even though the MultEQ filters above 24-30 kHz (adjustable) would be doing absolutely nothing. The issue is that doubling the sampling rate also doubles (roughly) the processing requirements needed. This is true for any kind of digital processing not just MultEQ. The AVR makers would have to add significant cost for more DSP processing and they have chosen not to do that. So they decided to use Audyssey at a max of 48 kHz. From an acoustic point of view this makes perfect sense for the reason I explained above.
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and below are from miniDSP support portal:

support.minidsp.com

On the topic of High Sample rate vs Room Correction

We get these questions very often so we thought that we'd take the time to do our best to answer them to the best of our effort. The standard questions are typical: - Can I get Dirac Live for 192k or 384kHz or MHz...? ANSWER: Unfortunately due t...
support.minidsp.com support.minidsp.com

On the topic of High Sample rate vs Room Correction​

Modified on: Tue, 12 Oct, 2021 at 1:13 PM

We get these questions very often so we thought that we'd take the time to do our best to answer them to the best of our effort. The standard questions are typical:
- Can I get Dirac Live for 192k or 384kHz or MHz...?
ANSWER: Unfortunately due to Dirac live high processing power requirements, above 96k processing would require a substantial amount of DSP power which we're not sure would be that beneficial in the end. There are always technical limitations when you add processing to a system and sample rate is one.



Dirac live does a lot more than high sample rate in a way that it's fixing room issues (i.e. 50% of your listening experience). Something that going from 48k to 384kHz will never be able to solve. A high sample rate just doesn't do much to solve the inherent issues of your room I'm afraid. For that reason, while we're biased, we're strong believers that a well-corrected room at 48k DSP is a lot better than an uncorrected room at 384kHz. We've done countless blind test DAC on this topic with extremely high-grade DACS. Room correction always wins so make sure to listen to it if you get a chance. :-)
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peng said:
I think Dirac itself can run at 96 kHz and possibly up to 192 kHz. If I remember right even Audyssey can do that.

The device itself is limiting it to lower sampling frequencies for various reasons, one being the DSP bottleneck.

I can use DLBC at 96 kHz sampling rate but it is the PC standalone version. That’s for playback, for calibration, I can only use 48 kHz because I have the Umik-1 that can only do 48 kHz.
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Would it be fair to say that saved project files would also need to proportionally increase as well? I'm not sure if that trade-off is worthwhile for the miniscule improvements with 96kHz.
 
EWL5 said:
Would it be fair to say that saved project files would also need to proportionally increase as well? I'm not sure if that trade-off is worthwhile for the miniscule improvements with 96
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For it to perform at 96 kHz, yes everything will need to be at that same sampling frequency. There is nothing to gain by resampling up and down. Regardless, I do not believe 96, 192 kHz would make any audible difference, not in DBT or even SBT anyway but if people feel good about it, then it is worth it. It's a function of money, if enough people are willing to pay more, and if manufacturers know that and expect more profits then they will do it, otherwise they won't.

I may purchase a Umik-2 eventually, in order to run DL in 96 kHz, until then, I do not, and will not have a single saved project ran in 96 kHz.

the 48 vs 96 vs 192 kHz sampling isn't much different than 95 dB vs 100 dB vs 105 dB vs 110 dB SINAD, some people will find reasons for the difference to mean "better" sound quality by arguing potentially lower noise, less ringing blablaba but those who make claims typically don't have to and won't provide any proof by credible documented comparison listening tests or other form of objective data that proves the same. The argument of hearing more "air" is easier said that proved. I felt I had heard excellent air in some CD quality stuff than 24b/192 kHz or higher resolution files, it really depends much more on the recording quality imo. So, I am believer in using higher sampling rate in the recording/mastering process, but for playback, 24b/48 kHz is good.
 
peng said:
the 48 vs 96 vs 192 kHz sampling isn't much different than 95 dB vs 100 dB vs 105 dB vs 110 dB SINAD, some people will find reasons for the difference to mean "better" sound quality by arguing potentially lower noise, less ringing blablaba but those who make claims typically don't have to and won't provide any proof by credible documented comparison listening tests or other form of objective data that proves the same.
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This foobar2000 ABX DBT by GoldenSound scored 18/20 (99.98% confidence) by simply comparing the differences between reconstruction digital filters.
The original audio file was at 44.1kHz, and was up-sampled to 176.4kHz using different filters.
One filter was Poly-Sinc_LP (similar to ESS DACs) and the other was Sinc-L ( similar to Chord MScaler).

 
peng said:
I felt I had heard excellent air in some CD quality stuff than 24b/192 kHz or higher resolution files, it really depends much more on the recording quality imo. So, I am believer in using higher sampling rate in the recording/mastering process, but for playback, 24b/48 kHz is good.
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Sure. OTOH, if one can run it all at 24/192 (or more), why not?
From miniDSP Support:
For that reason, while we're biased, we're strong believers that a well-corrected room at 48k DSP is a lot better than an uncorrected room at 384kHz.
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Fully agree but at what point does increasing the sampling rate while maintaining DSP make no audible difference? Is it different from without DSP?
 
My view is that effective DSP equalisation ( like Optimiser, Dirac or Audiolense ) improves transparency, and makes small differences upstream MORE audible.
 
welwynnick said:
The original audio file was at 44.1kHz, and was up-sampled to 176.4kHz using different filters.
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In what way does this relate to an actual use case? Were the files provided so we can see what the actual differences are?
 
kyuu said:
In what way does this relate to an actual use case? Were the files provided so we can see what the actual differences are?
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The rationale was explained in the video, and yes, the files were provided.
Personally I think the files should have been done at 88.2kHz, which is perfectly adequate IMHO, but whatever.
 
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