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Why Don't High SINAD Receivers Exist?

rhollan

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You're forgetting about the redirected bass from other channels. In the case of 7.1 you end up with 20.2dB required headroom. Then add 10dB for speaker trims and 10dB for EQ and your headroom requirement becomes >40dB.

Now if you send that data via the limited capabilities of HDMI you're making your system worse and not better. Better would be to keep the 32bit audio data right up until the DAC.
Er, that 10 dB I mentioned IS the redirected bass. The 10 dB implied gain on the sub channel is, well, implied. It does not take up dynamic range.

Speaker trims and EQ will eat some too, but thst's an argument for (a) well-matched speakers and (b) room treatment before EQ.

But it seems the crux of your argument is maintaining 32 bit resolution of signals with gain instead of 24 bits. This does not help.

You see, at best you have 24 bits of resolution to begin with. That is not bad:144 dB dynamic range, so losing 20 or even 30 dB is fine with a 115 dB SINAD. All you are doing in preserving 32 bits is raising the noise in your system. If you "overflow" in a processing stage and need 25, or 26, or 30, or 32 bits, you just divide the signal back down to stay in range. It's the same with digital volume control: bad with 16 bit data, OK wirh 24. Remember: you only need enough to match the SINAD of your worst stage. 115 dB SINAD means anything more than 19 bits resolution will be lost anyway.
 

rhollan

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The car is a room and those car-processors do "room" correction, there is even a Dirac version for cars. Can't see any difference there.
And yes a bigger DSP chip might be needed for 192kHz but so what? If you ask $5-10K or more for a AVR/P, at least put the best chips in it. It's not like those DSP chips cost a fortune, e.g. with 2x of these boards you have a 16 channel AVP for a BOM below $1000 (including case, power, connectors, etc).

In would agree that there is no need to process audio samples over 20kHz freq. But I just don't want the downsample "feature". The current situation is quite ridiculous: one has to pay a premium for 192kHz tracks, just to have them downsampled to 48kHz by a so called "highres" AVR/P. That's a wonderful (double) scam.
hanyway. Hires signals are not meant to be processed, other than, perhaps, bass management and IIR parametric equalization.
 

markus

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The car is a room and those car-processors do "room" correction, there is even a Dirac version for cars. Can't see any difference there.
And yes a bigger DSP chip might be needed for 192kHz but so what? If you ask $5-10K or more for a AVR/P, at least put the best chips in it. It's not like those DSP chips cost a fortune, e.g. with 2x of these boards you have a 16 channel AVP for a BOM below $1000 (including case, power, connectors, etc).

In would agree that there is no need to process audio samples over 20kHz freq. But I just don't want the downsample "feature". The current situation is quite ridiculous: one has to pay a premium for 192kHz tracks, just to have them downsampled to 48kHz by a so called "highres" AVR/P. That's a wonderful (double) scam.

Triple scam as those 192kHz files likely have undergone multiple A/D and D/A conversions during production and mastering ;)

Anyway, there's a lot more that goes into pricing of a SSP than BOM. You guys also overestimate the size of the market. Why don't we start a thread that consolidates the specs for an ASR-approved SSP? Then let's call one of the integrators like MDS and ask them for development timeline, process and pricing. From there it's "just" certification, production, fulfillment, accounting and support.
 
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markus

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rhollan

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+10dBFS plus 7 times 0dBFS is NOT +10dBFS. It's +20.14dBFS.

http://www.sengpielaudio.com/calculator-coherentsources.htm


At one point in your signal chain it has to.

The LFE signal is recorded 10 dB down. So, one adds 0dBFS + 7 * -10 dBFS for a total of +10 dBFS which has to be scaled on all channels. The 10 dB that the LFE channel is recorded down is made up in the analog gain stage after the D/A for the LFE channel.
 
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markus

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The LFE signal is recorded 10 dB down. So, one adds 0dBFS + 7 * -10 dBFS for a total of +10 dBFS which has to be scaled on all channels. The 10 dB that the LFE channel is recorded down is made up in the analog gain stage after the D/A for the LFE channel.

+10.14dB to be exact. But if you mix LFE and redirected channels you would need to reduce the other channels by another 10dB to keep the level relationship to the LFE signal which needs to be 10dB higher, hence the 20.14dB.

What about repurposing channels as sub outputs for multiple sub application? How would you approach this? Keep the fixed makeup gain for the subwoofer channel and supply enough subwoofer channels and "standby" DAC's just in case?

Another question, how would you communicate the speaker configuration to the separate upstream device that does bitstream decoding and object audio rendering?

How would you handle setups that have no sub(s), i.e. LFE needs to be redirected to speaker channels?
 
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rhollan

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That extra 10 dB that the LFE channel is down is regained in an analog gain stage if the LFE channel is isolated. If you have "large" speakers that are intended to carry their own bass AND a portion of combined LFE then yes: you have to give up another 10 dB of dynamic range on those channels. Signal summation will cost you dynamic range in a signal somewhere. I would argue it is a bad practice if excessive. By the way: Trinnov is very flexible here.

You have the same problem with excessive trim and EQ. You also have this problem in an integrated solution so it is not unique to separated processing stages. I my case, all my amps are "pro" amps that expect a +4 dBu nominal signal (with 20 dB headroom) except my center channel amp which is a consuner amp that expects -10 dBV, about 12 dB lower. To avoid excessive trim, I use an analog 10 dB pad before it.

Now: each stage that has to be aware of speaker configuration (size, distance, etc.) has to be aware of speaker configuration. A shared, global, "configuration" unit might have this data that all could access (ethernet over HDMI?) in some agreed-upon format, but I would argue that having to share data this way would be a bad system design.
 
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rhollan

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So back to square one ;)
I don't follow. I don't see how an integrated system avoids the problems you describe.

The only way I see is if one processing step increases dynamic range requirements and another reduces them in a known, commensurate manner, or is likely too. But I can't see that in practice or the need for processing parameters to be shared between stages.
 

Spocko

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As Amir will be reviewing the Trinnov Altitude 16 shortly, I welcome you to post questions and measurements requests you'd like to see! I am collecting those questions HERE for @amirm to consider.
 
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BullBuchanan

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I power my livingroom setup off an old laptop that feed into a pioneer elite sc-27 and out to the TV. Since the laptop doesn't output 4k, and the receiver doesn't pass it through, I'm looking to get a new mini pc, but I'd really like to be able to bypass needing a new receiver if possible. I don't know it for certain because I've never been able to find detailed measurements of the SC-27, but I have a feeling I could get better audible performance going with a separate dac/processor and dedicated amp. The detail I get from my THX 789 is considerably more than what I get from my speakers and I attribute that to a lower noise floor.

I was thinking one of the Nord Hypex Ncore multichannel amps would get me as close as possible to the point of diminishing returns, but I really don't know how to handle the dac/processing side. I saw the Okto DAC8 recommended, but are there better values to be had? Will there be any issues handling formats coming from streaming or blu-ray without the use of a separate receiver?

Setup
Windows 10 PC w/EAPO+Peace
SC 27 Receiver
Tekton DIs
 

3dbinCanada

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Not a fan of audiophiles and their alchemy.
 

rhollan

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Well, there's "audiophile" and "audiofool".

Purists wish to alter signal as little as possible and practicians wish to reduce all sources of distortion. There's a "give and take" struggle there.

I'm of the "room treatment then digital correction camp", the latter particularly in the lower frequencies.
 

3dbinCanada

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Well, there's "audiophile" and "audiofool".

Purists wish to alter signal as little as possible and practicians wish to reduce all sources of distortion. There's a "give and take" struggle there.

I'm of the "room treatment then digital correction camp", the latter particularly in the lower frequencies.

I belong to the camp your in. Unfortunately a multi use room like my greatroom (kitchen, dining, and living area combined in big room) doesnt lend well to having room treatments hung on the wall. I had to all my correction strictly through DSP. Below is the process I went through to where I'm at now.

https://www.audiosciencereview.com/forum/index.php?threads/3dbs-great-room-setup.17830/post-717548
 

rhollan

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I belong to the camp your in. Unfortunately a multi use room like my greatroom (kitchen, dining, and living area combined in big room) doesnt lend well to having room treatments hung on the wall. I had to all my correction strictly through DSP. Below is the process I went through to where I'm at now.

https://www.audiosciencereview.com/forum/index.php?threads/3dbs-great-room-setup.17830/post-717548
I hear you. I was in a similar "open concept" situation until I purchased my current home which has a dedicated theatre room.
 
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