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I live about 20 minutes from Amir and have a T778 on the way. I've already touched base with him and we should have some tests done as soon as my unit comes in
Nice. Could you please have him test what is known as "Rhye distortion"? There are reports that this issue affects even other brands like Arcam and Marantz (they all use probably the same chip/software). NAD could reproduce the issue but they didn't fix it in the past 6 months.

More details:
https://www.avsforum.com/forum/90-receivers-amps-processors/2895049-nad-758-v3-349.html#post59113332
https://www.avsforum.com/forum/90-receivers-amps-processors/2895049-nad-758-v3-395.html#post59708154
 

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Hello
I want to buy an av receiver and I'm looking at several options, one of them is nad t 778, I like it a lot but I have a series of doubts and if someone can solve it for me
1 this av will have option of the bass control of dirac?
2 if you don't have the option of bass control because it has dual subwoofer output ??
3 that double subwoofer output as it works today? I suppose as if it were one and having to adjust the phase by hand.
Thanks a lot
 

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Nice. Could you please have him test what is known as "Rhye distortion"? There are reports that this issue affects even other brands like Arcam and Marantz (they all use probably the same chip/software). NAD could reproduce the issue but they didn't fix it in the past 6 months.

More details:
https://www.avsforum.com/forum/90-receivers-amps-processors/2895049-nad-758-v3-349.html#post59113332
https://www.avsforum.com/forum/90-receivers-amps-processors/2895049-nad-758-v3-395.html#post59708154

Thanks for those links - I think the page linked in the second one is likely to be an accurate summary of the issue.

I'll admit that I don't know enough about signal processing to say why the filtering would result in a clipped signal if headroom wasn't created first.

It's interesting that the problem (for me at least) doesn't seem to be there when dealing with multichannel streams, when playing the exact same content through the same source. It really seems like they remembered to create headroom digitally before bass managing multichannel content, but entirely forgot that step when bass managing stereo content.

Of course it still could be a weird design/chip issue. I'm holding out hope that the engineers simply forgot that people don't always run their mains full range when listening to music :D
 

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Thanks for those links - I think the page linked in the second one is likely to be an accurate summary of the issue.

I'll admit that I don't know enough about signal processing to say why the filtering would result in a clipped signal if headroom wasn't created first.

It's interesting that the problem (for me at least) doesn't seem to be there when dealing with multichannel streams, when playing the exact same content through the same source. It really seems like they remembered to create headroom digitally before bass managing multichannel content, but entirely forgot that step when bass managing stereo content.

Of course it still could be a weird design/chip issue. I'm holding out hope that the engineers simply forgot that people don't always run their mains full range when listening to music :D
The issue is not limited to bass managed 2 channel content. The problem is the high pass filter pushing the signal over 0dBFS (= highly audible digital distortion). A small amount of headroom would fix this.
 

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The issue is not limited to bass managed 2 channel content. The problem is the high pass filter pushing the signal over 0dBFS (= highly audible digital distortion). A small amount of headroom would fix this.
I do understand what happens when the high pass pushes the signal above 0dBFS, which results in the awful digital distortion. In my case, with every receiver I've tried (Arcam AVR550, NAD T778, & Denon X3600H) the distortion issue disappeared if I set the output of my PC to 7.1 rather than stereo. In both cases, the high pass filter was active.

It seems to be treating PCM surround formats differently to PCM stereo formats.

Unfortunately, BluOS on the receiver is limited to PCM stereo only, so it suffers from the same issue when I stream music to the receiver directly.
 

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I do understand what happens when the high pass pushes the signal above 0dBFS, which results in the awful digital distortion. In my case, with every receiver I've tried (Arcam AVR550, NAD T778, & Denon X3600H) the distortion issue disappeared if I set the output of my PC to 7.1 rather than stereo. In both cases, the high pass filter was active.

It seems to be treating PCM surround formats differently to PCM stereo formats.

Unfortunately, BluOS on the receiver is limited to PCM stereo only, so it suffers from the same issue when I stream music to the receiver directly.
Interesting observation. The PC might alter the signal when repackaging into 7.1?
 

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Thanks @Lonestar1027, Amir has my AVR850 now. I’m interested to see the performance of the digital / DAC architecture on both of our units. I use a Node via coax for streaming and everyone else is HDMI so not too concerned analog performance which is what so many mainstream reviewers measure. The onboard amps of my Arcam are impressive. The 778 is intriguing as it has BlueOS, Dirac and Hypex amps along with solid DACs, I just hope for high performance integration of the “parts.” The X3600 results were impressive and used in 11 ch mode with an XTC or Monolith amp could be a great solution.
 

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Interesting observation. The PC might alter the signal when repackaging into 7.1?
I can't rule it out. For what it's worth though, two different windows PCs and a MacBook produce the same results when switching to 7.1 instead of stereo.

I don't have another device that would allow me to change the output format when playing a stereo file.

Have you seen any evidence of the distortion problem existing with surround formats?
 

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Hi all, quick question. I'm used to turn on dynamic volume with audyssey on my current denons. I'm going to buy this NAD anyway but I'd like to know how the T 778 manage to compress the dynamic range of the audio. I read from the manual there are two setups named Dolby and DTS - Dynamic Range Control. In particular I'm wondering why there are two separated features for the two distinct audio codecs. Do they work in a different way, with different processing? And then, what about PCM? Can I reduce the dynamic range for PCM inputs? The question comes as I own an apple tv which only output is PCM.

I understand this might a topic you don't wont to cover in a high-end amplifier but still, i'd like to use it in special occasions when I need to use the APTV a more quite environment.

Thanks.
 

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Hello
I want to buy an av receiver and I'm looking at several options, one of them is nad t 778, I like it a lot but I have a series of doubts and if someone can solve it for me
1 this av will have option of the bass control of dirac?
2 if you don't have the option of bass control because it has dual subwoofer output ??
3 that double subwoofer output as it works today? I suppose as if it were one and having to adjust the phase by hand.
Thanks a lot
The subwoofer outputs are split from a single channel. It's no different than using a Y-splitter.
 

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Hi all, how is everyone managing the integration of dual subs on the T778 since the sub outputs are not discrete? The DLBC module would be the best solution but as of now, just not sure when it will be available for NAD. I was thinking of getting a Minidsp 2x4HD for now to integrate then run Dirac after.
 

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Hi all, how is everyone managing the integration of dual subs on the T778 since the sub outputs are not discrete? The DLBC module would be the best solution but as of now, just not sure when it will be available for NAD. I was thinking of getting a Minidsp 2x4HD for now to integrate then run Dirac after.
In any case you need to measure with REW or similar. First step would be to find good locations for both subs. Depending on the result a miniDSP might not be needed at all.
 

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In any case you need to measure with REW or similar. First step would be to find good locations for both subs. Depending on the result a miniDSP might not be needed at all.
Like Markus said, I measured with REW and found the best location. Best here means limited and constrained to where you can actually place the subs, this will be a compromise. With miniDSP make fine tune adjustments to potentially maximise integration between them, then run Dirac. I used painters tape on the floor to keep track of placement according to the results I was getting. Nevertheless, room can really make or break things, depending on how flexible you can be with the location of the subs...
 

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I do understand what happens when the high pass pushes the signal above 0dBFS, which results in the awful digital distortion. In my case, with every receiver I've tried (Arcam AVR550, NAD T778, & Denon X3600H) the distortion issue disappeared if I set the output of my PC to 7.1 rather than stereo. In both cases, the high pass filter was active.

It seems to be treating PCM surround formats differently to PCM stereo formats.

Unfortunately, BluOS on the receiver is limited to PCM stereo only, so it suffers from the same issue when I stream music to the receiver directly.
While NAD doesn't communicate useful information whether they will fix this issue or not I went ahead and did some more testing to find out what's causing "Rhye distortion". Think I found it: https://www.avsforum.com/forum/90-receivers-amps-processors/2895049-nad-758-v3-396.html#post59724792
 

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While NAD doesn't communicate useful information whether they will fix this issue or not I went ahead and did some more testing to find out what's causing "Rhye distortion". Think I found it: https://www.avsforum.com/forum/90-receivers-amps-processors/2895049-nad-758-v3-396.html#post59724792
Thanks for your investigations Markus.

It really does seem like they only thought to digitally lower levels before the before crossovers are applied to surround content, but not with stereo. There obviously is a different signal processing flowchart for stereo content vs. surround (due to bass management and the LFE channel), but the headroom is clearly needed in both scenarios.

Perhaps they thought the extra headroom was warranted for surround content due to the LFE channel. With surround content, the sub signal is essentially the result of two upstream low pass filters (both the redirected mains bass, and the low passed LFE track). With stereo content, the LFE track is inactive so there's few filters applied upstream of the sub signal.

It seems like a huge oversight to assume that stereo content wouldn't also require some headroom before the crossovers.

This is assuming that the Windows or MacOS audio mixer is not responsible for lowering the levels while streaming files in surround containers, and it is indeed the NAD which is causing this difference.
 

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Hi all, how is everyone managing the integration of dual subs on the T778 since the sub outputs are not discrete? The DLBC module would be the best solution but as of now, just not sure when it will be available for NAD. I was thinking of getting a Minidsp 2x4HD for now to integrate then run Dirac after.
A MiniDSP 2x4 will be needed if:

Your subs are not equidistant from your seat, and

Your subs do not have a true time delay or do not have enough delay adjustment to amount for the difference in distance. Rythmik and PSA subs have delay adjustments. You'll want to investigate others - most have a phase adjustment, which is not suitable for this.
 

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A MiniDSP 2x4 will be needed if:

Your subs are not equidistant from your seat
Not true. At low frequencies the room swamps the response anyway. So while delay can be beneficial, time aligning subs based on their location is not a prerequisite.
 

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Not true. At low frequencies the room swamps the response anyway. So while delay can be beneficial, time aligning subs based on their location is not a prerequisite.
I just cited the general recommendation, but I should probably have said "may" need one. If your subs are on opposite ends of the room like mine, you only get acoustic coupling from 30Hz and down. However without auto room EQ, I just adjust delays until I get the best FR, and if that means they end up time aligned, great, but I personally haven't sought out that exercise specifically. Sometimes the best response isn't with time alignment.
 
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