DSpeaker Anti-Mode 2.0 Dual Core Review & Comments

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Reviewed by Russell Williams, 1st October 2012. This is the best subwoofer EQ device I've tested to date. That it goes so far beyond that is, quite simply, brilliant.
Read the full review...
 
Thanks Russell as ever for your informative and enjoyable write up.

"A point worth noting is that the VS8053 can only handle license free digital codecs, which means WAV (PCM & IMA ADPCM), Flac and Ogg Vorbis" As an aside, ALAC has been open source for a while now so I wonder why that is not in their list of formats.

Question -I assume these formats relate to the USB input only? Being a bit thick here but I assume my ALAC music from my Apple TV2 via optical out will be fine. I am happy with my anti-mode 8033 but am trying to understand if the Dual Core would easily let me equalise my 2.0 and 2.1 listening but still handle the .1 in my 5.1. I use an AV9 processor with active fronts. I have checked out the connection examples here DSPeaker-Anti-Mode 2.0 Dual Core connection examples but am still not clear.
 
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Question -I assume these formats relate to the USB input only?

Most USB audio devices support only PCM (of varying bit-depths, sample rates and number of channels), and any audio format decoding is performed on the PC. Anti-Mode 2.0 Dual Core is no different, so what formats the DSP's decode with their ROM firmware is more or less irrelevant when USB is concerned.
 
Most USB audio devices support only PCM (of varying bit-depths, sample rates and number of channels), and any audio format decoding is performed on the PC. Anti-Mode 2.0 Dual Core is no different, so what formats the DSP's decode with their ROM firmware is more or less irrelevant when USB is concerned.

So in what context is the unit limited to these formats then?
 
Good review again Russ :thumbsup:

I have a question for you. Taking the manual functions out of the equation for a minute, and going from a pure auto cal perspective, how would you say this compares to multi eq XT32. I am wandering if this device is worth investing in for someone that has that Audyssey system already, and only wants a fire and forget eq solution.
 
Great review, I'd love one of these to sort out my problem room but sadly it's about £500 more than I'd be prepared to pay for such a device.
Hopefully someone will come up with a much cheaper alternative soon.....Hopefully
Buy the Anti-Mode 8033 then. It's about £200, but it only does subwoofers.

Russell
 
Good review again Russ :thumbsup:

I have a question for you. Taking the manual functions out of the equation for a minute, and going from a pure auto cal perspective, how would you say this compares to multi eq XT32. I am wandering if this device is worth investing in for someone that has that Audyssey system already, and only wants a fire and forget eq solution.
I don't know is the short answer, as I've yet to try it. My prior experience was of the stand alone Audyssey unit that seemed to work wonders with film, but had nothing to do with music making. I've yet to hear an automated unit that applies EQ across the entire frequency range that sounds any different, but reports on XT 32 seem to indicate that it's very capable in this regard.

But you can't buy XT 32 on it's own, so the comparison is somewhat moot.

I can only compare to the SVS Audyssey unit and based on my experience of it and the meaurements, I'd lean toward the Anti-Mode. That it offers solutions to rooms that don't enjoy a preset idea of correct, only seals the deal in my book.

Russell
 
So in what context is the unit limited to these formats then?

None? (Or always, because Dual Core only takes PCM input.)

Take a step back and appreciate the difference between the Anti-Mode 2.0 Dual Core and the components used in its construction.

The components in Dual Core can with their ROM firmware do various things, for example decode compressed audio from their data input and play the result through their DAC's. When combined in Dual Core and given a different firmware to run, they perform filtering (and other processing) instead.

If Russell had some point other than just to give a very short overview of the vs8053 DSP, maybe he'll elaborate.
 
I don't know is the short answer, as I've yet to try it. My prior experience was of the stand alone Audyssey unit that seemed to work wonders with film, but had nothing to do with music making. I've yet to hear an automated unit that applies EQ across the entire frequency range that sounds any different, but reports on XT 32 seem to indicate that it's very capable in this regard.

But you can't buy XT 32 on it's own, so the comparison is somewhat moot.

I can only compare to the SVS Audyssey unit and based on my experience of it and the meaurements, I'd lean toward the Anti-Mode. That it offers solutions to rooms that don't enjoy a preset idea of correct, only seals the deal in my book.

Russell
The only caveat to that for me would be if you happen to have xt32 in your system already. Some processors are now carrying this system, and if you do have that, there is also the pro kit one could add. These two things seems to be in direct competition now. Ive used Audyssey plenty and its always impressed, and working on all speakers it definitely an added bonus IMO, but the lack of any manual tweaking means the AM definitely has a major appeal in that regard.

With XT32 essentially being free (dependant on your processing of choice), it does leave me with some serious head scratching as to what a person might or might not consider. Throw the Audyssey pro kit into the mix and it gets very interesting indeed. If I had xt32 already, would I buy a pro kit or an AM2. Tough call maybe I'd have no idea on how to call, especially given this glowing review.
 
I'd love to see a shoot out between XT32, plus pro, vs PBK, vs ARC, vs both Antimodes, purely for single sub calibration.

As an aside, do ARC and PBK work in the time dimension as i cant find much info about this on the web, and i still wonder if there is a bit of an overhang with my sub. Should get REW going really i suppose but im never confident i get the levels set properly.
 
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I understand that previous versions of anti mode only allow taming of peaks i.e. no ability to apply a smoothing boost. Is this any different?
 
Yes, it can boost dips gently and there is a certain amount of control in the advanced menu as to the limits. Increasing the boost, does reduce the output level of the Antimode as it seeks to maintain it's own dynamic headroom, but this can be off-set by increasing subwoofer gain.

Russell
 
I'd love to see a shoot out between XT32, plus pro, vs PBK, vs ARC, vs both Antimodes, purely for single sub calibration.

As an aside, do ARC and PBK work in the time dimension as i cant find much info about this on the web, and i still wonder if there is a bit of an overhang with my sub. Should get REW going really i suppose but im never confident i get the levels set properly.
PBK and ARC are the same thing in all practical terms, so if you had both, there's little point in running both. The EQ results of PBK/ARC and the Anti-Mode are very similar, from having tried both in this room.

Russell
 
Thanks Russell as ever for your informative and enjoyable write up.

"A point worth noting is that the VS8053 can only handle license free digital codecs, which means WAV (PCM & IMA ADPCM), Flac and Ogg Vorbis" As an aside, ALAC has been open source for a while now so I wonder why that is not in their list of formats.

Question -I assume these formats relate to the USB input only? Being a bit thick here but I assume my ALAC music from my Apple TV2 via optical out will be fine. I am happy with my anti-mode 8033 but am trying to understand if the Dual Core would easily let me equalise my 2.0 and 2.1 listening but still handle the .1 in my 5.1. I use an AV9 processor with active fronts. I have checked out the connection examples here DSPeaker-Anti-Mode 2.0 Dual Core connection examples but am still not clear.

Hi all. I am still not sure when the formats quoted come in to play, or if the Dual Core would/could equalise my 2.0/2.1 speakers, as well as the subwoofer in 5.1.
 
Hi all. I am still not sure when the formats quoted come in to play, or if the Dual Core would/could equalise my 2.0/2.1 speakers, as well as the subwoofer in 5.1.

I use a Dual Core to equalize my 2.1 system. See attached connection scheme.
 

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Reviewed by Russell Williams, 1st October 2012. This is the best subwoofer EQ device I've tested to date. That it goes so far beyond that is, quite simply, brilliant.
Read the full review...

Thanks for the detailed review, Russell.

You state that the Dual Core does not handle 24/96 inputs. However, my Dual Core does take all digital files up to and including 24/96 (though not 24/176/192). I have it connected to the output of my Squeezebox Touch via Toslink, and it performs quite well in my main system (see my prior post for the connection scheme). I'm not sure if the Dual Core then transcodes down, but 24/96 files actually sound better than when played through my former DAC, a Cambridge Audio 840-C.

Guido F.
 
Hi Russell,

Thanks for the great review.

I wonder if you could answer a question for me:

Im considering using this in place of something like an Audiolab M-DAC as a DAC/Pre in my system. I am a sound engineer, and therefore FIR/IIR filtering is something i'm very familiar with and would appreciate the benefits of at home.

However, I was wondering whether it is possible to have SPDIF input, through to an SPDIF output (at full scale) with EQ, while also EQing an analogue input to an analogue output?

The manual implies that you must select a single input source and a single output destination.
 
I have a couple of questions regarding the AM 2.0 DC.

1. I currently have a Bryston BDA-1 DAC. I was wondering if anyone knows if the DAC in the AM 2.0 DC is better than the Brystons or should I keep the Bryston as my DAC?

2. I was going to purchase acoustic panels for my walls, but I got roadblocked by my wife. I'm wondering if the AM 2.0 is a reasonable alternative to acoustic treatments?

Thanks
 
Thanks for the detailed review, Russell.

You state that the Dual Core does not handle 24/96 inputs. However, my Dual Core does take all digital files up to and including 24/96 (though not 24/176/192). I have it connected to the output of my Squeezebox Touch via Toslink, and it performs quite well in my main system (see my prior post for the connection scheme). I'm not sure if the Dual Core then transcodes down, but 24/96 files actually sound better than when played through my former DAC, a Cambridge Audio 840-C.

Guido F.

From the tweekgeek.com site:
DSpeaker Dual Core - Anti-Mode 2.0 Dual Core

What is the sampling frequency on S/PDIF, analog and USB?
The S/PDIF receiver supports the standard PCM rates 32kHz/44.1kHz/48kHz plus 96kHz with upto 24 bits, whatever comes from the source. The S/PDIF receiver can also be configured to support 192kHz. We are not sure if 192kHz will be enabled in the first firmware version.
Analog input rate is by default 48kHz.
Technically, USB Audio supports several configurations from 44.1kHz/32-bit through 192kHz/16-bit. The most useful rates will be provided, the default is 44.1kHz/48kHz with 24 bits.
The audio processing is performed in 32-bit domain, so 24-bit with 48kHz rate is optimal.
What USB version does the USB DAC mode support?
The hardware is USB 1.3 (Full Speed), several USB Audio configurations can be supported. The default is 24-bit 44.1kHz/48kHz
=============================

They also told me on the phone that USB reception will be upgraded in future firmware iterations so that very hi-rez files can at least be played back. However, they did say that it wont SOUND better, given that their DAC process will still use local clocking and still upsample everything to 6.1mhz.

I do have one question though that has not been answered anywhere. When you run the "typical" correction, what frequency range is impacted? Is it up to 250hz or up to the full 500hz that is possible in Advanced mode. I think there was a youtube video where this was discussed...
 
From the tweekgeek.com site:
DSpeaker Dual Core - Anti-Mode 2.0 Dual Core

What is the sampling frequency on S/PDIF, analog and USB?
The S/PDIF receiver supports the standard PCM rates 32kHz/44.1kHz/48kHz plus 96kHz with upto 24 bits, whatever comes from the source. The S/PDIF receiver can also be configured to support 192kHz. We are not sure if 192kHz will be enabled in the first firmware version.
Analog input rate is by default 48kHz.
Technically, USB Audio supports several configurations from 44.1kHz/32-bit through 192kHz/16-bit. The most useful rates will be provided, the default is 44.1kHz/48kHz with 24 bits.
The audio processing is performed in 32-bit domain, so 24-bit with 48kHz rate is optimal.
What USB version does the USB DAC mode support?
The hardware is USB 1.3 (Full Speed), several USB Audio configurations can be supported. The default is 24-bit 44.1kHz/48kHz
=============================

They also told me on the phone that USB reception will be upgraded in future firmware iterations so that very hi-rez files can at least be played back. However, they did say that it wont SOUND better, given that their DAC process will still use local clocking and still upsample everything to 6.1mhz.

I do have one question though that has not been answered anywhere. When you run the "typical" correction, what frequency range is impacted? Is it up to 250hz or up to the full 500hz that is possible in Advanced mode. I think there was a youtube video where this was discussed...
Thanks for the info. This parallels what the DSPeaker site says, as well as my experience with files up to 24/96. It's still not completely clear to me what actually happens to such files, internally, as they are processed by the DC. But the resulting sound is excellent in all respects.

I also tried the USB input (fed from my Squeezebox Touch via EDO app) and it sounded awful, including audible noise akin to hiss. Curiously, the USB input icon automatically changes to SPDIF as soon as signal reaches the DC, which suggests to me that the DC is converting USB to SPDIF internally, much as an external converter would. I tried this as well (the external route) via a Music Fidelity VLink II, which sounded very good, but not as good as the direct SPDIF. Will see what any updates to the firmware might bring.

The typical calibration appears to affect the range below 250Hz, as shown in the attached graph for my room.

All in all, a remarkable device.

Guido F.
 

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Thanks Guido,

This makes sense. I agree this device is great. Very disruptive to the status quo!
 
I just heard back from the folks in Finland and here is the reply to the frequency range for a TYPICAL correction:


The high limit for 'typical' is 150Hz. You see the limit as a dotted line in the measurement menu -> view before/after.
 
I use a Dual Core to equalize my 2.1 system. See attached connection scheme.

Thanks for that. Still not sure if I can do 2.1 stereo and the .1 in my 5.1. :confused:
 
Juanty,
the 2.0 option in Corrections menu covers 2.1 as well. Thus, if you choose that option and have the SW connected via the preamp or the DC, it will take the SW into account in the calibrations. I am sure of this for 2 reasons:
1) The manual says to turn up the SW to the max freq cutoff for the SW while playing the 2 main speakers, or even better disable the SW X-over altogether.

2) I could hear that the SW was indeed playing during the room correction process, so the Mic was picking up the output of the 3 speakers collectively. The only caveat is that you have to ensure the active SW is at appropriate volume setting.

===========================
The guys from Dspeaker are very responsive.
Q&A from Dspeaker:

What are the current max sample rates accepted into the USB and Toslink digital input channels? I was under the impression that Toslink was 24/96 and that USB was 24/48 and that possibly in a future firmware update, USB can go up to 24/192.
Currently the Toslink can go 24/96 (latest firmware) and USB 48/24.


Am I correct here? I hope so because I have at least 20 hi-rez albums of 176khz and higher that I wont be able to play otherwise from my laptop.

You should be able to play them, as USB should be able to send the data in 48/24. Also the ADC of the AM2DC is 192kHz, so you could also use analog input if nothing else works.

Please let me know the current status and future upgrade plans in the works.

Yes we will. We'll announce any new firmware features on our website, so stay tuned.

Any plans to allow Toslink and USB input to accept 24/192? I have a lot of hires albums and can make more from my SACDs.


It is difficult to say yet about 24/192 as we are still planning the 24/96. However, the analog input already is 192kHz and USB should be able to feed the data in 24/48 format. With 6.144MHz reclocking, 24/48 is essentially as good as 24/192 as there is no jitter (which is more relevant than the ultrasonics).
 
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