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TEAC NT 503 or UD-503: Opinions?


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got all the networking working, but no matter what i do, about 8 seconds in between songs.

both as dlna, as shared media, and usb.

 

laptop hooked up to usb works fine with both teac player and foobar.

 

which pretty much means the networking part of this thing is completely useless

 

Sounds like you are saying it doesn't support true gapless playback?

Regards,

Dave

 

Audio system

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  • 1 month later...

This “TEAC-QVCS” also provides 256 steps or 0.5dB steps of volume control, precisely throughout the effective range from -95dB to +24dB. Both the line output and headphone volume can be controlled to the exact level you wish with a remote control, something that's difficult to achieve with a motorized volume control. This feature becomes even more useful when connecting various types of headphones with different impedances.

 

Sorry to revive an old topic, but I recently purchased the UD-503, and have the XLR outs going to active speakers. Is the volume control on the UD-503 an "acceptable" way to control the volume, or should I be looking for a better solution?

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Sorry to revive an old topic, but I recently purchased the UD-503, and have the XLR outs going to active speakers. Is the volume control on the UD-503 an "acceptable" way to control the volume, or should I be looking for a better solution?
On paper yes, but maybe someone who owns one will give some better feedback.
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I have the NT version, but I presume it is extremely similar if not identical. The volume control is in fact in the analogue domain and is 100% transparent, and to my ears, at least, flawless. Each of my channels goes directly from the TEAC to an amp and speakers. I haven't had a chance yet to switch over to XLR from RCA connections, so if anything, it should sound even better.

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  • 2 weeks later...
I have the NT version, but I presume it is extremely similar if not identical. The volume control is in fact in the analogue domain and is 100% transparent, and to my ears, at least, flawless. Each of my channels goes directly from the TEAC to an amp and speakers. I haven't had a chance yet to switch over to XLR from RCA connections, so if anything, it should sound even better.

It's true.I use the internal controller 503 to the power amplifier 400 Wat.The quality of the internal controller is excellent.Clear,crisp, and distortion-free sound.Cranial with an external preamp Parasound for $ 3,500.,built-in controller 503 to win!!!

My system: Audio computer-based Snakeoil with linear supply>PPA v2+power supply for ti4700>, IS0 REGEN+power supply for ti4700>Teac NT503>power amplifier>

 

 

The best Linux system for music!!! :xhttps://www.snakeoil-os.net/   :x

 

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Use native software TeacAvrRemote.NT503 plays without pauses between tracks!!!

 

yes it works this way, requires a computer tied to the usb port. i'm ok with this. however when the nt503 is tied to

the network and you use the ios app, it will not play gapless.

 

whats the point of network enabled if you still need a pc.

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kevin gilmore

Of course not!Without a pause just for the grid connection on the UPNP network.Android or ios doesn't matter!The usb connection is of course also without a pause.

Of course, if the album was recorded without a pause! If recorded, with a pause,2-3 seconds.Checked in JRIVER 21.

My system: Audio computer-based Snakeoil with linear supply>PPA v2+power supply for ti4700>, IS0 REGEN+power supply for ti4700>Teac NT503>power amplifier>

 

 

The best Linux system for music!!! :xhttps://www.snakeoil-os.net/   :x

 

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I just received a new UD-503 in the mail. The first order of business is to confirm that it is in fact the US model. The sticker on the box says 120V~,60Hz,16W, and it came with a US power cord, but the seat for the power cord reads "INL-13- / 10A 250V~" and then after some logos "15A 250V", and above this is printed "Apparaten skall anslutas till jordat uttag," which is Swedish for "The appliance must be connected to a grounded outlet." Is this normal for a US model? I plugged it in, and it seems to be working fine.

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Regarding gapless, I found when playing DSD via the iOS remote into the NT-503, I get an audible gap, but I doubt it is more than 1.0 seconds; probably it is much less. There are third-party server bridging apps that allow a work-around and provide gapless playback, so it isn't a hardware issue in my case. I've also used this with Spotify, without a significant gap between tracks.

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My first impression: PCM playback on this device isn't all that pleasant. I tried listening with and without 4x upconversion on the DSP chip (FIR Sharp as the digital filter setting for the AK4490), and I experienced the same problem – seemingly with ultrasonic noise – that I did with the Bifrost 4490.

 

The Teac advertises -3dB at 80kHz, while the Bifrost gives -1dB at 100kHz; I was optimistic that this difference in analog LPF would mean that the Teac would sound fine to me, but alas the difference is not that great. The AK4490 has two switched-capacitor filters built in, and the one that it uses for all PCM input has a cutoff of 150kHz. But if I upconvert to DSD upstream of the chip, a 50kHz lowpass becomes available. When I set the UD-503 upconversion to DSD and the DSD filter to Cutoff 50kHz, the output is listenable and not fatiguing. As @Miska has pointed out, however, the DSD upconversion is not great. In particular, it allows high levels of imaging around low multiples of the input sample rate. As a consequence, I am letting iTunes convert all output to at least 192kHz sampling in order to minimize that imaging. Subjectively, native rate > DSD upconversion gives tone colors that are slightly dulled, while software upconversion to 192kHz makes for much purer colors.

 

So, iTunes > 192kHz > UD-503 > DSD > 50kHz filter sounds very good to me, but it certainly looks like a convoluted signal path and suboptimal compared to upconversion in software to DSD256. I'm looking into reasonably priced solutions that don't require me to convert my library from ALAC. Roon + HQPlayer is a kind of expensive route to go for that last little bit of sound quality. For experimenting, there is Mansr's SoX fork, the output of which I can play on the TEAC player, but if I'm going to follow that route with a significant number of files, I'll need to buy a very quiet external hard drive.

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Regarding gapless, I found when playing DSD via the iOS remote into the NT-503, I get an audible gap, but I doubt it is more than 1.0 seconds; probably it is much less. There are third-party server bridging apps that allow a work-around and provide gapless playback, so it isn't a hardware issue in my case. I've also used this with Spotify, without a significant gap between tracks.

 

 

Nt503 for use in IOS TEAC HR Remote !!!Only in this case will not pause between tracks!!!Repeat for bezdeliga playback in IOS to use the software

TEAC HR remote

!!!

Server built-in JRIVER22 great!

My system: Audio computer-based Snakeoil with linear supply>PPA v2+power supply for ti4700>, IS0 REGEN+power supply for ti4700>Teac NT503>power amplifier>

 

 

The best Linux system for music!!! :xhttps://www.snakeoil-os.net/   :x

 

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Hello,

 

This is my first post and I am sorry if this has already been answered but here is my question :

Does the upsampling, digital filtering etc is done by the AK4490 chip or is it done by a DSP chip with TEAC algorithm ?

 

By that I mean :

- If I turn off the upconverting on the TEAC, does it bypass the TEAC DSP chip and send NOS signal to the AK4490 which will upsample ? Or is it a complete NOS signal which will be converted in Analogue ?

- If I turn off Digital Filter on the TEAC, does it bypass the proprietary TEAC filter and use a preconfigure filter of the AK4490 ? Or will it turns the AK4490 in External Digital Filter mode and bypass all digital filter in order to use HQplayer filter ?

 

In summary I want to know if I can send a PCM signal upsampled and filtered with HQplayer and bypass everything until the delta-signal modulator ?

 

Thank you very much in advance !

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Hello,

Does the upsampling, digital filtering etc is done by the AK4490 chip or is it done by a DSP chip with TEAC algorithm ?

 

In summary I want to know if I can send a PCM signal upsampled and filtered with HQplayer and bypass everything until the delta-signal modulator ?

The "Digital filter" options select settings of the AK4490; the "Upconversion" options select a separate DSP. Whether that DSP is a solution implemented by TEAC or an off-the-shelf chip that they incorporated into the 503 isn't clear.

 

When you send PCM at 352/384 kHz, the PCM filters on the separate DSP are bypassed, as are those on the 4490. As long as you don't select DSD upconversion, the settings don't matter; the DSP will do nothing, and the 4490 will only do DS modulation followed by the 150kHz switched-capacitor.

 

I'm not completely sure what selecting digital filter off does when feeding lower-rate PCM to the unit, but somebody posted a spectrum from the UD-501 with digitial filter off, and it showed huge amounts of aliasing above the 22kHz Nyquist frequency of the input. As a result of that graph, my working hypothesis is that "digital filter off" performs sample-and-hold to 8x rate (352/384 kHz), maybe for the benefit of people who like the sound of an NOS DAC running at 44.1kHz. I avoid that setting, and I recommend you do the same.

 

I must ask, if you're using HQPlayer, why you want to use it to upsample to PCM instead of DSD. The latter option eliminates aliasing around 352/384 kHz and multiples, bypasses the modulator on the 4490 and gives the option of higher-performance DS modulators, and allows you to select between two switched-capacitor lowpass filters. There's no downside, if your computer can handle it.

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Hello Jay-dub and thank you very much for your very informative reply !

 

The "Digital filter" options select settings of the AK4490;

I tend to believe the same thing as the filters looks like the same to the ones proposed on the AK4490 (with one missing : the super slow roll-off). But how can you be sure ? Also I thought upsampling and digital filter go together ..? I thought the upsampling and DG filter was maybe performed by the FPGA chip and the DAC is runing in EXTERNAL DF mode ..?

 

the "Upconversion" options select a separate DSP. Whether that DSP is a solution implemented by TEAC or an off-the-shelf chip that they incorporated into the 503 isn't clear.
On the website I found that it is a FPGA chip programmed by TEAC.

 

When you send PCM at 352/384 kHz, the PCM filters on the separate DSP are bypassed, as are those on the 4490. As long as you don't select DSD upconversion, the settings don't matter; the DSP will do nothing, and the 4490 will only do DS modulation followed by the 150kHz switched-capacitor.

Agree with that, but is there a difference between : a)passing through the interpolator of the AK4490 but with no processing when sending in x8 PCM and in standard mode. Or b) having the chip in External DF Mode and going directly to the DS modulator, bypassing the interpolator ... ?

 

I must ask, if you're using HQPlayer, why you want to use it to upsample to PCM instead of DSD. The latter option eliminates aliasing around 352/384 kHz and multiples, bypasses the modulator on the 4490 and gives the option of higher-performance DS modulators, and allows you to select between two switched-capacitor lowpass filters. There's no downside, if your computer can handle it.

I will probably ending doing that. I am not currently using HQplayer, but get curious about all the DSD stuff and want to try by myself every configuration possible and maybe do some measurement also... It is more for curiousity than for pure SQ. Also how can you be sure that the DAC run in real bypass mode (volume and DS modulator) and not in Normal Path ??

 

There is so much uncertainty on how everything is handle by TEAC inside that I might go for a DIY or boutique solution based on AK4490 (as Mirand Audio DAC or KLEIN DAC II) as I can be sure of the mode the DAC is really running with those ... Too bad because the TEAC seemed to be a really versatile device.

 

(Please refer to the AK4490 diagram to understand what I am saying : https://www.akm.com/akm/en/file/datasheet/AK4490EQ.pdf)

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I tend to believe the same thing as the filters looks like the same to the ones proposed on the AK4490 (with one missing : the super slow roll-off). But how can you be sure ? Also I thought upsampling and digital filter go together ..? I thought the upsampling and DG filter was maybe performed by the FPGA chip and the DAC is runing in EXTERNAL DF mode ..?

 

The "upsampling" option is external to the DAC chip. At the beginning I suspected it is an AK4137 chip, but not sure anymore as I would have expected better performance from that chip. So maybe it is TEAC's FPGA or DSP stuff afterall.

 

The digital filter selections change the DAC chip filters, at least when the upsampling is off.

 

Agree with that, but is there a difference between : a)passing through the interpolator of the AK4490 but with no processing when sending in x8 PCM and in standard mode. Or b) having the chip in External DF Mode and going directly to the DS modulator, bypassing the interpolator ... ?

 

Since the AK4490 interpolator is 8x PCM, sending 8x PCM means running it in External DF Mode.

 

It is more for curiousity than for pure SQ. Also how can you be sure that the DAC run in real bypass mode (volume and DS modulator) and not in Normal Path ??

 

With DSD inputs, the DAC chip doesn't have 150 kHz filter option available in "normal mode". However, since the TEAC has it available, it must be running in Direct/Bypass DSD mode. I can also clearly see the difference between 50 kHz and 150 kHz modes in wide band (5 MHz) measurements when sending DSD256 there.

 

And generally they don't need the digital volume control of the DAC chip at all, because they have a separate analog volume control.

 

There is so much uncertainty on how everything is handle by TEAC inside that I might go for a DIY or boutique solution based on AK4490 (as Mirand Audio DAC or KLEIN DAC II) as I can be sure of the mode the DAC is really running with those ... Too bad because the TEAC seemed to be a really versatile device.

 

I've been just verifying the functionality with measurements to the extent possible. And seems to work pretty much the same way as UD-501 did, but of course with different chip.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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  • 10 months later...
On 11/18/2016 at 12:19 PM, Miska said:

 

The "upsampling" option is external to the DAC chip. At the beginning I suspected it is an AK4137 chip, but not sure anymore as I would have expected better performance from that chip. So maybe it is TEAC's FPGA or DSP stuff afterall.

 

The digital filter selections change the DAC chip filters, at least when the upsampling is off.

 

 

Since the AK4490 interpolator is 8x PCM, sending 8x PCM means running it in External DF Mode.

 

 

With DSD inputs, the DAC chip doesn't have 150 kHz filter option available in "normal mode". However, since the TEAC has it available, it must be running in Direct/Bypass DSD mode. I can also clearly see the difference between 50 kHz and 150 kHz modes in wide band (5 MHz) measurements when sending DSD256 there.

 

And generally they don't need the digital volume control of the DAC chip at all, because they have a separate analog volume control.

 

 

I've been just verifying the functionality with measurements to the extent possible. And seems to work pretty much the same way as UD-501 did, but of course with different chip.

Maybe the same in functionality as the 501, but big difference is that the analog outputs mute when you plug in a headphone. At least with the UD501 there is a setting to by pass this mode and then it is possible to have signals on analog outputs as well headphones. I presume in the 503 it has to with how the volume control is organised, something the 501 does not have for the analog outputs, only hp. 

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There are other possibilities. Maybe they considered ways of splitting the signal to two separate loads and decided (rightly or wrongly) that any of them would either compromise sound quality or introduce a vulnerability. Maybe an order came from the business office that this DAC was not to have signal-splitting functionality.

 

I must say I find your presence on this forum very strange. One post to advertise a streaming event, and then two years later you came back to necropost this topic (?!) You are, I presume, the Groot who has made a number of recordings for Pentatone. I'm a fan of the surround mixes but not always of the stereo versions. (Orchestral balances often seem a little off.) Welcome to the community, Erdo, but if you want to use this forum as a place to make announcements and pronouncements, I do recommend participating. Reply to some active threads, and get to know some of the more active members.

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10 hours ago, jtwrace said:

Has anyone compared this to the Mytek Brooklyn?  Do they work with the sMS or µRendu without any issue at all sampling rates?  

Also, is an actual remote control included with the dac?  

W10 NUC i7 (Gen 10) > Roon (Audiolense FIR) > Motu UltraLite mk5 > (4) Hypex NCore NC502MP > JBL M2 Master Reference +4 subs

 

Watch my Podcast https://www.youtube.com/channel/UCXMw_bZWBMtRWNJQfTJ38kA/videos

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