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Is there any obvious differences in quality between the SPDIF digital outs of the SB3 and the EMU 0404 USB (excepting the convenience of the SB3 and the ability to remove the laptop from the listening room)? Both would be feeding a digital signal into a Tact 2150 XDM which accepts optical, coax, or AES/EBU (though the SB3 does not have AES/EBU digital out).
The SB3 is attractive for its convenience and setup outside the listening area, and the EMU because it will handle Hi Rez files, but I'm wondering about technical qualities for redbook playback.
Thanks!
I have had the SB2 (not SB3) and E-MU 0404 USB in my system using S/PDIF output via Toslink. The digital part of my system is as follows:Network = Ethernet over power line using Netgear adapters
Dell 600m laptop with 1.6 GHz Pentium M processor with 1 GB of RAM
Drive with music is a RAID5 array in another computer, accessed from the laptop via the network.USB out into E-MU 0404 USB
Toslink out of E-MU 0404 USB into Behringer DEQ2496 Toslink in
Toslink out of Behringer into Benchmark DAC1 Toslink inWith the SB2 I was going from its Toslink out to the Toslink in of the Behringer above - same Benchmark DAC1 in both cases.
I have not compared them side by side. I got the E-MU when the display of the SB2 failed. Slim Devices wanted $90 to fix the SB2, so I got the E-MU instead.
I've found the sound of the E-MU to be very good and I prefer it to the SB2. I have found it to be able to reveal lots of detail without sounding the least bit harsh, somewhat like my experience with high-res. I have not used it on anything other than 44.1/16 files though.
I have had a number of problems with playback software though. In order to use S/PDIF with the E-MU 0404 USB, ASIO must be used in the playback software. Here is a summary of the problems I've had:
Foobar2000 0.9.5 with native ASIO : You set up ASIO in the menus, and specify which outputs are activated (Analog 1 = none, Analog 2 = none, S/PDIF 1 = left, S/PDIF 2 = right). Once this is done, playback is completely free of glitches of any kind. However, when you reboot, a bug in Foobar's ASIO support causes it to forget the configuration, even though the display looks like the config is remembered correctly. To fix the problem, one must delete the config, and re-enter it after each reboot. This is a known problem with the Foobar2000/E-MU 0404 USB combination and has been discussed here before and on hydrogenaudio.org - with no solution.
WinAmp 5.5 with Otachan ASIO : This remembers the ASIO setup after each reboot just fine. Initial setup is counterintuitive. You specify a channel offset of 2, meaning that the two active channels are 3 and 4 (S/PDIF 1 and S/PDIF 2 respectively). However there is a problem when gapless playback is selected. For the last song of a playlist, the last second or so is rudely chopped off. Disabling gapless playback fixes this, but causes an annoying gap when listening to live performances where consecutive tracks contain continuous music. I abandoned WinAmp because of this problem. I don't know if this is a WinAmp problem, an Otachan plug-in problem, or an E-MU driver problem.
MediaPortal with Symphy's Pure Audio ASIO plugin : This is an interesting player. It's for audio and video use, and appears that the UI was meant for a television. But the sound quality is great. The native ASIO support does not work at all for S/PDIF with the E-MU 0404 USB because of the inability to do channel assignment. Native ASIO is not bit perfect either. But Symphy's plug-in fixes this problem. It remembers the setup fine after reboot, and does not have the problem of truncating the end of the last song of a playlist either. However, I had a different problem. Every hour or two, the playback simply stops in the middle of a song, and I have to restart MediaPortal to get it going again. I don't know if the problem is with MediaPortal or the ASIO plugin. It doesn't seem to have anything to do with getting the music from a network drive, as I have not experienced this with Foobar2000 or WinAmp. At any rate, I abandoned MediaPortal because of this problem.
At present, I am limping along with Foobar2000, re-entering the ASIO configuration with each boot. I am thinking about using JRiver Media Canter. It would be great if it fixed all my problems.
The hang up is an issue with the thread priorities in winamp. I found you have to set one of the priorities lower than the others in Winamp.
I get zero problems with the ASIO at the critical and Winamp's general priority at realtime. Under "Player" set it to next to highest.
Bob
www.PlateauLight.com
Thanks for the tip. I just tried eight different combinations of player and ASIO priority, and still no go. Combinations tried were highest and second-highest on player priority, along with all four possibilities of ASIO priority for each player priority.
Maybe this is machine-dependent. JRiver is working great so far.
Andy,
Thanks for the great info. Since I planned on using Foobar with the EMU, if I purchased it over the SB3, you saved me a lot of time and frustration.
C.
[IF] you have any balanced cables around you may want to give the AES/EBU connection a try with your Behringer.
Dyn,
Does AES/EBU offer benefits over Coax or Toslink for transmission of SP/DIF?
Thanks.
That's a good idea. I believe the Benchmark DAC supports that as well.
I need to get the software situation cleared up though :-).
If your going with J River for a software solution I think your search is probably over....it just does too much right.
Hey Andy,
Do you adjust the volume with your software, or does the emu output a full unattenuated signal? and the benchmark or further up does the volume?
Do you adjust the volume with your software, or does the emu output a full unattenuated signal? and the benchmark or further up does the volume?
Hi Dawnrazor,
The only digital volume adjustment I do is in the Berhringer unit itself, to avoid digital clipping due to the digital filtering it provides (for the bass only). When this occurs, it's very audible as a crackling sound, but can be eliminated easily by looking at the VU meters of the Behringer, which have a digital clipping indicator, and adjusting the digital level to prevent the clipping. To help with that, I use a reference CD (Miles Davis' "Milestones" - the track "Two Bass Hit"). This song is recorded very loudly, and shows up digital clipping clearly on the Behringer's VU meters. I adjust the Behringer's digital volume control for the minimum attenuation that avoids digital clipping. This ends up being about 2 dB with my setup.
I have disabled the front panel volume control on the Benchmark, and instead use the volume control of my preamp, which can be controlled using the remote. However, I still use the multi-turn volume control trimmers on the Benchmark's real panel as follows:
1) To reduce overall volume such that the remote volume control of my preamp is in a range where it's not jumpy.
2) To get matched volumes between channels at a typical volume setting.
I do the first step empirically. I determine where the preamp's volume control needs to be set in order to get good control with the remote. Then I do a coarse tweak with the Benchmark's rear-mounted trimpots to get close.
To get precise channel matching, I first set the preamp's volume control to match that of typical listening. Then I play a mono WAV file of a sine wave of about 500 Hz. Using an AC voltmeter, I match the measured voltages of the preamp's left and right outputs with the voltmeter by varying the adjustment of the Benchmark's multi-turn pots in back. I can get channel-to-channel matching within hundredths of a dB at the normal listening level. This is possible even with a cheap AC voltmeter, as long as the display is digital (so you can read to the desired resolution).
In practice, this works great. Matching the two channels like this eliminates the "audiophile nervosa" effect from channel mismatch of conventional preamp volume control pots. Integrated circuit control of volume has better matching than motorized pots, but depends on using MOSFET switches in the IC, which cause distortion. The best approach is multi-position switched attenuators, but these get complex and expensive. Using a preamp with a motorized pot, combined with individually adjustable outputs of the DAC, works well for me as an economical solution. YMMV.
Since you are having problems with Foobar and Winamp, and no one else seems to have any issues with them, I am assuming that the creative drivers are part of the problem.
Here is a player with native ASIO support that has a trial. You might get lucky here but probably not. Its ASIO does not allow volume control, so you will be OK.
Its GUI is one of the best I have seen even better than Mp3toys IMHO. It is powerful and somewhat rather unintuitive in some areas due to its power.
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I recommended it to a gentlemen who was struggling with Foobar and his classical collection. It allows you to group albums by Composer, or Orchestra or Conductor among 15 other categories. I thought it would be perfect.
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Well he had problems getting the library set up and since he never told me how the hell he had his files configured I couldn't help.
It will set up a library based on file directory (say individual Artist/ album folders and .wavs ) or by tags if you have a .flac or mp3:
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For kicks, I have set up a library based on tags (for my spooky mp3s) and a library of my .wavs using the file structure option. Both worked great.
It has some interesting features that I haven't yet delved into, and its "playlists" (custom collections) are pretty interesting. It will also random all you albums or songs and you can get pretty specific on how that works.
May be worth a try. One day I plan to test it soundwise, since it supports Winamp DSP plugins, I wouldn't be surprised if it is the same asio driver as winamp, and sounds similiar. But i don't remember having to do ANYTHING as far as mapping. I did with Foobar. So the offset you mention might trip this up.
The demo shuts off after 15 minutes but you can restart it.
Let us know if this works.
Wow! Thanks for that info Dawnrazor. I will try it out this weekend also.
I just downloaded JRiver, and it seems to be working just fine for me. No truncation of the end of a track in gapless mode, and gapless mode seems to work perfectly. I'm going to do an extended listening session this evening, so I'll see how it goes.