Usb dac

Can I use a USB DAC (Meridian Explorer) by connecting to the airport express usb port?

Sorry, but no. The AirPort Express provides either an analog signal or a Toslink optical digital signal at its audio jack.
The USB port will only support a compatible printer.

Similar Messages

PROBLEM with 24/96 bit-rate USB DAC that worked fine until I plugged in two

I'm using a late 2008 15" Macbook Pro that I bought Jan 2009 right before the SD card models came out. Mac OSX 10.6.4 and iTunes 9.2 are installed. I have a Nuforce Icon HDP and a CEntrance DACport 24/96 bit-rate USB DAC (external sound cards) that still work great right now. Note - 16 bit/44 Khz is CD quality, and these 24/96 files are hi-resolution downloads that are closer to SACD or DVD audio.
I have been evaluating a new third 24/96 USB DAC plugged directly into the Macbook USB port (no hub) which has been working fine for over 2 weeks with 24/96 bit-rate lossless music files in iTunes. Audio Midi Setup is set for 24/96 to match the music file's resolution. After making suggestions to the manufacturer they made some improvements and sent me a new version to listen for the differences.
I listened to the new one for a day and it worked fine, but when I plugged in the original test sample into the second USB port at the same time as the new one in order to compare them, the sound became garbled and noisy. This never happened when either one was plugged in by itself. Note, if I make changes to the DAC's bit-rate in Audio Midi Setup I usually close iTunes first and then re-open it when I am done.
Upon hearing this new problem I unplugged both of the new USB DACs and plugged just one of them back in, but now it causes iTunes music playback to pause when one is plugged in and set for 24/96 in audio midi setup. Prior to plugging in both at the same time, either one worked just fine by itself. There is no reason when unplugging one or the other that they shouldn't start working again. But now I can't play any 24/96 music through either one anymore. In quicktime player the songs don't pause like with iTunes, but there is no sound coming through even with just one DAC plugged in at a time. Then I discovered that if I go into Audio Midi Setup application and lower the bit rate to 24/48 then iTunes will resume playing again and I can hear the music fine with iTunes or Quicktime player.
Basically, my Macbook Pro has now lost the ability to play 24/96 bit rate with either of these new USB DACs, when they both worked fine before I plugged them in at the same time. More confusing is that they use the same USB receiver chip as in one of the DACs that I mentioned above in the first paragraph, and both the HDP and DACport still work fine at 24/96 bit-rate. It's only these two new USB DAC that require me to lower the bit rate in Audio Midi Setup to 24/48 in order to listen through them, and this wasn't a problem as recently as yesterday!
The computer is acting like some setting needs to be wiped out - it's DEFINITELY NOT the USB DAC's that are faulty because they still both work fine at 24/96 on my wife's iMac (I'm afraid to try plugging in both at the same time to the iMac in case I screw that Mac up too).
I have reset the PRAM a few times, and this doesn't fix the problem. I thought maybe the com.apple.audiomidisetup.plist file could be corrupted and deleted it, but that didn't help. I don't know what to try next to get the Macbook to start fresh when trying to use these devices, short of having to re-install OSX.
Please, does anyone have a suggestion of how I can make the Macbook forget it ever saw these two USB DACs, and let me start fresh so they start working normally again? I do dozens of reviews for Head-fi.org and I can't afford to have my Macbook Pro compromised like this, which could prevent me from doing some reviews or evaluating new products.

UPDATE:
As noted in the first post, both of these newer USB DAC work fine at 24/96 on my wife's iMac, so it's clear that they didn't both become defective when I connected them to the Macbook Pro at the same time.
Resetting the Macbook Pro PRAM didn't help, so I removed the hard shell laptop case to get to the battery and remove it, then reset the SMC. After that I could use the original DAC at 24/96 again, but only IF it was plugged into the rear USB port. If I plugged it into the front USB port then the sound was garbled and filled with noise. This was a vast improvement over no sound at all.
Then I switched to the replacement/updated DAC, but I got no sound at all from either USB port at any bit-rate. And returning to the original DAC again now results in no sound again.
Sigh... I will reset the PRAM and SMC again, and see how far I can get with this. I'd still love some help. Thanks!

Possibility to use new Mac mini with an USB-DAC?

Anybody knowing anything about the possibility to use the new Mac mini with an USB-DAC. My reason for the question is that I've heard that USB3 doesn't handle sound. I'm thinking about to buy a mac mini and use it as an HTPC and signalsource together with NAIM DAC-V1.
// Rode orm

What I have heard that there are a few USB2 audio devices that
have had some issues with USB3, but USB3 itself is supposedly
backward compatible to USB2.
I would check with the makers of the DAC to see if there are any
USB3 issues.

Multiple Audio out: usb dac audio out and hdmi out

Good afternoon everyone, I have a 2010 mac mini with the following audio set up.
- A usb dac (xtreamer) connected to my stereo amplifier
- an Hdmi cable to the tv to get video and audio diretcly to the tv
Is it possibile by any chance to get simoultanesly audio output both from usb audio out and hdmi audio out???
In "midi preference panel" I tried to create a "virtual" sound card but i am not able to get sound simoultanesly. or via hdmi, or via usb.
this is quite disappointing because if I want to listen to music with the amplifier and then watching a simple video with mac mini and tv only I need to go back to the audio preferences and set up manually each time.
Is it possible to solve this issue???
Thanks a lot and gretings from Italy

I can select ATV audio out (Airplay setting) to either my Airport Express or my computer so it must be transmitting via airplay. Using a combination of Airserver, Airfoil and Airfoil Speakers I managed to get the ATV audio content on both my PC and my Airport Express at the same time but it cuts out the ATV audio.
My aim is to have a movie on ATV transmitting the Audio to ATV AND my stereo (which is hooked up to the Airport Express).
Cheers

Pulseaudio and USB Dac

How do I get pulseaudio to see my USB dac? It keeps playing sound through the sound card only.
I've added "load-module module-udev-detect" to default.pa, and changed my ~/.asoundrc to include:
pcm.pulse {
type pulse
ctl.pulse {
type pulse
pcm.!default {
type pulse
ctl.!default {
type pulse
aplay -L shows this, but I don't know what to add as the alsa-sinks:
default:CARD=CS46xx
Sound Fusion CS46xx, CS46xx
Default Audio Device
front:CARD=CS46xx,DEV=0
Sound Fusion CS46xx, CS46xx
Front speakers
rear:CARD=CS46xx,DEV=0
Sound Fusion CS46xx, CS46xx - Rear
Rear speakers
center_lfe:CARD=CS46xx,DEV=0
Sound Fusion CS46xx, CS46xx - Center LFE
Center and Subwoofer speakers
surround40:CARD=CS46xx,DEV=0
Sound Fusion CS46xx, CS46xx
4.0 Surround output to Front and Rear speakers
surround41:CARD=CS46xx,DEV=0
Sound Fusion CS46xx, CS46xx
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=CS46xx,DEV=0
Sound Fusion CS46xx, CS46xx
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=CS46xx,DEV=0
Sound Fusion CS46xx, CS46xx
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
iec958:CARD=CS46xx,DEV=0
Sound Fusion CS46xx, CS46xx - IEC958
IEC958 (S/PDIF) Digital Audio Output
null
Discard all samples (playback) or generate zero samples (capture)
default:CARD=default
USB Audio DAC , USB Audio
Default Audio Device
front:CARD=default,DEV=0
USB Audio DAC , USB Audio
Front speakers
surround40:CARD=default,DEV=0
USB Audio DAC , USB Audio
4.0 Surround output to Front and Rear speakers
surround41:CARD=default,DEV=0
USB Audio DAC , USB Audio
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=default,DEV=0
USB Audio DAC , USB Audio
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=default,DEV=0
USB Audio DAC , USB Audio
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=default,DEV=0
USB Audio DAC , USB Audio
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=default,DEV=0
USB Audio DAC , USB Audio
IEC958 (S/PDIF) Digital Audio Output

denisfalqueto wrote:
I think you can do it in pavucontrol. Theres the "Output devices" tab, which should be showing your devices. You can select the default one by pressing the green button with a check mark. This is what I do with my PCMCIA audio board.
Hope that helps.
I was trying to do it manually. pavucontrol doesn't work with the ubuntu font packages. :-(
10:48 PM:~ $ pavucontrol
pavucontrol: error while loading shared libraries: libglademm-2.4.so.1: cannot open shared object file: No such file or directory
# pacman -S libglademm
resolving dependencies...
looking for inter-conflicts...
:: cairo conflicts with cairo-ubuntu. Remove cairo-ubuntu? [Y/n] n
error: unresolvable package conflicts detected
error: failed to prepare transaction (conflicting dependencies)
:: cairo: conflicts with cairo-ubuntu

Delay on audio playback with FiiO e10 USB DAC

Hello,
My USB DAC playback is not working correctly. When I start the playback of any music stream, it takes a while (around a second or less) before I actually hear anything. The problem also occurs if short silent periods are present in the audio file/stream. The only way I can listen these problematic files/streams without the skipping is by keeping the master channel active all the time by playing some other audio along with problematic stream.
I have had this problem on openSUSE 12.3 too, and I was not able to solve it there either.
On Windows 7, however, this very same problem occurs only if I choose to use any of the 24 bit audio depth settings for the DAC.
Here is a screenshot to demonstrate the Windows scenario:
http://s10.postimg.org/xzwx302jd/fiio_delay.png
I have been trying out various things things without success, such as:
From Arch's PulseAudio wiki:
- Glitches, skips or crackling
- Setting the default fragment number and buffer size in PulseAudio
- Laggy sound
- Realtime scheduling
Here is some general information about my current Arch system and sound settings:
OS: Arch Linux x86_64
Kernel Release: 3.14.1-1-ARCH
WM: KWin
DE: KDE
Processor Type: Intel(R) Core(TM) i5-3570K CPU @ 3.40GHz
Sound settings: ALSA + PulseAudio
This problem did exist even before I installed Pulseaudio.
aplay -l :
**** List of PLAYBACK Hardware Devices ****
card 0: PCH [HDA Intel PCH], device 0: ALC898 Analog [ALC898 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 1: ALC898 Digital [ALC898 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 3: Audio [DigiHug USB Audio], device 0: USB Audio [USB Audio]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 3: Audio [DigiHug USB Audio], device 1: USB Audio [USB Audio #1]
Subdevices: 1/1
Subdevice #0: subdevice #0
pacmd list-cards (the last entry only)
index: 3
name: <alsa_card.usb-FiiO_DigiHug_USB_Audio-01-Audio>
driver: <module-alsa-card.c>
owner module: 9
properties:
alsa.card = "3"
alsa.card_name = "DigiHug USB Audio"
alsa.long_card_name = "FiiO DigiHug USB Audio at usb-0000:07:00.0-2, full speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:07:00.0-usb-0:2:1.1"
sysfs.path = "/devices/pci0000:00/0000:00:1c.7/0000:07:00.0/usb5/5-2/5-2:1.1/sound/card3"
udev.id = "usb-FiiO_DigiHug_USB_Audio-01-Audio"
device.bus = "usb"
device.vendor.id = "1852"
device.vendor.name = "GYROCOM C&C Co., LTD"
device.product.id = "7022"
device.product.name = "DigiHug USB Audio"
device.serial = "FiiO_DigiHug_USB_Audio"
device.string = "3"
device.description = "DigiHug USB Audio"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-usb"
profiles:
output:analog-stereo: Analog Stereo Output (priority 6000, available: unknown)
output:iec958-stereo: Digital Stereo (IEC958) Output (priority 5500, available: unknown)
off: Off (priority 0, available: unknown)
active profile: <output:iec958-stereo>
sinks:
alsa_output.usb-FiiO_DigiHug_USB_Audio-01-Audio.iec958-stereo/#0: DigiHug USB Audio Digital Stereo (IEC958)
sources:
alsa_output.usb-FiiO_DigiHug_USB_Audio-01-Audio.iec958-stereo.monitor/#1: Monitor of DigiHug USB Audio Digital Stereo (IEC958)
ports:
analog-output: Analog Output (priority 9900, latency offset 0 usec, available: unknown)
properties:
iec958-stereo-output: Digital Output (S/PDIF) (priority 0, latency offset 0 usec, available: unknown)
properties:
.asoundrc :
pcm.!default {
type hw
card 3
ctl.!default {
type hw
card 3
-- mod edit: read the Forum Etiquette and only post thumbnails http://wiki.archlinux.org/index.php/For … s_and_Code [jwr] --
Last edited by voodur (2014-04-23 22:24:35)

emeres wrote:Does this behaviour continue with pulseaudio killed? Use alsa directly to narrow the problem down first.
Yes. It did exist even before I installed pulseaudio.
emeres wrote:Check different resolutions, sample rates and channels. Try to determine the lowest buffer and period size, that the DAC accepts.
If by this you mean this, I went through that guide and it did not make any difference.
I found out that sometimes some audio formats, such as wav or mp3 do not produce this flaw. I gathered stats about this here. VLC was used for playback and displaying the information. Amarok behaves similarly.
emeres wrote: You probably load snd-usb-audio module with no options. Try loading it with "nrpacks" using different settings (default is 8, try 1, 2).
I am not sure if this is the way to do it, but I followed this guide and added file: "usbsound.conf" to "/etc/modprobe.d/". usbsound.conf has line: "options snd_usb_audio nrpacks=2". I tried changing the nrpacks to various values and booted after every change, but it did not make difference to the audio skipping.
What could it be that .xm (and every other tracker music format) and .ogg files are not playing like mp3 or wav?
Sorry for delayed reply and about that image
Last edited by voodur (2014-05-06 12:37:57)

Airport Extreme: Using USB to send music to a USB-DAC

Has anyone tried this? Can I send music data through the USB Port on the extreme to an external USB DAC (digital to analog converter)?
Would I still be able to use a USB hub and attach an external hardrive as well as a DAC?
Any help would be appreciated.
Thanks.

No there is no way to output music via the USB port on the AirPort Extreme base station (AEBS).

My new mac mini not recognizing usb dac

mac mini stopped recognizing external usb dac. i've reloaded osx 10.7.3 and run 10.7.3 combo. nothing, it appears audio mini setup is not working. the dac lists in devices in finder, but nothing in audio devices. have downloaded driver numerous times....nothnig.
can anyone help?
thanks

sorry...been away for a hile. it appearsbthatbit fixedbitself......until the power went out earlier today and its now doing the same thing. itnwill not connect to the dc. it is a eastern electric dac plus.

Connect to an USB DAC

Hello,
I'd know if it is possible to connect an USB DAC to an Apple TV ? I want to use Airplay…
Thank you…

Usb on the atv can only be used for firmware uodate nothing else

IOS7 issue with USB DACs?

Ive been running my iPad3 to an Audiolabs MDAC USB DAC via the CCK on ios5 and then 6 for awhile and its been working fine.
When using FlacPlayer or iPeng apps or itunes Home Sharing and hi-res FLACs/ALACs I would get 24/96 playback with no issues.
Since iOS7, with those same apps, while the dac detects a 24/88 or 24/96 signal, and it plays it, there is a white noise/stutter every other second in the tracks, though all 16bit and even 24/44 and 24/48 material is fine.
Is anyone else experiencing this since update to ios7.
Disappointed as this is something that used to work.

If I may act as a "Devils Advocate"... taking what you report as an OS issue, if that were so, it would suggest that any books created since iOS7 was introduced and had a movie a intro...would be reporting the issue here weeks ago.
I have just opened and checked three of my books with video intro..and none have any problems. I used iPad 2,3 and mini to check, and all show and allow manual control of the video.
Assuming your video is in the acceptable format..
Have you tried with default book from the template chooser, to recreate this apparent problem?
When you dropped in your video, did the application pop up the reprocessing message?
My (albeit often flawed) assumption is that your problems may arise from the video itself in some way.
Another issue which seems to affect some and not others is... using Preview to test a book. Advice suggests exporting the book and adding to iTunes > add to library and sync.
I prefer to export and use Dropbox, which has an option to open in iBooks in the iPad.
Although it may not be relevant, all my books have portrait mode disabled.

IPad's camera connector kit to use a USB DAC on iPod?

I understand that the iPad's camera connector kit can be plugged into an iPad to use a USB audio interfaces such as an external DAC. Will this be possible with an iPod 5.5G?

According to previoss reports, it does not work with an older iPod. However, there may be other third party products that offer this functionality for your older iPod.
B-rock

USB DAC on Z3 compact

Hi all
I have used my Audioquest Dragonfly 1.2 DAC without any problems on my Samsung S4. With my new Z3 compact, the Dragonfly connects and plays 24bit/96kHz files, but the Z3 cannot control the volume of the Dragonfly. The volume is always at maximum.
Is the Z3 not compatible with the Dragonfly? Has anyone got this combination to work?
Thanks a lot
Solved!
Go to Solution.

What you should use is a Magnetic Adaptor like the ones from Magnector on Amazon... I bought 2 of the newest, and they sent me 4 units (2 new, 2 old) and they are great for charging the phone without opening the flaps...
Also I charge the phone when plugged to a MHL cable and it charges faster than said connection. And it works when you are using the USB port as an OTG device. The magnetic port is one of the best features of our Xperias, its one thing that Samsung do not offer on their waterproof devices.
As an advice I would say that for Unplugging the magnetic adaptor, you should do it by separating them at an angle (as if you are breaking a cracker) because the magnet is pretty strong so I fear to have the connector pull off of my device.... its been previously posted here.

[SOLVED] How to achieve the best sound quality with USB DAC?

I want to take full advantage of my hardware. However, I'm having trouble understanding what settings I need to configure. I've spent several hours reading through wikis and searching online. Ultimately, I'm trying to achieve the highest quality audio output in stereo that's possible with my setup. USB > Behringer UAC202 > RCA out > Audioengine A2
I've had this DAC working for a couple of weeks, but I noticed that MPD was still eating a lot of CPU cycles upsampling music from 44.1 to 48khz. As I looked into it, it seems that this is unavoidable with ALSA, so I need to use OSS. Is that true? There's some conflicting ideas about when it's necessary to use OSS. I'm not using any legacy applications or old hardware. The "Hardware mixing" section in the ALSA wiki is pretty sparse.
I want all sound to go through the DAC, so I disabled all of the sound options in the BIOS (HDAudio and HDMI audio). I shouldn't need those enabled, right? (What's strange is that ALSA no longer wants to load without them.) I've added the OSS daemon and it starts, but I still don't have any sound in KDE. It no longer offers the "USB Audio CODEC" option. I'm ok with no system sound for now, if I can get MPD working.
/etc/mpd.conf contains:
audio_output {
type "oss"
name "UCA202"
device "/dev/oss/usb08bb2902-1/pcm0"
format "44100:16:2" # optional
mixer_device "/dev/mixer"
MPD is still starting as the root user. (I know, I should change that)
The daemon starts, but trying to play any track results in the following error in /var/log/mpd/mpd.log:
May 21 10:46 : output: "UCA202" [oss] failed to play: Write error on /dev/oss/usb08bb2902-1/pcm0: Input/output error
So... any help would be appreciated. Maybe, first of all, do I need to use OSS or should I switch back to alsa? Or maybe something else like jack?
*******EDIT*******
ALSA was the correct choice. The simple solution was to create /etc/asound.conf containing:
pcm.!default{
type rate
slave{
pcm "plughw:0,0"
rate 44100
Then change the audio output settings in mpd.conf to:
audio_output {
type "alsa"
name "MPD ALSA"
device "hw:0,0"
mixer_control "PCM"
With these settings the DAC receives 44.1khz audio and no upmixing occurs. Only one application can make audio output at a time. MPD uses much less CPU.
Last edited by ioos (2012-05-24 23:34:51)

Pajaro wrote:
For good sound quality, flat response is the most important thing. Take a look at qloud, http://aur.archlinux.org/packages.php?ID=34412, and find yourself a very good microphone.
Then you will probably want to set a global equalizer for your machine. You can try with alsa, http://krustev.net/w/articles/Global_eq … _for_ALSA/.
If ALSA isn't finegrained enough, jack has a lot more flexibility, except for the fact that it doesn't allow you to set default output device, so you will need a helper application to redirect all output connections to your equalizer.
Well, i would say this is quite subjective. Untouched signal is superior to any equalized sound in my opinion, but i know this debate is never satisfying

USB DAC Support

I recently got a new DAC with Mic/Line inputs to also record from analog sources. It's a Tascam UH-7000. My media-pc is mainly running Windows 8.1 but I can dual-boot to Arch.
Under Arch, the device is recognized as a USB device, its called TEAC (same company) with lsusb but with aplay -l or aplay -L there are no devices listed.
Does this simply mean the device is not supported on Linux or is there anything I can try to get it to work, at least as a DAC but input support would be grand too of course.
I would list more details but I'm unsure what else you'd need.
Thanks!

I have no idea if this can go anywhere, but I wrote an email to the driver developers asking whether they had plans to support linux. Here's my mail and their answer:
I own a Tascam UH-7000 USB device but unfortunately it doesn’t work under Linux. Seeing the driver and/or firmware carries your name (Ploytec) I was wondering if you could help me out.
Since the device also works on Mac OSX is there a chance this device can be supported on Linux too?
If the source code is available (or at least documented API or something) maybe the Linux community can pick up the development. There are so many nice tools on Linux for recording, editing audio and there are even distributions specifically for audio professionals. Not having Linux support is really a shame.
Thanks in advance!
A reply came from one of their managing directors:
There’s nothing special about audio on the UH7000, simple isocronous streaming.
The intense part would be doing the control panel, if there’s somebody having
a couple of months free time for that, I’m sure we can arrange the databooks.
Now, I'm not a developer myself and I know most open source developers are volunteers doing their thing in their free time, for free . Still, the reply gives me a glimpse of hope that maybe one day we can have Linux support for this device.
How do things like this go? I have a company possibly willing to work with a developer to support the device in Linux. Where of how would I find a developer who can work with them?
Last edited by mouseman (2015-05-05 15:40:53)

LP9 bug w/ USB DAC

We have a session w/ audio recorded in at 88.2 and all LP9 audio settings set appropriately.
In the Audio MIDI Setup, we have an M-Audio 610 (firewire) front end and a Benchmark DAC1 Pre (USB) back end.
In playback, the M-Audio is without issue at 88.2, and with LP9 closed, so is the Benchmark.
However, as soon as LP9 launches before selecting a session, it consistently flips the Benchmark into 96 in the A/M Setup.
Once the 88.2 session is up, if in the A/M Setup the Benchmark is switched back to 88.2, within a second it gets flipped back to 96. With LP closed, no issue.
If we "play" like this, we get LP errors saying incompatible output device setting, so LP can read the Benchmark's setting.
One guess is that LP is seeing the USB OUT device as if it's the OS 10.5.8 "Built-in Output", which has only 44.1 / 48 / 96 available as selections.

Servi wrote:
We have a session w/ audio recorded in at 88.2 and all LP9 audio settings set appropriately.
Meaning you have the sample rate in Logic set at 88.2.
What is set as Logic's audio device?
In the Audio MIDI Setup, we have an M-Audio 610 (firewire) front end and a Benchmark DAC1 Pre (USB) back end.
Unless you're used Audio/MIDI setup to create an "Aggregate Device" it's irrelevant to Logic.
Are you using an aggregate device setup?
In playback, the M-Audio is without issue at 88.2, and with LP9 closed, so is the Benchmark.
However, as soon as LP9 launches before selecting a session, it consistently flips the Benchmark into 96 in the A/M Setup.
Once the 88.2 session is up, if in the A/M Setup the Benchmark is switched back to 88.2, within a second it gets flipped back to 96. With LP closed, no issue.
If we "play" like this, we get LP errors saying incompatible output device setting, so LP can read the Benchmark's setting.
One guess is that LP is seeing the USB OUT device as if it's the OS 10.5.8 "Built-in Output", which has only 44.1 / 48 / 96 available as selections.
This is confusing, generally speaking you don't want your pro audio gear to be selected as anything in A/M setup, that's basically for the Mac's audio setup, Logic's audio system interfaces with core audio but operates independently.
1. What is the audio device selected in Logic.
2. Is Logic's sample rate set to 88.2?
3. Have you created an Aggregate device for your two audio components?
4. do you have the two devices m-audio and Benchmark clock locked to each other?
5. Am I totally misunderstanding you...?
pancenter-

Usb dac

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