@mhartzel, I tried it on both USB 3.0 and USB 2.0 ports, including a dedicated USB 2.0 controller card (the Syba). I may have disabled USB 3.0 in the bios at some point during troubleshooting, but I’m but I’m not entirely sure. I think I’m done testing this board unless someone else tells me it worked for them.
Most class compliant interfaces do work, but there can be strange effects. E.g., the Allen & Heath Qu16 digital console does work, but Open Broadcast Studio behaves funny with it, sometimes it just does not include the audio in the video stream. However, it seems to work flawlessly in Ardour. Some Tascam 4ch interface used to work perfectly, so does the old Alesis IO2, I will check out the Lexicon Alpha Studio soon.
ADAT clock sync works well for my old RME Digi (PCI), however kmix mixes it up – since it is a “mixer setting”, it sometimes gets lost when the mixer restores its settings after login. (Side note: The RME Digi only works up to Win7 32bit. So Linux and Ardour save me a lot of money here… the Digi is still good unless you really need 96k, which would limit it from 16 to 8 ch, or 24 to 12 resp.)
In general, I think: Ardour should recommend a few interfaces that work rock solid and for the others direct to some information source that knows about the details.
Things change rapidly sometimes, which might be due to USB Quirks added with distro/kernel updates.
1 Like
May I suggest Motu Ultralite AVB ?
USB Class compliant and also AVB streaming.
One of the good thing is the built in web server engine for controlling hardware routing, built in 48ch mixer with eq, compression and busses…
I have two and like them alot. Used to run Presonus Firepods, Focusrite Pro40, Pro24dsp and stuff.
@ahellquist Have you had the Ultralite AVB work on USB 3 ports? Fernando reported some issues with USB 3 here: http://linux-audio.4202.n7.nabble.com/Motu-1248-Full-success-td103577i20.html
The Ultralite looks excellent if it is stable…
@CTS: Ultralite AVB is stable and never had any issues related to the interface when on Linux + Ardour. Used it at a wedding I came out in with a digital dj mixer from my tablet and a couple mics. No issues at all for close to 4 hrs straight. The only issues I had were network related but that resolved rather quickly.
I had recently the opportunity to test a bunch of simple USB - audio devices with Linux. We needed at work a USB audio device with at least 1 Mic input, headphone output, zero latency audio routing from Mic to headphone output, driverless operation in windows 10 (using microsofts USB 2 Audio Class driver usbaudio2.sys that was released with windows 10 creators update (build 1709)). Unfortunately the windows USB 2 Class driver turned out to be a disappointment, none of the about ten USB Audio Class Compliant devices we tested worked with it. The reason for this is that microsoft says it has not yet implemented all Class Specification features in their USB 2 driver yet. The only devices that worked with a microsoft provided USB Audio Class driver were devices that were compatible with USB 1 Audio Class and it’s driver (usbaudio.sys) that was released in vista. Fortunately Linux USB Audio Class drivers seems to have been feature complete for many years now and these devices do work much better in Linux than in win10 with the microsoft class driver.
I tested all these Class Compliant audio devices with Manjaro Linux which has a recent kernel. I will post my findings here, one post per device. Sorry for the spam 
One more thing: I have previously said on this forum that some usb audio devices do not work on USB 3 ports and that a realtime kernel crashes a lot. Both of these seems to be a thing of the past now and recent kernels fix these. Manjaro’s realtime kernel works flawlessly (used it for days for audio recording / mixing and playing games in Steam) if you just remove VirtualBox’s “linux-rt-lts-manjaro-virtualbox-guest-modules”. Having this package installed made my computer fail to boot with the realtime kernel. I did the tests with my Asus ROG G75VW that had problems with USB 3 connected USB audio devices and hard disks about one year ago, now it worked flawlessly. I did change between USB 3 and USB 2 on this laptop by disabling / enabling xHCI controller in the EFI.
Test recording were quite short, about 1 minute in USB 2 mode and 5 minutes in USB 3 mode. Many of these devices are compatible with iPad’s also because they are USB Class Compliant.
Native Instruments Komplete Audio 6
6 x in, 6 out interface. Input volume adjustment is possible only on inputs 1 + 2.
Works in Linux with both a USB 2 and a USB 3 port (tested with Kernel 4.9.68-1-rt60-MANJARO) and Jack 0.125.0-3 (Manjaro package) with Sample Rate: 48000, Frames / Period: 1024, Periods / Buffer: 3.
The monitor knob is the “Zero Latency Monitoring feature” and affects what one hears from the headphones. It lets you mix Mic inputs 1 + 2 or line inputs 3 + 4 to the audio coming from the computer. Selection is done with a long press of the “On” knob in the monitor section. It cycles between 3 states: 1 + 2 / 3 + 4/ 1 + 2 + 3 + 4. There is a knob on the headphone section that lets you choose between outputs 1 + 2 or 3 + 4.
SPDIF In tested and it works. The device can be synced to incoming SPDIF signal with alsamixer. Go on top of the Audio Output that has text “Internal Clock” on top of it and change the state with cursor up / down keys on the keyboard.
I did not test midi on this device.
I had been thinking about buying this interface for my home audio work, but got a 18 input 20 output Behringer instead. Now that I got my hands on this Native Instruments device I’m beginning to wonder if I should buy it anyway. It’s a solid well built device with good features.
This device is USB 2 Audio Class Compliant and does not work in win10 without a proprietary driver.
cat /proc/asound/K6/stream0
Native Instruments Komplete Audio 6 at usb-0000:00:14.0-3, high speed : USB Audio
Playback:
Status: Stop
Interface 1
Altset 1
Format: S32_LE
Channels: 6
Endpoint: 1 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000
Data packet interval: 125 us
Capture:
Status: Stop
Interface 2
Altset 1
Format: S32_LE
Channels: 6
Endpoint: 2 IN (ASYNC)
Rates: 44100, 48000, 88200, 96000
Data packet interval: 125 us
cat /proc/asound/K6/usbmixer
USB Mixer: usb_id=0x17cc1001, ctrlif=0, ctlerr=0
Card: Native Instruments Komplete Audio 6 at usb-0000:00:14.0-3, high speed
Unit: 12
Control: name=“Monitor Control Playback Switch”, index=1
Info: id=12, control=1, cmask=0x0, channels=1, type=“INV_BOOLEAN”
Volume: min=0, max=1, dBmin=0, dBmax=0
Unit: 12
Control: name=“Monitor Control Playback Switch”, index=0
Info: id=12, control=1, cmask=0xf, channels=4, type=“INV_BOOLEAN”
Volume: min=0, max=1, dBmin=0, dBmax=0
Unit: 13
Control: name=“Audio Input Terminal Volume”, index=2
Info: id=13, control=12, cmask=0x9, channels=2, type=“S16”
Volume: min=0, max=1, dBmin=0, dBmax=0
Unit: 13
Control: name=“Audio Input Terminal Volume”, index=1
Info: id=13, control=11, cmask=0x3, channels=2, type=“S16”
Volume: min=0, max=1, dBmin=0, dBmax=0
Unit: 13
Control: name=“Audio Input Terminal Volume”, index=0
Info: id=13, control=8, cmask=0x20, channels=1, type=“S16”
Volume: min=0, max=1, dBmin=0, dBmax=0
Unit: 40
Control: name=“Internal Clock Validity”, index=0
Info: id=40, control=2, cmask=0x0, channels=1, type=“BOOLEAN”
Volume: min=0, max=1, dBmin=0, dBmax=0
Unit: 41
Control: name=“External Clock (S/PDIF) Validity”, index=0
Info: id=41, control=2, cmask=0x0, channels=1, type=“BOOLEAN”
Volume: min=0, max=1, dBmin=0, dBmax=0
Unit: 42
Control: name=“Audio Output Terminal Clock Source”, index=0
Info: id=42, control=1, cmask=0x0, channels=1, type=“U8”
Volume: min=1, max=2, dBmin=0, dBmax=0
cat /proc/asound/K6/midi0
Komplete Audio 6
Output 0
Tx bytes : 0
Owner PID : 7281
Mode : native
Buffer size : 4096
Avail : 4096
Input 0
Rx bytes : 0
Owner PID : 7281
Buffer size : 4096
Avail : 0
Overruns : 0
Soundcraft Notepad-5
Works in Linux with both a USB 2 and a USB 3 port (tested with Kernel 4.9.65-1-MANJARO) and Jack 0.125.0-3 (Manjaro package) with Sample Rate: 48000, Frames / Period: 1024, Periods / Buffer: 3.
The device has 1 mic channel and phantom power is always on, it can not be turned off. Not USB - powered, needs an external power supply. Hardware zero latency monitoring works. Adjustable High pass and Low pass filter
The Master stereo output of the mixer is routed to USB channels 1 + 2 and there are two return channels that the mixer routes to the headphones and the Master output.
I did not have the opportunity to connect anything to the Master XLR - outs, but I saw that master volume led display was showing a signal for both the mixer inputs and USB playback.
This device is USB 2 Audio Class Compliant and does not work in win10 without a proprietary driver.
cat /proc/asound/Notepad5/stream0
Soundcraft Notepad-5 at usb-0000:00:1d.0-1.2, high speed : USB Audio
Playback:
Status: Running
Interface = 1
Altset = 1
Packet Size = 56
Momentary freq = 47999 Hz (0x5.fff8)
Feedback Format = 16.16
Interface 1
Altset 1
Format: S32_LE
Channels: 2
Endpoint: 1 OUT (ASYNC)
Rates: 44100, 48000
Data packet interval: 125 us
Interface 1
Altset 2
Format: S16_LE
Channels: 2
Endpoint: 1 OUT (ASYNC)
Rates: 44100, 48000
Data packet interval: 125 us
Capture:
Status: Running
Interface = 2
Altset = 1
Packet Size = 56
Momentary freq = 48000 Hz (0x6.0000)
Interface 2
Altset 1
Format: S32_LE
Channels: 2
Endpoint: 2 IN (ASYNC)
Rates: 44100, 48000
Data packet interval: 125 us
Glensound Mini Cub
Works in Linux with both a USB 2 and a USB 3 port (tested with Kernel 4.9.65-1-MANJARO) and Jack 0.125.0-3 (Manjaro package) with Sample Rate: 48000, Frames / Period: 1024, Periods / Buffer: 3. Did not work at first, had constant xruns, but after connecting the device to other USB 2 ports and back, it suddenly started to work in any USB 2 port.
1 physical channel in and out. Recording appears on 2 USB channels and playback also on 2 USB channels. USB - powered. Hardware zero latency monitoring works. Blindlingly bright blue led on the front panel.
This device is USB 1 Audio Class Compliant and does work in win10 without a proprietary driver (using microsofts usbaudio.sys).
cat /proc/asound/MIC/stream0
STMicroelectronics STM32 Audio Class SPKR & MIC at usb-0000:00:1d.0-1.1, full s : USB Audio
Playback:
Status: Running
Interface = 1
Altset = 1
Packet Size = 294
Momentary freq = 48000 Hz (0x30.0000)
Feedback Format = 10.14
Interface 1
Altset 1
Format: S24_3LE
Channels: 2
Endpoint: 2 OUT (ASYNC)
Rates: 48000
Capture:
Status: Running
Interface = 2
Altset = 1
Packet Size = 294
Momentary freq = 48000 Hz (0x30.0000)
Interface 2
Altset 1
Format: S24_3LE
Channels: 2
Endpoint: 1 IN (ASYNC)
Rates: 48000
Steinberg UR22 MK2
Works in Linux with both a USB 2 and a USB 3 port (tested with Kernel 4.9.68-1-rt60-MANJARO) and Jack 0.125.0-3 (Manjaro package) with Sample Rate: 48000, Frames / Period: 1024, Periods / Buffer: 3.
Zero Latency monitoring is implemented with a “Mix” knob that lets one adjust the mix between audio from Mic Ins and Computer output.
No level meter, only a peak led on Mic Inputs that lights up when overloading the input.
Feels like a well built device.
I did not test midi with this device.
This device is USB 2 Audio Class Compliant and does not work in win10 without a proprietary driver.
cat /proc/asound/UR22mkII/stream0
Yamaha Corporation Steinberg UR22mkII at usb-0000:00:14.0-3.1, high speed : USB Audio
Playback:
Status: Running
Interface = 1
Altset = 1
Packet Size = 64
Momentary freq = 47997 Hz (0x5.ffe8)
Feedback Format = 16.16
Interface 1
Altset 1
Format: S32_LE
Channels: 2
Endpoint: 3 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Capture:
Status: Running
Interface = 2
Altset = 1
Packet Size = 64
Momentary freq = 48000 Hz (0x6.0000)
Interface 2
Altset 1
Format: S32_LE
Channels: 2
Endpoint: 1 IN (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
cat /proc/asound/UR22mkII/usbmixer
USB Mixer: usb_id=0x0499170f, ctrlif=0, ctlerr=0
Card: Yamaha Corporation Steinberg UR22mkII at usb-0000:00:14.0-3.1, high speed
Unit: 41
Control: name=“Clock Source 41 Validity”, index=0
Info: id=41, control=2, cmask=0x0, channels=1, type=“BOOLEAN”
Volume: min=0, max=1, dBmin=0, dBmax=0
cat /proc/asound/UR22mkII/midi0
Steinberg UR22mkII
Output 0
Tx bytes : 0
Owner PID : 5937
Mode : native
Buffer size : 4096
Avail : 4096
Input 0
Rx bytes : 0
Owner PID : 5937
Buffer size : 4096
Avail : 0
Overruns : 0
Yamaha AG03
Works in Linux with both a USB 2 and a USB 3 port (tested with Kernel 4.9.68-1-rt60-MANJARO) and Jack 0.125.0-3 (Manjaro package) with Sample Rate: 48000, Frames / Period: 1024, Periods / Buffer: 3.
This is a 2 channels to and from computer device and I expected it to be cheesy and feel cheap. It is made of plastic and metal and the knobs do feel a bit cheap but in my opinion they are alright for a device this cheap. I was amazed how many features can be squeezed in small device like this. If you never need any more than 2 channels to be recorded at a time then this device gives the most bang for the buck.
The device is a mixer and it lets one have many devices connected to it at once and it mixes these to a two channel mix that can be recorded on the computer. The Mic Input has lots and lots of gain, much more than any of the other 10 devices I tested. Volume from inputs 2 + 3 can be adjusted. Aux input is meant for a stereo device like a phone / tablet and its volume is not adjustable. There are two fixed non adjustable “effects”: comp + eq that felt quiet unusable and echo (called “Effect” on the mixer) that is an ok vocal reverb. The wet level is not adjustable and is a bit too wet for my taste.
There is a guitar input that if used replaces inputs 2 + 3. There is a three stage switch that lets you select what is sent to the computer:
- “Dry CH 1-2G” = Only Mic and Guitar channels are sent to the computer.
- “Input Mix” = what you can hear is sent to the computer.
- “Loopback” = same as “Input Mix” and also the sound coming from the computer is add to the mix.
There are connectors for a headset with 3.5 mm microphone and headphone connectors. I did not test this since I don’t have any device like this.
Zero Latency Monitoring is implemented like this:
- The Mic fader controls the amount of Mic input signal heard on the headphones and recorded.
- The Guitar / Keyboard knob controls the amount of input 2 + 3 signal heard on the headphones and recorded.
- The “Computer / USB” knob controls how much sound coming from the computer is heard on headphones and if the Loopback switch is on also how much of the signal is recorded.
This device is USB 2 Audio Class Compliant and does not work in win10 without a proprietary driver.
cat /proc/asound/AG06AG03/stream0
Yamaha Corporation AG06/AG03 at usb-0000:00:1d.0-1.3.1, high speed : USB Audio
Playback:
Status: Running
Interface = 1
Altset = 1
Packet Size = 64
Momentary freq = 48000 Hz (0x5.fffc)
Feedback Format = 16.16
Interface 1
Altset 1
Format: S32_LE
Channels: 2
Endpoint: 1 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Capture:
Status: Running
Interface = 2
Altset = 1
Packet Size = 64
Momentary freq = 48000 Hz (0x6.0000)
Interface 2
Altset 1
Format: S32_LE
Channels: 2
Endpoint: 2 IN (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
cat /proc/asound/AG06AG03/usbmixer
USB Mixer: usb_id=0x0499170d, ctrlif=0, ctlerr=0
Card: Yamaha Corporation AG06/AG03 at usb-0000:00:1d.0-1.3.1, high speed
Unit: 41
Control: name=“Clock Source 41 Validity”, index=0
Info: id=41, control=2, cmask=0x0, channels=1, type=“BOOLEAN”
Volume: min=0, max=1, dBmin=0, dBmax=0
Focusrite Scarlett 2i2 2nd Gen
Works in Linux with both a USB 2 and a USB 3 port (tested with Kernel 4.9.68-1-rt60-MANJARO) and Jack 0.125.0-3 (Manjaro package) with Sample Rate: 48000, Frames / Period: 1024, Periods / Buffer: 3.
The Zero Latency Monitoring is not adjustable and can only be turned on or off. If you want to have less Mic in the headphones the only way to do this is to use the Mic knob which also affect the recorded volume.
Level meter is implemented as a led lighted ring around the gain knob. The ring is green when there if signal present and red when overloading the input.
This device is USB 2 Audio Class Compliant and does not work in win10 without a proprietary driver.
cat /proc/asound/USB/stream0
Focusrite Scarlett 2i2 USB at usb-0000:00:14.0-3, high speed : USB Audio
Playback:
Status: Running
Interface = 1
Altset = 1
Packet Size = 64
Momentary freq = 48000 Hz (0x6.0000)
Interface 1
Altset 1
Format: S32_LE
Channels: 2
Endpoint: 1 OUT (SYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
Capture:
Status: Running
Interface = 2
Altset = 1
Packet Size = 64
Momentary freq = 48000 Hz (0x6.0000)
Interface 2
Altset 1
Format: S32_LE
Channels: 2
Endpoint: 2 IN (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000
Data packet interval: 125 us
cat /proc/asound/USB/usbmixer
USB Mixer: usb_id=0x12358202, ctrlif=0, ctlerr=0
Card: Focusrite Scarlett 2i2 USB at usb-0000:00:14.0-3, high speed
Unit: 41
Control: name=“USB Internal Validity”, index=0
Info: id=41, control=2, cmask=0x0, channels=1, type=“BOOLEAN”
Volume: min=0, max=1, dBmin=0, dBmax=0
Focusrite iTrack Solo
Works in Linux with both a USB 2 and a USB 3 port (tested with Kernel 4.9.68-1-rt60-MANJARO) and Jack 0.125.0-3 (Manjaro package) with Sample Rate: 48000, Frames / Period: 1024, Periods / Buffer: 3.
The Zero Latency Monitoring is not adjustable and can only be turned on or off. If you want to have less Mic in the headphones the only way to do this is to use the Mic knob which also affects the recorded volume.
Level meter is implemented as a led lighted ring around the gain knob. The ring is green when there if signal present and red when overloading the input.
This device is USB 2 Audio Class Compliant and does not work in win10 without a proprietary driver.
cat /proc/asound/Solo/stream0
Focusrite iTrack Solo at usb-0000:00:1d.0-1.3, high speed : USB Audio
Playback:
Status: Running
Interface = 1
Altset = 1
Packet Size = 64
Momentary freq = 48000 Hz (0x6.0000)
Feedback Format = 16.16
Interface 1
Altset 1
Format: S32_LE
Channels: 2
Endpoint: 1 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000
Data packet interval: 125 us
Capture:
Status: Running
Interface = 2
Altset = 1
Packet Size = 64
Momentary freq = 48000 Hz (0x6.0000)
Interface 2
Altset 1
Format: S32_LE
Channels: 2
Endpoint: 2 IN (ASYNC)
Rates: 44100, 48000, 88200, 96000
Data packet interval: 125 us
cat /proc/asound/Solo/usbmixer
USB Mixer: usb_id=0x1235800e, ctrlif=0, ctlerr=0
Card: Focusrite iTrack Solo at usb-0000:00:1d.0-1.3, high speed
Unit: 41
Control: name=“Internal Validity”, index=0
Info: id=41, control=2, cmask=0x0, channels=1, type=“BOOLEAN”
Volume: min=0, max=1, dBmin=0, dBmax=0
Audient iD4
Works in Linux with both a USB 2 and a USB 3 port (tested with Kernel 4.9.68-1-rt60-MANJARO) and Jack 0.125.0-3 (Manjaro package) with Sample Rate: 48000, Frames / Period: 1024, Periods / Buffer: 3.
There is one Mic input and one DI - input for a instrument. The signal from these is heard on both left and right on the headphone output unless the inputs are panned. This can be done by holding down MUTE and ID and turning the VOLUME knob. This affects only the headphone output, Mic is always channel 1 and DI channel 2 on the computer.
Zero Latency monitoring is implemented with a “Mix” knob that lets one adjust the mix between audio from Mic Ins and Computer output.
Feels like a well built device.
There are 5 leds that shows the monitoring volume while the VOLUME knob is turned and otherwise shows the output level. This means that when the Mix knob is between INPUT and DAW then the led meters shows the combined level of the signal coming in from the inputs and the computer. That is a bit strange design decision in my opinion.
This device is USB 2 Audio Class Compliant and does not work in win10 without a proprietary driver.
cat /proc/asound/iD4/stream0
Audient iD4 at usb-0000:00:14.0-3, high speed : USB Audio
Playback:
Status: Running
Interface = 1
Altset = 1
Packet Size = 64
Momentary freq = 47999 Hz (0x5.fff8)
Feedback Format = 16.16
Interface 1
Altset 1
Format: S32_LE
Channels: 2
Endpoint: 1 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000
Data packet interval: 125 us
Interface 1
Altset 2
Format: S16_LE
Channels: 2
Endpoint: 1 OUT (ASYNC)
Rates: 44100, 48000, 88200, 96000
Data packet interval: 125 us
Capture:
Status: Running
Interface = 2
Altset = 1
Packet Size = 64
Momentary freq = 48000 Hz (0x6.0000)
Interface 2
Altset 1
Format: S32_LE
Channels: 2
Endpoint: 2 IN (ASYNC)
Rates: 44100, 48000, 88200, 96000
Data packet interval: 125 us
cat /proc/asound/iD4/usbmixer
USB Mixer: usb_id=0x27080003, ctrlif=0, ctlerr=0
Card: Audient iD4 at usb-0000:00:14.0-3, high speed
Unit: 41
Control: name=“Audient Internal Clock Validity”, index=0
Info: id=41, control=2, cmask=0x0, channels=1, type=“BOOLEAN”
Volume: min=0, max=1, dBmin=0, dBmax=0
MOTU MicroBook IIc
MOTU says that this device is USB Class Compliant, but Zero Latency Monitoring is implemented using a DSP that requires proprietary drivers and software package “CueMix FX” (OS X 10.6.8 - 10.11, windows 7 - 10). That makes the device quite useless for our purpose. I did test recording 1 Mic with it (Kernel 4.9.68-1-rt60-MANJARO and Jack 0.125.0-3 (Manjaro package) with Sample Rate: 48000, Frames / Period: 1024, Periods / Buffer: 3). That resulted in more than 1000 xruns in 30 seconds, so the device does NOT work in Linux.
cat /proc/asound/IIc/stream0
MOTU MicroBook IIc at usb-0000:00:1d.0-1.3, high speed : USB Audio
Playback:
Status: Stop
Interface 1
Altset 1
Format: S24_3LE
Channels: 8
Endpoint: 3 OUT (SYNC)
Rates: 48000, 96000, 44100, 88200
Data packet interval: 125 us
Capture:
Status: Stop
Interface 2
Altset 1
Format: S24_3LE
Channels: 6
Endpoint: 4 IN (SYNC)
Rates: 48000, 96000, 44100, 88200
Data packet interval: 125 us
cat /proc/asound/IIc/usbmixer
USB Mixer: usb_id=0x07fd0004, ctrlif=0, ctlerr=0
Card: MOTU MicroBook IIc at usb-0000:00:1d.0-1.3, high speed
Unit: 1
Control: name=“USB Clock Validity”, index=0
Info: id=1, control=2, cmask=0x0, channels=1, type=“BOOLEAN”
Volume: min=0, max=1, dBmin=0, dBmax=0
Miditech Audiolink III
Works in Linux with both a USB 2 and a USB 3 port (tested with Kernel 4.9.68-1-rt60-MANJARO) and Jack 0.125.0-3 (Manjaro package) with Sample Rate: 48000, Frames / Period: 1024, Periods / Buffer: 3.
I don’t recommend this device for anybody because there is not enough of Mic and headphone amplifier gain for basic audio work. Also as soon as one starts Jack there is a silent but annoying 1 kHz tone on the headphones.
The Zero Latency Monitoring is not adjustable and can only be turned on or off. If you want to have less Mic in the headphones the only way to do this is to use the Mic knob which also affects the recorded volume.
This device is USB 1 Audio Class Compliant and does work in win10 without a proprietary driver (using microsofts usbaudio.sys).
The forum spamfilter didn’t let me post the full details of cat /proc/asound maybe because it printed a lot of text. So here is the short version:
The device supports playback sample rates 32 kHz, 44.1 kHz and 48 kHz and 8 and 16 bit sample sizes. Supported recording sample rates are: 8 kHz, 11.025 kHz, 16 kHz, 22.050 kHz, 32 kHz, 44.1 kHz and 48 kHz and 8 and 16 bit sample sizes.
cat /proc/asound/CODEC/usbmixer
USB Mixer: usb_id=0x08bb2900, ctrlif=0, ctlerr=0
Card: Burr-Brown from TI USB Audio CODEC at usb-0000:00:1d.0-1.3, full speed
Unit: 3
Control: name=“PCM Playback Volume”, index=0
Info: id=3, control=2, cmask=0x3, channels=2, type=“S16”
Volume: min=-32768, max=0, dBmin=-12800, dBmax=0
Unit: 3
Control: name=“PCM Playback Switch”, index=0
Info: id=3, control=1, cmask=0x0, channels=1, type=“INV_BOOLEAN”
Volume: min=0, max=1, dBmin=0, dBmax=0
I recently purchased a Behringer UMC404HD it interfaced fine with my JACK, Ardour set up in Mint 18.3 Cinnamon. I am just another person having problems with click, crackling and pops. To me it seems it only happens when I use Ardour. I have my quad core i5 CPU in performance mode but have read some folks may have solved the problem by changing buffer size and sampling rates. If you have experience in resolving the problem please let me know. Thanks!
I should add to this…
I while back I got hold of a Behringer XR18. This is an 18 channel mixer which also has 18 in and 18 out USB and it works beautifully in Linux. It even has a free downloadable Linux app to control the extremely capable, mixing and routing capabilities in the unit itself via the network.
It has 24-bit resolution at 44.1 or 48kHz.
As it’s, fundamentally, a standalone mixer, then this unit has some fantastic routing and mixing capabilities, and also some onboard FX. For each of the 18 inputs you can select whether it is an analogue input or a USB output from the computer. For analogue inputs, every channel has a gate (with sidechain), EQ, and compression, as well as an FX send to each of the 4 onboard FX slots (which can be loaded from a range of FX including guitar amps, vintage style compressors and reverbs, delays,chorus, etc.).
16 of the inputs are “combi” style connectors (Jack or XLR) and two are jack line level inputs. Combi inputs 1& 2 are Hi-Z so you can DI guitars and basses.
There’s 6 output busses in addition to the stereo master bus, and you can also route the output of any channel to the PC over USB. The mono busses and channels can be ganged together to form stereo pairs.
Controlling the unit is done using the free PC, Mac, or Linux (yes!!) app, and there’s also an iPad and Android app. Control is via the network, and the unit can either act as a station on an existing wireless or wired LAN, or can be switched to act as a standalone wifi network for stage use.
This gives a fantastic degree of flexibility. To give some examples of how I’ve used this.
I have this connected up with Jack, so I see an 18-input/output device (you can switch this to 2-channel mode if you want). I have a Mic on one channel (it has phantom power and phase options), and guitar and bass DI’d on channels 1 & 2. I also have a synth and a guitar FX processor plugged into other channels. I often do Google Hangouts, and sometimes I will mess around with my guitar during these. I have configured 2 stereo busses, one outputting to my desktop monitor speakers, and one acting as the mixing point for USB “send”. I have headphones plugged into the monitor jack on the unit which is on the master bus.
So this basically allows me to create separate mixes for:
- Headphones
- Monitor speakers
- PC
So if I’m in the middle of a Hangout, I can listen to my noodling on the guitar/bass and the Hangout audio on my headphones, whilst the hangout hears my Mic audio, but not the guitar/bass. I could also choose to let the hangout hear the guitar/bass if it’s a social hangout.
I can also push Pulseaudio sound out of the PC on a channel, and mix it into any of the busses. Again this allows me to selectively and separately mix the audio from (for instance) Youtube to the speakers, headphones, or back to the PC (e.g. to a Hangout). I’ve used this, for instance, to play a backing track on the PC and play along on the guitar, performing into the Hangout.
Conversely, I can also set the gate on the microphone to duck the mic based on the guitar or bass input, so if I am in a conference call, I can mess around on the guitar and they won’t hear the strings through the mic. Another use for this is so that, when performing to a Hangout, the mic doesn’t pick up any acoustic string noise or fret buzz.
Cheers,
Keith
P.S. I removed the links to the appropriate parts of the Behringer wiki as these seemed to trigger the spam filters.
hi i have a used lexicon alpha,berhinger uca 200,behringer uca 222. All soundcard work good in gnu linux debian stretch 9 with repository kxstudio. No problem with ardour 5.12,Mixbus 4,tracktion 6 and 7 Qtractor and various other programs dedicate audio production. I have a test maudio firewire solo with driver ffado. Are two days it’s work in debian stretch with repositories kx studio
gennaro