Getting Delta 1010 working in Ubuntu Studio 19.10

[ChuckLutz]

Hi folks,

I'm trying to get back into music recording with Linux; I'm not a total noob, but my first time around was with a very old P4 machine in which I just used the built-in soundcard input (sad, I know!) and mainly used Audacity on older Ubuntu Studio releases to record plain 2-track stereo jam sessions. I'm a techie guy (write mainly Python code these days for a living) and am familiar with Linux in general. However I've never mastered the audio stack (ALSA, Jack, etc.) beyond the basics. On that previous setup things must have been pretty "obvious" with the built-in sound card, since I was able to get it to work with a bit of brute-force fiddling with connections (probably in QJackCtrl if I recall correctly).

A while back I did a bit of research and got a used old Delta 1010 (not the LT version) off of eBay and picked up a newish machine (way better than the old P4 with 1G of ram anyhow!). I've recently installed Ubuntu Studio 19.10 on it. From what I can gather, the 1010 is pretty reliable and should work well under Linux, but I'm having a hard time getting going with it and to be honest I'm not quite sure where to start. I've screwed around in Ardour, envy24control, Qjackctrl, and some command line utilities, but I can't seem to get any sound coming into it.

Unfortunately most stuff on the web regarding this interface and Linux is really old, and it is hard to tell if it is still relevant (for example, there was some old stuff about using a '.asoundrc' file, but I've never used those, and couldn't tell if this is necessary to get the 1010 working). I'm hoping for a few pointers to get me in the right direction. I'm happy to put in the work (manuals, command line etc.) but I am well aware that Linux audio can be a "rabbit hole" so I'm hoping to try to minimize time wasted and I'm very grateful in advance for any advice.

I figured that it would be a good idea to try to do the most basic of tests - just to make sure that the interface is even operational (recall that I bought it used), and then move on to trying to use it in Ardour etc. I've noticed that the rack box (the actual I/O unit) has a light on the front, but it doesn't light up when I plug it in - hopefully that is a software-controlled thing and not a sign of a dead unit!

If I do a basic device listing, it seems that the system can see the 1010:

chuck@chuck-OptiPlex-7010:~$ aplay --list-devices
**** List of PLAYBACK Hardware Devices ****
card 0: PCH [HDA Intel PCH], device 0: ALC269VB Analog [ALC269VB Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: M1010 [M Audio Delta 1010], device 0: ICE1712 multi [ICE1712 multi]
Subdevices: 1/1
Subdevice #0: subdevice #0

I'm pretty old school and have a 24 channel hw mixer (an old Tascam) where all my instruments go; I have the Group 1 and 2 outputs of the mixer hooked into inputs 1 and 2 of the 1010. If I play one of the keyboards that is assigned to Groups 1 and 2 on the mixer, I can see the meter LEDs jump for those groups, so I should have signal coming into the 1010 on those two inputs.

With that, I attempted a basic audio capture command:

arecord -f S32_LE -c 2 -d 10 -r 44100 --device="hw:1,0" /home/chuck/test.wav
Recording WAVE '/home/chuck/test.wav' : Signed 32 bit Little Endian, Rate 44100 Hz, Stereo
arecord: set_params Channels count non available

I'm stuck at this point and don't know how to begin troubleshooting the 1010.

However, if I try it with the built-in soundcard, the command completes and I do get a .wav file (I actually have nothing hooked into the card right now, so it should be 10s of silence):

arecord -f S32_LE -c 2 -d 10 -r 44100 --device="hw:0,0" /home/chuck/test.wav
Recording WAVE '/home/chuck/test.wav' : Signed 32 bit Little Endian, Rate 44100 Hz, Stereo
chuck@chuck-OptiPlex-7010:~$

As I mentioned, I am far from an expert on the Linux audio "stack" - according to QJackCtrl, JACK is not started, but I was able to run the "arecord" above for the built-in sound card, so I assume that in such a case there is a lower-level (ALSA?) use of the card going on with "arecord".

Would anyone have any advice as to what to do next? Much thanks in advance for any tips.

Chuck

[bluebell]

Can you try with Audacity?

[merlyn]

It could be that the inputs and outputs are muted in alsamixer. Try

Code: Select all

alsamixer

Use 'F6' to select the 1010. You can unmute a channel with the 'm' key.

ALSA is a bit weird . I also have an ICE1712 card. This should work :

Code: Select all

arecord -f S32_LE -c 2 -d 10 -r 44100 --device="plughw:M1010" /home/chuck/test.wav

I don't know the exact details of the difference between hw: and plughw: but it works on my system.

[ChuckLutz]

Hi merlyn,

Thanks for the suggestions.

Turns out that may have been the case, but I am not out of the woods yet - I've been playing with the original envy24control, and I remembered that there is this updated version called "mudita24" that (if I understand it right) adds the display of the live audio level to the meters - that is, if there is a signal present, then I should see it jumping. Turns out that UbuntuStudio has mudita24 already on it so I started that up too. I also opened up alsamixer in a a parallel terminal window. I read a bit of the original windows-based Delta 1010 manual, and it reminded me that the hardware mixer / levels are unity gain. It turned out that the levels were all the way down, so I put them all the way up to 0db (the top of the fader range). I think I was also fooled by the envy24control buttons a bit b/c I couldn't initially tell when a button was pressed or not- by playing with the mute buttons and levels on envy24control/mudita24 and watching the alsamixer terminal, I figured out how to unmute things - also if I press mute on/off and play with the levels in envy24control or mudita24, then I see the changes in alsamixer in real time, which is a reassuring sign that something is working.

I gave the commands a quick spin:

chuck@chuck-OptiPlex-7010:~$ arecord -f S32_LE -c 2 -d 10 -r 44100 --device="plughw:M1010" /home/chuck/test.wav
arecord: main:828: audio open error: Device or resource busy
chuck@chuck-OptiPlex-7010:~$ arecord -f S32_LE -c 2 -d 10 -r 44100 --device="hw:M1010" /home/chuck/test.wav
arecord: main:828: audio open error: Device or resource busy

I hunted around and found this:
http://www.noah.org/wiki/%22arecord:_.. ... ce_busy%22

I had jack running at the time I tried to run the commands:

chuck@chuck-OptiPlex-7010:~$ fuser -fv /dev/snd/*
USER PID ACCESS COMMAND
/dev/snd/controlC0: chuck 18940 F.... alsamixer
/dev/snd/controlC1: chuck 1552 F.... jackdbus
chuck 3486 F.... envy24control
chuck 3938 F.... alsamixer
chuck 4121 F.... qasmixer
chuck 22450 F.... mudita24
/dev/snd/pcmC1D0c: chuck 1552 F...m jackdbus
/dev/snd/pcmC1D0p: chuck 1552 F...m jackdbus
/dev/snd/seq: chuck 3784 F.... qjackctl
chuck 18825 F.... audacity

I am re-learning some of the basic plumbing here - apparently with jack running, "arecord" can't access the card, but I am not sure why yet (I need to refamiliarize with some of the basic architecture of all of this stuff). I shut down jack and the "jackdbus" entries vanished from the list. I tried the command again:

chuck@chuck-OptiPlex-7010:~$ arecord -f S32_LE -c 2 -d 10 -r 44100 --device="plughw:M1010" /home/chuck/test.wav
Recording WAVE '/home/chuck/test.wav' : Signed 32 bit Little Endian, Rate 44100 Hz, Stereo

... so this time I got a .wav file. However, if I open it in the basic media player I don't hear anything. I did another basic test to see if audio is in fact working at all on my system - I tried to play a YouTube video or two, but in all cases I get errors similar to "An error occurred. Please try again later. (Playback ID: A_12FpOgx3u0cV3O)".

I should mention that to simplify things, I removed my mixer Group 1/2 outputs from the Delta 1010's inputs 1 and 2 and just directly plugged one of my keyboard's outputs into the Delta 1010 Input 1. I made sure the +4/-10 button is "out" which I think I read in the manual is the proper setting for direct instrument connections.

At this point I need to figure out if my audio works at all - I have an old pair of computer speakers I will hook up and I'll find some canned .wav file on the web to test playback with.

However, if I am right about the mudita24 ability to show levels jumping, then I would think that I would see that on the display when I hit some keys on the keyboard - but I don't.

Thanks again for your help,
Chuck

[merlyn]

You can get a test signal (pink noise) with ALSA by using speaker-test. This should work for you :

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$ speaker-test --device "plughw:M1010"

Use 'Ctrl-C' to stop it.

[channelite]

I have the M-audio Delta 2496 which is just like 1010, just 2 analog ins and outs on Linux Lite 4.8. I’m using the envy24control to turn on the volume. One thing about it I don’t like, is I have to launch envy24control every time I start up the computer and raise the volume. Is there a way that the setting can be remembered? Do I have to turn up the volume with envy24control every time the computer has been started?

[thumbknuckle]

I have the M-audio Delta 2496 which is just like 1010, just 2 analog ins and outs on Linux Lite 4.8. I’m using the envy24control to turn on the volume. One thing about it I don’t like, is I have to launch envy24control every time I start up the computer and raise the volume. Is there a way that the setting can be remembered? Do I have to turn up the volume with envy24control every time the computer has been started?

I had a system that behaved this way a few computers and installs ago. I think this works, or at least it did for me:

First make sure you have alsactl installed. I would think you should by default, but just try running it with no arguments. You should get back something like:

Code: Select all

"alsactl: Specify command..."

If that works get the soundcard configured the way you want it with envy24control and then run:

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sudo alsactl store

If I'm right you should be able to reboot and have the soundcard come back up the way you had it. I think. Let me know if this works for you.

[channelite]

I have the M-audio Delta 2496 which is just like 1010, just 2 analog ins and outs on Linux Lite 4.8. I’m using the envy24control to turn on the volume. One thing about it I don’t like, is I have to launch envy24control every time I start up the computer and raise the volume. Is there a way that the setting can be remembered? Do I have to turn up the volume with envy24control every time the computer has been started?

I had a system that behaved this way a few computers and installs ago. I think this works, or at least it did for me:

First make sure you have alsactl installed. I would think you should by default, but just try running it with no arguments. You should get back something like:

Code: Select all

"alsactl: Specify command..."

If that works get the soundcard configured the way you want it with envy24control and then run:

Code: Select all

sudo alsactl store

If I'm right you should be able to reboot and have the soundcard come back up the way you had it. I think. Let me know if this works for you.

Thanks,

Code: Select all

sudo alsactl store

fixed the problem. Volume is now turned up. No need for Envy24control every time.

[ChuckLutz]

HI folks,

I am catching up to my own thread here - thanks for all the replies. I am trying each out.

But first I have a BASIC SANITY CHECK... as I mentioned, I got my Delta 1010 used from eBay a while back. Trying to get it running with Linux is my first use of it. It occurred to me that I should make sure that the box even works in the first place! As my posts seem to show, the card appears to be recognized by the system.

However, I am not sure about the rack unit ("breakout box") itself.

Any users who have one of these and are using it with Linux - is the power light supposed to turn on when the rack unit is plugged in? Mine does not. I thought I heard somewhere that this is under software control and doesn't necessarily light up.

Apart from that question, does anyone know how I might be able to "prove" that the rack unit is working? Perhaps my problem is that although the card is "OK" maybe it is not receiving anything from the rack unit...

Maybe there is some command-line way to get the card to "ping" the rack unit (for lack of a better term)!

Thanks in advance,
Chuck

[kenzen]

The light comes on when the host machine it's connected to boots up. (Or at least that's how mine behaves...)
Ken

[ChuckLutz]

HI there,

Ah ok - I get no light whatsoever. At the risk of an idiotic question - you have it running with Linux, right?

Thanks,
Chuck

[ChuckLutz]

Hi folks,

Still at it (intermittently - as time allows...)

Trying another "basic sanity check" - I downloaded this simple .wav file of slow, solo clapping (applause) - it is 15s long:

https://bigsoundbank.com/detail-1767-applause-3.htm

I have the output channels 1 and 2 of my Delta 1010 box going out to an external mixer that then goes to two keyboard amps (that's all the monitoring I currently have).

Following the advice here:
https://www.alsa-project.org/wiki/SoundcardTesting

... I simply try to play that .wav file:

aplay -vv Downloads/1767.wav
Playing WAVE 'Downloads/1767.wav' : Signed 24 bit Little Endian in 3bytes, Rate 48000 Hz, Mono
ALSA <-> PulseAudio PCM I/O Plugin
Its setup is:
stream : PLAYBACK
access : RW_INTERLEAVED
format : S24_3LE
subformat : STD
channels : 1
rate : 48000
exact rate : 48000 (48000/1)
msbits : 24
buffer_size : 24000
period_size : 6000
period_time : 125000
tstamp_mode : NONE
tstamp_type : GETTIMEOFDAY
period_step : 1
avail_min : 6000
period_event : 0
start_threshold : 24000
stop_threshold : 24000
silence_threshold: 0
silence_size : 0
boundary : 6755399441055744000
#+ | 01%

What comes out of my speakers seems to be at the correct "pace" / tempo, and does seem to last the 15s of the original file.

However, it is VERY distorted sounding. I made a recording using my iPhone, and saved off the .aiff file from my phone to my machine here. Unfortunately it seems that a) this forum tool doesn't accept .aiff files and b) I tried converting it to .wav but I was further unable to attach it here due to file size restrictions. I even edited the file in Audacity - just visually (since my 1010 isn't working yet!) - down to the first 5s just to give an idea of how it sounds - which resulted in a 441K file, but this was STILL too big for this forum tool - I couldn't find any info on restrictions on attachment size. I gzipped it to no avail.

Any ideas on what would cause such distortion? Would it in fact help if I could find some way to share the resulting file from my iPhone?

Thanks again in advance for any help,
Chuck Lutz

[Pidgy]

Sorry to resurrect an old thread but I have a Delta 1010 which at one point had the issues that @ChuckLutz was having.
Specifically, the light went out and audio was very distorted, but the card still showed up in the os.

This problem turned out to be 2 power supply caps in the 19" rack part of the the unit. They were both swollen and had failed. They were
very easy to replace and my unit has worked without issues since I replaced them. I used a higher voltage cap than the original.

If you want to try this fix I'll supply pics. Just let me know.