Kevin Cosgrove | 20 Apr 01:27 2014
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[LAU] MIDI hardware with Linux


Could someone point me to some reference material to get MIDI 
hardware up and running on my Linux system?  I knew how to do this 
just fine about 3 rev's of Ardour and 1 major rev of the kernel ago.
Unfortunately, all the documentation I'm finding is 7-10 years old.

I'm trying to figure out how to interconnect the *MIDI* of the following:

- M-Audio Delta 1010 hardware MIDI ports
- Mackie Control Universal hardware MIDI ports
- Alesis QS8 hardware synth
- a2jmidid software MIDI interface
- Ardour jack-MIDI ports

What can you folks advise?

Thanks...

--
Kevin
Len Ovens | 19 Apr 19:13 2014
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Re: [LAU] MIDI over firewire on Linux?


On Sat, 19 Apr 2014, Jonathan E Brickman wrote:

> Thanks!  Do you know of any good such devices which are not very pricey?  I
> found a few out there after reading your post, but they seemed all to be in
> the high-port-count and higher-cost ranges.

Cheapest Audio plus midi has got to be the old ens1370 based cards... free 
if you can find it. Best 16 bit sound for the price. If you don't have a 
pci slot, then the next price range is going to be USB devices... because 
most new systems don't have a FW plug, so an adaptor is required. Also, 
USB covers more people's systems so scale brings prices lower too. USB 
midi ports can be had for less than $30. And of course there are lots of 
USB audio solutions out there including built right into the mic stuff, 
again under $100. USB audio is much more picky about system tuning for low 
latency.

Assuming you already have a FW port, it is a better solution than USB, but 
even a 2+2+midi FW unit will cost more than the USB 2+2+midi unit. The 
Focusrite line starts with the pro 14 at $260. They call it 8in and 6 out, 
but I only see 6 physical inputs (and two of those are s/pdif i/o so only 
4 i/o for analog) at $260... yet they sell the 2i2 for usb for $150 or so. 
Presonus is the same, their "Firestudio mobile" is similar features and 
price to the pro 14 and the price matches too. (just note that it seems to 
need breakout cables to get at all the ports... including MIDI) Their 
audiobox USB (2+2+midi) can be had for $140.

So, yes FW costs more, but works better/easier. In the end you have to 
decide how important moneywise these things are to you. The presonus USB 
has been very good and stable... except in USB3 ports on Intel USB3 ports 
(Continue reading)

Rafael Vega | 19 Apr 17:10 2014
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[LAU] Marking soundcard outputs as "default"

Hi,

I recently started using ALSA Jack plug-in bridge and have this issue: My laptop's sound card has 4 outputs: 1->L, 2->R, 3->Unused?, and 4->"woofer".

When any of the ALSA programs I use (flash player, for example) connect to Jack, they expose 2 outputs and auto connect to the sound card's 1st and 2nd output. I would like them to auto connect so that the 4th output (the woofer) is also used. right now, every time I hit play on grooveshark, for example, I have to go to patchage an make connections from output 1 and 2 in alsa bridge client to 4th output in the sound card.

Is there any way to make this automatic? perhaps by marking outputs 1, 2 and 4 as "default" or "autoconnect"? Maybe there is a hook that detects new clients and I can write a script that connects alsa clients the way I want?

Now that I think about it, this should not apply only to alsa bridge client but any jack client!

Regards,

Rafael.
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Jonathan E Brickman | 19 Apr 05:58 2014

[LAU] Concerning schedtool

I gave it a royal tryout just now.  Below is sound-patch startup code, it's actually a mashup of most of my START-INITIAL script which runs at boot, and the first patch script which it calls.  When I keep the schedtool prefixes, MIDI keyhits all make it in and synths all get their signal out through the sound card output.  When I don't, about one in 20-30 keyhits (and also occasionally key-ups) get lost, and there are cutouts both at high load and very low loads.  Some may notice I have let the latency rise to 4; this too seems to be indicated for the loads I'm using, though if I were using just one light load (say, a simple Yoshimi patch), it could be ratcheted down a lot.  Oddly, it does appear that a different patch which calls just Fluidsynth, functions as a heavier load than three Yoshimis; it is true that Fluidsynth is itself using a rather big four-voiced soundfont.  I have another which right now is two Yoshimis plus the same Fluidsynth an octave lower (octave handled by mididings), that one is higher yet, but the setup and weights below are handling it well.

The relative weights are definitely guesses based on behavior. I just increase the weight of anything which misbehaves, until it makes it worse, then reverse :-)  I did find earlier tonight that I had neglected my limits.conf, so this is what I have set now, which allows those numbers below to actually work:

<at> audio          -       rtprio          95
<at> audio          -       nice           -19
<at> audio          -       memlock         512000

I'm not sure the nice is at all needed, but it doesn't seem to be harming anything.

J.E.B.

------

jack_control start

jack_control ds dummy
jack_control dps capture 0
jack_control dps playback 0
jack_control dps rate 48000
jack_control dps period 64
jack_control eps realtime true
jack_control eps realtime-priority 50
jack_control eps clock-source 1

# Stop any running audio elements
echo "Stop any running audio elements..."
killall -9 -w yoshimi fluidsynth zita-j2a aj-snapshot calfjackhost non-mixer rakarrack mididings lashd
killall -9 -w yoshimi fluidsynth zita-j2a aj-snapshot calfjackhost non-mixer rakarrack mididings lashd

# Remove all connections
echo "Stopping a2j..."
a2j_control stop
echo "Stopping Jackd to remove all connections..."
killall -9 -w jackdbus
killall -9 -w jackdbus
jack_control stop
echo "Starting Jackd..."
jack_control start
sleep 2

# Running a2j -- needful to use ALSA MIDI devices with
# zita-a2j, which uses the 'dummy' driver in Jack
echo "Starting a2jmidid..."
a2j_control ehw
a2j_control start

# Start zita-j2a for audio out
echo "Starting Zita..."
nohup schedtool -R -p 80 -e zita-j2a -d hw:SB -r 48000 -p 64  &

# Start all relevant audio elements
echo "Start all relevant audio elements..."
nohup schedtool -R -p 75 -e mididings -f /home/jeb/Combine.py     \
    > /home/jeb/LOGS/Combine.log &
nohup schedtool -R -p 50 -e calfjackhost --client CalfSRO     \
    eq12:SRO ! reverb:SRO ! Compressor:SRO             \
    > /home/jeb/LOGS/calfjackhost-SRO.log &
nohup schedtool -R -p 50 -e yoshimi -N YoshSRO1 -j \
    -l /home/jeb/YOSHIMI/SROpart1.xmz  > /home/jeb/LOGS/Yoshimi-SRO1.log &
nohup schedtool -R -p 50 -e yoshimi -N YoshSRO2 -j \
    -l /home/jeb/YOSHIMI/SROpart2.xmz  > /home/jeb/LOGS/Yoshimi-SRO2.log &
nohup schedtool -R -p 50 -e yoshimi -N YoshSRO3 -j \
    -l /home/jeb/YOSHIMI/SROpart3.xmz  > /home/jeb/LOGS/Yoshimi-SRO3.log &
sleep 2

# And lastly, create jackd connections using aj-snapshot
echo "And lastly, create jackd connections using aj-snapshot..."
cp /home/jeb/AJSRO.xml /home/jeb/AJRunning.xml
nohup schedtool -R -p 75 -e aj-snapshot -d AJRunning.xml &

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Jonathan E Brickman | 19 Apr 03:43 2014

[LAU] MIDI over firewire on Linux?

I realized there was a possibly significant gap in my knowledge of hardware, so I searched, and found that there have been a few Firewire devices with both audio and MIDI.  The questions I have are (a) are any of them known to work well with current Linux, and (b) how is the MIDI supported on the backend, is it ALSA?

J.E.B.
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Filipe Coelho | 18 Apr 23:05 2014
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[LAU] ams-lv2 ingen patches

Hi there.

The ingen patches in this page are no longer available:
http://objectivewave.wordpress.com/ams-lv2/

I'd like to try them.
Can someone re-upload them please?
Thanks!


PS: UbuntuOne will go offline and stop working very soon

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Brett McCoy | 18 Apr 16:23 2014
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[LAU] Interview about theater music produced with Ardour3

I recently did an interview regarding the theater company I write music for. I am posting this because I produce all of my music with Linux audio (although Windows samplers like Kontakt are used, the real work I describe here is done with Linux). I also mention MuseScore and Ardour3, here, too :-)

http://www.art-stream.org/content/flair-dramatic-music-great-scott

(the singing your hear is me, I am a horrible singer but recorded scratch vocals for the actors, even Autotalent couldn't fix my warbling)

--
Brett W. McCoy -- http://www.brettwmccoy.com
------------------------------------------------------------------------
"In the rhythm of music a secret is hidden; If I were to divulge it, it would overturn the world."
    -- Jelaleddin Rumi
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Len Ovens | 17 Apr 22:39 2014
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Re: [LAU] hardware - Intel CPUs


On Thu, 17 Apr 2014, david wrote:

> On 04/17/2014 09:56 AM, Len Ovens wrote:
>> 
>> 
>> On Wed, 16 Apr 2014, david wrote:
>> 
>>> Then running your sound from stage to backhouse sound board back to
>>> stage and hearing it through headphones would give no latency at all.
>> 
>> Nothing I can hear. This is very important becasue if there was
>> noticable delay then stage sound would be ahead of the mains... might
>> give some interesting filtering from being out of phase.
>
> The mains are getting their signal via cable, so they're getting their 
> signal as quickly as the monitors on stage. The only change would be 
> physical distance from the different speakers, right?

The difference I was thinking of is between a DIed guitar amp and mains as 
these and the bass amp (sometimes KB too) have traditionally been faced 
towards the audience. We also have some pretty loud vocalists who's 
unamplified voice can be heard in the house.

--
Len Ovens
www.ovenwerks.net
Len Ovens | 17 Apr 21:53 2014
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Re: [LAU] hardware - Intel CPUs


On Wed, 16 Apr 2014, david wrote:

> On 04/16/2014 03:17 AM, Len Ovens wrote:
>> I was talking about through air. I can only play as fast as I can hear
>> and at thirty feet or so from the sound source, my instrument sounds
>> delayed from the rest of the band because I hear their sound that much
>> later and play my part that much later. I don't normally play at that
>> distance, but ten feet is pretty standard.
>
> You're a few orders of magnitude better than me! I don't think I'd notice 
> that much difference.

I used to play and try to adjust sound while practicing, so I had 50 feet 
or so of cable. By the time I got to the sound board it was getting hard 
to play.

> I use  headphone monitor.

I find monitoring with headphones when playing bass is hard. I use a small 
amp tilted back about 10 feet ahead of me... so 13-14 feet to my ear I 
guess. Bass needs some space to work. My amp just gives me enough for me 
and the board can add whatever they want to the mains. I go a bit heavy on 
the high end for my monitor and the mains can roll that off for house 
sound.

> That's my understanding, too. Things like staying away from USB mice and 
> keyboards, making sure the USB audio is the only device on that bus, yah?

On that irq... same for pci(e) really even prioritizing shared irqs, lone 
irqs are still best.

>> However, it looks like I can still get lots of MB with PCI slots in
>> them. I will probaly do that. Hopefully with three PCI slots I can get
>> one that is irq clean.
>
> My desktop has 2 PCIe slots; the other 4 are PCI. It also has 4 USB ports + 
> 2 more USB connector points on the mobo, 4 SATA connectors, an EIDE 
> connector, a floppy connector, plus the built-in audio and video and 
> Ethernet. I haven't checked to see what's sharing interrupts and what isn't. 
> I wouldn't be surprised if there aren't a bunch of shared interrupts!

cat /proc/interrupts will tell you that quick enough. For some reason a 
lot of MB share irq16 with 3 or 4 things... worse one of those tings is 
often one of the PCI(e) slots. No need really as most modern MB have 
access to 48 plus irqs. My old board has 24 but 2-7 are unused (for 
hysterical reasons?) and 10 and 13 seem to be skipped too. I think 20 goes 
to my internal audio which I have turned off (AC97) so it doesn't show.

I have noticed that on any of the MB I have checked or seen irq maps for 
that the internal audio always has it's own irq and quite often it is the 
highest one which in most systems these days has the most priority. I 
found this out because I had the card I was using for midi in the higher 
of the two slots and my audio below. I had trouble with xruns on the 
audio, but when I put the audio on the higher irq I had no more trouble.

I know that in theory that shouldn't happen because there are two part to 
the irq drivers, a stub to answer the IRQ and save enough info to work and 
the other part that the os prioritizes and does all the work. So the os 
should be able to prioritize by the module name. I just know what I have 
found works best.

I have heard the words "in a modern system" too many times. I think any 
system can do better audio if it is tuned/tweaked.

--
Len Ovens
www.ovenwerks.net
Kaza Kore | 17 Apr 13:02 2014
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[LAU] M-Audio MTRACK

Looking to buy a new USB audio interface in the next couple of days as am at a place where I will be staying for a little while (some of you may remember my previous post about the cards available on Amazon India as I'm currently travelling.) Well had a look at Ebay and come acros the M-Audio MTRACK. Sure it should be Linux compatible as it claims to be Class Compliant but I have a question about the headphone output which isn't answered specifically in the manual or FAQ.

Can anybody confirm whether it is a duplicate of the Main output audio? Or if it is individually configurable at all? I suspect the former, but I have been happily surprised in the past, finding a headphone output to actually be a stereo output of its own, rather than a Master duplicate. Feel this very unlikely in the case but thought I would ask just in case anybody knows (or wants to warn me off for other reasons)...


http://www.m-audio.com/images/global/manuals/M-Track_-_User_Guide_-_v1.1.pdf

http://www.m-audio.com/products/en_us/MTrack.html

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Carlo Ascani | 17 Apr 12:37 2014
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[LAU] Fwd: Music made in Linux - A song from an hobbyst

2014-04-16 20:52 GMT+02:00 F. Silvain <silvain@...>:
> Did you use a real analogue drum machine or are those sounds from a software
> synth? Also what tool did you use for the effect sound?

The electronic drums are samples played by hydrogen and processed by plugins
(calf mostly).
On the second part the real drum mixes with those electronic samples.

I used mostly calf plugins (eqs, comps and reverbs),
and a couple of effects from guitarix (distorsion and tube screamer)

> Thank you again.

That's a pleasure to share things with you guys!

> ...
>
> Ta-ta
> ----
> Ffanci
> * Internet: http://freeshell.de/~silvain

--
Carlo Ascani | carlorat.me
skype: carloratm

--

-- 
Carlo Ascani | carlorat.me
skype: carloratm

Gmane