Jonathan E. Brickman | 3 Mar 16:13 2015

I have been trying to strategize cutting the USB cord (right now, one USB, one Firewire...) for my portable synth on and off for a long time; I really don't like wearing out USB ports and cables !!!!  Was just perusing some of the AoIP conversation and thinking...and then a thought was given, did a search, and found:

So, at least in theory, for about $160 or so, I could get one of those and an RPi 2.0 starter kit, run netjack and qmidinet, and have a 96kHz wifi audio interface with MIDI...?  Anyone see an obvious catch?

Jonathan E. Brickman | | (785)233-9977
Ponderworthy |
Music of compassion; fire, and life!!!

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Len Ovens | 3 Mar 07:02 2015

[LAU] Mac and PC

I have been reading a manual for an AoIP (ravenna) SW sound card. There is 
a very interesting comparison between a Mac and a win-pc.

This is not a special card, just a standard GB ethernet card connected to 
a Ravenna i/o box (Horus).

 	48K = 64 i/o
 	96k = 32 i/o
 	192 = 16 i/o
 	384 = 8 i/o
 	64 samples - not recomended
 	128 - ok - but not default
 	256 -recomended
 	At 192k, smaller buffer sizes are ok, but 64 is minimum (like jack
 		at 32/2)

 	64 i/o at all sample rates
 	latency 32/48/64 (defaults to 32)

Is this the hw or the os? the mac gui shows if the computer is running ptp 
or not, but the windows one doesn't. maybe that is the difference.

Len Ovens
Ben Bell | 28 Feb 17:19 2015

[LAU] Focusrite Saffire Pro 40 - reliable?

Hi all,

My dual Delta 1010 setup is on its last legs and I'm thinking (again) of
retooling with a Saffire Pro 40. I've been bitten in the past by upgrades
like this which is why I'm still on the 1010s, but there's only so many
times I can repair them before I have to admit defeat.

Is anyone out there actively using a Saffire Pro 40 in a serious setting?
Is it reliable for low latency, with all channels working at once? Does it work
with any recent ffado, or do you need specific versions, patches or anything

I've read plenty of vague comments about which firewire chipsets might be
good or not, but are there known-good PCI Express firewire cards that are a
sensible price and work well?

I'm really hoping for "Yes, I've been using one for a year and it's rock
solid," rather than "I've googled and it claims to be supported," or "Yes,
it's fine if you pull the latest beta7 branch from git and only use Acme
2000 firewire interfaces." Sorry to sound unduly cautious or skeptical ;)

In short, if I bite the bullet, retire my Delta 1010s and buy a Pro 40, are
the drivers and tools now at the stage where I reasonably expect to be up and
running again in a day rather than still fighting problems a week later?

Chris Caudle | 27 Feb 17:06 2015

Re: [LAU] Problems with sample rate and recordings (Mackie onyx > 1640-i < Firewire > Debian )

> Date: Thu, 26 Feb 2015 17:18:35 +0100
> From: sub <subvertao@...>
> Message-ID: <54EF475B.8040601@...>
> i received a mail from the band in which they tell me
> that the mix is faster and at a higher pitch than the
> audio they recorded through a video camera at the
> concert.

Video cameras record at 48k rate, so someone will need to rate convert. 
Either the camera audio will need to be converted to 44.1k or your audio
will need to be converted to 48k.
Which you choose probably does not matter, but everyone involved needs to
know and agree on the chosen project sample rate.
If the final output will be a video then converting every thing to 48k is
probably the easiest choice.  If the final output will be a CD (or audio
only files) then probably 44.1k is the easiest choice.  If you will have
both a video and audio only files, my choice would be working in 48k and
then sample rate convert the final mix to 44.1k for CD.


Chris Caudle
tom haddington | 27 Feb 15:24 2015

[LAU] retuning bristol

Hey, Everybody--

i loaded bristol with a scala (.scl) file, this morning.  That much worked fine.  However, both my midi keyboard and the gui keyboard are nowhere near middle C.  i've even tuned my midi keyboard up (a maximum) three octaves, and my middle C is still two octaves too low.  Can this be prevented or adjusted in anyway?

Linux-audio-user mailing list
sub | 26 Feb 17:18 2015

[LAU] Problems with sample rate and recordings (Mackie onyx 1640-i < Firewire > Debian )

I am running linux on a Debian machine with kx repositories, connected
to a Mackie Onyx 1640-i.

Linux kx 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt4-3 (2015-02-03) x86_64
04:00.0 FireWire (IEEE 1394): Texas Instruments TSB43AB23
IEEE-1394a-2000 Controller (PHY/Link)

When i record multitrack audio using Ardour and jackd, i see that the
sample frequency is set to 44100.
The audio files are also at 44100, according to file:

root <at> kx:/home/humla/akt3bolger/interchange/akt3bolger/audiofiles# file
<head> bass-2.wav: RIFF (little-endian) data, WAVE audio, mono 44100 Hz

After doing a mix of the material on a separate machine using Ardour3 at
44100, and dumping the mix internally to a 44100 WAV, i received a mail
from the band in which they tell me that the mix is faster and at a
higher pitch than the audio they recorded through a video camera at the

When i playback audio from the linux machine (like mp3s ) through the
firewire and the Onyx console, i get random switches of speed and pitch.

I have been working in professional audio, and i know there must be a
problem with the sample frequency lock and synch between the two
devices, so that one is on the wrong sample frequency, and "translates"
the bitstream wrongly. Normally in a digital audio chain you have a
machine that is "master" and one that is "slave".

The problem here is that the console has no display showing the sample
frequency it is locked at, and the Linux machine does not show any
master / slave feature in jack or ardour.

There is something strange going on here.
The machine has also another analog audio card:

00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio
Controller (rev 02)

And i don't know how this could influence the setup.
Brendan Furneaux | 26 Feb 14:55 2015

[LAU] SPDIF plus NetJack for digital audio transfer between computers.

I play keyboard in a band, and use UbuntuStudio on a laptop for softsynths, effects, and most importantly, MIDI routing (via mididings).  Across the room is our studio computer, which runs Windows.  I am running audio out of my laptop via a Presonus FirePod (aka FP10), which works fine.  The studio computer has a Presonus FireStudio and Digimax, which are joined by an ADAT lightpipe.  Both the FirePod and FireStudio have SPDIF ports. At the moment, audio is traveling via audio cables from the hardware synths and laptop across the room to the studio computer.  However, it seems like I have everything in place to have a "simpler" setup, at least from a wiring and D/A/D conversion standpoint, by using NetJack to send digital audio via a crossover cable from my laptop to the studio computer.  The plan would be:

*The FireStudio (on the studio computer) must have its clock synced to the Digimax via ADAT.
*The FirePod (laptop) can be synced via SPDIF to the FireStudio.  Thus, all audio hardware will be on one master clock without any effort from the computers.
*Use the FirePod to capture audio from the hardware synths.
*Use NetJack(1|2) to route audio from the laptop (both captured from the hardware synths, and from softsynths) to the studio computer, via an Ethernet crossover cable.

The last step is where I am running in to trouble.

For starters, NetJack(1|2) does not seem to be designed for the use case where the sound cards on two networked computers are actually synced in hardware; thus the normal operating mode where only the master can access its audio hardware.  So far, my plan is to use NetJack2, with the laptop as master and the studio computer as slave. I have thought of either using audioadapter with the net backend on the slave, or of using netadapter with the alsa backend. Both of these do resampling to compensate for drifting clocks.  My first set of questions is: In the case that the clocks on the master and slave hardware are already synced, do audioadapter and netadapter still spend significant CPU time on resampling overhead?  Which one is likely to be more reliable/less resource intensive? Can netadapter receive MIDI? Is there some other software that would be better?

I currently have Jack 1.9.10 installed on both computers, and they work fine individually.  They are configured with static IPs on the crossover cable, and are at least able to ping each other.  However, I have not been able to get NetJack2 to connect using either computer as the master.

Linux-audio-user mailing list
Carlos sanchiavedraz | 25 Feb 19:59 2015

[LAU] Raspberry PI, internal audio IF + testing new USB IF

Hello dear all.

I'm testing a small interface for some musical and technological projects which has an instrument input jack on one side and no audio output, just USB on the other side. This IF works great but you have no physical audio output to monitor/hear sound, so I thought I could enable internal audio IF (as I've done on other devices) and that will be the audio output instead of needing another second IF to get audio from.

So I decided to rescue some past investigations I made when first researching about RPi, along with the great content generated by some dear great folks in here, and some new research one after each problem.

First started from a Raspbian 2012 version which I had already tweaked for a headless autojamming/instrument effects/looper, but got many issues making step by step progresses that I briefly point here FWIW:

- Enabled internal IF on Raspbian 2012 [1]

- Found mmap problem regarding using Jackd with internal IF [2], even tried a module somebody compiled

- Tried switching to the new platform version, Raspbian 2015 (raspberrypi 3.2.27+) but there wasn't even /proc/asound

- Couldn't get to load snd-bcm2835 kernel module for internal soundcard [4]

- Finally I got an error from the very raspi-config regarding internal audio IF [5]

Although I continue researching, I guess somebody here has already gone through some of this, but I'm wondering if it's already an abandoned thing by the makers themselves this thing of the internal IF thing.

Thanks as always.
Greetings all.


[1] enable internal souncard on Raspbian 2012

[2] enable mmap to run jack with internal soundcard on Raspbian 2012

[3] cat: /proc/asound: No such file or directory

[4] error when loading snd-bcm2835 kernel raspberrypi 3.2.27+ module for internal soundcard on Raspbian 2015

[5] error raspi-config when enabling audio via minijack on Raspbian 2015


C. sanchiavedraZ:
* Musix GNU+Linux:
Linux-audio-user mailing list
Hermann Meyer | 25 Feb 15:16 2015

[LAU] new Gxwavescharper.lv2

Fore those who love heavy distortion plugs :twisted: , I've uploaded a 
new, UI-less wavescharper lv2 plugin.
Get it here:

Will Godfrey | 25 Feb 10:06 2015

[LAU] Off topic - sorry

Is anyone able to post to any of the sourceforge lists?
My posts for the last three days (on different groups) seem to have been
silently ignored :(

Then again, this post will at lest prove whether I can post to any other list :?


Will J Godfrey
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.
Chris Bungue | 20 Feb 20:14 2015

[LAU] phase inversion plugin

Hi, I'm locking for a phase inversion plugin.
But I could not find anything.
Some ideas?

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