I'm the (un?)lucky owner of an M-Audio Fast Track Pro USB audio
interface and I'm having some serious problems getting this device to
record audio reliably under Linux.
I've been using arecord and occasionally Audacity for all of my
testing. My problem is this: Recording a take works about 80% of the
time. In the remaining ~20% of cases, the captured audio is extremely
loud with severe digital distortion. Once this problem shows up, it
persists for any subsequent takes. The only way I've found to make
the problem go away, at least temporarily, is to power-cycle the Fast
Track Pro.
I considered the possibility that this particular device might be
defective, but it seems to work wonderfully under Windows.
I'm calling out to other Fast Track Pro users in the hope that someone
out there has encountered the same problem and better still, found a
solution.
Any suggestions at all would be greatly appreciated!
.lewis
Hi.
Quite a shot into the dark, but: Is anyone here by chance into
eurorack modular and has tried the Expert Sleepers ES-8 USB Interface
with Linux yet? It claims to be class compliant, but you never know...
--
CYa,
⡍⠁⠗⠊⠕
Hi list,
I have been search the archives and the web a bit and it seems there is
no easy way to route html5 audio playback from firefox through jack on
Debian testing.
Is this still the case or has there been recent development that might
enable it? What would have to be done to make it work? This
functionality is the last piece in my puzzle to finally get rid of
flash.
thanks for pointers,
P
09-30-2017
Hi Bob,
Unfortunately, I am not the person to go into technical details about mixing. I know some of the basic concepts, but other than pan and volume, I have not used the other techniques.
Check out the Internet for sound mixing groups. At one time, I was working with Linux systems for music (personal pleasure use at home), and followed the Linux Audio group. I still do follow that group.
To post to that group, I use linux-audio-user(a)lists.linuxaudio.org <mailto:linux-audio-user@lists.linuxaudio.org>
There should be other Internet groups around. The people at the Linux audio group include mixing professionals. They should be able to listen and give suggestions on how to adjust your mix. The group is for people using digital audio workstations (DAW) running under Linux and Linux mixing tools. So some of their suggestions may only apply to Linux programs/tools.
But they are very helpful in listening to work and making suggestions.
I would think there are professionals who use the DAW and tools that you are using. Find the groups when they hang out. Mention your hearing issues. At the Linux audio group, there are blind people using command line Linux DAWs and tools for their mixing. As you are probably aware, musicians are very generous with their time and talents.
Good Luck,
Stephen.
From: Bob Ebdon [mailto:r.ebdon@ntlworld.com]
Sent: Saturday, September 30, 2017 7:54 AM
To: Stephen Stubbs <fartreader(a)gmail.com>
Cc: Bob Ebdon <r.ebdon(a)ntlworld.com>
Subject: Re: [CP] KISS or Complex?
Thanks Stephen. Just back on home computer after a few days away, so sorry for not getting back sooner. I needed to check a few things with my recording. As always, I value your comments.
One thing that worried me was your comments about panning the instruments. They are panned. I have two guitars panned 66R and L, fiddle is 33R, bass and lead vocal central. Autoharp mic and DI are 10L and 10R, vocal harmonies are 33L and R. Maybe this spread isn’t far enough? Maybe push the fiddle wider, it is in the same space as one of the harmony vocals at present? I have real problems with stereo placement as I am basically mono - totally deaf in one ear! This is why I had to come back to the recording and check that I had actually exported a stereo version, I could not tell just by listening whether it is stereo or mono, the only way I can check is if it sounds different when I swap the headphones round - and it does. I also have faders way down on the instruments relative to the vocals, and have written volume for the autoharp to bring it up as fills.
I appreciate your comments about reverb and compression. I have not yet discovered how to make my voice sound natural! I have on it some EQ, some compression (set at ratio 4.2 - too high?), a vocal rider and a C6 multi band compressor set to give a treble boost. Looks like this is all too much?
For reverb, I use Spaces. I have two, one set at about 2sec, the other at 6sec, and I use about -10dB of the first and -30dB of the second. Again - too much?
Any advice you can give is much appreciated.
Bob Ebdon
www.facebook.com/AutoharpBob <http://www.facebook.com/AutoharpBob>
On 28 Sep 2017, at 14:11, Stephen Stubbs <fartreader(a)gmail.com <mailto:fartreader@gmail.com> > wrote:
Why not do both?
Start out with the vocal and autoharp. Then start adding the other
instruments through the course of the song, to end up with everything
at the last.
On a mixing level, I prefer less reverb and compression. Why not pan
the backup instruments to different locations, and use volume to get
the depth of field? Go for the feeling that you and the autoharp at
the microphone, while the backup instruments are behind and around
you. I think it would give you more of a live performance sound.
For What It's Worth,
Stephen.
Hello,
This is a fast-paced 'organic techno' (of sorts) synth track. It was
fun to mix.
https://soundcloud.com/nominal6/too-late-for-goodbyes
Unfortunately the bass was mostly eaten by soundcloud. Almost sounds
AM-like. I'll have to figure this one out when mixing: why is the bass
largely diminished on soundcloud and how to compensate, if possible,
for this.
Cheers.
Hmmm. Those links are pretty "discutable". Telling that 128kbps is OK
for mobile and/or non critical material like rock or heavy metal seems
to me completely false: have you never heard "swiiiiiiish" in the hi
hats of such productions when using low bitrate mp3 (in which I include
128kbps)?
By the way, mp3 works on psycho-acoustics, and is based on the concept
of masking frequencies, so using a spectral view to explain that the
frequency content is or is not there is a non-sense when the principle
lies on the fact that those frequencies wouldn't have been "heard" in
the real world when positionned after certain other frequencies.
And I don't think "sacrifying" low and high ends means that low and high
ends aren't present, it just means that they are altered, and that you
can't check with a spectral view.
Still, I suggest using flac, bandcamp does accept that file format.
Le 29/09/2017 à 00:30, Paul Davis a écrit :
> There is no significant effect of MP3 encoing bitrate on low end
> frequency.
>
> https://www.mp3-tech.org/tests/pm/MP3-128k.htm
> https://www.mp3-tech.org/tests/pm/MP3-160k.htm
>
>
> On Thu, Sep 28, 2017 at 5:49 PM, Be <be.0(a)gmx.com
> <mailto:be.0@gmx.com>> wrote:
>
> Soundcloud uses 128 kbps MP3, so extreme highs and lows will be
> sacrificed. If you want to distribute your tracks as you hear them
> when you're making them, make them available as lossless
> downloads. Bandcamp is a good option for that.
>
>
> On 09/27/2017 11:09 PM, jonetsu wrote:
>
> Hello,
>
> This is a fast-paced 'organic techno' (of sorts) synth track.
> It was
> fun to mix.
>
> https://soundcloud.com/nominal6/too-late-for-goodbyes
> <https://soundcloud.com/nominal6/too-late-for-goodbyes>
>
> Unfortunately the bass was mostly eaten by soundcloud. Almost
> sounds
> AM-like. I'll have to figure this one out when mixing: why is
> the bass
> largely diminished on soundcloud and how to compensate, if
> possible,
> for this.
>
> Cheers.
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user(a)lists.linuxaudio.org
> <mailto:Linux-audio-user@lists.linuxaudio.org>
> https://lists.linuxaudio.org/listinfo/linux-audio-user
> <https://lists.linuxaudio.org/listinfo/linux-audio-user>
>
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user(a)lists.linuxaudio.org
> <mailto:Linux-audio-user@lists.linuxaudio.org>
> https://lists.linuxaudio.org/listinfo/linux-audio-user
> <https://lists.linuxaudio.org/listinfo/linux-audio-user>
>
>
>
>
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user(a)lists.linuxaudio.org
> https://lists.linuxaudio.org/listinfo/linux-audio-user
spectmorph-0.3.4 has been released.
Overview of Changes in spectmorph-0.3.4:
----------------------------------------
* Added optional ADSR Envelope
* Make LV2 and VST plugin stereo to allow supporting stereo in the future
* LV2 plugin description fixes
* Added about dialog to plugin/smjack UI
* Remove BEAST plugin (plugin code will be moved to BEAST)
* Fixed compilation for newer g++ >= 6 (std::fabs)
* Get rid of some malloc() calls in linear morphing
What is SpectMorph?
-------------------
SpectMorph is a free software project which allows to analyze samples of
musical instruments, and to combine them (morphing). It can be used to
construct hybrid sounds, for instance a sound between a trumpet and a flute; or
smooth transitions, for instance a sound that starts as a trumpet and then
gradually changes to a flute.
SpectMorph ships with many ready-to-use instruments which can be combined using
morphing.
SpectMorph is implemented in C++ and licensed under the GNU LGPL version 3
Integrating SpectMorph into your Work
-------------------------------------
In order to make music that contains SpectMorph, you currently need to use
Linux. There are four ways of integrating SpectMorph sounds into music you
create:
- LV2 Plugin, for any sequencer that supports it.
- VST Plugin, especially for proprietary solutions that don't support LV2.
- JACK Client.
- BEAST Module, integrating into BEASTs modular environment.
Note that at this point, we may still change the way sound synthesis works, so
newer versions of SpectMorph may sound (slightly) different than the current
version.
Links:
------
Website: http://www.spectmorph.org
Download: http://www.spectmorph.org/downloads/spectmorph-0.3.4.tar.bz2
There are many audio demos on the website, which demonstrate morphing between
instruments.
--
Stefan Westerfeld, Hamburg/Germany, http://space.twc.de/~stefan
Starting Aeolus from command line returns this:
Reading '/usr/share/aeolus/stops/Aeolus/definition'
Reading '/home/david/.aeolus-presets'
Segmentation fault
Running on Debian Testing. It's been awhile since I played anything with
Aeolus, so I have no idea how it might have been broken.
Just tried reinstalling, no luck. Then removed Aeolus and stops packages
completely, but trying to reinstall using Synaptic reported that I had
"held broken packages' and wouldn't complete. Command line APT was able
to reinstall them.
But I still get seg faults and it won't start. I miss my 'real' organ
sounds!
Ideas?
--
David W. Jones
gnome(a)hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com
Dear List!
this is my first post to this list, but the issues and solutions i faced
with the MotuAVB devices plus the light push of a fellow linux-user made
me subscribe to this list to share my experiences.
I am a happy user of a Motu 24ao and Motu Ultralite AVB under
Linux(Ubuntu 16.04) in combination with jack/cadence.
almost everything went rather fine until i updated the firmware last
week on my UltraliteAVB:
the current firmware v1.2.9+479
<http://cdn-data.motu.com/downloads/audio/AVB/firmware/io/170621/MOTU%20AVB%…>
(Release Date 2017-06-21) removes the possibility to change the USB-mode
(44/48; 44-96; 44-192) ultimately disabling me to use more then
24channels via USB/class compliant.
With the previous firmware v1.2.8+378
<http://cdn-data.motu.com/downloads/audio/AVB/firmware/io/160901/MOTU%20AVB%…>
(Release Date 2016-09-01)this feature is still available, allowing for
64ch USB-i/o(44/48), 32ch USB-iO(44-96khz) or 24ch (44-192kHz).
According to Motu and tested by myself, a downgrade of the firmware is
possible without bricking the device.
The sad truth is, that the users are locked to this firmware until motu
brings back the feature to change the USB-Mode- unless one loves to
settle with 24ch i/o between computer and device.
regards,
Peter