Hi,
My laptop (running ubuntu) has the following (audio) specs. I know it's
probably subjective, but is this considered "good enough" for pro audio
work or should I invest in an external sound card? (Or for that matter a
better laptop?!) And if so, which soundcard is recommended?
Thanks a lot,
Peter
Memory: 3.8 GiB
Processor: Intel® Pentium(R) Dual CPU T3400 @ 2.16GHz × 2
00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio
Controller (rev 03)
Subsystem: Toshiba America Info Systems Device ff66
Flags: bus master, fast devsel, latency 0, IRQ 46
Memory at d6700000 (64-bit, non-prefetchable) [size=16K]
Capabilities: <access denied>
Kernel driver in use: snd_hda_intel
00:1c.0 PCI bridge: Intel Corporation 82801I (ICH9 Family) PCI Express
Port 1 (rev 03) (prog-if 00 [Normal decode])
Hey hey,
I want to sample the waveform of an oscillator of a hardware synth. One period
of the waveform must have 65536 points in the end. Will the following process
work:
Samplerate 48kHz, Oscillator pitch 71.0449218750000Hz, becasue:
71.0449218750000Hz / 48000Hz = 97/65536
In theory I can now create an empty table with 65536 entries and fill every
97th entry (with wraparound) with successive samples from the original audio
recording.
I see a problem: the soundcards ADC must average/interpolate the digital
samples from the analogue input arriving in between samples. I can't tune my
synth to 48000/65536Hz.
I can increase my samplerate to 96kHz and work through the sample process as
above, but will that help? Am I completely on the wrong track for creation of
an adequate copy of an analogue waveform?
If a correct process is more complicated and requires more mathematics, I will
drop the idea as such.
Ta-ta
----
Ffanci
* Internet: https://freeshell.de/~silvain
Twitter: http://twitter.com/ffanci_silvain
I have a cheap (hipstreet i8) android tablet I have been playing with. It
has Android 4.4.4, kernel 3.10.20 and says the cpu is an Intel product.
I have tried all the audio related applets I could find and have generally
been dissapointed with the latency. I hit the drum on a drum kit and the
sound is delayed enough to make it hard to play though it does help to not
listen to the audio and just play. I mention this as background for the
rest.
One of the apps is "WiFi Audio" http://www.ajeetv.info/wifiaudio/ (the
link is only the PC send end there is catually very little info about
this)
Anyway this app works quite well. Considering the background info above
about how much latency there is in android audio to begin with, I would
say that the WiFi transfor adds no noticable delay to the already long
android delay. The quality is good with drop outs only when I have the
android device obscured from the AP by a metal desk or something like
that. Certainly if the android audio latency could be cleared up this
would make a usable personal monitor for a musician for inear use.
In my case the audio chain looks something like this:
player -> pulse -> jack -> alsa -> out.
Jack is set to 128/2 which does some interesting things to pulse (It
forces pulse to deal with it at that latency)
The chain to my android is:
player -> pulse -> WiFi Audio -> android.
According to Pavucontrol, WIfI Audio if being fed by "Monitor of Jack
Sink".
The point of this is that if I had gotten a tablet/phone that was on the
ubuntu touch dev list, which while using the Android kernel, bypasses
most of the android audio stuff and goes straight to the Alsa bit. There
may be a usable monitoring system for Audio over WiFi.
--
Len Ovens
www.ovenwerks.net
Much more practical than audio over wireless is ardroid a remote control
Ardour. It just works really nice. There are a number of other remote
controls out there for the Android one of which looks like a complete MC
set up (TouchDAW) but I am realizing (besides not getting it to work) that
it is really more than I want or need in a touch control. And would likely
lead to finger trouble. The other nice toy was goOSC which can be set up
however someone wants it. It will take me a while to get it doing
something useful as the learning curve is a bit bigger. But very flexable.
A standalone GUI on the computer for creating layouts would be great... I
should make one :)
The only thing I would like with ardroid is one fader/peek meter so I can
adjust the input level which at my instrument. I realize the meter would
be slow and laggy and so would suggest only a static peek with reset. It
could even be text.
Most of the fader like controls I have seen on these soft controlers jump.
That is they don't move it you touch the control away from where the
control is but if you then slide your finger up to where the "knob" is,
the center of the knob will jump to where the finger is when the finger
reaches the edge of the knob. This may be something to do with the android
GUI.
I would suggest that a better way would be to take a touch anywhere on the
fader to be as if it was where the fader level already is and move in
whatever direction the finger moves from there... if the GUI allows such
things.
--
Len Ovens
www.ovenwerks.net
New builds available at: https://forge.ircam.fr/p/OM/downloads/
OpenMusic homepage: http://repmus.ircam.fr/openmusic/home
Main news:
- new package scripts for RPM and DEB, to take care of necessary
(32-bit) dependencies for OM. (OM is still 32-bit only from lack of
access to 64-bit lw-compiler.)
- the music-fonts OM uses (omfonts) get installed as part of the main
package, ie. the extra package is no longer needed
- a non-standard dependency: libsdif.so ("SDIF: Sound Description
Interchange Format" - http://sdif.sourceforge.net/) - is provided as
RPM- and DEB-packages at the download site
Other news:
- PortMidi handles midi i/o, using the same code on all 3 supported
platforms (Linux, OSX, Windows)
- fluidsynth is no longer part of the OM-application, instead users are
expected to connect to their preferred MIDI-synth to play back MIDI
from OM
- various new features and bug-fixes
Packages are checked on Fedora 20+21, and Ubuntu 14.04.1.
Thanks for all bug-reports.
-anders
Something has been bugging me...
Ardour is doing 32-bit FP math to handle samples internally.
And yet, when I change the volume on the master fader, even if I
compensate for the volume change on my RME Multiface's physical volume
knob for the headphone monitor, I'd almost swear I can hear some sort of
change in the audio. I'm not using any post-fader plugins. I'm
probably just fooling myself, but is it possible to change the perceived
audio by just changing the output level over a 4-5db range when you've
got a precision like Ardour is using, and you're listening through
decent 24-bit converters?
In a related question (which was the real reason I wanted to know), what
kind of levels should we be shooting for in the master output anyway?
Of course, in the old analog tape world, they used to shoot for zero,
maybe a little hotter if you wanted some saturation. In digital, they
say mix to about -3 or -4db. And mastering houses seem to want more
headroom than that (-6 or 7db), even though in priciple, with 24-bit
resolution, they could just adjust it themselves without changing it
timbrally (true?). Why do they care what your maximum level is as long
as it's 24-bit and it's not clipping?
Commercial CD's these days seem to be mastered with the idea that if the
level ever fell below -1db, that would be a sin against God.
Suppose I'm just trying to produce a demo that can be burned to CD,
uploaded to a streaming service, or given to a mastering house. Is
there a different ideal level for those various uses?
I suppose the question I'm wondering is, if you've got it just the way
you like it at some particular master fader setting, is anything harmed
by making it a little hotter before you export, especially if you're
going to be putting a track on CD where it will be only 16-bit? Or
should you just keep it where it is, even if you're wasting some headroom?
--
+ Brent A. Busby + "We've all heard that a million monkeys
+ Sr. UNIX Systems Admin + banging on a million typewriters will
+ University of Chicago + eventually reproduce the entire works of
+ James Franck Institute + Shakespeare. Now, thanks to the Internet,
+ Materials Research Ctr + we know this is not true." -Robert Wilensky
Hi,
I need to calculate the total playing time of the audio files in a
particular folder. These are variously encoded--flac, ogg, and even some
mp3.
Anyone know how to do this, preferably from the cli? Or, perhaps one of
the music player apps can provide this datum for a folder?
tia
Janina
--
Janina Sajka, Phone: +1.443.300.2200
sip:janina@asterisk.rednote.net
Email: janina(a)rednote.net
Linux Foundation Fellow
Executive Chair, Accessibility Workgroup: http://a11y.org
The World Wide Web Consortium (W3C), Web Accessibility Initiative (WAI)
Chair, Protocols & Formats http://www.w3.org/wai/pf
Indie UI http://www.w3.org/WAI/IndieUI/
Can anyone suggest a program that monitors a MIDI stream and sends it straight
to a .mid file without adding or subtracting anything at all?
--
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.
On Wed, 4 Feb 2015 00:01:15 +0100 (CET)
"F. Silvain" <silvain(a)freeshell.de> wrote:
> Will Godfrey, Feb 3 2015:
>
> > Can anyone suggest a program that monitors a MIDI stream and sends it straight
> > to a .mid file without adding or subtracting anything at all?
> In addition to arecordmidi smfrec, part of Midish (http://www.midish.org). This is also a CLI tool.
> ...
>
> Ta-ta
Thanks again. Actually arecordmidi told me exactly what I needed to know - A
sequencer (unnamed to protect the guilty) is silently inserting extra CCs at
start and end of run :(
--
It wasn't me! (Well actually, it probably was)
... the hard part is not dodging what life throws at you,
but trying to catch the good bits.