Hi,
I do have what I consider as a big problem with MP3 encoding, and that
is artifacts and degenerated sound quality. End results are things like
stronger esses, more "noise" (IE. something like schhhhhh upper
overtones on disted guitars) and a loss of clarity. The thing is that a
finished 320 kbps MP3 should not sound (significant) different than an
original wave file IMO. Note that this artifacts are usually quite
subtle, but very audible for the person(s) that recorded it and the mixer.
I usually do mixes in 24/44.1 and I'm using dithering when exporting to
a 16-bit stereo file. The artifacts, especially stronger esses are more
audible without dithering and it should really not matter.
I'm using Mixbus (Ardour) for exporting the wav files to 16-bit waves
and Audacity for converting the file to a MP3. If I upload a lossless
file to soundcloud, then the same problem comes there when Soundcloud
converts it to it's player.
Unfortunately, I don't have a raw 24/44.1 file and a MP3 I legally can
post right now. But I hope that someone have suggestions about how to
make the best possible MP3 files from a wave file. :-)
Thanks,
Jostein
On Tue, April 2, 2013 6:29 am, Florian Paul Schmidt wrote:
> On 04/02/2013 02:35 PM, Peder Hedlund wrote:
>> The same was true for a Stradivarius and a cheap beginners violin,
>> though IIRC the violin player was correct.
There is a difference between electric guitars (solid body) and acoustic
instruments. Though certainly good PUs make a difference. (noisy pots
don't help either)
> I found that for classical guitars it's significantly higher though: In
> the 1500-2500 euro range (I only found this out because I got the chance
> to play some really nice classicals). A guitar student of mine likes my
> 250 euro yamaha classical guitar better than either the Hans Herb I had
> for a while or the Bertrand Martin I play at the moment (they were lent
I actually like the sound of the guitars I have made in Quebec that are
sub $500 over some more expensive acoustics. They have a very live sound,
I think because there is a very thin coat of finish. I have both classical
and folk instruments. I have a more expensive Yamaha 12 string that sounds
no better than the $100 El Degas plywood model I had when I was 16. (which
was very playable and sounded ok, but was picked as the only good one out
of a rack of 10 or more)
The big difference is that my cheaper guitars are not finished as well as
the more costly guitars... things like nut and fret ends not rounded as
nice etc.
--
Len Ovens
www.OvenWerks.net
"Impedances
Mic in:
3.4 kilohms
Channel insert return:
5 kilohms
All other inputs:
20 kilohms or greater" -
http://www.mackie.com/products/802vlz3/pdf/802VLZ3_OM.pdf
"Nominal Output Level
PHONES/LINE OUT jack:
-14 dBu
Output Impedance
PHONES/LINE OUT jack:
22 ohms" -
http://lib.roland.co.jp/support/en/manuals/res/62121932/BR-80_e04_W.pdf
This is as wanted, a low impedance output -> high impedance input, since
the mixer and the BR-40 do work, it must be the cable.
Unlikely that the cable is to long, but many jacks tend not to fit to
many sockets. Even if it should be the correct cable and even if it's
not broken, it could be, that the jack must not go completely plugged
in. I experienced often that 1/4 jacks have different construction
forms, so that they don't fit to each other.
-------- Forwarded Message --------
From: Ralf Mardorf
To: linux-audio-user
Subject: Re: [LAU] S/PDIF in with ESI Juli@ and JACK
Date: Wed, 03 Apr 2013 22:20:50 +0200
On Wed, 2013-04-03 at 13:44 -0600, Daniel Worth wrote:
>
>
> My analogue ins and outs work fine, but only 2 ins show up in
> JACK, even though I'd imagine I'd get one more for my S/PDIF
> in.
>
>
>
>
> S/PDIF on these cards is only two channels in/out. These are not ADAT
> connections.
Yes but the OP at least should get the 2 analog and the 2 SPDIF
channels, assumed the card is capable of using analog and SPDIF at the
same time ;).
Is the driver for this card ok?
For my ICE1712
^^
stereo cards jack does show 12 capture and 10 playback channels.
Hello all,
I am giving Ardour 3 a try. This is a wonderful piece of software.
I just note the following problem: I create a studio under Claudia, add
Ardour as an app, and then just try to record midi data from my keyboard
on a midi track named 'Synth'.
The clip does appear and I can see the notes getting drawed on the
track, but in the end I get the following message:
[ERROR]: cannot open MIDI file
/home/user/ladish-projects/Ardour3_7174/interchange/Ardour3_7174/midifiles/Synth-1.mid
for write
and the clip disappears.
The midifile directory does exist but the correct path is:
/home/user/ladish-projects/Ardour3_7174/interchange/Ardour3/midifiles/
I don't know is the directory is created by Ardour, or Claudia or
Ladish. But it may also be that I am doing something wrong. I tried to
redefine the file locations in the Session>Properties but this did not
solve the problem.
Any idea?
Thanks in advance,
Yvonnick
Hi,
what abx tester are you using? (if any)
I am trying to build abx tester from http://phintsan.kapsi.fi/abx.html
with no success.
debian testing here, amd64 release.
/r
Hi,
This is a 2-part question:
I'm playing in a little duo and we are using a Boss BR-80 which
provides us with the backing-band duties on stage. The problem here is
that it's only one output (Line/Phones - 22 ohms) that gives way to low
output for line level, are someone aware of a decent pre amp (DIY is ok)
that can be plugged into line input level in a mixer?
The second question is about a backing-band software: In addition to the
main output, it's nice when everybody can have his own mix (mono is ok).
I'm not aware of a good SW based backing-band solution for Linux, are
you? Right now, I can only think about something like playing a
multi-track from the command line, IE. SOX (one track for each
monitoring plus one or two for mixer input). Programs like Audacity and
Ardour can't be used here, they take to long time to load and are also a
gigantic overkill for this usage. The song need to be triggered (IE.
with help from a script) from the space bar by a finger or foot, and the
next song must be ready for playing when the current one is finished. I
will not have any problem to fix the command line/script solution by myself.
Any thoughts or solutions?
Jostein
Hi folks!
I'm a new Linux audio convert.
I'm in the process of building my system right now!
I'd like some suggestions about usb or firewire audio interfaces.
First of all, I'll tell you what my ideal setup would be.
I'd like more than two simultaneous analog inputs. I'd like midi
input, and even hardware controller capabilities.
I know that some units have this all built in to one. I'm very
intrigued by the Tascam 1884, an older firewire unit which has 8
analog inputs with preamps. It also offers midi i/o and works as a
midi control surface in a Mackie Emulation mode. I don't know if it
has any linux support. This would be the dream, but I have to start
somewhere.
I have available to me a Tascam us122 which I could start with if it
is supported.
I have looked at a few pages detailing supported audio devices, but
they seemed a little older so I thought I should come to the list to
get up to date info!
In any case, your suggestions are welcome!
Thanks!
Rusty
On Tue, April 2, 2013 7:06 pm, Ralf Mardorf wrote:
> On Tue, 2013-04-02 at 18:16 -0700, Len Ovens wrote:
>> RME has no choice really. If they want their gear to be used to make
>> bluray sound tracks, they have to support 192k. This is the
>> certification
>> needed by equipment for that use. True, the sound is not any better than
>> if the studio used 48k and resampled the finished product to 192k (maybe
>> worse). but this is not about sound quality or RME doing marketing... it
>> is Hollywood doing the marketing...
>
> People are convinced it does sound better,
"People" are convinced of many things. The list is rather long and
probably includes most of what is wrong in this world these days... or for
the past 10000 years for that matter. (the 10000 number is drawn from thin
air and is in no way related to any understanding I have of our species
origin)
> http://forum.dvdtalk.com/hd-talk/578983-more-96-192khz-blurays-please.html
> , I don't think they are all wrong. Perhaps the players really do sound
> better at 192 KHz, because of better matching filters or what ever, just
I believe the new DVDs include other audio differences besides just sample
rate/depth. Multi-channel (surround) for one.. combining that to stereo
would have some sonic difference as well even if A/Bing on stereo only.
Setting the Bluray players up to play blueray disks 1 or 2 db louder would
be enough to make them sound better... Just remixing for surround is a
remix and the mix will probably be "just different". That difference is
"Fresh" and new to the listener. People who buy a new system also have a
new set of speakers generally, to handle the extra channels if nothing
else. Too many new/different things for a good test.
Testing using a blueray player would not be the way to go, rather testing
each of the different tech changes from one to the other with known
program material at known levels.
--
Len Ovens
www.OvenWerks.net
