On Wed, 2011-10-19 at 19:00 +0000,
linux-audio-user-request(a)lists.linuxaudio.org wrote:
> Message: 21
> Date: Wed, 19 Oct 2011 17:09:05 +0100
> From: Folderol <folderol(a)ukfsn.org>
> Subject: Re: [LAU] Top DSP plugins?
> To: linux-audio-user(a)lists.linuxaudio.org
> Message-ID: <20111019170905.35d0425b@debian>
> Content-Type: text/plain; charset=US-ASCII
>
> On Wed, 19 Oct 2011 08:39:32 +0200
> Jeremy Jongepier <autostatic(a)gmail.com> wrote:
>
> > On 10/18/2011 07:45 PM, S. Massy wrote:
> > > - Finally, the only thing which seems to have been left out by
> others is
> > > the TAP scaling limiter which is great for final limiting if
> handled
> > > gently.
> > >
> >
> > Oh yeah true! Most of the time the LADSPA Fast Look Ahead Limiter
> is
> > sitting in my Master bus.
> >
> > Best,
> >
> > Jeremy
>
> Forgot about that one! I use it quite a lot as a final 'haircut'.
> Yoshimi can
> produce some quite high random peaks, and judicious use of this can
> often give
> me an extra 3-4dB headroom.
>
> --
> Will J Godfrey
If a special synth sound or e.g. an effect like a phaser should cause
such peaks, I would use a compressor for the channel(s) not a limiter
for the master bus.
IMO a limiter can be used the way Fons was suggesting, to protect
against a maximum credible accident and yes, than it should be used for
the master bus. But my philosophy differs. Compress critical sound
material for the channels, if needed. Good usage of EQs very often could
help, so that there isn't the need to compress those signals.
When we don't had computers and analog equipment was expensive, we need
to find other solutions. Today there are some easy solutions, but we
risk that this over-processing became audible.
Again, the usage Fons was suggesting is uncritical, but using a limiter
instead of a compressor, for often repeating peaks isn't good, since the
limiter would cause distortion.
2 cents,
Ralf
On Wed, 2011-10-19 at 19:00 +0000,
linux-audio-user-request(a)lists.linuxaudio.org wrote:
> Message: 7
> Date: Wed, 19 Oct 2011 09:11:52 -0400
> From: Brett McCoy <idragosani(a)gmail.com>
> Subject: Re: [LAU] Top DSP plugins?
> To: Colin Fletcher <colin.m.fletcher(a)googlemail.com>
> Cc: linux-audio-user(a)lists.linuxaudio.org
> Message-ID:
> <CAPNMgw5eQkS6NBnGHtZw38VvgCjtJBdtm6
> +DBG6O6WgsyMe2Yw(a)mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> On Wed, Oct 19, 2011 at 9:07 AM, Colin Fletcher
> <colin.m.fletcher(a)googlemail.com> wrote:
> > On 19/10/11 00:38, Ralf Mardorf wrote:
> >
> >> Jackconvolver is very good regarding to it's sound quality, but
> AFAIK
> >> there's no usable GUI for real-time control...
> >
> > Have you seen the "IR: LV2 Convolution Reverb" by Tom Szilagyi?
> > (http://factorial.hu/plugins/lv2/ir). It's an LV2 wrapper for
> > zita-convolver: it works well for me.
>
> Yes, it's been mentioned a couple of times. I LOVE the plugin and use
> it extensively. It's great for mixing orchestral music. There is also
> a standalone GUI for jconvolver called jcgui --
> http://jcgui.sourceforge.net/
Sometimes I'm using jcgui, but for example, one of the issue is, that it
does use the same waveform for both stereo-channels.
>
>
On Wed, 2011-10-19 at 19:00 +0000,
linux-audio-user-request(a)lists.linuxaudio.org wrote:
> Message: 5
> Date: Wed, 19 Oct 2011 14:07:46 +0100
> From: Colin Fletcher <colin.m.fletcher(a)googlemail.com>
> Subject: Re: [LAU] Top DSP plugins?
> To: linux-audio-user(a)lists.linuxaudio.org
> Message-ID: <4E9ECBA2.1010706(a)googlemail.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> On 19/10/11 00:38, Ralf Mardorf wrote:
>
> > Jackconvolver is very good regarding to it's sound quality, but
> AFAIK
> > there's no usable GUI for real-time control...
>
> Have you seen the "IR: LV2 Convolution Reverb" by Tom Szilagyi?
> (http://factorial.hu/plugins/lv2/ir). It's an LV2 wrapper for
> zita-convolver: it works well for me.
>
> Colin.
Wow, thank you Colin.
I hope it ships with the bookmarked waveforms. Lexicon 480l, I mentioned
it in my posting and I've forgotten to mention the PCM 70, I own some
Yamaha SPX myself. The GUI does look promising.
Cooooooooooooool :)
Ralf
On Wed, 2011-10-19 at 12:00 +0000,
linux-audio-user-request(a)lists.linuxaudio.org wrote:
> Message: 19
> Date: Wed, 19 Oct 2011 11:23:09 +0200
> From: Jeremy Jongepier <jeremy(a)autostatic.com>
> Subject: Re: [LAU] Top DSP plugins?
> To: linux-audio-user(a)lists.linuxaudio.org
> Message-ID: <4E9E96FD.8070908(a)autostatic.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> On 10/19/2011 10:52 AM, Ralf Mardorf wrote:
> > +1, regarding to Jamin is good and it does work without pumping.
>
> I'm not so fond of JAMin. It has flaws and consumes way too much CPU.
> About a year ago Patrick Shirkey mentioned that there was no other
> JACK
> application that attempts to provide a complete mastering chain. But
> that was about a year ago, at the moment it is perfectly possible to
> create a similar tool chain with the help of plug-ins. I prefer
> plug-ins
> then, more flexible.
>
> Best,
>
> Jeremy
Yes, it does consume too much CPU resources. Do we have a similar
three-band compressor? I don't need the limiter and the EQ.
- Ralf
>
>
Hello,
I've been trying to find a way to create an alsa to jack bridge. I found
instructions at the following URL:
http://alsa.opensrc.org/Jack_and_Loopback_device_as_Alsa-to-Jack_bridge
I compiled the aloop module and it works fine, but when I try setting up
the asoundrc as described, and use the device, I get.
ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave
aplay: main:654: audio open error: Invalid argument
Does anyone know what this uninformative error mean? Has anybody ever
had luck with a similar setup?
Thanks.
Cheers,
S.M.
--
Hi
I'm teaching a course in electronic music, and one of the subjects I'd
like to cover is what compression does to the music. I googled a bit abd
found this:
http://www.youtube.com/watch?v=u5gdwpPrv_8
Basically it's a matter of loading the original + the mp3 encoded
version of the same track, inverting the phase in one of the two clips
and listening to the artifacts.
I did some tests, and the results are scary. Lame (128 kbps)and oggenc
(q=3) are different but both horrible, the artifacts are very ugly and
distorted and are as loud as -19 dB!. My favorite mp3 encoder, gogo, has
another strange result: The artifacts sound almost like the mp3, which
should mean it changes the audio much more! However I don't really hear
that much difference between lame and gogo encoded files...
This got me thinking if this is even a realistic test. Assume for
instance the encoder introduces a simple, constant delay in the encoded
audio. This will result in a lot of sound slipping through the
invert-the-phase-of-one-of-the-signals test. Although it could be said
it alters the audio dramatically, when aligning the files and comparing
them sample for sample, it has no impact on the perceived quality of the
encoded audio. I didn't fiddle with delaying the gogo encoded file, though.
My question is: is this really a fair way to judge the artifacts
introduced by encoding?
--
Atte
http://atte.dkhttp://modlys.dk
> Message: 1
> Date: Tue, 18 Oct 2011 13:59:18 +0200
> From: "Jostein Chr. Andersen" <jostein(a)vait.se>
> Subject: Re: [LAU] Top DSP plugins?
> To: linux-audio-user(a)lists.linuxaudio.org
> [snip] *Mainly Fons and LinuxDSP stuff [snip]
REVERBS IMO are a weak point for any OS.
If possible avoid using any Linux reverb, better use some stand alone
19" stuff. But if you need to use a Linux reverb the best choices are
Jackconvolver for impulse response and LinuxDSP for an algorithm reverb.
Jackconvolver is very good regarding to it's sound quality, but AFAIK
there's no usable GUI for real-time control, while LinuxDSP has a usable
GUI, but it still has limited abilities. Don't expect a Rev-1/7 or 480L
or even a SPX 90 II. I suspect there are a lot of low cost reverbs,
perhaps from Alesis, Behringer some other vendors and even Lexicon, that
are easier to use than a Linux reverb. Perhaps the Arts reverb will run
under wine, dunno, this VST anyway isn't that much better. If I wouldn't
have a reverb, I would take a look at Alesis or Behringer, as long as I
don't have the money for something better.
2 Cents,
Ralf
On Tue, 2011-10-18 at 22:56 +0000,
linux-audio-user-request(a)lists.linuxaudio.org wrote:
> Message: 22
> Date: Tue, 18 Oct 2011 21:10:19 +0100
> From: Harry van Haaren <harryhaaren(a)gmail.com>
> Subject: Re: [LAU] Top DSP plugins?
> To: f.rech(a)yahoo.fr
> Cc: linux-audio-user(a)lists.linuxaudio.org
> Message-ID:
> <CAKudYbO7Vaheir9zJ2iDMvNx17M1Tzu2jtv7a2j71=VrQVp8GA(a)mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> On Tue, Oct 18, 2011 at 6:08 PM, fred <f.rech(a)yahoo.fr> wrote:
>
> > it is that easy : tempo is maths !!
http://en.wikipedia.org/wiki/Dyscalculia
Some people are geniuses, still they could suffer from dyscalculia.
2 cents from a dyslexic :p,
Ralf