16 Jan
2013
16 Jan
'13
4:46 p.m.
Hello
My music server is headless (no graphical environment).
I'm using mpd + JACK to play music.
Now I need some software equalizer (JACK based of course), but running
in daemon mode.
JackEQ is the application I currently installed. It could be run in
daemon mode like that:
$ jackeq -d
But I don't understand how to use it further more:
- how to properly connect it (to mpd in my case) ?
- how to adjust bass and treble ?
Can you suggests any other EQ software based on your practice?
16 Jan
16 Jan
10:45 p.m.
On Wed, January 16, 2013 7:46 am, Stefan Stefanov wrote:
Hello
My music server is headless (no graphical environment).
I'm using mpd + JACK to play music.
Now I need some software equalizer (JACK based of course), but running
in daemon mode.
JackEQ is the application I currently installed. It could be run in
daemon mode like that:
$ jackeq -d
But I don't understand how to use it further more:
- how to properly connect it (to mpd in my case) ?
- how to adjust bass and treble ?
Can you suggests any other EQ software based on your practice?
Nama or ecasound are both set up to do audio chains and can have effects
in them. I think ecasound would be the best for set and forget though, as
it would not need input once running and could be run with & to background
it.
Ecarack looks interesting too. (comes with jackrack and takes a saved
jackrack config and runs it from terminal)
--
Len Ovens
www.OvenWerks.net
17 Jan
17 Jan
12:28 a.m.
Hello Stefan!
I can't suggest a specific text-based EQ, but there is ecasound, which can
host ladspa plugins, so you could use any EQ of your choice (or even more than
one). Adjusting can of course be a little tiresome the first time round,
because you have a lot of parameters to the good EQs. But I suspect, the
effort would be worth it. If you need something even more powerful, and
perhaps a little more convenient in its interface, there is Nama.
then of course you could try to use csound, sc3 or clm (not sure about sc3),
those are fully fledged synthesis languages. If you know one of them: fine. If
not, best leave those as a measure of last resort. :-)
Warm regards
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
11:12 a.m.
Hello and thanks for response
I just installed ecasound, but I cannot understand how to use it for EQ.
My current mpd.conf is configured to use jack:
...
audio_output {
type "jack"
name "JACK Audiolab MDAC"
device "hw:2,0"
format "*:24_3:2"
mixer_type "disabled"
}
...
Here is quote from
http://mpd.wikia.com/wiki/Music_Player_Daemon_Is_There_An_Equalizer
"It is also possible to configure the pipe output and use another application
to process the audio. For example, ecasound can be used together with
ladspa plugins
to provide equalizer and some other features."
So, if I pipe output from mpd to the ecasound then is it possible
to use similar audio settings as with "audio_output" section in mpd.conf?
Could you please point me to the right direction how to use ecasound
to process audio and how to use it as EQ?
2013/1/17 Julien Claassen <julien(a)mail.upb.de>de>:
Hello Stefan!
I can't suggest a specific text-based EQ, but there is ecasound, which can
host ladspa plugins, so you could use any EQ of your choice (or even more
than one). Adjusting can of course be a little tiresome the first time
round, because you have a lot of parameters to the good EQs. But I suspect,
the effort would be worth it. If you need something even more powerful, and
perhaps a little more convenient in its interface, there is Nama.
then of course you could try to use csound, sc3 or clm (not sure about
sc3), those are fully fledged synthesis languages. If you know one of them:
fine. If not, best leave those as a measure of last resort. :-)
Warm regards
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
11:30 a.m.
Hello Stefan!
Ecasound's basic concept is chains. A chain looks like this:
input -> possible processing options -> output
So you can use jack intput and output:
ecasound -f:16,2,44100 -i jack_multi,mpd:out1,mpd:out2 -o \
jack_multi,system:playback_1,system:playback_2
At the end of this line you can add effects. For example a filter. You can
use LADSPA unique IDs or the labels. I prefer IDs, since I always forget upper
and lower case letters. So the GLAME Highpass IIR filter (label: highpass_iir
and ID 1890). It has two parameter cutoff and poles/stages. You can find that
out with analyseplugin from the basic ladspa suite or look it up in your
favourite graphic ladspa host. So at the end of the ecasound line we have:
-eli:1890,50,2
or
-el:highpass_iir,50,2
Adds a highpass filter with 50Hz cutoff frequency and two stages/poles.
My favourite EQ is still Fons Adriaenson's 4-band parametric EQ (1970). I
think the package is called 4-band parametric filter (or just filter). It has
a lot of options, you can try them in your favourite graphical application
first and then copy all the values, that you found. probably easiest for you.
If you need more help ask. - one last note: the -f:16,2,44100 is the
ecasound audio formation option, see the manpage. It's
bitrate,number-of-channels,samplerate. So this is CD-quality. Adapt to your
needs.
Warmly yours
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
9:51 p.m.
Hi Julien
In process of accomplish my task with EQ I discover a problem playing
music with ecasound.
If I execute: ecasound -i test.wav -o
jack_multi,system:playback_1,system:playback_2
then I get this error:
"ERROR: Connecting chainsetup failed: "All audio objects must have a common
... sampling rate; sampling rate of audio object "jack_multi" differs from
... engine rate (48000 <-> 44100); unable to continue."
"
I run jack like this: /usr/bin/jackd -P 85 -d alsa -P hw:2,0 -n 3 -p 256
(test.wav format is 16,2,44100)
Regards
Stefan
2013/1/17 Julien Claassen <julien(a)mail.upb.de>de>:
Hello Stefan!
Ecasound's basic concept is chains. A chain looks like this:
input -> possible processing options -> output
So you can use jack intput and output:
ecasound -f:16,2,44100 -i jack_multi,mpd:out1,mpd:out2 -o \
jack_multi,system:playback_1,system:playback_2
At the end of this line you can add effects. For example a filter. You can
use LADSPA unique IDs or the labels. I prefer IDs, since I always forget
upper and lower case letters. So the GLAME Highpass IIR filter (label:
highpass_iir and ID 1890). It has two parameter cutoff and poles/stages. You
can find that out with analyseplugin from the basic ladspa suite or look it
up in your favourite graphic ladspa host. So at the end of the ecasound line
we have:
-eli:1890,50,2
or
-el:highpass_iir,50,2
Adds a highpass filter with 50Hz cutoff frequency and two stages/poles.
My favourite EQ is still Fons Adriaenson's 4-band parametric EQ (1970). I
think the package is called 4-band parametric filter (or just filter). It
has a lot of options, you can try them in your favourite graphical
application first and then copy all the values, that you found. probably
easiest for you.
If you need more help ask. - one last note: the -f:16,2,44100 is the
ecasound audio formation option, see the manpage. It's
bitrate,number-of-channels,samplerate. So this is CD-quality. Adapt to your
needs.
Warmly yours
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
9:57 p.m.
On Fri, January 18, 2013 7:51 am, Stefan Stefanov wrote:
Hi Julien
In process of accomplish my task with EQ I discover a problem playing
music with ecasound.
If I execute: ecasound -i test.wav -o
jack_multi,system:playback_1,system:playback_2
then I get this error:
"ERROR: Connecting chainsetup failed: "All audio objects must have a
common
... sampling rate; sampling rate of audio object "jack_multi" differs from
... engine rate (48000 <-> 44100); unable to continue."
"
I run jack like this: /usr/bin/jackd -P 85 -d alsa -P hw:2,0 -n 3 -p 256
(test.wav format is 16,2,44100)
Jack is probably started with a sample rate of 48000.
You can of course start jack with a sample rate of 44100 or you can tell
ecasound to resample from 44100 to 48000.
Probably easier to tell jack to use 44100 though.
Regards
Stefan
2013/1/17 Julien Claassen <julien(a)mail.upb.de>de>:
--
Patrick Shirkey
Boost Hardware Ltd
Hello Stefan!
Ecasound's basic concept is chains. A chain looks like this:
input -> possible processing options -> output
So you can use jack intput and output:
ecasound -f:16,2,44100 -i jack_multi,mpd:out1,mpd:out2 -o \
jack_multi,system:playback_1,system:playback_2
At the end of this line you can add effects. For example a filter. You
can
use LADSPA unique IDs or the labels. I prefer IDs, since I always forget
upper and lower case letters. So the GLAME Highpass IIR filter (label:
highpass_iir and ID 1890). It has two parameter cutoff and poles/stages.
You
can find that out with analyseplugin from the basic ladspa suite or look
it
up in your favourite graphic ladspa host. So at the end of the ecasound
line
we have:
-eli:1890,50,2
or
-el:highpass_iir,50,2
Adds a highpass filter with 50Hz cutoff frequency and two
stages/poles.
My favourite EQ is still Fons Adriaenson's 4-band parametric EQ
(1970). I
think the package is called 4-band parametric filter (or just filter).
It
has a lot of options, you can try them in your favourite graphical
application first and then copy all the values, that you found. probably
easiest for you.
If you need more help ask. - one last note: the -f:16,2,44100 is the
ecasound audio formation option, see the manpage. It's
bitrate,number-of-channels,samplerate. So this is CD-quality. Adapt to
your
needs.
Warmly yours
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user(a)lists.linuxaudio.org
http://lists.linuxaudio.org/listinfo/linux-audio-user
9:59 p.m.
Yes, but it is very ugly to change rate in config file for every song I play.
2013/1/17 Patrick Shirkey <pshirkey(a)boosthardware.com>om>:
On Fri, January 18, 2013 7:51 am, Stefan Stefanov wrote:
Hi Julien
In process of accomplish my task with EQ I discover a problem playing
music with ecasound.
If I execute: ecasound -i test.wav -o
jack_multi,system:playback_1,system:playback_2
then I get this error:
"ERROR: Connecting chainsetup failed: "All audio objects must have a
common
... sampling rate; sampling rate of audio object "jack_multi" differs from
... engine rate (48000 <-> 44100); unable to continue."
"
I run jack like this: /usr/bin/jackd -P 85 -d alsa -P hw:2,0 -n 3 -p 256
(test.wav format is 16,2,44100)
Jack is probably started with a sample rate of 48000.
You can of course start jack with a sample rate of 44100 or you can tell
ecasound to resample from 44100 to 48000.
Probably easier to tell jack to use 44100 though.
Regards
Stefan
2013/1/17 Julien Claassen <julien(a)mail.upb.de>de>:
--
Patrick Shirkey
Boost Hardware Ltd
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user(a)lists.linuxaudio.org
http://lists.linuxaudio.org/listinfo/linux-audio-user Hello Stefan!
Ecasound's basic concept is chains. A chain looks like this:
input -> possible processing options -> output
So you can use jack intput and output:
ecasound -f:16,2,44100 -i jack_multi,mpd:out1,mpd:out2 -o \
jack_multi,system:playback_1,system:playback_2
At the end of this line you can add effects. For example a filter. You
can
use LADSPA unique IDs or the labels. I prefer IDs, since I always forget
upper and lower case letters. So the GLAME Highpass IIR filter (label:
highpass_iir and ID 1890). It has two parameter cutoff and poles/stages.
You
can find that out with analyseplugin from the basic ladspa suite or look
it
up in your favourite graphic ladspa host. So at the end of the ecasound
line
we have:
-eli:1890,50,2
or
-el:highpass_iir,50,2
Adds a highpass filter with 50Hz cutoff frequency and two
stages/poles.
My favourite EQ is still Fons Adriaenson's 4-band parametric EQ
(1970). I
think the package is called 4-band parametric filter (or just filter).
It
has a lot of options, you can try them in your favourite graphical
application first and then copy all the values, that you found. probably
easiest for you.
If you need more help ask. - one last note: the -f:16,2,44100 is the
ecasound audio formation option, see the manpage. It's
bitrate,number-of-channels,samplerate. So this is CD-quality. Adapt to
your
needs.
Warmly yours
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user(a)lists.linuxaudio.org
http://lists.linuxaudio.org/listinfo/linux-audio-user
10:06 p.m.
Hello Stefan!
You won't have to change the samplerate for every song you play, since your
setup will be different in the end:
mpd -> ecasound -> mpd or jack
So mpd will read the songs and convert to the samplerate of JACK like any
good music player. So Ecasound wil receive the correct signal. There is no
other way. If that wouldn't work, then you couldn't use mpd through JACK now.
At least not, without reconfiguring. So don't worry. For the test with a file:
1. Check the samplerate of your JACK server. (say it's 44100Hz)
2. use sndfile-resample or resample (from Fon Adriaenson) like this on your
test.wav file:
sndfile-resample -c 0 -to 44100 test.wav resampled.wav
or:
resample --lips --16bit --wav --rate 44100 test.wav resampled.wav
Then you can test Ecasound with this, until you have found your bearings.
Warmly yours
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
10:14 p.m.
On Thu, 17 Jan 2013 21:59:36 +0100, Stefan Stefanov <selectany(a)gmail.com>
wrote:
Yes, but it is very ugly to change rate in config file
for every song I
play.
http://linux.seindal.dk/2004/11/04/resampling-audio-data/
9:40 p.m.
On Wed, 16 Jan 2013 17:46:55 +0200 Stefan Stefanov
<selectany(a)gmail.com> wrote:
My music server is headless (no graphical
environment).
I'm using mpd + JACK to play music.
Now I need some software equalizer (JACK based of course), but running
in daemon mode.
My I suggest the possibility that you are looking at the wrong tools
for your problem?
I think you are not looking for an equalizer. You are looking for
digital room correction (aka DRC) fitting your speakers with their
characteristics into your room with its characteristics and make it
sound right[TM] and without any quirky "characteristics".
The tools for this (as far as I remember, its been a while since I set
mine up):
- aliki to record the impulse-response of your setup
- drc to compute the correction impulse to be folded onto your signal
- jconvolver to actually fold your music signal with the correction
pulse. This will be your work-horse.
- And of course jack to plug it all toghether.
The first two steps would be done on your main machine (but playing the
measurement-sweeps through your actual speakers). jack and jconvolver
would run on your headless music server.
The results are far better then anything you can achieve with an EQ,
regardless how many bands it might have. An EQ can never compensate for
the time-effects of your setup.
Have fun,
Arnold
9:55 p.m.
Hi Arnold
You are right, I would like/prefer as you said to " fitting my
speakers with their
characteristics into my room".
I'll do some researches how exactly to accomplish this.
Regards
Stefan
2013/1/17 Arnold Krille <arnold(a)arnoldarts.de>de>:
On Wed, 16 Jan 2013 17:46:55 +0200 Stefan Stefanov
<selectany(a)gmail.com> wrote:
My music server is headless (no graphical
environment).
I'm using mpd + JACK to play music.
Now I need some software equalizer (JACK based of course), but running
in daemon mode.
My I suggest the possibility that you are looking at the wrong tools
for your problem?
I think you are not looking for an equalizer. You are looking for
digital room correction (aka DRC) fitting your speakers with their
characteristics into your room with its characteristics and make it
sound right[TM] and without any quirky "characteristics".
The tools for this (as far as I remember, its been a while since I set
mine up):
- aliki to record the impulse-response of your setup
- drc to compute the correction impulse to be folded onto your signal
- jconvolver to actually fold your music signal with the correction
pulse. This will be your work-horse.
- And of course jack to plug it all toghether.
The first two steps would be done on your main machine (but playing the
measurement-sweeps through your actual speakers). jack and jconvolver
would run on your headless music server.
The results are far better then anything you can achieve with an EQ,
regardless how many bands it might have. An EQ can never compensate for
the time-effects of your setup.
Have fun,
Arnold
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user(a)lists.linuxaudio.org
http://lists.linuxaudio.org/listinfo/linux-audio-user
18 Jan
18 Jan
12:36 a.m.
On Thu, Jan 17, 2013 at 10:55:27PM +0200, Stefan Stefanov wrote:
You are right, I would like/prefer as you said to
" fitting my
speakers with their
characteristics into my room".
I'll do some researches how exactly to accomplish this.
As Arnold already pointed out, DRC is the tool to do this.
There's nothing else that comes close. DRC has excellent
documentation, also explaining what can be done and what
can't be done. Make sure to read it.
Ciao,
--
FA
A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)
1:04 a.m.
On Fri, 18 Jan 2013 00:36:38 +0100, Fons Adriaensen <fons(a)linuxaudio.org>
wrote:
On Thu, Jan 17, 2013 at 10:55:27PM +0200, Stefan
Stefanov wrote:
A work around of a cheap, resampling, soundcard will fix issues regarding
to the used microphones and pre-amps too?
--
FreeBSD 9.1-RELEASE amd64
You are right, I would like/prefer as you said to
" fitting my
speakers with their
characteristics into my room".
I'll do some researches how exactly to accomplish this.
As Arnold already pointed out, DRC is the tool to do this.
There's nothing else that comes close. DRC has excellent
documentation, also explaining what can be done and what
can't be done. Make sure to read it.
Ciao,
4420
days inactive
4422
days old
linux-audio-user@lists.linuxaudio.org
13 comments
7 participants
participants (7)
-
Arnold Krille -
Fons Adriaensen -
Julien Claassen -
Len Ovens -
Patrick Shirkey -
Ralf Mardorf -
Stefan Stefanov