LinuxMusicians

Musinux wrote: 2. If you have this running, and if you're brave enough, see https://www.ludd.ltu.se/~torger/brutefir.html , which let's you try eq changes in realtime (and much more)

CrocoDuck wrote:

Musinux wrote: 2. If you have this running, and if you're brave enough, see https://www.ludd.ltu.se/~torger/brutefir.html , which let's you try eq changes in realtime (and much more)

That's pretty interesting! I did not know about that program. I reckon that if you export the impulse response of your room from REW, if it isn't too long, you can calculate an inverse of it and use that program to convolve it to your audio signals, thus (theoretically) compensating perfectly for your room acoustics.

Musinux wrote:1. See http://www.jackaudio.org/faq/routing_alsa.html . Change your .asoundrc to the jack-bridge and REW can "see" your jackd.

2. If you have this running, and if you're brave enough, see https://www.ludd.ltu.se/~torger/brutefir.html , which let's you try eq changes in realtime (and much more)

# ------------------------------------------------------
# Custom asoundrc file for use with snd-aloop and JACK
#
# use it like this:
# env JACK_SAMPLE_RATE=44100 JACK_PERIOD_SIZE=1024 alsa_in (...)
#

# ------------------------------------------------------
# playback device
pcm.aloopPlayback {
  type dmix
  ipc_key 1
  ipc_key_add_uid true
  slave {
    pcm "hw:Loopback,0,0"
    format S32_LE
    rate {
      @func igetenv
      vars [ JACK_SAMPLE_RATE ]
      default 44100
    }
    period_size {
      @func igetenv
      vars [ JACK_PERIOD_SIZE ]
      default 1024
    }
    buffer_size 4096
  }
}

# capture device
pcm.aloopCapture {
  type dsnoop
  ipc_key 2
  ipc_key_add_uid true
  slave {
    pcm "hw:Loopback,0,1"
    format S32_LE
    rate {
      @func igetenv
      vars [ JACK_SAMPLE_RATE ]
      default 44100
    }
    period_size {
      @func igetenv
      vars [ JACK_PERIOD_SIZE ]
      default 1024
    }
    buffer_size 4096
  }
}

# duplex device
pcm.aloopDuplex {
  type asym
  playback.pcm "aloopPlayback"
  capture.pcm "aloopCapture"
}

# ------------------------------------------------------
# default device
pcm.!default {
  type plug
  slave.pcm "aloopDuplex"
}

# ------------------------------------------------------
# alsa_in -j alsa_in -dcloop -q 1
pcm.cloop {
  type dsnoop
  ipc_key 3
  ipc_key_add_uid true
  slave {
    pcm "hw:Loopback,1,0"
    channels 2
    format S32_LE
    rate {
      @func igetenv
      vars [ JACK_SAMPLE_RATE ]
      default 44100
    }
    period_size {
      @func igetenv
      vars [ JACK_PERIOD_SIZE ]
      default 1024
    }
    buffer_size 32768
  }
}

# ------------------------------------------------------
# alsa_out -j alsa_out -dploop -q 1
pcm.ploop {
  type plug
  slave.pcm "hw:Loopback,1,1"
}

lilith wrote:As Room Equation Wizard does not use Jack, is there any way to use REW with a software Equalizer plugin to test the settings, i.e. if the signal becomes more linear with EQ on?

CrocoDuck wrote:

lilith wrote:As Room Equation Wizard does not use Jack, is there any way to use REW with a software Equalizer plugin to test the settings, i.e. if the signal becomes more linear with EQ on?

Just for curiosity, I am not very familiar with REQ, but what can you export from REW? Can you export the whole impulse response in some way?

lilith wrote:

CrocoDuck wrote:

lilith wrote:As Room Equation Wizard does not use Jack, is there any way to use REW with a software Equalizer plugin to test the settings, i.e. if the signal becomes more linear with EQ on?

Just for curiosity, I am not very familiar with REQ, but what can you export from REW? Can you export the whole impulse response in some way?

You can export EQ settings as a file based on your measurement for various hardware or plugins (e.g. miniDSP). I was just interested in equalizing three frequencies on which I have have strong room modes (~40, 50 and 120 Hz). You can also show the correction settings as a table (frequency, Q-factor, damping) which you can use for any EQ (plugin or hardware) by setting the parameters by hand.

CrocoDuck wrote:

lilith wrote:

CrocoDuck wrote:
Just for curiosity, I am not very familiar with REQ, but what can you export from REW? Can you export the whole impulse response in some way?

You can export EQ settings as a file based on your measurement for various hardware or plugins (e.g. miniDSP). I was just interested in equalizing three frequencies on which I have have strong room modes (~40, 50 and 120 Hz). You can also show the correction settings as a table (frequency, Q-factor, damping) which you can use for any EQ (plugin or hardware) by setting the parameters by hand.

The feature list suggests that it is possible to export the measured impulse response as WAV. If you manage to do that, it would be fun to try to invert it and use the inverse in a convolution reverb. Octave can be used for that pretty easily. This should give you the highest degree of compensation for the transmission line you measured. This means that, if you put your head where the microphone was during the measurement, then the compensation will be not optimal as your Head Related Transfer Functions are not the mic response, but maybe it will work OK anyway, especially at low frequency I think.

If you have time to spare, try to give us your impulse response, if you figure out how to do it. I can try to calculate the inverse for you and we can give it a spin, and compare it with the EQ solution.

lilith wrote:
That sounds interesting. What kind of convolution plugin can I use for that in Linux? I'll check how to do this in REW. Anyway ... My monitors (Neumann KH80) have in in-built room correction, however I'm still waiting for the control app which is always shifted. It should come out in summer 2018.

lilith wrote:[
What do I have to inlcude here that REW sees the input channel? BruteFIR looks pretty complicated, thanks for the tip.

sudo modprobe snd_aloop

options snd_aloop pcm_substreams=2 index=4

# playback devices
aplay -l 
# recording devices
arecord -l
# for advanced devices like hdmi
aplay -L
arecord -L

pcm.!default {
    type plug
    slave { pcm "rawjack" }
}
pcm.rawjack {
    type jack
    playback_ports {
        0 system:playback_1
        1 system:playback_2
    }
    capture_ports {
        0 system:capture_1
        1 system:capture_2
    }
}

pcm.jack {
    type plug
    slave { pcm "rawjack" }
    hint {
 	description "JACK Audio Connection Kit"
    }
}

Musinux wrote:

lilith wrote:[
What do I have to inlcude here that REW sees the input channel? BruteFIR looks pretty complicated, thanks for the tip.

Although I'm pretty sure you don't want to do it that way:


For this asoundrc, you have to load the snd-aloop module with

Code: Select all

sudo modprobe snd_aloop

or fill /etc/modprobe.d/sound.conf with

Code: Select all

options snd_aloop pcm_substreams=2 index=4

you can leave out the pcm_substreams and index options, but this is needed i.e. for consistent numbering of soundcards after reboots...

You have to point the !default device in .asoundrc (see below) to the loopback input device.

check with

Code: Select all

# playback devices
aplay -l 
# recording devices
arecord -l
# for advanced devices like hdmi
aplay -L
arecord -L

You'd have to route the physical in/outs to the loopback and out of it back to the physical in/outs.


To avoid timing issues and the unforseeable latency of alsa loopback in general, try it with the rather simple approach from the jack faq:

Code: Select all

pcm.!default {
    type plug
    slave { pcm "rawjack" }
}
pcm.rawjack {
    type jack
    playback_ports {
        0 system:playback_1
        1 system:playback_2
    }
    capture_ports {
        0 system:capture_1
        1 system:capture_2
    }
}

pcm.jack {
    type plug
    slave { pcm "rawjack" }
    hint {
 	description "JACK Audio Connection Kit"
    }
}

The !default entry ensures that after saving the .asoundrc everything uses the alsa->jack bridge.
Depending on your interface and setup, you only need to change the ports (system:playback_1...) to your channels.
Then in REW you just select the "default" device and that should be it.

If this doesn't work, you're probably missing some package named alsa-plugins or alsa-jack.

I know the faq mentions something about a more robust approach using the loopback method, this can be true for studio or recording stuff, but REW is "dumb" picking alsa devices, sample rates and such.

Btw, I have to use jack2 with REW when measuring since jack1 will kick REW because of timing issues.

pcm.!default {
    type plug
    slave { pcm "rawjack" }
}
pcm.rawjack {
    type jack
    playback_ports {
        0 system:playback_1
        1 system:playback_2
    }
    capture_ports {
        0 system:capture_1
        1 system:capture_2
    }
}

pcm.jack {
    type plug
    slave { pcm "rawjack" }
    hint {
 	description "JACK Audio Connection Kit"
    }
}

lilith wrote: is enough? I'll give it a trial as soon I find some time and do some reading first.