19 Dec
2018
19 Dec
'18
1:37 a.m.
I told some friends to come over tomorrow to make a recording ... and, of
course, I can't get things to work on the computer. Hope someone can set me
straight.
I'm using pulse and audacity ... yeah, I know someone will tell me that
will never work, but it's worked just fine in the past. So, lets continue :)
I'm using a Audiobox USB interface with a single mic. I have tried
different mics and cables, so I'm pretty sure that's not the problem.
In the pavucontrol program the audiobox shows and I select
Analog Stereo Output + Digital Stereo (IEC958) Input
Now, when I bring up audacity I have 3 inputs available. Well, this time
... sometimes I have a list as long as my arm. But, for now lets use the
short list:
HDA Intel ...
Pulse
Default
Both Pulse and Default appear to work as recording devices. Funny, though
is that at other times I've gotten a raft of HDA lines, USB lines, etc.
Doesn't seem to make sense to me.
First off, lets test playback of a track in audacity to phones. Great ...
perfect. Audacity (and everything else on the system)plays though the
device into the phones. I'm happy.
Now, I will record. I get the waveform. I speak into the mic ... and then I
play back ... and it is just awful, scratchy crap.
So, if you've read this far my problem is that I am getting crappy, very
crappy, recording quality with the audiobox interface and audacity.
Suggestions?
I should add this output as well:
arecord -l
**** List of CAPTURE Hardware Devices ****
card 0: USB [AudioBox USB], device 0: USB Audio [USB Audio]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 2: PCH [HDA Intel PCH], device 0: ALC887-VD Analog [ALC887-VD Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 2: PCH [HDA Intel PCH], device 2: ALC887-VD Alt Analog [ALC887-VD Alt
Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 4: U0x46d0x802 [USB Device 0x46d:0x802], device 0: USB Audio [USB
Audio]
Subdevices: 0/1
Subdevice #0: subdevice #0
Thanks to all.
--
**** Listen to my FREE CD at http://www.mellowood.ca/music/cedars ****
Bob van der Poel ** Wynndel, British Columbia, CANADA **
EMAIL: bob(a)mellowood.ca
WWW: http://www.mellowood.ca
19 Dec
19 Dec
2:32 a.m.
On Tue, 18 Dec 2018 16:17:25 -0700, Bob van der Poel wrote:
I told some friends to come over tomorrow to make a
recording ... and,
of course, I can't get things to work on the computer. Hope someone
can set me straight.
What kind of recording? Multi-track? Did you test another app? Did you
test without pulseaudio?
FWIW did you test different USB ports?
What kernel do you use? Assuming a vanilla kernel, with or without
'threadirqs'?
Do you have got an alternative, such as an iPhone, if a few hours
shouldn't be enough to get it working?
Did you test different apps?
Sample rate? Bits? Dithering? Frames/Period? Periods/Buffer?
5:43 p.m.
This looks odd:
On Tue, December 18, 2018 5:17 pm, Bob van der Poel wrote:
In the pavucontrol program the audiobox shows and I
select
Analog Stereo Output + Digital Stereo (IEC958) Input
Any idea why shows digital input and not analog stereo input? I just
found a picture of a PreSonus Audiobox, appears pretty simple, no digital
input at all, at least on the external controls nothing to set sample
rate. Only software controls for setting the sample rate I presume?
Check the sample rate setting of your recording software, and see if you
can find the info for that interface under /proc/asound to check the
currently in use sample rate of the device. I don't know why it would be
incorrect, especially with PulseAudio disabled. Pulse will use sample
rate conversion by default if an application opens a connection at a
different rate than the sample rate Pulse is using as default, but using
pasuspender to launch your audio applications would eliminated Pulse rate
mismatch as a problem.
--
Chris Caudle
5:46 p.m.
"Chris Caudle" <chris(a)chriscaudle.org> writes:
This looks odd:
On Tue, December 18, 2018 5:17 pm, Bob van der Poel wrote:
That could actually be related to the problem: maybe the chipset
supports an IEC958 input and selecting this combination synchronizes the
chipset to the non-existing input.
--
David Kastrup
In the pavucontrol program the audiobox shows and
I select
Analog Stereo Output + Digital Stereo (IEC958) Input
Any idea why shows digital input and not analog stereo input? I just
found a picture of a PreSonus Audiobox, appears pretty simple, no digital
input at all, at least on the external controls nothing to set sample
rate. Only software controls for setting the sample rate I presume?
20 Dec
20 Dec
8:14 p.m.
Okay ... good news! Got it all working. And, even better ... I think I
might know what's going on (maybe not).
Hooked the same stuff up today. Presonus Audiobox USB with 2 mic inputs,
microphone and phones. Now, this time I did one thing first: I turned the
inputs in pulse for the audiobox OFF. Then I launched "pasuspender
audacity". Did a few tests and all works fine. Even changed the input from
44100 to 48000 and back a few times, worked fine on either setting
(assuming that Audacity sets the interface correctly?).
I think the key is to disable the pulse stuff and that a conflict between
the 2 was screwing things up. But, I *thought* that pasuspender was
supposed to do that?
Only one more issue is that I can't adjust the gain in audacity for the
mic. Seems to be a very low signal ... might have to run it though an amp
or mixer anyway, but amplifying it after recording seems to be fine.
Thanks for the suggestions!
On Wed, Dec 19, 2018 at 8:46 AM David Kastrup <dak(a)gnu.org> wrote:
"Chris Caudle" <chris(a)chriscaudle.org>
writes:
--
**** Listen to my FREE CD at http://www.mellowood.ca/music/cedars ****
Bob van der Poel ** Wynndel, British Columbia, CANADA **
EMAIL: bob(a)mellowood.ca
WWW: http://www.mellowood.ca
This looks odd:
On Tue, December 18, 2018 5:17 pm, Bob van der Poel wrote:
That could actually be related to the problem: maybe the chipset
supports an IEC958 input and selecting this combination synchronizes the
chipset to the non-existing input.
--
David Kastrup
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user(a)lists.linuxaudio.org
https://lists.linuxaudio.org/listinfo/linux-audio-user
In the pavucontrol program the audiobox shows and
I select
Analog Stereo Output + Digital Stereo (IEC958) Input
Any idea why shows digital input and not analog stereo input? I just
found a picture of a PreSonus Audiobox, appears pretty simple, no digital
input at all, at least on the external controls nothing to set sample
rate. Only software controls for setting the sample rate I presume?
8:48 p.m.
On Thu, December 20, 2018 12:13 pm, Bob van der Poel wrote:
Now, this time I did one thing first: I turned the
inputs in pulse for the audiobox OFF. Then I launched "pasuspender
audacity"
That is strange that modifying the PulseAudio settings first makes a
difference.
All I have to go on is the man page for pasuspender, which states:
"pasuspender will suspend access to the audio devices"
Good to know that you figured out how to work around it.
Only one more issue is that I can't adjust the
gain in audacity for the
mic.
The interface only has hardware knobs to control gain, Audacity will not
be able to change the gain of the mic preamp.
Or did you mean you wanted to change the gain of the recorded track after
recording?
I don't recall if you mentioned what type of mic you use, the Audiobox
only has a maximum gain of 35dB. Should be more than enough for close
position on instruments with a condenser mic, would not be enough to
record a small chamber group from mid auditorium with ribbon mics. For a
mic relatively closely positioned to your saxophone that should be enough
gain available.
The instrument input actually has more gain available, up to 50dB, so
depending on what you need using the unbalanced input might work better.
--
Chris Caudle
21 Dec
21 Dec
5:19 a.m.
That is strange that modifying the PulseAudio settings first makes a
I sort of figured that.
Or did you mean you wanted to change the gain of the recorded track after
difference.
All I have to go on is the man page for pasuspender, which states:
"pasuspender will suspend access to the audio devices"
Good to know that you figured out how to work around it.
Not saying that one really needs to do this ... just that it worked for me
today. I've not had a chance to try it a 2nd time, as they say "'tis the
season to be weary", but I will next week when things settle down a bit.
Only one more
issue is that I can't adjust the gain in audacity for the
mic.
The interface only has hardware knobs to control gain, Audacity will not
be able to change the gain of the mic preamp.
recording?
That's easy to do and works fine.
I don't recall if you mentioned what type of mic you use, the Audiobox
only has a maximum gain of 35dB. Should be more than
enough for close
position on instruments with a condenser mic, would not be enough to
record a small chamber group from mid auditorium with ribbon mics. For a
mic relatively closely positioned to your saxophone that should be enough
gain available.
I was doing my testing with a cheap dynamic. I'll have a go later with a
condenser. I will want to use my ribbon for sax recording, but at a close
distance. So, it should be fine. And, if not, I can always stick an amp
(well, mixer) into the loop.
The instrument input actually has more gain available, up to 50dB, so
depending on what you need using the unbalanced input
might work better.
That's interesting. I have a 3pin to 1/4 cable laying about. I'll give that
a go and see what happens. Would this effect the quality of the signal?
So much easier, for an acoustic guy like me, to not get involved in all
this recording stuff :)
--
**** Listen to my FREE CD at http://www.mellowood.ca/music/cedars ****
Bob van der Poel ** Wynndel, British Columbia, CANADA **
EMAIL: bob(a)mellowood.ca
WWW: http://www.mellowood.ca
8:57 a.m.
On Thu, 20 Dec 2018 20:18:31 -0700, Bob van der Poel wrote:
On Thu, 20 Dec 2018 12:48:14 -0600, Chris Caudle
wrote:
Yes, it does!
In my experiences it makes a big difference. For connecting gear such as
e.g. a mixing console and 19" effects it usually makes no audible
difference when doing it balanced or unbalanced, but for a dynamic
microphone there are poles apart when doing it one way or the other.
depending on what you need using the unbalanced
input might work
better.
That's interesting. I have a 3pin to 1/4 cable laying about. I'll give
that a go and see what happens. Would this effect the quality of the
signal? So much easier, for an acoustic guy like me, to not get
involved in all
this recording stuff :)
You now better ;), in the end humans waste way too much time by
self-deluding themself. We don't own the time to become an expert for
everything, but we waste less time for some things, once we are experts
for things, that are important to our lives.
5:14 p.m.
On Thu, December 20, 2018 9:18 pm, Bob van der Poel wrote:
The instrument input actually has more gain available,
up to 50dB, so
> depending on what you need using the unbalanced input
> might work better.
That's interesting. I have a 3pin to 1/4 cable
laying about. I'll give
that a go and see what happens. Would this effect the quality of the
signal?
There are a lot of "it depends" caveats regarding any quality changes.
You definitely cannot do that for condenser microphones which need phantom
power, and you lose the noise rejection features of a balanced interface
when you change to the unbalanced interface. For a short microphone
cable the noise rejection features should not make a noticeable
difference.
One thing which can make a difference with ribbon and dynamic mics is the
input impedance of the amplifier. This may not be the place to delve into
the specifics of the circuit theory and math, but the summary is that if
the microphone is designed with the expectation of a typical microphone
input impedance in the 1200 to 2000 Ohms range, the frequency response may
be different, likely with a peak in the frequency response, when used into
an instrument input with a typical 500,000 Ohm input impedance. I love
this stuff, I would pull out some connectors and make adapters with
matching resistors, but spending time on that kind of stuff is one of the
reasons why I do not have good instrument skills, so learning circuit
theory and soldering skills is probably not the way you want to spend your
winter holidays. :)
tl;dr version: try your normal microphone cable and just turn up the gain
knob, you'll probably be fine. If it seems not enough gain, try your
adapter cable and see how it sounds if you can do that in 15 minutes time
and not waste a lot of your day messing with equipment instead of playing
your sax. If you can get your mixer connected in less time than messing
about with adapter cables and listening tests, just do that, you know it
will work.
--
Chris Caudle
10 Jan
10 Jan
4:14 a.m.
Hey, me again. Back at it ... same problems :) But, I _think_ I figured out
the problem ... it's the USB port. If I use the "easy to get at port",
I'm
using a USB3, and it creates the problem. If I use a harder to reach port,
USB2, all works fine. Don't need to use pasuspender, don't need to turn off
pulse audio ports, etc. Just run audacity and it all works.
To recap, I'm using a Presonus Audiobox USB connected to my PC and running
Audacity to do some simple recording. The USB3 port gives me scrambled
recordings ... the USB2 port works fine.
So, do I have a bum port or is this normal?
On Thu, Dec 20, 2018 at 8:18 PM Bob van der Poel <bob(a)mellowood.ca> wrote:
That is strange that modifying the PulseAudio settings first makes a
I sort of figured that.
Or did you mean you wanted to change the gain of the recorded track after
--
**** Listen to my FREE CD at http://www.mellowood.ca/music/cedars ****
Bob van der Poel ** Wynndel, British Columbia, CANADA **
EMAIL: bob(a)mellowood.ca
WWW: http://www.mellowood.ca
difference.
All I have to go on is the man page for pasuspender, which states:
"pasuspender will suspend access to the audio devices"
Good to know that you figured out how to work around it.
Not saying that one really needs to do this ... just that it worked for me
today. I've not had a chance to try it a 2nd time, as they say "'tis the
season to be weary", but I will next week when things settle down a bit.
Only one
more issue is that I can't adjust the gain in audacity for the
mic.
The interface only has hardware knobs to control gain, Audacity will not
be able to change the gain of the mic preamp.
recording?
That's easy to do and works fine.
I don't recall if you mentioned what type of mic you use, the Audiobox
only has a maximum gain of 35dB. Should be more
than enough for close
position on instruments with a condenser mic, would not be enough to
record a small chamber group from mid auditorium with ribbon mics. For a
mic relatively closely positioned to your saxophone that should be enough
gain available.
I was doing my testing with a cheap dynamic. I'll have a go later with a
condenser. I will want to use my ribbon for sax recording, but at a close
distance. So, it should be fine. And, if not, I can always stick an amp
(well, mixer) into the loop.
The instrument input actually has more gain available, up to 50dB, so
depending on what you need using the unbalanced
input might work better.
That's interesting. I have a 3pin to 1/4 cable laying about. I'll give
that a go and see what happens. Would this effect the quality of the signal?
So much easier, for an acoustic guy like me, to not get involved in all
this recording stuff :)
--
**** Listen to my FREE CD at http://www.mellowood.ca/music/cedars ****
Bob van der Poel ** Wynndel, British Columbia, CANADA **
EMAIL: bob(a)mellowood.ca
WWW: http://www.mellowood.ca
4:41 a.m.
On Wed, 9 Jan 2019 19:13:50 -0700, Bob van der Poel wrote:
So, do I have a bum port or is this normal?
It's normal (but that doesn't necessarily meant that the port is not
defect ;). Some USB ports work less good than other. Even while I
sometimes have read claims about USB 3 causing issues, it's unlikely
the used USB version. I had a mobo with USB 2 only and on some USB
ports an audio device (a Presonus and/or Focusrite, I don't remember
which audio device) worked ok, while others ports caused crackled sound
and those USB ports where not defect. On my new mobo I connect my USB 2
Focusrite to an USB 3.0 port. It works without issues.
--
pacman -Q linux{,-rt{-cornflower,,-securityink,-pussytoes}}|cut -d\ -f2
4.20.arch1-1
4.19.13_rt10-0
4.19.10_rt8-0
4.19.8_rt6-0
4.18.16_rt9-1
4:42 a.m.
On Wed, 9 Jan 2019, Bob van der Poel wrote:
Hey, me again. Back at it ... same problems :) But, I
_think_ I figured out the
problem ... it's the USB port. If I use the "easy to get at port", I'm
using a
USB3, and it creates the problem. If I use a harder to reach port, USB2, all
works fine. Don't need to use pasuspender, don't need to turn off pulse audio
ports, etc. Just run audacity and it all works.
To recap, I'm using a Presonus Audiobox USB connected to my PC and running
Audacity to do some simple recording. The USB3 port gives me scrambled recordings
... the USB2 port works fine.
So, do I have a bum port or is this normal?
Totally normal., Most mother boards do this two ways:
- one of your USB ports is probably connected to IRQ16 for legacy
purposes, along with two or three other internal bits.
- even if that is not the case, it may have an internal hub and be
sharing that port with another physical USB plug (with your
mouse?) or even an internal mic or whatever.
I had a laptop (netbook?) with one USB on the left and two one the right.
To get stable sound, I had to use one of the right hand USB ports for
audio and leave the second right hand empty and put a hub on the lefthand
side to use with everything else. The lefthand USB was also used
internally as well as being irq16 while the two on right were from the
same hub (so I could only use one) but had a clear irq.
Also, some MB have bios settings for irq choosing. It may be possible to
allow the MB to asign USB or other irqs or not (or leave certain irq
unassigned). I have found that letting the Linux kernel assign the irq
rather than bios gives a more inteligent assignment. If you have a desktop
with spare slots, I would get a PCIe USB card and use that USB for audio
only... trying different PCIe slots to find the one that works best.
Finally, if you use the rtirq script at boot.... it may seem nice to do:
"usb, snd, whatever" but you may be better doing: "usb2, snd, whatever,
usb" to make sure that your audio IF is actually higher priority than your
mouse (assuming you are using USB2 for audio). I have an old ice1712 in a
PCI slot (yes I found a new MB with 3 PCI slots in it) and so put
"snd_ice, snd_ens, etc" because my ensoniq sound card can not be before
the ice or it interferes with it (and I am only using the ensoniq for it's
MIDI ports).
Some of these tweaks may be less needed if you use an RT kernel, but I am
using a lowlatency kernel and these kind of tweaks allow me to run the
ice1712 with jack at 16/2 and no xruns (so long as pulse is not bridged,
pulse is fine down to 64/2 though... 128 with skype).
--
Len Ovens
www.ovenwerks.net
11:54 a.m.
I don't have a USB3 port on my laptop and only use a Behringer UAC-202 on a laptop
USB2 port. There, Audacity records just fine.
I also have a powered USB2 hub. When I plugged the UAC into the hub, the input signal
level dropped by half plus there was noticeable power hum. Unplugging the hub's power
supply fixed the hum but not the signal level.
So it's possible the USB3 port and/or internal hub behind it has a flakey connection
somewhere. I think Ralf mentioned that.
But glad you can record. The only Presonus I have here is Firewire and we no longer have
anything with Firewire around.
On January 9, 2019 4:13:50 PM HST, Bob van der Poel <bob(a)mellowood.ca> wrote:
Hey, me again. Back at it ... same problems :) But, I
_think_ I
figured out
the problem ... it's the USB port. If I use the "easy to get at port",
I'm
using a USB3, and it creates the problem. If I use a harder to reach
port,
USB2, all works fine. Don't need to use pasuspender, don't need to
turn off
pulse audio ports, etc. Just run audacity and it all works.
To recap, I'm using a Presonus Audiobox USB connected to my PC and
running
Audacity to do some simple recording. The USB3 port gives me scrambled
recordings ... the USB2 port works fine.
So, do I have a bum port or is this normal?
On Thu, Dec 20, 2018 at 8:18 PM Bob van der Poel <bob(a)mellowood.ca>
wrote:
after
Audiobox
with a
That's interesting. I have a 3pin to 1/4 cable laying about. I'll give
--
David W. Jones
gnome(a)hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com
Sent from my Android device with F/LOSS K-9 Mail.
That is strange that modifying the PulseAudio settings first makes a
for me
difference.
All I have to go on is the man page for pasuspender, which states:
"pasuspender will suspend access to the audio devices"
Good to know that you figured out how to work around it.
Not saying that one really needs to do this ... just that it worked today. I've not had a chance to try it a 2nd
time, as they say "'tis
the
season to be weary", but I will next week
when things settle down a
bit.
> > Only one more issue is that I can't adjust the gain in audacity
for
the
> > mic.
>
> The interface only has hardware knobs to control gain, Audacity
will not
be able
to change the gain of the mic preamp.
I sort of figured that.
Or did you mean you wanted to change the gain of the recorded track
recording?
That's easy to do and works fine.
I don't recall if you mentioned what type of mic you use, the > only has a maximum gain of 35dB. Should be
more than enough for
close
> position on instruments with a condenser mic,
would not be enough
to
> record a small chamber group from mid
auditorium with ribbon mics.
For a
> mic relatively closely positioned to your
saxophone that should be
enough
gain
available.
I was doing my testing with a cheap dynamic. I'll have a go later condenser. I will want to use my ribbon for sax
recording, but at a
close
distance. So, it should be fine. And, if not, I
can always stick an
amp
(well, mixer) into the loop.
The instrument input actually has more gain available, up to 50dB,
so
> depending on what you need using the
unbalanced input might work
better.
that a go and see what happens. Would this effect
the quality of the
signal?
So much easier, for an acoustic guy like me, to not get involved in
all
this recording stuff :)
--
**** Listen to my FREE CD at http://www.mellowood.ca/music/cedars
****
Bob van der Poel ** Wynndel, British Columbia,
CANADA **
EMAIL: bob(a)mellowood.ca
WWW: http://www.mellowood.ca
--
**** Listen to my FREE CD at http://www.mellowood.ca/music/cedars ****
Bob van der Poel ** Wynndel, British Columbia, CANADA **
EMAIL: bob(a)mellowood.ca
WWW: http://www.mellowood.ca
1945
days inactive
1967
days old
linux-audio-user@lists.linuxaudio.org
12 comments
6 participants
participants (6)
-
Bob van der Poel -
Chris Caudle -
David Kastrup -
David W. Jones -
Len Ovens -
Ralf Mardorf