Rustom Mody:
> On Thu, Feb 2, 2012 at 7:41 PM, Daniel Worth <pipemanmusic(a)gmail.com>
> wrote:
>
>>
>> This is easily solved with qjacktl's patchbay. It's clients like
>> this that
>> it was designed for.
>>
>> http://www.rncbc.org/drupal/node/76
>>
>
> This looks very useful --- just what I was asking for.
>
> However...
>
> When I tried this a few days ago, firefox showed as
> alsa-jack.rawjackC(somenumber)
> and audacity showed as portaudio(something)
>
I'm not sure if I have followed all of the discussion. But if your
problem is to record firefox, it's not very difficult:
1. Start jack
2. Start pulseaudio using jack sinks:
$ pulseaudio -L module-jack-sink -L module-jack-source
[Automatically: Put "'sleep 5 &&
/home/kjetil/bin/startpulseaudio.sh' &" under
qjackctl->options->"execute scripts after startup",
where "startpulseaudio.sh" is a script containing "pulseaudio -L
module-jack-sink -L module-jack-source"]
3. Start firefox
4. Play something
5. Record from jack, for instance using jack_capture, like this:
$ jack_capture
(no options!)
On Thu, Feb 2, 2012 at 7:41 PM, Daniel Worth <pipemanmusic(a)gmail.com> wrote:
>
> This is easily solved with qjacktl's patchbay. It's clients like this that
> it was designed for.
>
> http://www.rncbc.org/drupal/node/76
>
This looks very useful --- just what I was asking for.
However...
When I tried this a few days ago, firefox showed as
alsa-jack.rawjackC(somenumber)
and audacity showed as portaudio(something)
Now both are showing as alsa-jack.rawjacksomething.
My .asoundrc file (which I dont pretend to understand) is thusly:
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"
}
}
pcm.!default {
type plug
slave { pcm "rawjack" }
}
Hi
I have a RME HDSPM pci card. it has MADI in and MADI out connected to
an RME ADI-628. The outputs are working well, but I cannot get
anything in the inputs. The inputs appear in the RME ADI-648, but I
can't see them in a jack app or with jack_meter. I have set the level
to 100% to all channels in 'amixer' (which I believe are input and
output joined). I'm I missing something ? What's the best way to test
this ?
best,
Miguel Negrão
[re-arranged top-posted]
On 01/10/2012 02:59 PM, Vytautas Jancauskas wrote:
> On Tue, Jan 10, 2012 at 2:47 PM, Robin Gareus <robin(a)gareus.org> wrote:
>
>> Hi *,
>>
>> Is there a command-line tool akin to `sndfile-spectrogram` that
>> generates an image (preferably PNG) of an audio wave-form?
>>
>> I found endless GUI apps, but command-line tools are scarce.
>>
>> TIA
>> robin
>>
> It's fairly trivial to write one your own with python using scipy.
> Should be in the ballpark of 20 lines or so.
Thanks, indeed. Python's also the only solution I found:
https://github.com/endolith/freesound-thumbnailer/blob/master/wav2png.py
It's also pretty straight-forward in C; but before whipping up
`sndfile-waveform`, I'd thought I ask around.
robin
Am 02.02.2012 04:12, schrieb Ross Hamblin:
> On 02/02/12 15:56, mn0 wrote:
>> IRQ fiddling is fun. Does anyone want to employ a perfectly trained pci
>> card dis- and re-mounter?
>> I can also restore BIOSes with corrupted checksums...
>> I don't know what to do... one of the cards' interrupts is magically
>> attracted by hdd controllers, either it's ata and usb or it's ide.
>> BIOS options are totally pointless, yet changing something but in an
>> unforeseeable manner.
>> I could disable the IDE controller if there wasn't the os on an ide hdd.
>> This would also be a stupid approach.
>> I'll try again in a few hours...
>> /mn0
>>
>>
>
> I probably missed something as I lost track with the split threads, but
> did you check your interrupts in linux to see whether there are any free
> IRQs? If there are free IRQs you could try using the BIOS option to
> reassign the card to a free interrupt. If Linux is running in PIC
> instead of APIC mode there will be way less interrupts to choose from -
> if is APIC then cat /proc/interrrupts will show IRQs above 24 (IIRC)
> otherwise you could look at enabling APIC which should allow the card to
> take a higher and hopefully free IRQ.
> Sorry for the noise if this was already covered.
>
> HTH
> Ross.
>
Not covered, yet.
APIC is enabled, but thanks for "cat /proc/interrrupts", makes it a
little easier.
There are indeed free interrupts/numbers not listed in /proc interrupts.
Only my BIOS settings are ignored when trying to assign certain devices
to IRQs. The BIOS options are also the wrong way: I can assign an IRQ#
to one or a group of devices... one of the devices is always grouped
with hdd. So I could assign a different IRQ# to the group, but don't
split it, to have the card on a single IRQ.
/mn0
>
> Message: 23
> Date: Wed, 1 Feb 2012 16:31:28 -0500
> From: Joe Hartley <jh(a)brainiac.com>
> Subject: Re: [LAU] Multiple ICE1712 cards setup with jack2
> To: linux-audio-user(a)lists.linuxaudio.org
> Message-ID: <20120201163128.7f5ca69bcd62e4fb3dae49ce(a)brainiac.com>
> Content-Type: text/plain; charset=US-ASCII
>
> On Wed, 01 Feb 2012 21:48:16 +0100
> mn0 <mn0(a)fukked-up.de> wrote:
>
>>> Hope this helps, or at least gives you a starting point for troubleshooting.
>> Oh yes, I found the shared IRQs and I know for certain, it's possible to
>> use jack 1.9.7 with .asoundrc and two ICE1712.
>
> While I agree that the Jack: JackGraphManager::AssertPort error would seem
> to be some sort of pointer issue,
For this specific issue, please test with latest SVN (1.9.9)
> odd things happen when the hardware
> configuration is less-than-stable. Sharing an IRQ with both a SATA and
> a USB controller sets up a situation where I can odd things happening.
>
> Changing slots is one wat of getting a different IRQ; your motherboard
> may also have a way of setting or reserving IRQs in its BIOS. I recall
> having to try a few different slots before the cards got unique IRQs.
>
> I did some searching, and found that you're not the only one with a
> similar issue. This user had different hardware but very similar
> errors: http://ubuntuforums.org/showthread.php?t=1580566
>
> One last thing to check, based on my searches - the AssertPort error
> has been associated with some plugins. You haven't mentioned a client
> program, so I'm not sure if jack dies before you can start one, or if
> there's a correlation here. If the errors actually happen after you
> start something like Ardour, I'd see what happens if you disable the
> plugins. There are some that will absolutely hammer my system every
> time I try to use them. They might be from the C* set, but I really
> can't remember off the top of my head.
>
Stéphane
Hello Everyone.
I am a mere lurker, sadly no musician, but do really love listening. I
run linux mint and am thinking about treating myself to the above sound
card to improve my listening experience. I'll be running the output to
my Kenwood hifi amp using the phonos I expect.
Is the above sound? as in sensible and workable. I'd rather not get
into manually messing with my distro settings if that is possible. Are
there any gotchas to be considered?
Can I just go ahead, buy, and plug and play?
many thanks for any help offered.
n
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Hello everyone!
I had strange goings-on this morning. Yesterday night, I left some sound
playing, which looked like it had finished allright. But on another console I
saw, that a paused mplayer had some errors and wouldn't unpause again. I quit
it and restarted. No luck, I discovered, that ALSA wasn't working properly.
So I rebooted a couple of times. The first time, no Delta soundcard at all.
I turned the computer off, felt, that the Delta was properly screwed in its
place. Yep. Rebooted, found the soundcard.
But the alsamixer settings were a bit wrong. The S/PDIFF items at thwe
beginning were set to IEC958 in-l and r, although I'd ALWAYS left them sitting
on pcm out. So were the "hw in" elements all set to the same in-l and in-r
device. The internal clock was set to iec958 input and couldn't be changed,
the default rate was set to 96000, which is always 48000.
I rebooted and could changed the internal clock to 48000, I heard a pop,
when I did it. Only after that, were aplay and mplayer -ao alsa,default
showing movement. Before then, they started and paused at 0, unable to move
forward.
Still: No Sound! I can start my jack as well. All the volume levels are
check. ALL, that I can see in the playback view.
In jack I tried using mplayer - with -ao jack) and a direct connection of
in/out ports of the card itself. No luck!
So please, what else can I do, to eliminate possibilities, that would
frighten me.
Warmly yours
Julien
=-=-=-=-=-=-=-=-=-=-=-=-
Such Is Life: Very Intensely Adorable;
Free And Jubilating Amazement Revels, Dancing On - FLOWERS!
====== Find my music at ======
http://juliencoder.de/nama/music.html
.....................................
"If you live to be 100, I hope I live to be 100 minus 1 day,
so I never have to live without you." (Winnie the Pooh)
Concerning laptops with firewire I have a Dell 1420 laptop with a Ricoh
firewire controller that runs very well. The controller is:
03:01.0 FireWire (IEEE 1394): Ricoh Co Ltd R5C832 IEEE 1394 Controller (rev
05)
This runs fine with AVLinux 5.0.2 and a Focusrite Saffire Pro24 firewire
interface. I've run it down to 2 ms latency recording a stereo track with
no xruns. Possibly, could do even better but I have not pushed it since
that was adequate for my needs.
The only issue I've had is that occasionally the Focusrite device is not
detected meaning that /dev/fw1 is not created when I turn the Focusrite
device on. This problem is fixed either by power cycling the Focusrite or
rebooting. I'm not sure if this problem is hardware or software, and might
possibly be related to the the Ricoh controller, but there are many other
possibilities. At any rate, once the Focusrite is detected as /dev/fw1 it
runs well.
Of course, my experience may have limited relevance to what Dell currently
offers.