Linux-audio-user

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20661 discussions

[linux-audio-user] A most bizarre problem.

by Malcolm Baldridge

A few weeks into my admittedly bizarre Jack-based telephone logging + music-on-hold system, I ran into an interesting "bug". But first, the hardware in play: Motherboard: ECS P4IBMS (Pentium 4, i845 chipset) 00:1f.5 Multimedia audio controller: Intel Corp. 82801BA/BAM AC'97 Audio Card: Intel 82801BA-ICH2 Chip: Realtek ALC200/200P rev 0 ALSA-1.0.3b w/1.0.4 drivers + libraries, jackd is from late April 2004 CVS. My jackd invocation line is as follows: jackd -R -d alsa -S -C -p 4096 -M -n 2 -r 11025 I have an AT&T MERLIN phone system in my closet along with my main home server (see above hardware description). I have my PCM (Line out) plugged into the MERLIN system's music-on-hold port, that works well. I have two phone lines hooked up to JD Audio Phone Patch boxes to provide local + remote side level correction (trans-hybrid technology), which feeds into my Line In jack [each patch box provides a mixed in/out audio source on a mono output, so I have Line 1 feeding Left, and Line 2 feeding Right]. This too works beautifully with my vox-activated audio logging program which I'd adapted from one of the jackd sample clients. [It's ended up quite something else, but not too bad if I say so myself.] PROBLEM: When I have someone on hold on Line 2, say, and someone else is talking on Line 1, the person on Line 2 hears the music-on-hold *AND* the conversation of Line 1!! It gets better. If I dial out on Line 3 [lines 3+4 are outbound only], put that call on hold, and there are people talking on Lines 1+2, the caller at the other side of Line 3 hears not only the music-on-hold, but BOTH Line 1 + 2's conversations!!! YIKES!!! This was a big shocker. Naturally, I never did this sort of testing, being more concerned with DC bias, filtering curves and bandpass settings, and which compressors to use for the online-browsable phonelogs as well as the losslessly-compressed (FLAC of course) data archival logs. Obviously the PCM output is being mixed into whatever feeds into the Line input of the AC97 codec. This is annoying, and I can see no way of switching it off. I've explored every Mute/on/off option in the alsamixer, alas the dark fire did not avail me. I see nothing about "hardware monitoring" as an option for this chipset, however Hammerfall and others are mentioned as supporting this feature; nothing in the intel-8x0 driver suggest this is a supported feature. I'm relatively new to PC audio hardware, but has the LINE-IN been fed into the PCM output mixer from the beginning? Is this a funky chipset bug or weird issue or interaction? I know that the output mixer is fed audio from the CD and the WAVE device, but it seems a bit strange that the LINE-IN port is also routed through in this way. I can't try this under Windows, as this machine has been kept clean from the Dark Software of Udûn since its deployment. I obviously can't mute the PCM output, since that's the music-on-hold. After much experimentation, I figured I'd try the Microphone input instead. It seems noisier, even with the +20dB turned off. The impedance is quite a bit different, though fortunately I can get by with the gain set to the bare minimum [3 in the ALSA mixer, which is the smallest non-zero value I'm allowed]. But the cross-surveillance bonus to the music-on-hold problem is gone. Maybe I can do something tricksy like emit the music-on-hold on the right channel only, and hook up Line 2's audio feed into the left channel of the LINE-IN. This is a cruel hack which depends on the mic+mix functions in the codec doing their job properly, which is a huge ask. Ah well. I'm back in action, only now I have only one channel of recording capability. :( :( Is there a mixer option/setting to disable this "pass-through" routing of LINE-IN to the LINE-OUT sound port? In Mordor where the PC platform lies, =MB= -- A focus on Quality.

21 years, 5 months

Re: [linux-audio-user] jack and sblive inputs

by michelangelo79＠libero.it

---------- Initial Header ----------- >From : linux-audio-user-bounces(a)music.columbia.edu To : "A list for linux audio users" linux-audio-user(a)music.columbia.edu Cc : Date : Sat, 22 May 2004 07:00:02 -0500 Subject : Re: [linux-audio-user] jack and sblive inputs > To use a mic just use a stereo to dual mono splitter in the line input > and use either the left or right channel. Thanks, this is a practical solution, but wishing to use both mic and line-in? Is it possible that there is no way to enable the mic plug? Michelangelo

21 years, 5 months

[linux-audio-user] jack and sblive inputs

by michelangelo79＠libero.it

Hi People! I've been searchin around to solve my little problem but couldnt find any solution yet. I have a Creative Sound Blaster Live! sound card and the problem is that when i star jack deamon, i have only 2 input channel and 2 output channel. Not bad for the output (even if my sound card has more, i need only that two). Too bad for the input! I noticed that the two output channel are not really two separate channel, but left channel and right channel of the same output plug in sblive. In the same way the two input channel are left channel and right channel of the line-in plug. I wonder how can I add a third input channel for the microphone plug? (just one 'cause mic is mono). To start jack i use qjackctl, i tried to put 3 input channels in the setup section, but deamon do not start with this configuration! : configuring for 44100Hz, period = 1024 frames, buffer = 2 periods Couldn't open hw:0 for 32bit samples trying 24bit instead Couldn't open hw:0 for 24bit samples trying 16bit instead ALSA: cannot set channel count to 3 for capture ALSA: cannot configure capture channel cannot load driver module alsa 13:33:10.468 JACK was stopped successfully. While when starting jack deamon with only 2 channels (one plug) the command line used by qjackctl is: jackd -v -R -t500 -dalsa -dhw:0 -r44100 -p1024 -n2 -i2 -o2 Any help is apriciated! :) Thank you all! Michelangelo

21 years, 5 months

[linux-audio-user] M-Audio Delta 66 breakout box cable

by Cal

howdy folks, WRT an M-Audio Delta 66, can anyone tell me if the 15 way cable connecting the card to the breakout box is a straight through cable, or does it have funky things like special shielding or oddball connection patterns? cheers, Cal

21 years, 5 months

[linux-audio-user] Using .PAT files with Jack, and drawbars

by Rob

I just got an Edirol PCR-30 for my birthday, so I've dug out my 10 years worth of GUS patches and have been playing them through Timidity. Unfortunately, the Intel chipset on my laptop doesn't seem to want to give ALSA more than one pcm client, so I can't use Timidity and Jack at the same time. I can kill Timidity when I want to switch over to, for example, ams, but this seems to make Rosegarden (and sometimes ALSA Patch Bay) unstable and will be a huge pain if I ever want fake analog synth and fake piano in the same song. So.... Is there an easy way to do one of the following in free software: (a) make Timidity use Jack, (b) make Fluidsynth use .PAT files, or (c) losslessly and automatically convert a couple hundred .PAT files to soundfonts? I did just spend about an hour googling for any of the above solutions but to no avail. There was a widely-pirated sound format converter I remember under Windows, Awave maybe? but I don't know of a Linux equivalent whether free or not. Any ideas would be appreciated. Oh yeah, and the PCR-30 has eight sliders right on the front of it (plus a bunch of knobs and buttons.) I know they're sent over MIDI because one of the sliders affects the filter cutoff in one of the ams examples. This makes me think of drawbars. Any suggestions for a drawbar organ simulator that lets you assign MIDI controllers to each of the drawbars and then move them in realtime? Rob

21 years, 5 months

[linux-audio-user] Problem recording 24/7

by H. T. Hind

Hi, I'm trying to record 24/7 using Fedora Core 1, JACK 0.98, ecasound 2.3.3 , python2.2.3, alsa-1.0.1 and 2.4.22-1.2140.nptl.caps.rhfc1.ccrma kernel and delta 1010. I'm using the Planet CCRMA packages and kernel. Cron kicks off a python script every hour, that starts ecasound and records 4 channels for 1 hour . The script also invokes lame to compress the recorded files to mp3. The Python script invokes jack_connect to connect the ecasound instance with JACK. After a couple of days of perfect recording, jack_connect just seems to hang while connecting and then all the subsequent instances of recording invoked from CRON pile and no more recording happens till JACK is restarted. The JACK logs show the following entries - subgraph starting at ecarecord_26319 timed out (subgraph_wait_ fd=13, status = 0, state = Triggered) at 357895053385 client waiting on 13 took 1009935 usecs, status = 1 sig = 357894043446 awa = 0 fin = 0 dur=0 subgraph starting at ecarecord_26319 timed out (subgraph_wait_ fd=13, status = 0, state = Triggered) at 357896063568 client waiting on 13 took 1009961 usecs, statu s = 1 sig = 357895053603 awa = 0 fin = 0 dur=0 subgraph starting at ecarecord_26319 timed out (subgraph_wait_ fd=13, status = 0, state = Triggered) The ecasound log for the instance ecarecord_26319 shows - (ecasoundc_sa) Error='read() error', cmd='engine-halt' last_error='' cmd_cnt=2192 last_cnt=2191. (ecasoundc_sa) Error='sync error', cmd='cs-disconnect' last_error='' cmd_cnt=2193 last_cnt=2191. (ecasoundc_sa) Error='read() error', cmd='cs-disconnect' last_error='' cmd_cnt=2193 last_cnt=2191. I can't figure out what could be causing the jack_connect instance to stall , whether it is JACK , jack_connect or ecasound or python I can't figure out. Any pointers or suggestions on how to solve the problem ? hth ---

21 years, 5 months

[linux-audio-user] clavier -o /dev/midi ==>> clavier: `/dev/midi': No such device

by Mark Knecht

Hi, I wonder if anyone has tried building clavier and getting it to work with Alsa? I just get the error message in the subject line. Is /dev/midi not appropriate? thanks, Mark

21 years, 5 months

RE: [linux-audio-user] midi filter to convert NRPN's to CC's

by Matthew Allen

> -----Original Message----- > From: linux-audio-user-bounces(a)music.columbia.edu [mailto:linux-audio- > user-bounces(a)music.columbia.edu] On Behalf Of James Stone > Sent: Thursday, May 20, 2004 12:07 AM > To: A list for linux audio users > Subject: Re: [linux-audio-user] midi filter to convert NRPN's to CC's > > I think you should be able to do it with pd fairly easily.. You can > certainly perform mathematical functions on midi inputs. Using pd purely > as a midi router (and modifier) should not be a problem. I was thinking about PD, but never having used NRPN's I don't actually know how you would get them into PD to do the tweaking. Once they were in there I could certainly set up an environment that scaled and tweaked ranges of numbers. m. _________________________________________________ Scanned on 20 May 2004 16:43:02 Scanning by http://erado.com

21 years, 5 months

RE: [linux-audio-user] midi filter to convert NRPN's to CC's

by Matthew Allen

The problem will actually be getting the NRPN's into PD. There are built in pd objects for [ctlin] which gives you 3 outlets, one for the Value, one for CC# and one for channel. There is a [midiin] object that might give you raw Midi info that you can parse, but like I said in my first email, I have never really used NRPN's so I don't know how you would go about getting them into PD. You could search the PD list archives here: http://iem.at/mailinglists/pd-list/ Or just join the list its self. It is full of a bunch of very helpful folks (many of whom are also here). David Mccallum has done a couple of abstractions dealing with NRPN's including it appears [nrpnin] and [nrpnout]. So this would be a good place to start. http://mentalfloss.ca/sintheta/html/downloads.html m. > -----Original Message----- > From: linux-audio-user-bounces(a)music.columbia.edu [mailto:linux-audio- > user-bounces(a)music.columbia.edu] On Behalf Of Chris Pickett > Sent: Thursday, May 20, 2004 10:53 AM > To: A list for linux audio users > Subject: Re: [linux-audio-user] midi filter to convert NRPN's to CC's > > Matthew Allen wrote: > >>-----Original Message----- > >>From: linux-audio-user-bounces(a)music.columbia.edu [mailto:linux-audio- > >>user-bounces(a)music.columbia.edu] On Behalf Of James Stone > >>Sent: Thursday, May 20, 2004 12:07 AM > >>To: A list for linux audio users > >>Subject: Re: [linux-audio-user] midi filter to convert NRPN's to CC's > >> > >>I think you should be able to do it with pd fairly easily.. You can > >>certainly perform mathematical functions on midi inputs. Using pd > > > > purely > > > >>as a midi router (and modifier) should not be a problem. > > > > > > I was thinking about PD, but never having used NRPN's I don't actually > > know how you would get them into PD to do the tweaking. Once they were > > in there I could certainly set up an environment that scaled and tweaked > > ranges of numbers. > > NRPN's are encoded by 4 CC's, two for the parameter number and two for > the data value, and they appear in four-byte chunks in the midi stream. > I'll take a look at PD, and post again if I get anywhere or get stuck. > Thanks for the suggestions ... three votes for PD in a row sounds good > to me :) > > Cheers, > Chris > > > _________________________________________________ > Scanned on 20 May 2004 16:59:20 > Scanning by http://erado.com _________________________________________________ Scanned on 20 May 2004 18:22:47 Scanning by http://erado.com

21 years, 5 months

[linux-audio-user] Steinberg The Grand

by Carl Hetherington

Hi, Has anyone had any luck running Steinberg's The Grand under vstserver? vstserver works beautifully for the Native Instruments B4 and the Steinberg EP, but The Grand gets close and then wine starts complaining. I'll post more details if people are interested, but I'd just like to know if anyone has managed to get it going. Cheers Carl

21 years, 5 months

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Linux-audio-user