Linux-audio-user October 2017

linux-audio-user@lists.linuxaudio.org

32 participants
39 discussions

jack freewheeling / has excessive timing errors

by Athanasios Silis

Hi all, may I ask what could be the cause of jack freewheeling unexpectedly and having timing errors (and xruns of course). the console window outputs these error messages: JackPosixSemaphore::TimedWait err = Connection timed out JackFreewheelDriver::ProcessSync: SuspendRefNum error JackAudioDriver::ProcessGraphSync: ProcessWriteSlaves error, engine may now behave abnormally!! JackPosixSemaphore::TimedWait err = Connection timed out JackAudioDriver::ProcessGraphSync: SuspendRefNum error, engine may now behave abnormally!! If I happen to run zita-a2j and zitaj2a instances at the time this happens, they will report: Detected excessive timing errors, waiting 15 seconds. This may happen with current Jack1 after freewheeling. Detected excessive timing errors, waiting 15 seconds. This may happen with current Jack1 after freewheeling. Detected excessive timing errors, waiting 15 seconds. This may happen with current Jack1 after freewheeling. Detected excessive timing errors, waiting 15 seconds. This may happen with current Jack1 after freewheeling. Starting synchronisation. Starting synchronisation. The setup I have contains the following cards: 0 [DMX6Fire ]: ICE1712 - TerraTec DMX6Fire TerraTec DMX6Fire at 0xec00, irq 16 1 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf9ff8000 irq 45 2 [DSP24 ]: ICE1712 - Hoontech SoundTrack Audio DSP24 Hoontech SoundTrack Audio DSP24 at 0xe400, irq 17 3 [DSP24_1 ]: ICE1712 - Hoontech SoundTrack Audio DSP24 Hoontech SoundTrack Audio DSP24 at 0xd880, irq 18 I run jack on a combination of cards #2 and #3. Analytically: It contains 2x hoontech dsp c-ports 2000 (each with 8x ins and 8x outs - these cards are identical with each other and very similar to the Midiman L1010 - same ICE1712 chip), joined as a unified sound card with 16x ins and 16x outs through ~/.asoundrc as described here http://bandshed.net/forum/index.php?topic=1208.0. I have unfortunately deviated from the guide, in that I have not used the SPDIF output of the 1st card to syncronize the clock of the 2nd one. This was not intentional but some unexpected error shows up in the envy24control when I try to set the SPDIF settings as shown in the link. So I can only assume my changes have not been registered (they definitely do not survive a restart of the envy24control mixer). The bottom line is the 2 sound cards may very well be running out of sync. Is that a reason that could cause the freewheeling? I do not get xruns messages in qjackctl often, but when a message appears the counted xruns are in the order of ~1600 or so. I am junning jackd2 with there parameters: jackd -S -R -dalsa -C capture16 -P playback16 -r48000 -p256 -n2 Should be reasonble to avoid these many xruns. No? Additionally, there are alot of plugins loaded. 2x mpd's , many calf racks (~21 calf plugins in total), pulseaudio jack module and some times zita-a2j and zita-j2a to connect in the setup the additional soundcards (DMX6Fire and HDA_Intel). The problem is that because of these timing errors some (but never all!) calf racks crash and I would like to avoid that. So what could be triggers for these jack error messages ? thank you in advance for your help.

6 years, 6 months

[Music] Musace

by jonetsu

Hello, Basically created with the limitation of using only the u-he ACE synth but then the Bazille synth also has some contribution in two tracks. The bass is ACE 'Color of Chloe' preset, with appropriate mixing-time adjustments. Also created as a mixing exercise. This is a short song-like piece with a simple bright theme. Of course the drums and percussion were not made using the synths. https://soundcloud.com/nominal6/musace Cheers.

6 years, 6 months

[LAA] [LAD] Guitarix 0.36.0 released

by Hermann Meyer

Guitarix release 0.36.0 Guitarix is a tube amplifier simulation for jack (Linux), with an additional mono and a stereo effect rack. Guitarix includes a large list of LV2 and LADSPA plugins, and support LADSPA / LV2 plugs as well in it's racks. The guitarix engine is designed for LIVE usage, and feature ultra fast, glitch and click free preset switching and is full Midi and remote controllable (the Web UI is not included in the distributed tar ball). This release add a couple of features to the lately introduced DrumSequencer plugin. It add 2 more toms, it add beat repeat/skip switches, it add the possibility to add a sequence (Preset) to the current one, switch to previous/next preset, . . or load a slected preset (sequence) all controllable with Midi CC. Also it introduce new Midi Note ON to CC support, means, you could control toggle switches with Midi Note On messages now (continues control). Also it introduce a new option to load Stereo plugs as Mono plugs, eg. before the amp. This is in special useful for the famous rkrlv2 plugs: https://github.com/ssj71/rkrlv2 + new configure options --fontdir (set font install directory) and --no-font-cache-update (dont automatically update the font cache) + fix build with --no-lv2-gui option + add new style "Gold" + fix GxAmplifier re-init convolver on buffersize change ( Ardour loops) + fix jack transport support + also fix midi clock and bpm sync support + fix load online preset in remote UI + fix trim label in Lv2/Ladspa plugins beside that a couple of changes have been made under the hut. Refer to our project page for more information: http://guitarix.org Download Site: http://sourceforge.net/projects/guitarix/ Forum: https://guitarix.sourceforge.io/forum/ Consider visiting our forum or leaving a message on guitarix-developer(a)lists.sourceforge.net enjoy and keep on rooking. regards hermann

6 years, 6 months

Hoontech DSP24 with DSP C-Port 2000: how do I use the line in/out of the pci card in JACK?

by Athanasios Silis

Hi all , I am using the Hoontech DSP24 with DSP C-Port 2000 for music production. This is similar to Midiman L1010 card , which has the same chipset (ice1712) and a fairly similar card architecture/implementation. So if you are an owner of ice1712 chipset, please do not skip this email :) Personally, I've been using the external 1U c-port 2000 (based on the ICE1712) for the past 10+ years in linux and i'm very satisfied with it :) So far I had not bothered with capture/playback from the pci card (DSP24, based on SigmaTel STAC97xx as I read in this link http://www.st-audio.de/products/dsp24.html). I would like to do that now. Of course I want to use these playback/capture ports from within jack $ cat /proc/asound/cards 0 [DMX6Fire ]: ICE1712 - TerraTec DMX6Fire TerraTec DMX6Fire at 0xec00, irq 16 1 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf9ff8000 irq 45 2 [DSP24 ]: ICE1712 - Hoontech SoundTrack Audio DSP24 Hoontech SoundTrack Audio DSP24 at 0xe400, irq 17 3 [DSP24_1 ]: ICE1712 - Hoontech SoundTrack Audio DSP24 Hoontech SoundTrack Audio DSP24 at 0xd880, irq 18 where for card2: $ cat /proc/asound/pcm 02-00: ICE1712 multi : ICE1712 multi : playback 1 : capture 1 02-01: ICE1712 consumer : ICE1712 consumer : playback 1 : capture 1 02-02: ICE1712 consumer (DS) : ICE1712 consumer (DS) : playback 6 I start up jack with master card the #2 as: $ pasuspender -- jackd -S -dalsa -dhw:DSP24,0 -r44100 -p64 -n2 not sure if this is correct, but I get to see all the channels I expect at least (which is weird because here I see that device 2:0 apparently has only 1p and 1c port ?!?!?!? Can someone explain?) To do that I tried using zita-a2j and zita-j2a. the commands are the following : #playback zita-j2a -d hw:DSP24,1 -c 2 -j DSP24old_out -r44100 -p1024 -n2 #capture zita-a2j -d hw:DSP24,1 -c 2 -j DSP24old_in -r44100 -p1024 -n2 I am not sure if device hw:2,1 is correct here either. If someone could explain to me the differences of "ICE1712 multi" , "ICE1712 consumer" , "ICE1712 consumer(DS)", I would appreciate it! right so, I can start the playback port fine - the zita-j2a outputs in cli "Starting synchronisation" every second or so, but it seems to have started up at least. However I do not get any sound from the line out jack of the pci card. I checked the levels in alsamixer, "PCM" and "Master" are at 100% and not muted. What can I not hear anything? The capture port is non existent for some reason. $ Can't open ALSA capture device 'hw:DSP24,1'. How can I check if a capture port exists for this card? or how can I verify it is not used by someone else (f.e. pulseaudio server, etc) ? Thank you in advance for your help!

6 years, 6 months

Re: [LAU] firewire-thunderbolt adapter?

by David W. Jones

On October 3, 2017 1:02:25 PM HST, Bernardo Barros <bernardo.barros(a)nyu.edu> wrote: > On 10/3/17 18:49, David W. Jones wrote: > > rds, for example. > > > > I wonder - could an adaptor connect a Firewire device to USB-C or > Thunderbolt port? > > It's possible on mac-OS, you can plug any firewire sound-card with an > adapter to Thunderbolt 3. > > I just want to know if that's the case on Linux, since no laptop has > firewire anymore. Sending back to the list. I wonder if one of the FW=>Thunderbolt adaptors would work to connect to a non-Mac? I'd like to know, too, since I have a Firewire audio device sitting at home unused since the demise of my old Toshiba laptop and its Firewire port. -- David W. Jones gnome(a)hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com Sent from my Android device with K-9 Mail.

6 years, 6 months

Linux Audio Berlin meeting @ c-base

by Daniel Swärd

Hi. Time for my monthly spam to invite you to the Berlin monthly meeting at c-base. Hope to see you there. :-) Cheers! /Daniel

6 years, 6 months

Re: [LAU] Trying to start Aeous gives me seg faults

by gnome＠hawaii.rr.com

> On 09/18/2017 06:39 AM, Fons Adriaensen wrote: >> On Mon, Sep 18, 2017 at 07:31:41AM +0200, Hermann Meyer wrote: >> >>> Am 18.09.2017 um 06:44 schrieb david: >>>> Starting Aeolus from command line returns this: >>>> >>>> Reading '/usr/share/aeolus/stops/Aeolus/definition' >>>> Reading '/home/david/.aeolus-presets' >>>> Segmentation fault >>> >>> First shot in such cases is remove/rename config files in home >>> '/home/david/.aeolus-presets' >> >> No, this is due to recent versions of the font libs requiring >> more stack space. >> >> In main.cc, line 253, increase the stack size of the interface >> thread: >> >> iface->thr_start (SCHED_OTHER, 0, 0x00020000); >> >> This should solve the issue. > > Thanks. I see that aeolus is up to 0.9.5 according to this: > > http://kokkinizita.linuxaudio.org/linuxaudio/downloads/index.html > > Line 253 in main.cc there matches your line above. So I should increase the 0x00020000? Or is that the value it should be at and my current version has something smaller? > > Thanks. Sorry for delay, but I downloaded the Aeolus source from the link above and tried running make while in the source directory. I got the following error: g++ -O3 -Wall -MMD -MP -march=native -DVERSION=\"0.9.5\" -DLIBDIR=\"/usr/local/lib64\" -c -o main.o main.cc main.cc:28:23: fatal error: clthreads.h: No such file or directory compilation terminated. <builtin>: recipe for target 'main.o' failed make: *** [main.o] Error 1 Ideas? Also, I can report something else weird. On my desktop machine, running Debian Testing + KXStudio repositories, Aeolus starts up and runs fine. On my laptop, also running Debian Testing + KXStudio repositories, Aeolus dies with the seg fault reported above. David W. Jones gnome(a)hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com

6 years, 6 months

Re: [LAU] [CP] KISS or Complex?

by Stephen Stubbs

09-30-2017 Hi Bob, Unfortunately, I am not the person to go into technical details about mixing. I know some of the basic concepts, but other than pan and volume, I have not used the other techniques. Check out the Internet for sound mixing groups. At one time, I was working with Linux systems for music (personal pleasure use at home), and followed the Linux Audio group. I still do follow that group. To post to that group, I use linux-audio-user(a)lists.linuxaudio.org <mailto:linux-audio-user@lists.linuxaudio.org> There should be other Internet groups around. The people at the Linux audio group include mixing professionals. They should be able to listen and give suggestions on how to adjust your mix. The group is for people using digital audio workstations (DAW) running under Linux and Linux mixing tools. So some of their suggestions may only apply to Linux programs/tools. But they are very helpful in listening to work and making suggestions. I would think there are professionals who use the DAW and tools that you are using. Find the groups when they hang out. Mention your hearing issues. At the Linux audio group, there are blind people using command line Linux DAWs and tools for their mixing. As you are probably aware, musicians are very generous with their time and talents. Good Luck, Stephen. From: Bob Ebdon [mailto:r.ebdon@ntlworld.com] Sent: Saturday, September 30, 2017 7:54 AM To: Stephen Stubbs <fartreader(a)gmail.com> Cc: Bob Ebdon <r.ebdon(a)ntlworld.com> Subject: Re: [CP] KISS or Complex? Thanks Stephen. Just back on home computer after a few days away, so sorry for not getting back sooner. I needed to check a few things with my recording. As always, I value your comments. One thing that worried me was your comments about panning the instruments. They are panned. I have two guitars panned 66R and L, fiddle is 33R, bass and lead vocal central. Autoharp mic and DI are 10L and 10R, vocal harmonies are 33L and R. Maybe this spread isn’t far enough? Maybe push the fiddle wider, it is in the same space as one of the harmony vocals at present? I have real problems with stereo placement as I am basically mono - totally deaf in one ear! This is why I had to come back to the recording and check that I had actually exported a stereo version, I could not tell just by listening whether it is stereo or mono, the only way I can check is if it sounds different when I swap the headphones round - and it does. I also have faders way down on the instruments relative to the vocals, and have written volume for the autoharp to bring it up as fills. I appreciate your comments about reverb and compression. I have not yet discovered how to make my voice sound natural! I have on it some EQ, some compression (set at ratio 4.2 - too high?), a vocal rider and a C6 multi band compressor set to give a treble boost. Looks like this is all too much? For reverb, I use Spaces. I have two, one set at about 2sec, the other at 6sec, and I use about -10dB of the first and -30dB of the second. Again - too much? Any advice you can give is much appreciated. Bob Ebdon www.facebook.com/AutoharpBob <http://www.facebook.com/AutoharpBob> On 28 Sep 2017, at 14:11, Stephen Stubbs <fartreader(a)gmail.com <mailto:fartreader@gmail.com> > wrote: Why not do both? Start out with the vocal and autoharp. Then start adding the other instruments through the course of the song, to end up with everything at the last. On a mixing level, I prefer less reverb and compression. Why not pan the backup instruments to different locations, and use volume to get the depth of field? Go for the feeling that you and the autoharp at the microphone, while the backup instruments are behind and around you. I think it would give you more of a live performance sound. For What It's Worth, Stephen.

6 years, 6 months

Music: Too Late for Goodbyes ?

by jonetsu

Hello, This is a fast-paced 'organic techno' (of sorts) synth track. It was fun to mix. https://soundcloud.com/nominal6/too-late-for-goodbyes Unfortunately the bass was mostly eaten by soundcloud. Almost sounds AM-like. I'll have to figure this one out when mixing: why is the bass largely diminished on soundcloud and how to compensate, if possible, for this. Cheers.

6 years, 6 months

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Linux-audio-user October 2017