Linux-audio-user September 2004

linux-audio-user@lists.linuxaudio.org

161 participants
177 discussions

by Greg Reddin

This is something that has bugged me ever since I built my Linux DAW. When I boot up, when I start jack, when I open a web page with Flash, when I get a licq message -- anything that uses my audio driver, I hear a fairly loud low-end pop. Is this common? Is it a known issue I can fix? I figure it's a problem somewhere in the following (backwards) chain: Delta1010LT hardware (SPDIF out) -> Delta1010 alsa driver -> alsa library -> jack. I suspect it's not jack since it happens with other audio output as well. I don't remember, but I don't think it was happening when I was using the analog outs. Of course I can check that pretty quickly when I get my Linux box hooked back up tomorrow. Thanks, Greg __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! http://promotions.yahoo.com/new_mail

20 years, 2 months

[linux-audio-user] LAC 2004 recordings now online

by Joern Nettingsmeier

hi everyone! thanks to the work of stephan römer at zkm, the recordings of the linux audio conference 2004 are now completely edited and tagged. they are being uploaded as i type. you will find them at http://linuxaudiodev.org/contrib/zkm_meeting_2004/recordings/ , the corresponding slides are at http://linuxaudiodev.org/contrib/zkm_meeting_2004/slides/ . best regards, and sorry it took so long (my responsibility), jörn ps: if you have more photos, recordings or other lac memorabilia to share, throw them my way so that i can upload them. -- "90% of all networking problems are routing problems. 9 of the remaining 10% are routing problems, but in the other direction. The final 1% might be something else, but check the routing anyway." - Anthony Stone's networking words of wisdom Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Germany http://spunk.dnsalias.org (my server) http://www.linuxaudiodev.org (Linux Audio Developers)

20 years, 2 months

[linux-audio-user] Just bought a Delta 1010LT, now what..?

by Glenn McCord

I'm now the proud owner of a Delta 1010LT. Unfortunately getting it to run on Linux is proving a chore. So far I've googled a big (maybe I'm a useless googler), and looked at the alsa-project website So from those things I figured I would do the following... modified my /etc/modules.d/alsa file to this _____________________________________________________ # Alsa 0.9.X kernel modules' configuration file. # $Header: /home/cvsroot/gentoo-x86/media-sound/alsa-driver/files/alsa-modules.conf-rc,v 1.1 2002/12/21 06:31:52 agenkin Exp $ # ALSA portion alias char-major-116 snd # OSS/Free portion alias char-major-14 soundcore ## ## IMPORTANT: ## You need to customise this section for your specific sound card(s) ## and then run `update-modules' command. ## Read alsa-driver's INSTALL file in /usr/share/doc for more info. ## ## ALSA portion alias snd-card-0 snd-ice1712 # OSS/Free portion - card #1 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss alias /dev/mixer snd-mixer-oss alias /dev/dsp snd-pcm-oss alias /dev/midi snd-seq-oss # Set this to the correct number of cards. options snd cards_limit=1 _________________________________________ Run the following command #modprobe snd-ice1712;modprobe snd-pcm-oss;modprobe snd-mixer-oss;modprobe snd-seq-oss ________________________________________ made a .asoundrc file with the following in it pcm.ice1712 { type hw card 0 } ctl.ice1712 { type hw card 0 } Simple version just so I can get some playback _______________________________________ I did the whole modules-update and restart of alsa (yes I'm using Gentoo) I get this error when I try to run aplay root@upstairs glenn # aplay -D ice1712 test.wav ALSA lib pcm.c:1972:(snd_pcm_open_noupdate) Unknown PCM ice1712 aplay: main:507: audio open error: No such file or directory _______________________________________ Basically I've just been trying to follow this http://alsa-project.org/alsa-doc/doc-php/template.php?company=Midiman&card=… I was using a Soundblaster Live! and wanted a 1010LT with the purpose of doing live multitrack recording with Ardour. Please tell me I can do it because that was the main purpose of buying the card in the first place. Cheers for any direction pointing. Glenn

20 years, 2 months

[linux-audio-user] Commandline tools for some audio operations?

by Mikhail Ramendik

Hello, This time I am encoding from a cassette with a rather low noise level. And I'd like to do several things automatically, from the command line, rather than call up Audacity and GWC and do some manual selection. So, which command line tools(if they are available, of course) will: - Do lowpass filtering? - Cut off the silence in the beginning and the end? - Split a file into songs based on long periods of silence in betweeh them? For the latter two points, the silence is not absolute, there is some noise at about -40 db. The tool I want would take a "noise level" value in db as a parameter. The REALLY good tool would leave about a second of "silence" before/after a song intact, because it may contain the beginning or end that are below the threshold. The PERFECT tool would also be able to look at a file with just noise and give me that db value, but it may be too much to ask, so I can find out in Audacity as well. So - are these tools available? Or am I out of luck? Of course, I can use GWC to split a file into songs and Audacity to cut out the silence manually, but that's quite a lot of mouse-dragging for an operation that could be automatic. Yours, Mikhail Ramendik

20 years, 2 months

[linux-audio-user] metronome-free MIDI recording

by Hans Fugal

I'm curious how everybody handles MIDI recording. I find it hard to believe that everyone records to a straight tempo using the metronome, only to add ritardandos and accelerandos later in some artificial way by editing the tempo track. That may work for some kinds of music (dance music perhaps), but certainly not everyone works that way.In particular, I'm recording organ music, which is similar to piano music or indeed almost any classical music in its need for expression. But if I turn off the metronome and record at a free tempo I find that the transports and bar information are more or less useless. It makes it very hard to re-record segments or set up loops, etc. How do you handle MIDI wrt tempo? I'm using Rosegarden, but not entrenched yet. -- De gustibus non disputandum est.

20 years, 2 months

[linux-audio-user] DAW Dillema -- Seeking Advice

by Greg Reddin

I've brought up discussions like this before, but it's been a while and I'm finding myself at a point of decision again. So please bear with me. I'd like some advice on what my next step is. Basically, here's my problem. I love the Linux audio world, but I am an artist and I really need something that will facilitate making music. So I'm faced with the following options: 1. Use Ardour and participate in its development. 2. Use Audacity and participate in its development. 3. Use something else and participate in its development. 4. Roll my own. 5. Buy a mac and use Digital Performer or something else. To examine these options further: For option 1, Ardour has a lot of features, but seems to lack in stability and usability. I have to restart ardour/jack several times during a session because one or the other becomes unresponsive or flaky. The transport even completely stopped working last time I used it. I lost 2 - 4 hrs work and have not been able to get it working again. Granted, I have not tried too hard to receive help with it, but I just haven't had good luck with its stability yet. Perhaps, my problem is more with usability than stability. It may be intuitive to some people to use the middle mouse button or ctl+right button combinations, but I have a really hard time getting around in ardour. Another thing about ardour that makes it hard for me to adopt it wholeheartedly is the way it is developed. It seems, IMHO, that Release 1.0 should've come out a long time ago, like after real-time multitrack recording, editing, and mixing were available. Or maybe start over, do a refactor, then release when those features are working again. There's something psychologically limiting (to me) when a product reaches version 0.9beta19 and still doesn't seem ready for a "release". To me, that seems to create a culture where things move very slowly and gives the impression that it will never really be production-ready. I recognize that there are very differing opinions on what a "release" actually means in open source. I also recognize that ardour doesn't have my name on it anywhere so I can't really complain unless I'm contributing to its development. I'm not trying to start a war, just to figure out what direction I need to settle on, so I'll shut up about that. For option 2, audacity seems to be stable and easy to use. But it lacks some essential features, like real-time effects processing. The mezzo thing looks promising, but there doesn't seem to be much momentum behind it right now. Is there an option 3? Is there another Linux DAW solution that provides (or seeks to provide) multitrack recording, real-time mixing, automation, etc.? I would typically omit option 4 right off the bat. The open source culture frowns on reinventing something that already exists. But there's a few reasons why I'm actually considering this option. First, the problems I have with ardour and audacity don't seem likely to change. Please don't misunderstand what I'm about to say. I'm not trying to offend anyone, but these are just my observations. If they are incorrect, please correct me. I don't gather that there's much momentum to build audacity into a real-time professional DAW solution. And it seems like ardour's development has been in a rut for a while. Development is happening, and new things are being added, but the stability and usability doesn't seem to be improving. So, if I'm trying to build a professionally viable DAW for Linux I could come to the conclusion that there's not currently a workable solution. Second, I'm not convinced that "three" DAWs for Linux is an unhealthy number. Look at how many different commercial solutions are available -- each one doing things a bit differently and appealing to a different user base. Maybe if there was another project with a healthy development cycle, good stability, and essential features, it would encourage the others to compete and help push Linux over the edge and into professional viability. There are other reasons, but things like coding style, object model, or testing strategy are not valid reasons in and of themselves to start a new project. But truthfully, given my limited skillset and other factors, it would likely be years before a new project was able to compete even with what's already out there, much less surpass them. So, unless I get an overwhelming response to this option, I'll probably not consider it much further. That brings me to option 5. I've only considered this because I'm ready to actually spend some money in the interest of making music instead of twiddling with code and configs. But, I'm not a big fan of ProTools, Logic, or Cubase. So Digital Performer seems to be my best option on a mac -- and I really don't know much about DP (my prior experience is mostly with Sonar and I'm simply not willing to invest further in a Windows-based platform). So, I'm not certain that I will be satisfied even if I spend a wad of cash on a Mac and some DAW software. And of course, this thread of logic implies that I have some money to throw at it, which, for the time being, is not the case. Now, I think I'm something of a poster-child for Linux audio. I'm enough of a tech-head that I can write some code and diagnose problems. I can wade through a mass of complex logic and find what I need (usually). My sessions are few and far between enough that I can experiment with stuff between them. And they are low-risk enough (meaning that I don't get paid or get paid very little for them) that lack of stability is not a huge risk for me -- just a frustration. I suspect that by the time I get to a point to actually charge real money for my services something in Linux might be ready for prime time whether it be ardour, audacity, or something as yet undeveloped. And I want to contribute financially and/or intellectually to whatever I end up with. So, given what little you know of me and what I'm looking for, what would you suggest? Would you recommend that I start following Ardour and/or Audactiy with more interest? Is there something else I don't know about? Have I actually found a need for something new? Or should I (for the time being) punt and invest in a Mac-based commercial solution? If you've gotten this far, thanks for bearing with me. Your help is greatly appreciated. I know I made some statements that could be considered controversial. I hope I have not offended anyone and apologize if I have. Greg __________________________________ Do you Yahoo!? Take Yahoo! Mail with you! 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20 years, 2 months

[linux-audio-user] Errors loading ALSA for HDSP

by Janina Sajka

I'm attempting to get my musical activities squared away on a new dual Opteron system. I'm currently running single processor with the (almost stock) Fedora Core 2 kernel, 2.6.8-1.541. I have alsa 1.06a (including the tools) installed. I compiled these from the CCRMA sources. All the modules appear to be loaded: snd_seq 80832 0 snd_hdsp 68492 0 snd_rawmidi 30116 1 snd_hdsp snd_seq_device 9356 2 snd_seq,snd_rawmidi snd_pcm 127116 1 snd_hdsp snd_timer 36104 2 snd_seq,snd_pcm snd_hwdep 10760 1 snd_hdsp snd 70664 7 snd_seq,snd_hdsp,snd_rawmidi,snd_seq_device,snd_pcm,snd_timer,snd_hwdep soundcore 10593 1 snd snd_page_alloc 11024 2 snd_hdsp,snd_pcm However, I cannot start alsa: service alsasound start Starting sound driver snd-hdsp [ OK ] Starting sequencer [ OK ] Restoring sound driver settings /usr/sbin/alsactl: load_state:1134: No soundcards found... [FAILED] If I run hdsploader I get either no message, or perhaps a message about an inability to access a certain memory range. Of course, as I write this message, I can't seem to get that error up. 'll try and capture that message in a followup, in case this is at all helpful. Needless to say, I'm hoping someone has a suggestion that will resolve this. --

20 years, 2 months

RE: [linux-audio-user] Starts with a pop

by Reuben Martin

>> --- Reuben Martin <MartinR(a)jbu.edu> wrote: >> > My guess is that devices on sound cards have a default sample rate, >> > bit rate or some other type of setting and when something accesses >> > the device and changes one of those settings, it results in that pop >> > sound. >> >> I didn't think about that. I'll have to check and see if the 1010lt >> has a default sample rate. The bad news is, if you are right, then >> driver modifications won't fix the problem. > >Not necessarily. The driver can mute the output, change the card setting, and unmute it again. I don't think muting the output will change anything if this is truly the source of the problem. This a matter of changing the digital signal, and nothing to do with the audio that is within the digital signal. -Reuben

20 years, 2 months

[linux-audio-user] asfxload and 2.6 kernel

by Jeremiah Benham

I have noticed since I switched to the 2.6 kernel tree I have been having troubles with asfxload and sfxload. When I try to launch it hangs. The program just sits their running and can't be killed. It does not use to many clock cycles but I can't seem to kill it no matter what I do. I tried killall asfxload and other tricks. I also tried to kill it with kde's process management program but it still did not die. A few times it complained that /dev/sequencer was not available no such device. I was wondering if there is something I am doing wrong or if I should file a bug report. Anyone else experience this? sound card = SB live Kernel = 2.6.2 (same problem with other versions of 2.6) asfxload = ver.0.5.0c launching asfxload by itself complains: No Emux synth hwdep device is found so I modprobe snd-hwdep and snd-emux-synth but it still complains Thanks, Jeremiah

20 years, 2 months

[linux-audio-user] ice1712: No H/W Monitoring TerraTec EWS88D with Alsa 1.0.4-1

by gm_tarkin＠comcast.net

I am unable to get Hardware monitoring working with my EWS88D card, yet it works perfectly with my Delta 1010LT (they are in the same machine running FC1 CCRMA). I checked the bug list last night on Alsa. I could find nothing for EWS88D or "PCI - ice1712" category related to H/W monitoring. The Alsa soundcard matrix says the EWS88D supports "H/W mixing". Is this the same as H/W monitoring? At present my theories are: 1. Perhaps there are different variationson the envy24/ice1712 chip and the EWS88D uses one that lacks H/W monitoring. 2. Terratec implementation of the envy24/ice1712 is 'broken'. 3. There is bug in ice1712 driver with regards to the EWS88D Any assistence ya'll can provide is appreciated. Regards, Robb NOTE: While the Alsa sound card matrix for the EWS88D states that "[To] IEC-60958 TOSLINK Output" works for the EWS88D. I can confirm that [Aio] should be added to the notes section. I've got the EWS88D ADAT In and ADAT Out plugged into a Berhinger ADA8000 via Lightpipe. Recording and playback work brilliantly with the ADA8000 set as Master for wordclock purposes. In fact, I will add this to Alsa Mantis system. . . silly me. Below is detailed information from ALSA, JACK, and ENVY24CONTROL. ALSA VERSION: rpm -qa |grep alsa: alsa-driver-1.0.4-1.cvs.rhfc1.ccrma alsa-firmware-1.0.4-1.cvs.rhfc1.ccrma alsa-kernel-2.4.26-1.ll.rhfc1.ccrma-1.0.4-1.cvs.rhfc1.ccrma alsa-lib-1.0.4-1.cvs.rhfc1.ccrma alsa-lib-devel-1.0.4-1.cvs.rhfc1.ccrma alsa-oss-1.0.4-1.cvs.rhfc1.ccrma alsa-tools-1.0.4-1.cvs.rhfc1.ccrma alsa-utils-1.0.4-1.cvs.rhfc1.ccrma clalsadrv-1.0.0-1.rhfc1.ccrma JACK - TERRATEC EWS88D (hw:0) - with hardware monitoring enabled: ==================================================== /usr/bin/jackstart -R -t200 -dalsa -dhw:0 -r44100 -p1024 -n2 -H -M OUTPUT: loading driver .. apparent rate = 44100 creating alsa driver ... hw:0|hw:0|1024|2|44100|0|0|hwmon|swmeter|-|32bit control device hw:0 configuring for 44100Hz, period = 1024 frames, buffer = 2 periods ICE1712?: (0) cannot set input monitoring (No such file or directory) ICE1712?: (1) cannot set input monitoring (No such file or directory) ICE1712?: (2) cannot set input monitoring (No such file or directory) ICE1712?: (3) cannot set input monitoring (No such file or directory) ICE1712?: (4) cannot set input monitoring (No such file or directory) ICE1712?: (5) cannot set input monitoring (No such file or directory) ICE1712?: (6) cannot set input monitoring (No such file or directory) ICE1712?: (7) cannot set input monitoring (No such file or directory) COMMENT: Hardware monitoring for the ice1712 driver on the EWS88D? seems broken, for lack of a better word. JACK - M-AUDIO DELTA1010LT (hw:1) - with hardware monitoring enabled: ==================================================== /usr/bin/jackstart -R -t200 -dalsa -dhw:1 -r44100 -p1024 -n2 -M -H OUTPUT: loading driver .. apparent rate = 44100 creating alsa driver ... hw:1|hw:1|1024|2|44100|0|0|hwmon|hwmeter|-|32bit control device hw:1 configuring for 44100Hz, period = 1024 frames, buffer = 2 periods ENVY24CONTROL - EWS88D ==================================================== envy24control -c 0 using --- input_channels: 0 --- output_channels: 0 --- pcm_output_channels: 8 --- spdif in/out channels: 2 Monitor Mixer Tab: * Digital Mixer * PCM Out 1 ... PCM Out 8 * H/W In 1 ... H/W In 8 [MISSING] * S/PDIF In L & S/DIF In R Patchbay/Router Tab: * H/W Out 1 ... H/W Out 8 [MISSING] * S/PDIF Out L & S/PDIF Out R Hardware Settings Tab: * Master Clock * Rate State * Actual Rate * Volume Change * S/PDIF Output Settings o Consumer o Professional Analog Volume [MISSING] COMMENT: Hardware monitoring envy24control corroberates the lack of hardware monitoring on the EWS88D. No input or output channels, and missing tabs and controls when comparted to DELTA1010LT. ENVY24CONTROL - DELTA1010LT ==================================================== envy24control -c 1: using --- input_channels: 8 --- output_channels: 8 --- pcm_output_channels: 8 --- spdif in/out channels: 2 Monitor Mixer Tab: * Digital Mixer * PCM Out 1 ... PCM Out 8 * H/W In 1 ... H/W In 8 * S/PDIF In L & S/DIF In R Patchbay/Router Tab: * H/W Out 1 ... H/W Out 8 * S/PDIF Out L & S/PDIF Out R Hardware Settings Tab: * Master Clock * Rate State * Actual Rate * Volume Change * S/PDIF Output Settings o Consumer o Professional Analog Volume Tab: * DAC 1 ... DAC 8 * ADC 1 ... ADC 8 * IPGA 1 .... IPGA 8

20 years, 2 months

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Linux-audio-user September 2004