Linux-audio-user

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20661 discussions

[linux-audio-user] Help for isisalsa driver coding

by Felipe Sologuren

Hello: Sorry by my english. I'm request for people could help in coding of isisalsa driver (for Guillemot Maxistudio Isis). I'm not a developer, only entusiast user, and people developing in this project "know little of alsa coding rules" and/or got little time. The status is only playback and still have bugs for work with alsa applications, this is the area for my request. The reversing is complete. Maybe the card isn't a professional audio device, but offer the possibility of several users to switch their environment to linux (for me it does) and cause guillemot closed win2k/xp support for this card. Feel free to forward (please). http://uk.guillemot.com/brand/gamme/son/msisis/isis.html http://fr.guillemot.com/brand/gamme/son/msisis/isis.html http://sourceforge.net/projects/isisalsa/ irc://freenode/isisalsa http://www.estudiohum.cl/Isis.htm Thanks for hear us ;) Felipe Sologuren Sound Engineer

21 years, 7 months

RE: [linux-audio-user] snd-hdsp+cardbus=distortion -- the sagacontinues (cardbus driver=culprit?) UPDATE: 99.9% sure it is the cardbus driver yenta_socket

by Ivica Ico Bukvic

> -----Original Message----- > From: Russell King [mailto:rmk@arm.linux.org.uk] On Behalf Of Russell King > Sent: Tuesday, March 30, 2004 3:01 AM > To: Ivica Ico Bukvic > Cc: 'A list for linux audio users'; alsa-devel(a)lists.sourceforge.net; > linux-kernel(a)vger.kernel.org; linux-pcmcia(a)lists.infradead.org > Subject: Re: [linux-audio-user] snd-hdsp+cardbus=distortion -- the > sagacontinues (cardbus driver=culprit?) UPDATE: 99.9% sure it is the > cardbus driver yenta_socket > > On Tue, Mar 30, 2004 at 12:52:11AM -0500, Ivica Ico Bukvic wrote: > > 6) Pester alsa-dev, lau, and kernel/pcmcia people to death begging for > help > > :-) > > > > IN-PROGRESS :-) > > What needs to happen is that the card driver author needs to investigate > what is going on, and, if it seems related to the core PCMCIA core or > the socket driver, we need to get involved. > > IOW, linux-pcmcia people don't debug card drivers. > To add to what Tim mentioned, I think that the driver is fine as it does work on select notebooks and desktops (the card can be plugged into either PCI card or PCMCIA cardbus). Yet, in these select instances it does not work even though neither the cardbus driver nor the actual card driver do not report any particular problems. Hence the only logical explanation is that there is something wrong with the pcmcia controller driver. This card does tax the throughput of the cardbus like no other card I can think of, hence the problem may be more widespread, but exhibits itself just in this case where the cardbus is being pushed to its limits. Yet, the hardware is not the issue when the same notebook/soundcard combo works flawlessly in WinXP. Hope this helps! Ico

21 years, 7 months

Re: [linux-audio-user] snd-hdsp+cardbus=distortion -- the sagacontinues (cardbus driver=culprit?) UPDATE: 99.9% sure it is the cardbus driver yenta_socket

by Tim Blechmann

hi russel > The majority of PCMCIA is the same between the two kernels. There > have been some cleanups and changes to the way card events (insertions > and removals) occur, and some setup changes to the cardbus bridge to > turn on some extra features. > > However, if you're saying that 2.4 and 2.6 behave the same way, then > logically it isn't something that any of these changes have caused. i tried to use the pcmcia-cs driver again (2.4.24-ck1 ... they didn't work with the hdsp, either (afaik, the pcmcia-cs driver never worked with the hdsp, but i'm not sure about that) the dmesg output was: PCI: Found IRQ 5 for device 02:00.0 PCI: Sharing IRQ 5 with 01:00.0 PCI: Found IRQ 11 for device 02:00.1 PCI: Sharing IRQ 11 with 00:1f.3 PCI: Sharing IRQ 11 with 00:1f.5 PCI: Sharing IRQ 11 with 00:1f.6 O2Micro OZ6933 rev 01 PCI-to-CardBus at slot 02:00, mem 0x20000000 host opts [0]: [pci/way] [pci irq 5] [lat 168/176] [bus 3/6] host opts [1]: [pci/way] [pci irq 11] [lat 168/176] [bus 7/10] ISA irqs (default) = 3,4,7,9,10 PCI status changes cs: cb_alloc(bus 7): vendor 0x10ee, device 0x3fc5 cs: cb_free(bus 7) cs: cb_alloc(bus 7): vendor 0x10ee, device 0x3fc5 PCI: No IRQ known for interrupt pin A of device . ALSA ../../alsa-kernel/pci/rme9652/hdsp.c:5031: unable to grab memory region 0x0-0x1bff RME Hammerfall-DSP: no cards found i have no idea of the differences between the cardbus drivers of the kernel and of the pcmcia-cs project ... anyway, the hdsp is working on a certain area of memory, that the pcmcia-cs driver can't grab the kernel driver can ... i disabled acpi and apic and used pci=biosirq as kernel flag maybe this can give you a hint ... anyway, i hope thomas can comment on that... cheers and thanks for your help... Tim mailto:TimBlechmann@gmx.de ICQ: 96771783 -- The only people for me are the mad ones, the ones who are mad to live, mad to talk, mad to be saved, desirous of everything at the same time, the ones who never yawn or say a commonplace thing, but burn, burn, burn, like fabulous yellow roman candles exploding like spiders across the stars and in the middle you see the blue centerlight pop and everybody goes "Awww!" Jack Kerouac

21 years, 7 months

Re: [linux-audio-user] snd-hdsp+cardbus=distortion -- the sagacontinues (cardbus driver=culprit?) UPDATE: 99.9% sure it is the cardbus driver yenta_socket

by Tim Blechmann

> What needs to happen is that the card driver author needs to > investigate what is going on, and, if it seems related to the core > PCMCIA core or the socket driver, we need to get involved. he pointed out that it's probably a kernel problem ... we don't want you to solve problems of the alsa driver, but we suspect there is a problem with the cardbus bridge... basics: 1.) the card is a combination of 2 devices: a interface device, either a pci or a cardbus card, and a dsp device, either digital or analog audio ... no problems have ever been reported about the pci interface, so it's likely to be a problem with the cardbus interface ... although i have to admit, that there are only 3 reported systems that have the problems, we experience ... but these systems are completely different ones (p4 / amd64, different cardbus bridges) 2.) the dsp device has an internal matrix mixer, that's independant from the computer (this audio data isn't sent to the computer at all) ... in fact it's possible to set the matrix mixer and unplug the dsp device from the computer ... there are three problems, that show the situation: 3.) if we start an audio application that produces interrupts on the hdsp device, some of the audio data that's been transfered to the computer will be copied or mapped to the buffer of the audio data that's been transfered from the computer to the corresponding output. this could either happen - inside the device itself (i doubt that, since the same driver, the same firmware, the same devices are working without and problems on other linux machines or with windows on the same machine) - inside the cardbus bridge - inside the kernel 4.) if we start the audio playback, the audio playback seems to be distorted: one block of samples will be followed by one block of samples of 0 (zero) ... note that these block are not the blocks of data, that are sent to the hdsp during one interrupt. it's usually 32 samples of sound followed by 32 samples of silence, the driver sends blocks of 64 to 8192 during one interrupt depending on user settings... iirc one sample is 24 bit on the other hand the blocks of silence aren't missing, but the alsa layer will adapt, so that it takes twice as long as it should to play back a soundfile (i'm not sure if it's because of the alsa soundfile player or of the driver) 5.) the reason why we (or at least i) think, that it's a problem with the cardbus interface, is because of jack's output ... jack is both an audio connection interface for different softwares and a very good alsa implementation... using jack as super user in realtime mode, it complains about latency problems: delay of xxx usecs exceeds estimated spare time of yyy; restart ... at the default sample rate of 48000 xxx only about 0.06 % bigger than yyy ... on the other hand, i only get this error on sample rates of more than 32000 ... although the sound at 32000 is distorted, as well... other thoughts i had: 6.) the distortion isn't affected by the block size / the number of interrupts ... it is affected by the samplerate ... that's why i think, there are always 32 samples (768 bit) followed by 32 samples of silence (768 bit zero ???) 7.) the problems occured on different cardbus bridges: ENE C1410 (ico / mandrake) o2micro6933 (myself / gentoo) Texas Instruments PCI1250 (timothy / red hat) i don't know what's the reason for the problems, the latency or the mapping problem, or if there is something else that results this problem ... but since it's a software and not a hardware issue (works fine with windows), i'd like to get into it, and at least try to solve it ... but since i'm neiter a kernel hacker nor i have any idea about the hardware internals, i'd need some help from some people, who are able to help me... i don't want some people to solve my problems for me, but i'd appreciate any help of people, who know what's going on inside the computer... cheers ... Tim mailto:TimBlechmann@gmx.de ICQ: 96771783 -- The only people for me are the mad ones, the ones who are mad to live, mad to talk, mad to be saved, desirous of everything at the same time, the ones who never yawn or say a commonplace thing, but burn, burn, burn, like fabulous yellow roman candles exploding like spiders across the stars and in the middle you see the blue centerlight pop and everybody goes "Awww!" Jack Kerouac

21 years, 7 months

[linux-audio-user] Stuttering via HDSP, Cardbus, Multiface?

by Timothy Polashek

Hello! Has anyone experienced stuttering and very slow playback via RME multiface, cardbus, and Alsa on a notebook, or other computer? I am using the CCRMA low latency kernel on Fedora. How can I fix it? My machine has is 2.8 Pentium 4, plenty of horsepower, and two pcmcia slots (only one works with hdsploader for some reason). I know that I have alsa installed correctly because when I modify the modules.conf file to use the internal sound card and speakers, everything is very fine playing audio back via xmms, audacity, csound(from the command line), and pd. However, once I switch over to the hdsp in the modules.conf file, everything runs but extremely slow... Audacity will take a minute to launch and then will play back a 2 minute file over 5 minutes, the sound out of the multiface is stuttering and distorted. (Very beautiful in a twisted sort of way!) My guess is that the problem is the pcmcia bus is somehow 2000 times slower than it should be.. csound seems to confirm this hypothesis from the command line when attempting realtime synthesis: $ csound -d -o devaudio toot01.orc toot01.sco Using default language 0dBFS level = 32767.0 Csound Version 4.23f03 (Nov 18 2003) orchname: toot01.orc scorename: toot01.sco orch compiler: 32 lines read sorting score ... ... done Csound Version 4.23f03 (Nov 18 2003) displays suppressed 0dBFS level = 32767.0 orch now loaded audio buffered in 1024 sample-frame blocks hardware buffers set to 2048 bytes writing 2048-byte blks of shorts to devaudio SECTION 1: ftable 1: new alloc for instr 1: /dev/dsp: could not write all bytes requested /dev/dsp: could not write all bytes requested /dev/dsp: could not write all bytes requested /dev/dsp: could not write all bytes requested /dev/dsp: could not write all bytes requested /dev/dsp: could not write all bytes requested [1]+ Stopped csound -d -o devaudio toot01.orc toot01.sco Thanks for the help! -Tim

21 years, 7 months

[linux-audio-user] high quality drum samples where to download

by michael heubeck

hi, where can i download very good (acoustic and others) drum-samples? _______________________________________________________________________ ... and the winner is... WEB.DE FreeMail! - Deutschlands beste E-Mail ist zum 39. Mal Testsieger (PC Praxis 03/04) http://f.web.de/?mc=021191

21 years, 7 months

Re: [linux-audio-user] Beginner sampler questions

by davidrclark＠earthlink.net

Tim Hall wrote: > The difference is that a > synth that uses samples uses relatively short bursts of sound, mostly the > attack portion, that the ear uses to differentiate instruments and various > loop portions, the difference is made up with synthesised sound. > This kind of thing exists in soundfonts, usable by fluidsynth and editable > with smurf/swami (in theory) The term "wave table" refers to looking up the sequence of values in a table or an array, not where the samples came from nor how they were created. A wave table synth is a type of sampler synth. There is no requirement that the violin begin with a sample of the attack of a real violin, then transform into an oscillator. This could be done, and the values could be stored in a table, subsequently played out a wave table synth. But this doesn't determine that such a synth has the name "wave table synth." One can record knee slaps and put them into a soundfont, and then play them out their SB Live! card with no synthesis anywhere in this process. Longer notes of extended samples can be created by looping back through the sample. One doesn't need to resort to synthesized sound for the remainder of a note, yet still legitimately refer to their synth as a "wave table synth." Chris wrote: > a sampler accomplishes basically the same > thing as a wavetable synth -- it uses sound samples to generate > tones, doing frequency shifting and interpolation as necessary. I would recommend considering a wavetable synth to be a type of sampler synth. > And as I understand it, the main difference between a sampler and > a wavetable synth is the lack of constraints on the samples used > -- with a sampler, anything at all could be a perfectly good > sample, including samples of almost arbitrary duration (and thus > size). This is not really correct, but an implementation detail. ALL sampler synths, including wavetable synths, have limitations on the samples. Now sometimes you'll see marketingspeak: "Limited only by the capabilities of your machine." There may be no hard- coded limits, but there are indeed limits. > But that brings my > first question -- if you don't own/play the instruments in question, > where do you get the samples? I've done a lot of web searching, > and found tons of drum loops and bass lines that are two measures > long and so forth, but don't find much in the way of e.g. individual > notes on basses. When someone asks "Where does one obtain samples," many immediately advise the questioner to go to the many sample libraries which exist, sample CD's, sample loops, the Internet sources, etc. One can also record one's own samples. One can also simulate instruments through physical modelling. One can also record one's own samples of *whatever can be sent through the audio path*. One can mix samples, including individual notes. For masochists, one can type in a table of values, convert this to an audio format, then use that as samples. One can "rip" them off CD's, videotapes, radio broadcasts (the legality of which depends on the source and the use of the material). There are *many* sources of samples. The main limitation is the composer's imagination. A simple example: Record yourself humming into a mic. Go into a WAV editor and give this a guitar envelope. If you can, build an SF2 font. Now play some guitar melody. Send it through an effects processor. Those who have never done anything like this will learn a lot. Now repeat with some other sustained sound... In some wave editors you can keep just the envelope. So take a spoon and tap on something. Keep this envelope and apply it to your humming or other sustained sound. Rip a track of a CD and isolate one sound that you really like that is a solo part, and edit it down to one second. Apply the envelope to this. Build a font and play something. (Specimen by Pete Bessman can be used to do this sort of thing without having to build fonts.) > you're gonna be > spending hours and hours trying to find samples that work. In my opinion, this is always true for great music. You can either spend years learning how to play an instrument, or you can spend years learning how to sythesize, record, or manipulate samples. Otherwise your stuff sounds like everybody else's and/or you begin to repeat yourself.

21 years, 7 months

[linux-audio-user] what's the best option here? / SosPubs article

by Kent, Gary

Steve, Sorry to be a pest, but does the Hammerfall Lite fall right into the box on Linux? If I'm seeing right there is not the possibility with this card to have playback without a computer since it is nothing but optical ports, with no 1/8" or 1/4" jacks? This Lite card is what I have been leaning towards but want to be really sure about it because I am researching a whole new setup based around a good optical card. Just out of curiosity, what preamps and such do you interface to this card with? Mark Knecht: You said that you didn't use the Lite card much under Linux but that it worked ok, and sorry to be a pest here, but were there deficiencies somewhere with this card that caused you to yank it out? >> I also own a Hammerfall Light. It worked under Linux, but I didn;t use it much. Some time ago I removed it and it's sitting on my desk begging for a machine to go into. That wouldn't be a sales offer, would it? ;>) Many thanks again, gk

21 years, 7 months

[linux-audio-user] snd-hdsp+cardbus=distortion -- the saga continues (cardbus driver=culprit?)

by ico＠fuse.net

Hi all, NOTE: I am cc-ing this to the kernel list in hope someone there might have a better insight in this. For the kernel people who intend to respond to this, I would greatly appreciate it if you could CC me, as I am not subscribed to the kernel list. Summary: snd-hdsp (RME Multiface cardbus pro-audio soundcard) works in Linux but the sound is trashed (distorted). In Windows on the same notebook, everything works fine. The problem has been now reported on 2 completely different notebooks (Acer 1400 with 02 cardbus and my eMachines m6805 with ENE CB1410 cardbus controller). I suspect at this point that the culprit is most likely the cardbus driver (yenta_socket in my case). --------------------- After digging some more I am absolutely confident that my problem with the hdsp_multiface+laptop(cardbus) is definitely not only similar but identical to the problem Tim Blechmann reported in January. The scary part is that my laptop is completely different brandname than his and uses entirely different cardbus (IIRC he has Acer 1400 with the O2 cardbus; I have eMachines m6807 with ENE CB1410 cardbus). I think that this now has to have something to do with the current state of the kernel cardbus drivers (pcmcia-cs has not been updated since December so I would assume that they are no better than the ones that are found in the kernel) and possibly the updated hdsp driver (although not entirely sure on this last one). Here's the current scoop on the problem: Windows XP -- stuff works great, everything as expected. The only thing is that when the computer goes to standby/hibernate, upon resuming the sound is all distorted (just like Tim reported it -- slower, full of artifacts, but you can still discern the original sound's content); this most likely has to do with the crappy BIOS my notebook has (esp. in respect to the ACPI and APIC -- DSDT table is trashed etc.). After distortion occurs, overclocking the computer seems to speed the sound up bringing it closer to the desired playback speed but the artifacts remain. Miller Puckette suggested that perhaps the hdsp is not getting proper clock info from the CPU, something that I have not investigated as of yet as I do not currently have access to an external equipment that would provide Word Clock functionality. Although this also sounds a bit weird as the soundcard works just fine upon first boot (prior to suspending the computer). No matter how many times I reconnect the card and/or mess with it before suspending the computer, everything continues to work as expected. Linux: Mandrake Community 10.0 Kernel 2.6.3 Plenty of RAM and other junk IRQ for cardbus and hdsp is shared on 11 (them sharing the same irq IIRC should be normal behavior) Alsa 1.0.2 and 1.0.3 tested (1.0.2 came with the system, 1.0.3 compiled from source) Latest Jack and alsaplayer packages compiled from source Hw:0 onboard via82xx Hw:1 snd-hdsp ACPI and APIC are disabled due to BIOS issues with the laptop and because even with the pci=noacpi flag in lilo the system still freezes when inserting cardbus. I saw somewhere a kernel patch that would enable use of cardbus with a limited acpi presence (pci=noacpi) but have not tried using it just yet mainly since presence of acpi should not have any positive bearing on resolving this issue (if anything, it would make it even worse due to IRQ shuffling). Modprobing goes without a hitch, pcmcia service automatically starts, the cardbus interface using yenta_socket driver. Snd-hdsp also works without a hitch and configuring the soundcard is all ok (hdsploader, hdspconf, hdspmixer all check-out fine). aplay D plughw:1 (or plughw:hw:1 forget the exact syntax currently booted into Windoze) <soundfile> plays the sound distorted similarly like in Windows after resume. Jackd d alsa d hw:1 with various flags and sampling rates of either 44100 or 48000 works without any dropouts. Just like with Tim, no distortion is coming through until the sound is played. During the sound playback, the distortion is identical to the one during the playback without jackd. Connecting simple clients like jack_metro plays stuff, but distorted. Alsaplayer when connected via jackd also works but again distorted. I have to put the playback at 200% speed to get the right tempo of the song but the sound of the songs singer is now very high-pitched (chipmunk?). Distortion persists no matter what. Reconnecting cardbus and all that works but the distorted sound persists. Hdspmixer shows the sound levels as expected and they reflect the fact that the sound is being played slower than it should be and that it is distorted. Messing with hdspconf during playback makes no difference. Adjusting the sampling rate though does alter the sound of distortion when playing (just like in Tims case), but does not alleviate it. /var/log/syslog lists no complaints and/or problems. (I will check more thoroughly for the boot-time stuff). I am aware of the fact that the BIOS is somewhat trashed (manufacturers fault) but not to the point where machine does not behave normally, esp. in Windows. Some have suggested switching distros, but my understanding is that Tim was running gentoo and I am running Mandrake and were both having the same problem Tim, I believe also used 2.4 kernel without success, as well as pre-1.0 Alsa drivers. --------------------------------------- TODO: 1) provide detailed lspci 2) thoroughly check /var/log/syslog for anything suspicious 3) try pcmcia-cs (most likely wont work as Tim already tried that and it made no difference on his laptop, also the package wasnt updated since Dec.) 4) try playback with an external Word Clock source 5) provide downloadable examples of the distorted sounds 6) Pester alsa-dev, lau, and kernel/pcmcia people to death begging for help :-) 7) Pester eMachines to update BIOS (I may retire before this one happens, though) 8) Something else? I would appreciate any help with this one especially now that we know that the problem is not related to one particular notebook/cardbus controller I will provide additional info as soon as I get home (sometime early next week). Many thanks! Best wishes, Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico/ ico(a)fuse.net

21 years, 7 months

[linux-audio-user] what's the best option here? / SosPubs article

by Kent, Gary

Thanks for this valuable input. So...if the HDSP9652 is not the best ADAT card available for Linux, would the Digi9636 (the HammerFall Lite) be a better option? I found a discussion of this very thing from earlier in the month, but am wondering why the AP2496 limits your options for real-time recording work? The AP2496 has no optical inputs and the DiO 2496 from M-Audio seems to be the only card they have that does have them, and that is only 2 X 4 in/out. I don't seem to have many options in choosing ADAT cards here. Mark Knecht from earlier March 8th post: >> Don't get the stand alone HDSP9652 or HDSP9636 until you make sure the Linux drivers will do what you need. You had said this on a previous post and I was wondering what you meant by stand alone. Which RME card seems to work the best for recording work? I'm looking at that new M-Audio Octane as a possibility. Am I correct in that the RME cards have the converters on them, or are the converters on the preamp/optical out interface? Thanks for bearing with me on this, gk

21 years, 7 months

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Linux-audio-user