Linux-audio-user

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20702 discussions

Re: [linux-audio-user] Just how bad is USB audio?

by Patrick Shirkey

On 1/25/2003 - 18:09:59, Dan Armak said: > > CPU utilization is the main problem. >From my experience it is not actually the main problem. Rather it is conflicts with the vga device interrupts which cause sound problems. Saying that, I have successfully recorded 4 channels over nfs through my usb device with arecord as normal user. That's about as interrupt intensive as I have found so far. I have also had it going at 64 bytes/period and had no dropouts AFAICR. If you want a reasonably priced option then a usb device should see you right. If you want to buy a cheap one then send me the cash and I'll post it to you. Warranty is still valid, just that you will have to post it back to me first. (Takes about 3-4 days). -- Patrick Shirkey - Boost Hardware Ltd. For the discerning hardware connoisseur Http://www.boosthardware.com Http://www.djcj.org - The Linux Audio Users guide ======================================== Being on stage with the band in front of crowds shouting, "Get off! No! We want normal music!", I think that was more like acting than anything I've ever done. Goldie, 8 Nov, 2002 The Scotsman --

23 years

[linux-audio-user] Just how bad is USB audio?

by Dan Armak

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I'm considering buying a USB sound card for my laptop. (I have no desktop box worth talking about atm.) Everyone says a USB card is a Bad Thing, but I've found few numerical measures, and I don't have much of a choice anyway (unless someone releases a firewire card, or a pcmcica one under $1000)... If someone can point me to some more concrete data, it would help me make my choice. (Which is basically to buy or to suffer crappy sound, as I see no real alternative...) CPU utilization is the main problem. I probably wouldn't count as a hardcore gamer, since I have a p3-900 with 256mb ram and it's enough to run all the games I need (mostly ps1 emulations, but some sims/fpss too). So what performance hit should I expect from using a usb audio device? The only review on the net that actually mentioned numbers was one of the Extigy on Tom's H/w Giude, and it said 20% cpu of a p4-2Ghz, but that was using 6 or 8 channels with special effects requested by something on the order of UT2003, which is out of my league anyway. I don't have or plan to have any other high-bandwith devices on my usb (1.1) bus, just a keyboard/mouse/joystick. Also I only have 2 speakers atm, so I'll only be streaming two channels. Therefore bandwith shouldn't be a problem as I understand it. OTOH I'd want to do some none-cpu-intensive work (writing, reading) while listening to mp3s and copmpiling stuff in the backgruond. I would probably accept some slowdown in the compiling, as long as the audio didn't skip and responsiveness wasn't affected (I already use a kernel with all the low latency/preempt/O1 scheduler etc. patches and have a prelinked system because I really hate responsiveness issues)... Finally as to the card itself: AFAICS there are the costly ones (Extigy and Audiotrak Maya 5.1 usb), which have lots and lts of features most of which I don't really need, and some cheap ones like the iMic or Xitel Hi-Fi-Link which only have stereo rca input/output. The latter type is much closer to what I'd be using of all those features, but unfortunately I cannot find them in any shops here in Israel, and if I order over the internet I'll have to pay for shipping and won't have warranty coverage. So I'd be willing to pay for the maya card if I cuold be sure I'll be using it well, preferrably also when I build my next desktop system in a year from now. Any recommendations, info etc. will be most welcome :-) - -- Dan Armak Matan, Israel Public GPG key: http://www.gentoo.org/~danarmak/danarmak-gpg-public.key -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.1 (GNU/Linux) iD8DBQE+MphWUI2RQ41fiVERAsISAJ9Xxgimu2DETubVEdQ5faUUqjQTOACfWd+E jT5Lw1WKA2aBteKTHPmn7Tg= =wUPc -----END PGP SIGNATURE-----

23 years

[linux-audio-user] MTC synchronized recording - possible?

by Olivier

Hi all, I have 2 MTC synchronized 4tracks that have 8 tracks together (mono). I would like to transfer all of the tracks to my computer so I can play around with some effects. I only have 1 input on my soundcard, so I was wondering if I could record one track at a time using MTC. This way all tracks would still be synchronous after transferring, because they are all recorded with the same MTC master. Is there a program available that can do this with Linux? Is this anyway a viable idea? regards, Olivier

23 years

[linux-audio-user] Re: [linux-audio-dev] sooperlooper goes PD

by Steve Harris

>It might be nice if this sort of functionality melded into ssm, where >you could see the loop displayed around the circle... There would be a >visual cue then for when the loop was about to repeat, which can be hard >to find when the loop is > 10s long... Yes, this is like dream land. I've been thinking of a way of squeezing enough spiral loops type displays onto a screen so that I can watch 4+ loops at once. You either make them side by side, and they all have to be small or make them coaxial, but then one is small and one is huge. Spiral Loops is just the most natural way to work with synced loops. - Steve

23 years

[linux-audio-user] Control|24

by Linux Rob

Hey! I have a load of questions and I hope it's not too much trouble to get them answered. Does Ardour or any Linux program support Digidesigns Control|24? If not, how would it get support? What control surfaces are supported? I know a mackie controller is supported... Also, can Ardour recieve 96Khz from the HDSP 9652? What's the status with the HDSP 9652 driver? Where can I check the status of the HDSP 9652 driver? thanks... rob

23 years

[linux-audio-user] Soundless ALSA Newbie

by Sakari Aaltonen

I have been trying to get ALSA (0.9.0rc5) working with an on-board ALC650 chip, ALSA because that is what the chip's manufacturer, Realtek, offers on their website. (I assume that this means that the sound code in kernel 2.4.20 won't work with the chip.) I compiled everything and made 'insmod' with 'snd' and 'snd-via82xx', with no error messages that I could see (a lot of other modules were loaded automatically.) I also ran the make-dev's script. Yet, no sound. So I looked at Sound-HOWTO's. They stress the importance of seeing kernel messages about sound. Well, I don't see any. Then, because ALSA apparently mutes everything by default, I tried compiling two mixers from the ALSA site. No success, because I was missing some libraries. Also, I was pretty confused about what version of ALSA they were supposed to work with. Of course, I can start looking for those libraries, but while I do that, could some kind soul please explain how I can find out whether my actual sound setup is likely to work at all? Or would anyone know of a mixer program that would not need libraries that I will not have? If it matters - my sound sources are DVD and TV... Thank you, Sakari Aaltonen

23 years

[linux-audio-user] Recording from an SBLive's LiveDrive input in Ardour

by JW

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, (I hope it's ok to post messages like this to this list? If not, please steer me in the right direction...) I'm using Ardour on SuSE 8.1 (the version that comes with SuSE, not CVS). I have a Creatoive SoundBlaster Platinum, with the original edition of the LiveDrive (No IR). Using alsa 0.9.x that also came with SuSE. Inside Ardour, I can record from the sound card's microphone just fine. By setting "Capture" to the line-in (on the PCI card) with alsamixer, I can then record from the PCI card's line-in just fine, without changing any settings in Ardour. I might point out that I am using headphones connected to the LiveDrive, and not using external speakers at all. So, I thought I could just set LiveDrive-Line-In to "Capture" in alsamixer and get going. When I do that, I can hear my guitar in my headphones just fine, just like I can with the line-in or mic-in. However, no sound gets recorded. I am not sure what this means. Ardour/Jack has to be listening to alsa somehow, or I wouldn't be able to shift between mic-in and line-in with alsamixer. Why, then, does moving alsamixer over to the LiveDrive's line-in produce silence in ardor? Remember, it's not sielent in the system, as I can hear it in through headphones. It's just that Ardour doens't record it. Any ideas? Does anyone out there use a LiveDrive with ardour? Thanks! JW - - -- - - ---------------------------------------------------- Jonathan Wilson Cedar Creek Software http://www.cedarcreeksoftware.com -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQE+MIZDQ5u80xXOLBcRAjwkAKCMf9SnuDyG9ePJKHoroX5DEBZsTACdEyT9 QefQohXNIm7NixDSPIlQeq0= =l6jz -----END PGP SIGNATURE-----

23 years

Re: [Fwd: Re: [linux-audio-user] i'm a graphic designer - use me! ;-)]

by iriXx

>>NI Spektral Delay (never entirely sure what its going to do, but its got a >>lovely interface - oh please please someone code something like this for >>GNU/Linux! ;-) > > > It's called freqtweak and it rules. http://freqtweak.sourceforge.net/ > > - Steve oh yes indeedy!!!! my prayers have been answered :o).... thanx! m~ -- iriXx www.iriXx.org copyleft: creativity, technology and freedom? info(a)copyleftmedia.org.uk www.copyleftmedia.org.uk _ ( ) ascii ribbon against html email X / \ cat /dev/sda1 > /dev/dsp *** stopping make sense ***

23 years

[linux-audio-user] Bundled reply of you guess what :)

by Roel de Wit

Hi, > The Nyquist frequency is equal to (sample rate / 2). > Logically, this is because the A/D converter needs at > least 2 samples to detect 1 cycle at the Nyquist > frequency. Frequencies higher then the Nyquist will > actually appear as lower frequencies to the A/D > converter and is called aliasing. Normally, A/D > converts run the signal through a highpass filter > first to avoid these problems. > I guess the interesting problem is aliasing. Right? Yes but did you ever actually here problems at 48 Khz yourself ?? I didn't.. The sounds for the piano where sampled at 48 Khz but the Audigy 2 which was used for this (not the best ADC arround) internally processes the input sound at 192 or 96 Khz and probably downsamples it in hardware with anti-aliasing (24 bit accuracy I think). Anyhow theoretically there might be a problem but practically I've never encountered it. > Please do not think you need to change ANYTHING about your samples to > sound like Bardstown. Every piano sounds different, so every library will > sound different, so every mp3 will sound different. I know of many game > developers and people writing scores for TV and movies that would seem to > disagree with your opinion, but we're all entitled. Right? ;-) If I would change something it would be in an additional soundfont (that's if the samples themself are changed) for the Audigy (2) initially. Regarding 44.1 Khz samples, we can downsample with anti-aliasing during the installation process if needed.. When not changing the sampled themself changes can be made in realtime in hardware/software with filters and EQ settings. I agree that every piano sounds different.. Even if you bought two the same acoustic instruments the sound is likely to be slightly different (hearable). What you would prefer when you would buy a piano like Grandest Piano if it offered soundbank in multiple formats ? One installer that installs all formats in one go and might write 1.5 gig of soundbanks or more to your harddisk or.... One data file for which somebody has to pay and a lot of free downloadable installers for each format ? I'm asking since we can't make an installer with the current installation system which allows you to install in several formats since we rely on a 'patching' system (like DeltaX under Linux) for part of the installation proces.. If a person would select multiple formats we'll need a lot of patchfiles to cope with this. > I disagree that the Bosey is 'filtered' or 'dull', but I do agree that it > is not as bright as your library. Don't get fooled though, and listen to > what others say. I think there is no right answer here, like there is no > right piano. People tend to get blind because a piano is lot's of Mb/Gb, read some propaganda about a soundbank (e.g. an multivelocity sampled, multilayer, 127 realistic velocity levels using system XXX for obtaining the YYY acoustic realism effect blah blah..).. Just read the text on www.studioax.com and you'll see what I mean. > From what I can hear, you are making great progress, especially for a > soundfont in less than 200MB. (Or whatever it ends up being.) I'd suggest > you do your best work and make the library sound as much like the original > piano you sampled as possible. There will be a 220 MB version at maximum (initially) and several smaller versions down to 40 MB (only one sampled velocity layer) for maximum compatibillity with any system.. I'm currently looking at Halion and EXS for destination formats.. From what I saw I think that EXS is no longer available for Windows.. Does anybody know if this is correct... Also when anybody on this list would buy Grandest Piano, would you use it under Linux or Windows ? If you want to use it under Linux which softsynths do you think would be good candidates ? I can't promise I'll support any of them initially but there might be free update patches (or free downloadable installers if we go for the data file + installer). > In real life the Bosey doesn't sound like the German Steinway, which > doesn't like the American Steinway, which doesn't sound like the Young > Chang, which doesn't sound like the Yamaha, which doesn't sound at all like > the Baldwin that Bruce Hornsby plays. Why should your library of a sampled > Yamaha sound like Kips Bosey? I don't think so! Not at all! It doesn't have to sound like a Yamaha even.. As long as it sounds like a piano.. If there should be variations which should allow you to easily switch between several types of piano while using the same soundbank that's only a pro. > If you want to hear a really dark piano, go to digital shootout and > listen to any of the Malmsjo's. They are very dark, and yet there are people > that love them. This is the way that piano sounds in real life. I think they > work for certain types of music. Once v0.2 is complete, which I hope will be the last beta containing only the biggest version of the piano I'll look into adding variation soundbanks. > I think your samples will work well also. Just keep up the good work and > listen to lots of libraries. If pointing you or others towards sites like > the Digital Piano Shootout helps, then great. If not, don't worry about it. It surely helps.. It basically allows me to listen to several types of pianos (whether I like them or not) and see if it's possible to make variations similar to well known pianos.. Not that these would sound exactly like them but it allows for pianos with different types of sound. --- Roel / Utopia Sound Division http://www.utopiasd.com

23 years

[linux-audio-user] Pianotest1 recording as requested by Mark and answers to questions by others.

by Roel de Wit

Hi, I've uploaded the MP3 recording of this MIDI file to http://project-portability.foad.nl/mp3/pianotest1.mp3 Note though that it was sampled from Grandest Piano beta v0.1 and thus will not neccesarily represent the v1.0 release of the piano. Next week two new sample layers will be added and the amount of velocity layers will be increased to 127 (which will fully replace the current not so perfect ones). Also the release values of each note (or layer.. not sure yet) will be adapted.. I'll provide the website with a recording from v1.0 when it arrives so I prefer to keep this recording low profile. The recording was made without any reverb active or any post-processing so that it shows the actual samples the best way. Regarding 16 bit vs 24 bit.. I personally don't see the use of 24 bit samples too much yet. When mixing them is concerned the more bits the better. Regading looping etc.. The samples are all unlooped with a maximum of 16 seconds (which is often used). > Yeah, but what is a tiMIDIty version, explicitly? If you want to > please the Linux community, just send us 88 (more?) samples, one for > each key, and chuck the looping. Just let them all ring out till > silence. Better yet, 4-5 samples for each key at different striking > pressures would be great. Don't worry about how or what software will > sort this all out, we'll figure out something. With all the sampled velocity layers and huge amount of filtered velocity layers one soundfont won't simply work on any 'claimed' soundfont compatible softsynth or device. E.g. LiveSynth Pro a very good SF2 softsynth can't handle the piano correctly (yet). Our Utopia Live! soundfont (GM/GS) can't be played back (or even loaded) on anything else than SB Live!/Audigy hardware. It's the main drawback of 'complicated' soundfonts.. I have all the velocity layers sampled at 88 notes at my harddisk but the Audigy is stuck at around 330 MB (maximum loadable soundfont size under Windows 2000/XP). That's why I choose for around 25-28 samples per velocity layer. The size of the full 88 key sampled piano with 5 sampled velocity layers would be at least 800 MB assuming I'd stick to 48 Khz stereo samples.. It might be done in the future, but not now, unless there is a demand for it. I don't think the difference would be noticable too much.. I'm willing to add around 3 samples to each layer if the beta testers think this is needed.. > A good idea would be to double all of these samples, one recorded > with the sustain pedal open, and one closed. This would allow the > creation of the interactive effect between the strings which is > so noticeably lost in most sample sets. This makes the set too > large you say? Not even. I wouldn't sweat if it was a couple of That would mean that there would be 3 instruments (MIDI wise) which a composer would have to switch between.. I'm not sure if this is usefull. > GIG even. Hey, it's only one DVD. Lets start putting today's machines' > capabilities to good use. In fact, these samples should be AIFF, WAV, > or raw, 96k/24bit, mastered digitally with mics and equipment that can > register frequencies approaching the nyquist. That way, when we lower > notes an octave or two for effect, we will still have high end sizzle. Try to explain to me what the use is of 96 Khz samples ? It's just marketing nothing more.. 48 Khz makes it possible to represent up to 24 Khz of sampled frequencies, far about what you can hear. More important is that the device or softsynth used mixes at higher bitrates and that the DACs are of high quality (unless you render directly to a .WAV/whatever file). > Yeah, make that, and I would buy it. $20.00 - $30.00 say. As long as > there were no restrictions through licensing, other than, of course, on > the resale of the package as a sample set. No restrictions, other than you are not allowed to distribute any part of what's in the package that will be sold. --- Roel / Utopia Sound Division http://www.utopiasd.com

23 years

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Linux-audio-user