Linux-audio-user June 2004

linux-audio-user@lists.linuxaudio.org

140 participants
148 discussions

[linux-audio-user] Good mixer for Delta 1010LT...

by Russell Hanaghan

Hi folks, Finally took the plunge and bought a 1010LT... I have it working on my previous MDK 10 setup but still using the old 2.4.22mm2 kernel. Can't seem to get 2.6 to work quite as faithfully yet. Anyhooo, I was really happy with QAmix before as a nicely laid out mixer app but alas...it seems it is no good with the Delta?? Is there anything out there other than the regular Alsamixergui or Gnome alsa-mixer that has some flexibility here? R~

20 years, 6 months

[linux-audio-user] psychoacoustics

by Matthew Barber

->Psychoacoustics is just a bit too frightening for me I'm afraid. Although I would urge everybody to read Yost's book and if they like this stuff try Moore's psychoacoustics book. Prepare to be amazed though!<- And if you really want the skinny, try something like "Signals, Sound and Sensation" by William Hartmann... it's mostly about signal and signal processing, not as much about psychoacoustics per se. But it's a-pretty-pretty good. Matt

20 years, 6 months

[linux-audio-user] [ANN] Specimen 0.4.0 "Smooth"

by Pete Bessman

Specimen is a MIDI controllable SoftSampler for GNU/Linux systems. Features added since 0.3.0 include: * Portamento * Per-parameter LFOs with delay and attack * LFOs may be "global" (voice independent) or per-voice * LFOs may be constrained to oscillate between 0 and 1 instead of -1 and 1 * ADSRs have gained delay and hold phases * Most parameters can be velocity sensitive to a user-specified degree * LFO/ADSR amounts for pitch are now entered in half-steps * A sample's pitch may be adjusted within a user-definable range You can download the latest tarball (some assembly requried) from www.gazuga.net, or directly below: http://www.gazuga.net/files/specimen-0.4.0.tar.gz The focus of the 0.4.x series is the GUI, so if you've got a gripe or a wish about that, now's the time to pipe up. As always, my inbox is wide open to suggestions and bug reports. -Pete

20 years, 6 months

Re: [linux-audio-user] Strange XMMS persistance

by David Baron

On Monday 21 June 2004 03:03, linux-audio-user-request(a)music.columbia.edu wrote: > Update to the latest xmms-jack plugin and xmms? I have them and don't > see this problem. I have whatever is on Sid.

20 years, 6 months

[linux-audio-user] Re: producing a drum sample library for hydrogen

by Juhana Sadeharju

>From: R Parker <rtp405(a)yahoo.com> > >I have been discussing the design of a drum sample >library for Hydrogen with Comix (Hydrogen author), Such a sample set would be useful outside of Hydrogen as well. How about "public domain" license? We just have to make sure people can get the originals if they wish. Is there GPL equivalent for samples? Here are my non-professional tips: Tip 1: Listening to existing libraries may not help. It could be that most drum samples are not done as good as possible. Tip 2: Provide as many snare (say) samples as possible. If you want create a limited preset for Hydrogen, you can do that later but leave an option for us others who want a randomizing element to the drums. Press record and hit the snare for an hour in various ways. Tip 3: Group the samples so that all samples from one drum kit are together. Tip 4: No effects whatsoever at recording time. Reverbs, gates etc can be added later, but cannot be removed if they already are there. Tip 5: If you record stereo, make the other channel so that it can be played mono. Channels should not be mixed for obtaining the mono version. First mic is for mono and the second mic is to get a different perpective from the same sound. The first mic should be placed to best position. Snare is often recorded both from top and bottom (where the spring is heard best). Stereo effects can be done with these later; the mics does not need to be in a true stereo configuration. Tip 6: When two close mics are used to record a drum, check any phasiness effects by mixing the mics. Fix it by moving the second microphone. It may not be possible to fix it later. Tip 7: The overall stereo mic could be in a human-ear-stereo configuration above the drum kit. Those mics should be kept in fixed position during all recordings. So, for each drum we would get both the mono, the second mic, and overall stereo recording --- four channels in total. That should give freedom at mix time. Tip 8: When samples are trimmed, don't cut decays too early. I use spectrum view in editing long decays because the spectrum display can see throught the waveform noises. Listening to the sample helps but not many editors have +64 dB output gains (Audacity has ok +36 dB max output gain). Leave good space to the start as well; trigger point can always be adjusted later in software. >mic pattern. When sequencing ride cymbals they need to >bleed from the listeners Hard left side across the >stereo field and into the Hard right side. If we don't >treat cymbals as stereo images they could fail to >sound like a live player when sequenced. Why? Do you sit on drummer's lap in concerts? Does your drummer's cymbal walk from left to right when hit? :-) As far as I know the cymbal sound (one cymbal) comes from one fixed position. If I stand at 50 feet from the drummer, the sound is practically mono. Any left to right movement coming from speakers is just an effect. Effects can be done later. Though, use of two mics makes panning effect less dull. >from 1 in 2 out reverbs. In addition to dry mono snare >drums, we intend to produce a number of reverb >processed (wet) stereo samples. I can help here and apply some of my own reverbs. I can also help in trimming and in labeling the samples from the raw recordings (if you don't have time). I can also provide space from here ftp.funet.fi. >The drums will sound great because Bill, >Bobby and I know what we're doing. If we don't know >what we're doing someone should shoot us--we've got >about 90 years of combined experienced with playing >and recording drums. I have recorded zero seconds of drums, but giving "tips" won't harm anyone. If I'm mistagen, I'm sure you see it instantly. Juhana

20 years, 6 months

[linux-audio-user] Strange XMMS persistance

by David Baron

Once connected through the "biojack" interface to jack, XMMS is stuck in place. If one closes it and later opens another instance, this one will connect with jack. BOTH remain as active processes. Do this enough times and of course, the whole thing falls. How might one correct this problem?

20 years, 6 months

[linux-audio-user] rosegarden silent

by David Corkill

Hi folks If anybody can help with this little problem of mine I would be most grateful. Rosegarden-4-alsa, installed on Mandrake 10, is silent. There is no audio output when playing midi tracks. My soundcard is ensoniq CT5880, the driver is snd ens1371(alsa driver). My system is (I hope!) configured for alsa sound support, from the original OSS default. The Rosegarden > Studio > Manage MIDI Devices has 64:0 ES1371 listed as play and record devices. Importing bank, program and controller data from a rosegarden file does not help. Rg is silent on playback of both created and imported midi tracks. Timidity and XMMS do have audio output. I have just installed RTSynth in the hope that a soft synth will provide a means for Rosegarden to play midi but I don't know how to configure it for recognition by Rg. Any information would be appreciated, although please keep in mind that I am a newbie. Note - I have googled about for information on the problem for many hours, only to find that other newbies are also stuck with this situation. I could find no solutions posted anywhere. David

20 years, 6 months

[linux-audio-user] [ANN] BLOP LADSPA Plugins 0.2.8

by Mike Rawes

Hi After way too long, a new release of the BLOP LADSPA Plugin set. Orginally named Bandlimited LADSPA Oscillator Plugins, but there's more than oscillators... they're more useful in a modular host, such as Spiral Synth Modular, gAlan, Alsa Modular Synth etc. Plugins in the set: New since last release: * Quantiser (quantise to arbitrary values in a range) * Signal Tracker (envelope follower, sample+hold, track+hold, ...) * DAHDSR envelope generator * Difference and Ratio utility plugins Existing ones, with less bugs I hope: * Bandlimited DCOs (wavetable based): - Sawtooth - Square - Pulse with variable width - Triangle with variable slope (Saw->Triangle) * Analogue-style step sequencer * Fast'n'Dirty Moog 4 Pole implementation * Two ADSR envelope generators * Random wave generator * A mono amplifier * Non-bandlimited square and pulse clock oscillators * 1V / Octave CV to Frequency converter * Control rate to Audio rate interpolator * Sum and Product utility plugins Some plugins come in control and audio rate variations. RDF metadata is also included if your host supports it. All plugins pass the demolition test: http://www.ecs.soton.ac.uk/~njl98r/code/ladspa/demolition.html Website: http://blop.sourceforge.net - Myk

20 years, 6 months

[linux-audio-user] Sample Rates / Sample Sizes

by davidrclark＠earthlink.net

Starting a new thread originating as: Re: producing a drum sample library for hydrogen I can easily tell the difference, as can many, between 16- and 24-bit samples, but not equally for all instruments, of course. I highly, highly recommend using 24-bit samples for any widely used library, especially given the amount of processing that most of us do. I myself also use 96,000 samples/second because I always do quite a bit of processing, but that's a lot tougher sell for a library. --------------------- Someone said that 44,100 samples/second is a quirk. It's not merely a quirk. 44,100 = 2^2 * 3^2 * 5^2 * 7^2 which has some useful properties compared to 48,000 = 2^7 * 3 * 5^3. Lots of small primes that differ is sometimes better than a bunch of powers of two, despite the advantage primarily associated with FFT's that some focus on. Looked at from a prime number standpoint (therefore certain types of algorithms), 48,000 is truly bizarre compared to 44,100. 96,000 is yet ANOTHER power of two on top of 48,000! Whether a particular sample rate is a good one or not is dependent upon more than just what is customary and standard.

20 years, 6 months

[linux-audio-user] running gag: m-audio quattro

by metafor＠gmx.net

hi list i followed the instructions to get the card working (thanks josh) but i'm still don't get any results. i've patched the kernel 2.6.5-mm4 with the two patches mentioned on the list. i can load the snd-usb-audio module, dmesg means that it's registered (module) , but i don't get this error-messages: config 1 interface 1 has no altsetting 1 config 1 interface 2 has no altsetting 1 config 1 interface 4 has no altsetting 2 config 1 interface 5 has no altsetting 2 config 1 interface 7 has no altsetting 2 config 1 interface 8 has no altsetting 2 is there a way to see if the pc atleast realise that there is a card conected? any module which has to be loaded beside snd-usb-audio? by the way i can't compile nvidia with this patched kernel, running nv therefore, anybody already solved this OT? are this 2 special patches http://marc.theaimsgroup.com/?l=linux-usb-devel&m=108178375704810&q=raw http://marc.theaimsgroup.com/?l=linux-usb-devel&m=108178386018353&q=raw allredady integrated in some standard kernel release and also the mm4? i could then go directly to 2.6.6 or 2.6.7 which would solve also my nvidia problem. sorry to bother you again with this quattro issue, but if someone can give a simple step to step advise, it would be really helpfull. thanks marc

20 years, 6 months

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Linux-audio-user June 2004