Linux-audio-user

linux-audio-user@lists.linuxaudio.org

3 participants
20662 discussions

by Brent Busby

I've been trying to get DOS midi applications to talk to hardware in dosemu, and since I normally use native Alsa for everything, I'm not that familiar with midi through the OSS emulation. My USB midi interface has three subdevices (in/out pairs) which show up in Alsa separately, but I have only one /dev/midi3 (the one that maps to my device), and one /dev/snd/midiC3D0. Is there a way to access individual subdevices through the OSS emulation? -- - Brent Busby + =============================================== + "The introduction of a new kind of music must -- Studio -- + be shunned as imperiling the whole state, for -- Amadeus/ -- + styles of music are never disturbed without -- Keycorner -- + without affecting the most important political -- Recording -- + institutions." --Plato, "Republic" ----------------+ ===============================================

7 years, 7 months

A new, decentralized video streaming system alternative

by Phil CM

#Peertube <https://framapiaf.org/web/timelines/tag/peertube> /beta/ has been launched last week, the source code and docs are here <https://github.com/Chocobozzz/PeerTube>, there are already many open public instances <https://instances.joinpeertube.org/instances>, a lot of which are dedicated to music, like the one hosting mine: https://exode.me/videos/watch/fed67262-6edb-4d1c-833b-daa9085c71d7 (all my videos are here <https://exode.me/videos/search?search=yphil>) This is part of a large-scale "De-google-ify Internet <https://degooglisons-internet.org/?l=en>" operation that has been going on for years, but is now somehow in full motion. Mastodon <https://mastodon.social> has replaced Twitter Diaspora <https://diasporafoundation.org> has replaced Facebook (who ? :p) Peertube <https://github.com/Chocobozzz/PeerTube> will now replace Youtube. That's it, gone :) -- yPhil

7 years, 7 months

[Music] Spring Theme

by jonetsu

Hello, This is a simple, upbeat and straightforward track, largely based on synths, with a touch of acoustic guitar. No angst in this one, or maybe just a little touch of drama. The synth solo was made using Repro-1 seconded by Monique. The piano parts were pianoteq flanked by an electric piano from Discovery Pro. The sequenced foundation was made using Bazille. The guitar part was played on a Shiraki guitar, with some fair bit of treatment. As always, created in Bitwig, mixed and masterized in Mixbus32C. Enjoy. Comments of all kinds welcomed. https://soundcloud.com/nominal6/spring-theme Cheers.

7 years, 7 months

in search of IR response files

by Ede Wolf

I am looking for response files for ir.lv2 - the only convolution reverb so far I have found working within ardour. Looking for natural, long, lush reverbs and thought to have found a good starting point with openairlib.net. However, all big reverbs I've found so far, do produce lots of kind of rythmic clicking (or audible transients) during playback, thus rendering them unusable. The shorter ones partly work, but I just happen to be looking for large spaces. Cathedrals and the like. Why not the whole universe? Any sources or hints, what I may have to look out for? Wrong file format? Or what I may be doing wrong otherwise? It does not seem to be a CPU or load issue at all. And the settings I have been playing with did so far not affect the issue notably. Any ideas? Thanks

7 years, 7 months

Linux Audio Berlin meeting @ c-base 2018-04-04

by Daniel Swärd

Hi all. On Wednesday the 4th of April the monthly Berlin meeting is taking place at c- base. I'll as usual be in the mainhall from 20:00. See you there! :-) Cheers /Daniel

7 years, 7 months

Cleaning RAM

by Will Godfrey

While most issues with computers slowing down are well known, and pretty much a solved problem. There is one that is only slowly becoming noticeable as systems get progressively faster. This is the problem of memory slowing down as it ages. Every time a bit is flipped the underlying structures are stressed and an infinitesimal change takes place that makes it slightly more difficult for the next change. We are of course talking incredibly tiny amounts here so it's hardly surprising that it's not really been noticed up to now. However, it is something that has interested our most {cough} mature {cough} developer, Mary (she of ion trap fame). Once Mary gets interested in something there is simply no stopping her, so we just left her too it, and concentrated on our latest software build problem that only appears on the latest 'felt cowboy hat' distribution. Our Mary is a really methodical person, and she had this weird setup where she was rapidly switching RAM in various bit patterns and then every few hours, stopping, letting it rest then measuring the switching speed. This was fully under software control of course in an Automated Recursive Sequencing Environment. Once she had several sticks showing measurable speed drops (takes several months) she put half of them to one side as a control, then tried out every idea she could think of to get the others to 'relax'. The first was quite obviously thermal - the Applied Heat Neutralising Optimiser - but was inconclusive, and applied long enough it produced dead cells in the RAM. Then it was mechanical - her Vibrating Axis Guided Uniform Exciter idea. Nope, simply shook things apart. A sort of combination of both of these was to attack the problem with a Selective Acoustic Wavefront - quite ineffective unfortunately. Of course, some electronics can be altered with light, so Mary tried using Phased Optical Resonance Notching, although that raised a few eyebrows in the workshop. With no real breakthrough Mary moved on to trying software methods. Unfortunately I'm not permitted to explain how these work, so you'll have to try to work it out from their names. The first was Branched Asynchronous Selective Insert Code. Then there was the Binary Indexed Latency Extractor. The last of these was Consolidated Ram Accretion Pruner. Almost a year had passed now and Mary was showing visible signs of desperation (something quite shocking to witness). Almost in panic she tried combining hardware and software in just about every configuration. This morning she suddenly got up and almost ran out of the workshop without a word. She'd left her last scribbled note out though - it read: V.A.G.U.E Hardware OK, A.H.N.O C.R.A.P software tests A.R.S.E. We just hope she'll calm down by lunchtime. -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song.

7 years, 7 months

Syncing MP4 video files with LTC on CH1, and a BWF WAV containing timecode metadata

by robertlazarski .

Hello! I am a long time Linux user and I used to participate in the LAU group many years ago. A while ago, I decided to do my music "out of the box" with mostly analog gear. I am returning once again to LAU. My goal here is to edit several cameras videos together using different angles, into one video. And use the WAV generated from the F8 as the audio. Actually I plan on having some professional mastering done on the tracks from the F8. But I am doing the video editing part. I need to sync videos from multiple Android phone cameras that are recording some "live" music instrument performances in my home studio. The videos contain CH1 as LTC audio (sounds nasty!) , and a scratch track on CH2 - both coming from a Zoom F8 outputs. The phone camera inputs are from an Irig DUO over USB. The F8 is recording the performances via its 8 inputs, to a BWF WAV file on a removable SD that contains all 8 tracks. I can see the timecode metadata of the WAV using bwftool. I can see all the tracks in Ardour when importing the WAV (compiled latest release from source on OpenSuse LEAP 42.2) and also Audacity. About the LTC and timecode: Lots of people use "Plural Eyes" and closed source software to sync audio and video. I believe LAU will understand why that is not an option for me. Really though I have a personal interest in timecode, and some equipment that uses it. I want to try it. Nice cameras like REDD can accept timecode on a BNC input, or generate it. Phone cameras and even many Cannon cameras AFAIK do not have that capability. While not ideal, it is common I believe to record the timecode as LTC audio on one of the stereo tracks on the camera. These performances are mostly under 5 minutes and I don't expect drift to be an issue - if so I will need a better camera later on. My idea was to startup XJadeo with the "-l" (LTC) option, and receive events via "mplayer -ao jack F8.WAV" . I have it backwards though: The MP4 loaded into Xjadeo has the LTC audio track. The WAV has the timecode metadata. And I have several MP4 files with LTC. Ardour doesn't read MP4 files. I am stuck on the next step here. I am a little weak in my video skills, I am trying to use this project to learn more. Any advice? Well, besides skipping timecode ... I got that a lot already on the "gearslutz" forum :-) .

7 years, 7 months

Yoshi/Zyn Instument envelopes

by Will Godfrey

I've recently pinned down a number of oddities concerning these, and thought what I've learned would be interesting to anyone working on their own instrument patches. The first thing to keep in mind is that amplitude envelopes (particularly release time) set the point at which a note ceases. Frequency/filter envelopes can be shorter, so their effect stops part way, but if they are longer, the last part will be ineffective. Across all three engines, and kits (if kit mode is active) it is whichever is the longest that sets the overall time of the note, and you may well hear others stop if the times are sufficiently different. Also, within AddSynth itself, it is which ever voice has the longest envelope that sets the overall voice time, and if you set voices with very different characteristics you can hear the shorter ones finish before the overall sound stops. Bear in mind, that each voice can also have a start delay set, so you can get a late sound pickup that is then the last bit you hear, even if it's quite short. However, if the start time of one voice is after all the others others have finished it will never sound. This sort of idea works best with 'Forced Release' disabled. An unexpected twist to this, is that taking the combined voice envelope time against the main AddSynth envelope, although attack and decay times follow the above pattern, it is which ever has the *shortest* release that sets the AddSynth time as a whole. This can really catch you out! With regard to the Modulator amplitude envelopes. They don't change the overall time, but if they are shorter than their voice length (or any voice that the modulator is slaved to) the modulation may end a bit strangely. If they are longer, then part of their action will be missed. Finally, there is what I think is a bug (that goes back to Zyn 2.2.1). If an AddSynth voice is enabled, it's amplitude envelope time is active, even if the envelope is apparently deactivated and not editable. Oh, and by default all the voice times are quite long, so again you could be puzzled as to why a sound is longer than you expected. This has always been there, so I don't believe it should be changed. To do so would quite likely alter many existing instrument patches, but do keep it in mind. In the latest Yoshimi commit, there is a new instrument in my 'Companion' bank called 'AddSynth Morph' that demonstrates some of these points - I think it sounds nice too :) -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song.

7 years, 7 months

Soundcard Oscilloscope

by Will Godfrey

Has anyone got any experience of these? Are they in fact any good? I would imagine that running one at 96k, 16bit should give a good enough bandwidth and resolution for checking most audio kit. Which rather begs the question, why are almost all digital scopes only 8bit... unless you spend a fortune on them? -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song.

7 years, 7 months

Sound from/to custom ADC/DAC

by pepijn de vos

Hello, For a university project we're building a custom audio system with our own input and amplifier. We will most likely use an FPGA that communicates sound data over SPI to a Raspberry Pi. On the Raspberry Pi the sound can be further processed by for example Sonic Pi. Sonic Pi uses SuperCollider which uses JACK which uses ALSA. At some point in this chain we need to be able to interface with our FPGA. Initially I thought it would be easy to write a JACK client, and it is. The problem with that seems to be that JACK is in control of the sampling rate. So if I read data from the FPGA into a buffer and the clocks drift, I get overruns or underruns. I found a few potential solutions. What alsa_in and alsa_out do is resample between the two clocks. Maybe a bit of work, but definitely works. There is some business about clockmaster in JACK, which seems to be something different, but maybe I don't understand it. There is a freerunning mode, which makes it OK to do IO in the callback. I'm not sure if this plays well with SuperCollider. It seems that in this case the processing is directly driven by how fast I get data from the FPGA, which is what I want. If all of the above turns out to be bad ideas, I need to look at a different location in the chain. It would make sense to write an ALSA driver for what is pretty much a custom sound card. However, it seems that writing an ALSA driver is orders of magnitudes more complex than registering a callback with JACK. Any ideas what would be the easiest way to get sound from our FPGA into SuperCollider and back? Regards, Pepijn de Vos

7 years, 7 months

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Linux-audio-user