Linux-audio-user March 2018

linux-audio-user@lists.linuxaudio.org

47 participants
41 discussions

M-Audio Fast Track Pro: unreliable, distorted recording

by Lewis Pike

I'm the (un?)lucky owner of an M-Audio Fast Track Pro USB audio interface and I'm having some serious problems getting this device to record audio reliably under Linux. I've been using arecord and occasionally Audacity for all of my testing. My problem is this: Recording a take works about 80% of the time. In the remaining ~20% of cases, the captured audio is extremely loud with severe digital distortion. Once this problem shows up, it persists for any subsequent takes. The only way I've found to make the problem go away, at least temporarily, is to power-cycle the Fast Track Pro. I considered the possibility that this particular device might be defective, but it seems to work wonderfully under Windows. I'm calling out to other Fast Track Pro users in the hope that someone out there has encountered the same problem and better still, found a solution. Any suggestions at all would be greatly appreciated! .lewis

5 years, 8 months

Zoom U-25 Audio Interface

by Max

Nowhere on this page it says that this device is class compliant, but it claims that there is no driver necessary for OS X and iOS which seems a good indicator for it. What do you you think of this interface? https://www.zoom.co.jp/products/production-recording/audio-interfaces/u-24-… 89 € there is also a version with digital interfaces: https://www.zoom.co.jp/products/production-recording/audio-interfaces/u-44-… 149 €

5 years, 10 months

Save the date: Linux Audio Conference 2018

by David Runge

[Apologies for cross-postings] [Please distribute] Save the date: Linux Audio Conference 2018 The 2018 Linux Audio Conference will take place at c-base, Berlin 7th -11th June 2018 The LAC 2018 will feature a full program of talks, workshops and music. The official call for papers and works will be sent shortly after the winter break - so use the quiet of the holidays to think about possible submissions. All relevant information will be made available on: http://lac.linuxaudio.org/2018/ The LAC 2018 Organizing Team -- https://sleepmap.de

5 years, 11 months

Help getting sound from Cakewalk UA-25 EX to Ardour

by Hernán Labbé Grünberg

Hi everyone, This is my first email to this list. Apologies in advance if I break any conventions I am unaware of. I am a long-time musician, but just getting acquainted with audio in linux. I have a very simple goal of getting sound from a plugged instrument to my computer via a usb-interface, the Cakewalk UA25-EX, which I know works with linux. So far the instrument's output is received by the interface, and the interface is recognized by the computer. I asked in the ardour forums on how to get ardour to recognize that the input for audio is the usb interface. They suggested <https://community.ardour.org/node/15577> I use ALSA instead of jack to connect to Ardour. The interface of Ardour allows me to select the Cakewalk interface as input and output, but when I play the instrument I see that while the interface gets the signal, the audio is not received by Ardour. If I change the input to my laptop's microphone, then I do get a signal. The problem, then, lies in the connection between the interface and Ardour (or the computer?) I checked the Alsamixer levels and they are not at 0. The advice from the Ardour forum was to contact this mailing list, as I probably need some help checking the connection of the usb interface to the computer. Thanks in advance

5 years, 11 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

6 years

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 :-) .

6 years

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.

6 years

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.

6 years

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

6 years

ECC or non-ECC ram ?

by Ivan K

I can't recall ever seeing this discussed here. I have always purchased the non-ECC RAM because it is less expensive, but do people have opinions/information on this? Does anyone here always purchase ECC RAM? Thanks.

6 years

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Linux-audio-user March 2018