Linux-audio-user February 2014

linux-audio-user@lists.linuxaudio.org

108 participants
83 discussions

[LAU] Focusrite Scarlett 2i2 and KXStudio

by Clifford Dunn

Hi list, This is my first post as I'm a newcomer to pure Linux use. My issue is that I'm using the newest LiveCD install of KXStudio, and trying to use my Focusrite Scarlett 2i2 interface. Sound comes out, and the sound quality is good, but it gets a lot of interrupts. This is visible by the input gains flashing amber, and audio clicks and pops somewhat regularly. I've seen this same problem on various forums, but no solutions. Does anyone have any advice? Thanks! Clifford Dunn Flutist/Composer http://www.myspace.com/clifforddunn http://www.youtube.com/user/beatleboy07 https://www.soundcloud.com/clifford-dunn

11 years, 11 months

[LAU] On the fly bass downmixing to mono (mpd or alsa)?

by Matt Garman

Hi, I started the thread "How is the bass mixed? Per-channel frequency analysis? Histogram?" a few weeks ago. One suggestion that came up a few times was to simply downmix the bass to mono before sending it out. How would I go about this? Currently, this system doesn't have any audio. What I intend to do is send audio out over USB to a DAC. Ideally, I'd like to do on-the-fly remixing of only "bass frequencies" to mono. Ideally the frequency should be configurable, but I think about 150Hz or below is a good start. If possible, I'd like to keep things simple, and have only ALSA and MPD running. As far as I can tell, I don't think I can do this at the MPD level. I saw an example where I can easily downmix all output to mono. I'm not opposed to that, but it would be nice to selectively downmix only the lower (subwoofer) frequencies. Thanks again! Matt

11 years, 11 months

Re: [LAU] On the fly bass downmixing to mono (mpd or alsa)?

by Matt Garman

On Mon, Feb 10, 2014 at 6:08 AM, Shani Hadiyanto Pribadi <shanipribadi(a)gmx.net> wrote: > Now this is where I lost you, how many speakers do you have exactly, > and how many amps for them? Doesn't it mean that you need as many amps > for each speakers you have if you ditch the speaker selector? > > If indeed you want to control it like that then you also need a soundcard > with as many output channels as the number of power amps channels for > the speakers. I currently have three pairs of speakers, and intend to add a fourth as soon as this is all squared away (the wiring for the fourth is already in place, I just need to connect them to speakers). The existing three pairs of speakers are the in-ceiling speakers I've been talking about. The fourth pair of speakers will be traditional full-range speakers. That's one reason why I want to send the full-range signal to the power amp (and in turn the speaker selector). I think you might be confusing the AVR with the speaker selector; they are two different devices. The speaker selector is a passive device (it doesn't require external power). It simply allows an amplified, speaker-level signal to be multiplexed. Without such a device, I would indeed need to have a 1:1 power amp to speaker pair ratio. Here's a concrete example of a speaker selector, Monoprice product 8229: http://www.monoprice.com/Product?c_id=109&cp_id=10903&cs_id=1090305&p_id=82…

11 years, 11 months

[LAU] Steinberg UR22 error -5 after patching

by Patrick

As some other users I'm trying to get the Steinberg/Yamaha UR22 to work. I added Clemens' patch to quirks-table.h and compiled a new kernel (although I noticed later that patching alsa sources and creating a package should've been enough?). The USB LED on the card now stopped blinking after booting just as it does on windows as soon as I installed the drivers there, but still it is not recognized as a (working) sound card. dmesg output is: [ 7803.974700] snd-usb-audio: probe of 1-2:1.0 failed with error -5 [ 7803.994164] snd-usb-audio: probe of 1-2:1.1 failed with error -5 [ 7803.994182] usbcore: registered new interface driver snd-usb-audio I tried to disable onboard/HMDI sound devices, which did not help. (onboard via BIOS and modprobe.d conf) Does anyone have an idea what could help to do next? Patrick

11 years, 11 months

[LAU] On the fly bass downmixing to mono (mpd or alsa)?

by Shani Hadiyanto Pribadi

On 10 February 2014 06:51, Matt Garman <matthew.garman(a)gmail.com> wrote: > Sure, no problem. What I have is: > (1) Linux server with a media collection in a basement network closet > (2) A powered subwoofer on the main floor, with only a single coax > run between the sub and network closet in the basement > (3) In-ceiling speakers on the main floor, all wires terminating > in same network closet Are the in-ceiling speakers connected in pairs? If they are then you need 3 channels, L,R,M. where M = L+R. If you have a soundcard with at least 3 channels and insist on using alsa output then you can use http://alsa.opensrc.org/Low-pass_filter_for_subwoofer_channel_%28HOWTO%29 to upmix from 2 channels to 3 channels in ALSA. (you can ignore the part about low pass). Or you could instead use the jack mpd output plugins and just connect the channels like MPD_out_L -> HW_L -> HW_M MPD_out_R -> HW_R -> HW_M There is also jackminimix for a OSC controlled mixer. I think ecasound can also be used for that purpose. > Note that in this solution, I'm sending the full-range signal to both > the in-ceiling speakers and the sub. The sub has a builtin crossover, > and the speakers can handle the full signal. > > As a side note, if you read the AudioCircle thread above, you'll see I > already have an AVR that can send a bass signal over coax... my goal > is to get rid of the AVR though, as I have more speakers than it has > outputs (would rather not use the AVR and speaker selector), plus I > want to move control off the AVR and onto the Linux server (mpd). Now this is where I lost you, how many speakers do you have exactly, and how many amps for them? Doesn't it mean that you need as many amps for each speakers you have if you ditch the speaker selector? If indeed you want to control it like that then you also need a soundcard with as many output channels as the number of power amps channels for the speakers.

11 years, 11 months

[LAU] [Fwd: Re: Bitwig at long last...?]

by Ralf Mardorf

-------- Forwarded Message -------- From: Ralf Mardorf <ralf.mardorf(a)alice-dsl.net> To: linux-audio-user <linux-audio-user(a)lists.linuxaudio.org> Subject: Re: [LAU] Bitwig at long last...? Date: Fri, 07 Feb 2014 18:19:21 +0100 Mailer: Evolution 3.10.3 On Fri, 2014-02-07 at 16:54 +0100, Jeremy Jongepier wrote: > On 02/07/2014 04:27 PM, Ralf Mardorf wrote: > > Aaargh, I own a nice single coil Ibanez guitar, but I love to play the > > SG's of friends :). The fingerboard of a SG is close to a classical > > guitar, the mechanics are bad, but the "rest" is amazing. It's my > > favourite guitar, just because I grow up as a musician by imitating > > Jimi, I bought this single coil Ibanez. Today I would buy a SG :). > > As the owner of a 1970 SG I can say that the fingerboard doesn't feel at > all like the one from a classical guitar. Also the mechanics of my SG > are in pretty good shape for a guitar of 44 years old. Just to make > clear that not every SG has the same fingerboard or bad mechanics. And > then I'm not talking about how different they can sound. My SG has P90's > but I've owned one with humbuckers too which sounded completely > different. And then I almost forget the player. Robby Krieger sounds > completely different than say Tony Iommi or Angus Young. > On-topic, I think the same way about DAW's. I consider generalizing a vice. :) No comment ;), no, a comment, I suspect you're aware about what I'm talking about ;). The fingerboard is more like a classical guitar, than a Stratocaster or Stratocaster alike guitar and all SG humbuckers from different ages have a special unique sound _and_ a Schaller M6 Mini or derivative definitively is better than an original (Conclusion or what ever is the name) opened SG mechanic.

11 years, 11 months

[LAU] ams-lv2 1.0.2 and a tutorial

by Aurélien Leblond

Hello all, Today I just released the version 1.0.2 of the ams-lv2 plugins. ams-lv2 is a port of alsa-modular-synth in the lv2 format to create modular synth primarily with Ingen. Source code and more information available on the website: http://objectivewave.wordpress.com/ams-lv2/ To celebrate this release, I created two videos: The first is a demo of 3 synths created with ams-lv2 and Ingen (a bass, a lead and a pad) soloed and inside a mix: http://www.youtube.com/watch?v=LWfF71NerkQ I created as well this first tutorial on the basics of ams-lv2: http://www.youtube.com/watch?v=IuQZajaSw6M I would as well like to take advantage to thank a lot of people who helped with this project: - David for his patience to answer my 1000+ questions and for all his work on lv2 and Ingen - Robin for creating the sisco.lv2 plugins, they are featured in the demo and they are a great help when creating modular synths - Harry for the dials used in the GUI of ams-lv2 - and finally Fons for creating the great alsa-modular-synth in the first place - the whole Linux Audio community in general I really hope you enjoy these plugins and find them useful somehow! Aurélien

11 years, 11 months

Re: [LAU] ams-lv2 1.0.2 and a tutorial

by Al

CV is different than MIDI. Atte <atte(a)youmail.dk> wrote: >On 02/09/2014 04:31 PM, Paul Davis wrote: > >> How would the VST spec allow for control voltage? >> >> (hint: it won't) > >Not sure what you're talking about. > >AMS when ran standalone is (in my setup) controlled by midi. That's what >I thought might be possible with AMS as VST. But if you say so... > >-- >Atte > >http://atte.dk http://modlys.dk >_______________________________________________ >Linux-audio-user mailing list >Linux-audio-user(a)lists.linuxaudio.org >http://lists.linuxaudio.org/listinfo/linux-audio-user

11 years, 11 months

[LAU] Pulseaudio and Jack

by Max

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Which of the suggested option is the one people here on the list favour? http://jackaudio.org/pulseaudio_and_jack -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.14 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/ iEYEARECAAYFAlL1L3UACgkQ3EB7kzgMM6Ln1ACfVyT8RXVcBzHiIkyQJU9LWBWK 4p4Anj08pRcsv5ytIxvtrebWhbkQulGA =JPXo -----END PGP SIGNATURE-----

11 years, 11 months

[LAU] How is the bass mixed? Per-channel frequency analysis? Histogram?

by Matt Garman

I have a collection of FLAC files, all ripped from my CD collection What I would like to do is run an analysis across all the music to determine how the bass/lower frequencies are generally mixed. For example, how much content below (for example) 150 Hz is on the left channel versus the right channel? I'm not sure if "histogram" is the right word, but in my mind what I'd like to see, per-channel, is something like this: 150--125 Hz: x samples 125--100 Hz: y samples 100--80 Hz: z samples ... Then I can look at the two channels of a song, and if the histograms are approximately the same, I can assume the bass was mixed equally to both channels. I am a programmer, and thought it would be easy to quickly hack something up that would do this, but I have no experience with signal processing, and as I started reading about this, I quickly got in over my head! So I was hoping there might already exist a tool that has this functionality. Note that I don't need any kind of graphical output, as this needs to be wrapped up in some kind of batch processing script---I have about 11,000 files to analyze! The motivation for this is: I have a hardware DAC (digital audio converter) in one part of my house, and a subwoofer in another. There is a single coax run between the DAC and subwoofer, so I can only send one channel. If the overwhelming majority of my music has the bass mixed equally, sending only one channel isn't a problem. But if I choose the "L" channel to send to the sub, and much music has the bass mixed only to the "R" channel, then I won't be able to hear the low frequencies. I want to find out how often this might happen. Thanks, Matt

11 years, 11 months

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Linux-audio-user February 2014