Linux-audio-user February 2015

linux-audio-user@lists.linuxaudio.org

69 participants
58 discussions

OT? Downloaded youtube videos with subtitles

by Ivan K

I was very excited to find a video of the complete Tristan and Isolde on YouTube recently, complete with English subtitles. I used the youtube-dl utility to grab the video, but when I began playing the copy on my hard-drive with totem, to my horror, the English subtitles were not there! To repeat, the subtitles are on youtube.com but not in the .webm file that I downloaded with youtube-dl. Can someone coach me how to download a version with the English subtitles? Or are there subtitles in my downloaded version and totem is just not playing them? Thank you for your help. P.S. On the totem menu under View, there is a command called Subtitles but this prompts me for another file. Is there a subtitle file in addition to the video file that I need to download from youtube? Thanks again.

8 years, 5 months

re Zoom R16

by reSet Sakrecoer

Hi all! I found this old discussion about the Zoom R16 in the archive of LAU. http://linuxaudio.org/mailarchive/lau/2012/3/14/188926 Strangely, in that archive there is a missing reply from Mr. Brett McCoy. (You can find the missing one a the bottom of this archive: http://linux-audio.4202.n7.nabble.com/Zoom-R16-td7748.html ) Mr. McCoy claims to have enabled the Zoom R8 (yes, there is quite a difference between r16 and r8) as a multi-input soundcard for ardour and also as a control surface in 12.04 64bits. There is no information as to which driver is being in use in his post. And searching the web only leads me to various WIP to get the R16 running on linux. My question is, has anyone got any news about this device and a possible drivers? Specifically to Mr. Brett McCoy: what driver did you use to make jack aware of your r8? Thank full for any hints, yours, -- Set Hallström AKA reSet Sakrecoer http://sakrecoer.com

8 years, 5 months

Focusrite Saffire Pro 40 - reliable?

by Ben Bell

Hi all, My dual Delta 1010 setup is on its last legs and I'm thinking (again) of retooling with a Saffire Pro 40. I've been bitten in the past by upgrades like this which is why I'm still on the 1010s, but there's only so many times I can repair them before I have to admit defeat. Is anyone out there actively using a Saffire Pro 40 in a serious setting? Is it reliable for low latency, with all channels working at once? Does it work with any recent ffado, or do you need specific versions, patches or anything else? I've read plenty of vague comments about which firewire chipsets might be good or not, but are there known-good PCI Express firewire cards that are a sensible price and work well? I'm really hoping for "Yes, I've been using one for a year and it's rock solid," rather than "I've googled and it claims to be supported," or "Yes, it's fine if you pull the latest beta7 branch from git and only use Acme 2000 firewire interfaces." Sorry to sound unduly cautious or skeptical ;) In short, if I bite the bullet, retire my Delta 1010s and buy a Pro 40, are the drivers and tools now at the stage where I reasonably expect to be up and running again in a day rather than still fighting problems a week later? Cheers, bjb

8 years, 11 months

qjackctl: disable baloon popups?

by IOhannes m zmoelnig

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 this is something that has annoyed me endlessly for years, and i still haven't found a solution. the issue is minor (that's why i haven't asked before), but annoying nevertheless. so: i'm using qjackctl with the "system tray icon" enabled. when i close the application window, qjackctl keeps running and is accessible via the system tray. cool. that's what i want. however, every single time i close the main qjackctl window, a balloon pops up in my notification area, telling me that the "program will keep running in the system tray" and how to really shut it down. the first time i read this, this information was informative (i suppose, it has been years). but no longer. i already know that the program will keep running. that's why i'm using the system tray. there's no need to tell me again and again (even within a single session!). there *must* be simple way to "not show this message again", without completely disabling tray notifications. but how? btw, this is on Debian/sid with xfce4. fgasdmr IOhannes -----BEGIN PGP SIGNATURE----- Version: GnuPG v1 iQIcBAEBCAAGBQJU0eVfAAoJELZQGcR/ejb4SY0P/3NphEAkGRawvdm4espj7i1J CIIGtUFTyvsKwFXEqXtAdlHz3/mt9pv1H17LEWdIxOfJ+ziY5dOC1HbAnNSPXmte 91AGmrFCNC8uKSrvCx+kySJEiU4AbNaSY0g8ORKMvOuCPAdO//9qXFBcSRaPX5eb rT7xtGKK+NYTmFSiPIypjW0ttGabtIjJE+Pg8e674rep+Q63ovxPBTBzkVcvJt4J udTKPc8+V7vZvC0Q0y4/vwiih5l0+7RypNxuapm+KY8zSgSkGUc2Wug2XMPhaLid Zmej+Kn/72PZflC5p7uo9JOZkXcShO7SX6296oaFqhFdtxlkUxmggZf/D6eeIGhA 91bACU/+GVgfXE8QUXX0XaifvstBaEwUMdpLIB9lLhC6sdROpl6hHFz/JMJSJYqt pX0oFI1YNzaT65E0BzOfQa4yxuyNO4BoYZ/C7wg4WtSMd/zHaiK5z9K0nxh1ruDM 5BNgeYNtvQ++FO3V0/eVkOwqab6hdS8/MfLRbro/BmJbNsg4MTDUxHZiL1AzXhIz 6l/SbMsdExY3NgbDQiVmfoW6P9MGDrBO/7n6DZj6g7O5lmtEdR9jhPKpFN3ApJzr +yVkxrcppph9eNMMUtAWnXh+XBGuuvfdrhakHoAZIaDGa3P7f6GsBjufyrysKGT1 SRBESzDoiLj+5K3rCOYr =ertZ -----END PGP SIGNATURE-----

8 years, 11 months

Re: [LAU] OM6.9 on Arch

by David Jones

On Feb 14, 2015 5:48 AM, Len Ovens <len(a)ovenwerks.net> wrote: > > On Sat, 14 Feb 2015, anders.vinjar(a)bek.no wrote: > > > The real pain here is having to do 32-bit at all! If we only got lw to > > consider 64-bit a 'professional' feature, and not just a high-end > > 'enterprise' feature... :-/ Even my wife's years-discontinued little netbook is 64-bit. > With some distros talking about dropping support for 32bit kernels, 64bit > is just where the world is going anymore. 32bit versions are just > outdated. > > As someone who uses old computers for servers etc. I find this anoying... I do wish kernel makers would also stop requiring PAE support. I have a couple of boxes that don't support PAE. Musix is 32-bit only. I expect Debian will be producing 32-bit kernels for a good long while yet. They just seem to change slower than others. David W. Jones gnome(a)hawaii.rr.com authenticity, honesty, community http://dancingtreefrog.com

9 years

Re: [LAU] Problems with sample rate and recordings (Mackie onyx > 1640-i < Firewire > Debian )

by Chris Caudle

> Date: Thu, 26 Feb 2015 17:18:35 +0100 > From: sub <subvertao(a)inventati.org> > Message-ID: <54EF475B.8040601(a)inventati.org> > i received a mail from the band in which they tell me > that the mix is faster and at a higher pitch than the > audio they recorded through a video camera at the > concert. Video cameras record at 48k rate, so someone will need to rate convert. Either the camera audio will need to be converted to 44.1k or your audio will need to be converted to 48k. Which you choose probably does not matter, but everyone involved needs to know and agree on the chosen project sample rate. If the final output will be a video then converting every thing to 48k is probably the easiest choice. If the final output will be a CD (or audio only files) then probably 44.1k is the easiest choice. If you will have both a video and audio only files, my choice would be working in 48k and then sample rate convert the final mix to 44.1k for CD. -- Chris Caudle

9 years, 1 month

Raspberry PI, internal audio IF + testing new USB IF

by Carlos sanchiavedraz

Hello dear all. I'm testing a small interface for some musical and technological projects which has an instrument input jack on one side and no audio output, just USB on the other side. This IF works great but you have no physical audio output to monitor/hear sound, so I thought I could enable internal audio IF (as I've done on other devices) and that will be the audio output instead of needing another second IF to get audio from. So I decided to rescue some past investigations I made when first researching about RPi, along with the great content generated by some dear great folks in here, and some new research one after each problem. First started from a Raspbian 2012 version which I had already tweaked for a headless autojamming/instrument effects/looper, but got many issues making step by step progresses that I briefly point here FWIW: - Enabled internal IF on Raspbian 2012 [1] - Found mmap problem regarding using Jackd with internal IF [2], even tried a module somebody compiled - Tried switching to the new platform version, Raspbian 2015 (raspberrypi 3.2.27+) but there wasn't even /proc/asound - Couldn't get to load snd-bcm2835 kernel module for internal soundcard [4] - Finally I got an error from the very raspi-config regarding internal audio IF [5] Although I continue researching, I guess somebody here has already gone through some of this, but I'm wondering if it's already an abandoned thing by the makers themselves this thing of the internal IF thing. Thanks as always. Greetings all. +info: [1] enable internal souncard on Raspbian 2012 https://duckduckgo.com/?q=%22raspberry+pi%22+enable+internal+soundcard&t=de… http://www.raspberrypi.org/forums/viewtopic.php?f=66&t=18573 http://stackoverflow.com/questions/12144972/jack-audio-connection-kit-jackd… [2] enable mmap to run jack with internal soundcard on Raspbian 2012 https://martinezjavier.wordpress.com/2013/04/27/mmap-support-for-raspberry-… https://www.hifiberry.com/forums/topic/jackd/ [3] cat: /proc/asound: No such file or directory https://duckduckgo.com/?q=%22cat%3A+%2Fproc%2Fasound%3A+No+such+file+or+dir… http://forums.debian.net/viewtopic.php?t=74914 http://forum.ubuntu.ru/index.php?topic=99987.105 http://forums.debian.net/viewtopic.php?p=415341 [4] error when loading snd-bcm2835 kernel raspberrypi 3.2.27+ module for internal soundcard on Raspbian 2015 https://duckduckgo.com/?q=%22libkmod%3A+ERROR+..%2Flibkmod%2Flibkmod.c%3A55… http://www.raspberrypi.org/forums/viewtopic.php?f=44&t=36445&p=305808 [5] error raspi-config when enabling audio via minijack on Raspbian 2015 https://duckduckgo.com/?q=raspberry+pi+%22there+was+an+error+running+option… http://www.hifiwigwam.com/showthread.php?115856-Raspberry-Pi-2/page3 -- C. sanchiavedraZ: * NEW / NUEVO: www.sanchiavedraZ.com * Musix GNU+Linux: www.musix.es

9 years, 1 month

retuning bristol

by tom haddington

Hey, Everybody-- i loaded bristol with a scala (.scl) file, this morning. That much worked fine. However, both my midi keyboard and the gui keyboard are nowhere near middle C. i've even tuned my midi keyboard up (a maximum) three octaves, and my middle C is still two octaves too low. Can this be prevented or adjusted in anyway? Thanks! bill

9 years, 1 month

Problems with sample rate and recordings (Mackie onyx 1640-i < Firewire > Debian )

by sub

Hei. I am running linux on a Debian machine with kx repositories, connected to a Mackie Onyx 1640-i. Linux kx 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt4-3 (2015-02-03) x86_64 GNU/Linux 04:00.0 FireWire (IEEE 1394): Texas Instruments TSB43AB23 IEEE-1394a-2000 Controller (PHY/Link) When i record multitrack audio using Ardour and jackd, i see that the sample frequency is set to 44100. The audio files are also at 44100, according to file: root@kx:/home/humla/akt3bolger/interchange/akt3bolger/audiofiles# file bass-2.wav <head> bass-2.wav: RIFF (little-endian) data, WAVE audio, mono 44100 Hz After doing a mix of the material on a separate machine using Ardour3 at 44100, and dumping the mix internally to a 44100 WAV, i received a mail from the band in which they tell me that the mix is faster and at a higher pitch than the audio they recorded through a video camera at the concert. When i playback audio from the linux machine (like mp3s ) through the firewire and the Onyx console, i get random switches of speed and pitch. I have been working in professional audio, and i know there must be a problem with the sample frequency lock and synch between the two devices, so that one is on the wrong sample frequency, and "translates" the bitstream wrongly. Normally in a digital audio chain you have a machine that is "master" and one that is "slave". The problem here is that the console has no display showing the sample frequency it is locked at, and the Linux machine does not show any master / slave feature in jack or ardour. There is something strange going on here. The machine has also another analog audio card: 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 02) And i don't know how this could influence the setup.

9 years, 1 month

SPDIF plus NetJack for digital audio transfer between computers.

by Brendan Furneaux

I play keyboard in a band, and use UbuntuStudio on a laptop for softsynths, effects, and most importantly, MIDI routing (via mididings). Across the room is our studio computer, which runs Windows. I am running audio out of my laptop via a Presonus FirePod (aka FP10), which works fine. The studio computer has a Presonus FireStudio and Digimax, which are joined by an ADAT lightpipe. Both the FirePod and FireStudio have SPDIF ports. At the moment, audio is traveling via audio cables from the hardware synths and laptop across the room to the studio computer. However, it seems like I have everything in place to have a "simpler" setup, at least from a wiring and D/A/D conversion standpoint, by using NetJack to send digital audio via a crossover cable from my laptop to the studio computer. The plan would be: *The FireStudio (on the studio computer) must have its clock synced to the Digimax via ADAT. *The FirePod (laptop) can be synced via SPDIF to the FireStudio. Thus, all audio hardware will be on one master clock without any effort from the computers. *Use the FirePod to capture audio from the hardware synths. *Use NetJack(1|2) to route audio from the laptop (both captured from the hardware synths, and from softsynths) to the studio computer, via an Ethernet crossover cable. The last step is where I am running in to trouble. For starters, NetJack(1|2) does not seem to be designed for the use case where the sound cards on two networked computers are actually synced in hardware; thus the normal operating mode where only the master can access its audio hardware. So far, my plan is to use NetJack2, with the laptop as master and the studio computer as slave. I have thought of either using audioadapter with the net backend on the slave, or of using netadapter with the alsa backend. Both of these do resampling to compensate for drifting clocks. My first set of questions is: In the case that the clocks on the master and slave hardware are already synced, do audioadapter and netadapter still spend significant CPU time on resampling overhead? Which one is likely to be more reliable/less resource intensive? Can netadapter receive MIDI? Is there some other software that would be better? I currently have Jack 1.9.10 installed on both computers, and they work fine individually. They are configured with static IPs on the crossover cable, and are at least able to ping each other. However, I have not been able to get NetJack2 to connect using either computer as the master.

9 years, 1 month

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Linux-audio-user February 2015