Linux-audio-user August 2003

linux-audio-user@lists.linuxaudio.org

100 participants
114 discussions

[linux-audio-user] USB audio capture line/mic: A/D quality, how set record levels?

by Mannr＠uwaterloo.ca

I am considering getting a USB audio interface. My goal is to capture audio for making CDs. I will capture from line level, eg., from minidisc or phonograph, and from microphone inputs, eg., for recording live shows to a laptop. It should be completely USB powered for maximum flexibility. Digital I/O would be great, but is not required. I am not looking for professional quality. But I would like something pretty good, eg., at the level of consumer minidisc recorders. That is what I am using to record now. I don't need any mixing capability. Plain capture at 44.1kHz/16bit and optionally 48.0kHz would be fine. I've seen lots of gear, eg., Edirol, M-audio, etc. But I have several questions: - What is the A/D quality of some of the retail units? Retail units: Edirol UA-1A, UA-3D (edirol.com), MAudio Transit (m-audio.com), Soundblaster Extigy, Soundblaster MP3+ Higher quality? Tascam U-122, Edirol UA-20?, MAudio MicPre, Professional level (>500USD) SoundDevices USBPre (http://www.sounddevices.com/products/usbpremaster.htm) - Is the Mic preamp any good? Ie., as good or better than the one in my consumer minidisc? - What is the "native" capture mode of these units? 16bits is OK, but I have heard reports that some, like the Soundblaster, capture at 48 or 96kHz and resample for 44.1kHz. I am looking for true sampling at 44.1kHz. - How can I control record level on these units? Some have level controls, but it is unclear what level these control, the mic preamp or the line level, or both. The only one I can see a clear "block diagram" for is the USBPre. This shows separate mic and line level gain controls, as it should have. (See user guide PDF file.) If there are no knobs, can the level be set with software? Is this what the Linux "xmixer" program does? The point is that I want to: - adjust the line level gain *before* A/D is done - adjust the mic level gain *within the mic preamp*, to avoid clipping So the unit must have internal gain control. Here is what I have learned so far. Please help... Richard Encl: list ------------------------------------------------------------------------------ Product Cost |USB |LineI/O |MicI/O |DigiI/O |Sample |Depth |Quality self level? level? (native) 16/24 Consumer/ (street) power control control exact-bit Audioph/ | | copy?| Pro phantom power?| ------------------------------------------------------------------------------ Edirol UA-1A Y I/O, N none none 44.1k 16 Consumer UA-3D Y I/O, Y I/O, YN I/O, N ?(1) 16 Consumer UA-20 Y I/O, Y none I/O, N 44.1k 16/24 ? Edirol UA-5 9VDC I/O, Y I/O, YY none 44/48/96(2) 16/24 ??? Creative MP3+ Y I/O, N I/O, NN I/O, N 48k(3) 16 Consumer Creative Extigy Y I/O, Y I/O, YN I/O, N 48k(3) 16/24 Consumer AFAICT all above units sample at 48/96K and resample for 44.1K. So they will add artifacts, and they will not make an accruate digital copy either! (1) claims 32/44.1/48k. Is this 48k native sample rate? (2) "real time sample rate conversion", is this 96k native sample rate? (3) rumored to be resampled. MAudio Transit $100 Y I/O, N I/O, NN I/O, ? ? 16/24 Consumer MAudio MicPre $180 Y I/O, ? I/O, YY none 44.1/48 16 ??? (150?) MAudio Duo $350 9VAC! I/O, ? I/O, YY I/O, Y 44.1/48 16/24 Pro Tascam US-122 $270 Y I/O, ? I/O, YY none 44.1/48 16/24 Pro? (200?) USBPre $700 Y I/O, Y I/O, YY I/O, Y 44.1/48 16/24 Pro Possibilities: ------------- MAudio Transit would be great if it had: - reasonable quality A/D - true level controls (from software) - true hardware sampling at 44.1, 48k and 96k. MAudio MobilePre looks good if I am happy with 16bit sample depth and no digital I/O. Tascam US-122 looks good as well, but is more expensive.

21 years, 3 months

[linux-audio-user] RE: Invert And Mix

by Jamie Lutch

what works for me is to generate a sine wave (I used wavelab in windows), then copy and paste it until it is several hours long, then insert 10 seconds of silence at the beginning. Start rolling your sine wav recording from another machine with a known good soundcard, and start your test machine recording in the silent period. Its super easy to sync up to your original wav by deleting all the silence at the beginning.

21 years, 3 months

[linux-audio-user] [OT] Linux soundapps sites updated (again)

by Dave Phillips

Greetings: I've added some new & updated listings to the pages since posting the last announcement. Some of the interesting items include Drums++ (a programming language for percussion), GuitarCodex Plus (excellent chord/scale utility for guitarists), Pymprovisator (accompaniment software), and updates for Mixxx and the Music Kit. Just FYI... Best regards, == dp

21 years, 3 months

[linux-audio-user] Noise removal, what filter is best?

by Tim Beauregard

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm looking for a filter/software to remove noise from a sound file. I'm not an audio expert so will try to describe the noise in a way that you out there might be able to put a name to it and recommend a suitable filter/software to remove it. It is a short sharp blip, fairly high frequency, sounds like a compass point stabbing glass. It occurs as often as every minute or so in the two hour sound file. I obtained this file by recording DAB radio here in the UK. I'm presuming the noise is derived from errors in the transmission or analogue to digital conversion phase of the broadcast. I think it is due to a signal exceeding line volume. I've tried the gramofile filter 'Conditional Median Filter II', which failed to remove the noise. If necessary I could email a sample of the noise. I'm using Debian 3.0r0, 2.4.18 kernel. Thank you in advance :-) Tim Beauregard -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.0.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQE/Snz3sUUdIDHrdAURAuF/AJ9+QZHf4lZMgDLkAm+wv1Ov3dTgqACghi2f 1q7lXukyBjtYvG0mTFPv+po= =Mu6g -----END PGP SIGNATURE-----

21 years, 3 months

[linux-audio-user] cannot get mplayer to resume an interrupted stream

by Alex Timmer

hi all, i use mplayer to successfully play an ".asx" stream [internet radio broadcast]. but i have every few hours a short interruption in my internet connection, upto about 10 secs. network connection is automatically restored. after each interruption in the stream mplayer's cache goes empty, so even with the huge default cache of 8192KB (> 1min audio) i end up having to quit and restart mplayer. 1) is there any way to get mplayer to try to retry/reconnect after it detects an interrupt in the media stream? 2) is there any other program that can play .asx streams and will survive these temporary interuptions of the stream (with caching)? cau, ALEX ________________________________________________________________________ Want to chat instantly with your online friends? Get the FREE Yahoo! Messenger http://uk.messenger.yahoo.com/

21 years, 3 months

Re: [linux-audio-user] alsa driver install help

by Jamie Lutch

>You need the kernel headers package and the ncurses-devel package. I found the ncurses-devel, but a search through the suse install finds nothing like 'kernel headers'. Is there an explicit name for this module? thanks, jamie

21 years, 3 months

[linux-audio-user] Dual Soundcard Woes

by Peter Hutnick

Gang, I'm not sure this is the best place to ask this, but here goes . . . I have a "VIA Technologies, Inc. VT82C686 AC97 Audio Controller (rev 80)." according to /proc/pci using the via82cxxx_audio module that comes stock with Red Hat 9. (The 2.4.20-8 version due to some buggyness in the 2.4.20-20.9 versions V4L drivers.) I also have a "Burr-Brown Japan PCM2702" (according to /proc/bus/usb/devices) sound device built into my home stereo receiver. It seems to use a driver simply called "audio" that is part of the USB driver set. I want to use the VT82cxxx for everything except a fixed set of multimedia apps (xawtv/motv, mplayer, xine, xmms). My first problem is that the USB device seems to be /dev/dsp and the sound card (actually on board audio) is /dev/dsp1. I'd like this to be the other way around. There is no entry in /etc/modules.conf for the USB device . . . my first instinct was to just switch them around there. The other thing is that control over which device gets used seem to be flakey. For instance the motv -C switch seems to do nothing. I always get the output from the soundcard. With mplayer using -ao oss:/dev/dsp[1] seems to give me control over which device is used about 90% of the time. The other 10% it just seems to "do what it wants to." I'm about as sure as I can be that this isn't just fat-finger syndrome. Thanks in advance for any and all advice. -Peter PS: I suspect someone will suggest "Just use Alsa." but there doesn't seem to be an Alsa driver for the USB device. -P PPS: Anyone know of any good "CD Player" app that supports CDDA? I'd like to be able to use my CDROM->USB->Home Stereo.

21 years, 3 months

[linux-audio-user] Re: Small Problem with Alsaplayer

by BartonBosch＠aol.com

In a message dated 8/26/03 7:25:32 PM Pacific Daylight Time, markknecht(a)comcast.net writes: >I'm noting this evening that on a newer RPWL CD (Stock) with no >dead time between tracks that alsaplayer is making a small, >somewhat frustrating dead spot between each track while I'm >listening. I noticed the same thing the other day. Is this a problem that was not present in previous versions of alsaplayer? Barton

21 years, 3 months

[linux-audio-user] how to invest sensible to get more stability for JACK

by Hartmut Z Noack

Hello, i am working quite intense on lerning ARDOUR by doing. Using its beta i exrerience some bugs that are more or less tolerablein such great an App for no cost (crashes when trying some adventurous routing-stuff, some lesser gui-issues). After all there are only 2 really serious problems: 1.)Importing .wav-files fails too often (regardless if these files are written by Samplitude, Audacity, rezound or even Ardour itself) How is that on your setups out there? 2.)Many xruns, (some after 2 seconds, others after 2 mins, mostly within 40 secs or so - sometimes i had testrecordings of unbroken 10 minutes) I use Ardour on SuSE 8.2 pro with jack0.74 as root only on runlevel 3 in Windowmaker, all networking is off, no kdeinit is running. System is: Athlon XP 1800, 512MB DDR, 60GB Samsung HD/IDE, Terratec EWX 24/96 I think about wether i buy another IDE-HD and spread the installation on 2 disks to get more bandwith or getting another 512 RAM to let Jack run without HD or investing in a SCSI-HD for installation and Projects and using the IDE-Disk for Archiving only. What would you prefer, how are your Experiences about this Bandwithissues? Thank you

21 years, 3 months

[linux-audio-user] drums++

by Robert Jonsson

Hi all, This appeared on freshmeat a while ago. http://dpp.mikekohn.net/ I don't know if this guy is a reader so I thought I would mention it :) It seems it is a programming language specifically for scripting drum-sequences for midi ! That seems like one specialized language :) I'm putting it down as another cool thing I need to try... /Robert

21 years, 3 months

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Linux-audio-user August 2003