Linux-audio-user

linux-audio-user@lists.linuxaudio.org

20661 discussions

by Juhana Sadeharju

>From: Malcolm Baldridge <linux-audio(a)paypc.com> >Subject: Re: [linux-audio-user] Ardour, Jack, and 2.6 kernels + XRuns > with the Audiophile 24/96 M-Audio > >1) You can't use more than one drive per IDE channel. [ ... ] >It's VERY bad on performance to mix very slow devices (such as CD-ROMs) >with fast ones: they hold a lock on the bus during all operations, I recently removed CD-ROM device completely. I ended up doing that because I got disk errors. The disks are perfectly ok, however. Some example symptoms were: -When the CD-ROM device slided in the disk, Alsa audio play was interrupted for 7 seconds -File copy sometimes made errors -- such as the highest bit of one byte of 300 MB file turned from 0 to 1 -File was temporarily corrupted -- error stayed at the same place, no matter how many times the file was verified; after the boot, the file was again ok; I made sure the caches were emptied, and so, it looks like a certain bit pattern or certain address caused the change in data I'm still testing and waiting for errors to happen. This problem has driven me to verify that files are copied ok, that the created tar.bz2 packages are ok (both the tar.bz2 file and its content). I also have set Emacs to make the numbered backup copies and make them by moving (not by copying). Backups, flac, mp3 and ogg codings need to be verified too. The point is this: how you would know about the rare bit changes if you never verify the files? In audio work such a bit changes can go unnoticed. How many of you verify, e.g., that a simple edit to a file did not made a bit error? I'm sure that no one verifies. Use of a dither makes the detection quite hard. >3) Good/correct IDE drivers for the chipset. With Linux at least, you have >pretty good control over this: just make sure you build your kernel with the >IDE chipset driver compiled into the kernel statically [it needs to activate >prior to the mounting of /]. How to know what drivers there already are? How to know if the driver in use is not the correct one? Can the default IDE drivers work for a IDE controller with possible limitations? That is, the problem with IDE driver could go unnoticed without an expert help. How to drop the speed of the IDE? Reliability vs. speed. >If not, you can set them up with: > >hdparm -c1 -u1 -d1 -m16 /dev/hda I made the change, but the speed stayed the same: % hdparm -t /dev/hda: Timing buffered disk reads: 64 MB in 2.23 seconds = 28.70 MB/sec Regards, Juhana

21 years, 5 months

[linux-audio-user] Ogg stream through authenticating firewall

by Pete Harlow

To play an Ogg Vorbis stream through a firewall requiring authentication try : wget --proxy-user="myusername" --proxy-passwd=mypassword --ignore-length -O - http://ogg.smgradio.com/vc32.ogg | ogg123 -d arts - Replace "-d arts" with the appropriate system on your box - see man ogg123 Replace "http://ogg.smgradio.com/vc32.ogg" (Virgin Classic) with your favourite. Enjoy! Pete. This e-mail and any attachment may contain confidential and privileged material intended for the addressee only. If you are not the addressee, you are notified that no part of the e-mail or any attachment may be disclosed, copied or distributed, and that any other action related to this e-mail or attachment is strictly prohibited, and may be unlawful. If you have received this e-mail by error, please notify the sender immediately by return e-mail, and delete this message. Thales, its subsidiaries and/or its employees shall not be liable for the incorrect or incomplete transmission of this e-mail or any attachments, nor responsible for any delay in receipt.

21 years, 5 months

[linux-audio-user] determining source of latency (system info included)

by Nick Salamander

i am experiencing the "system can't keep up" messages like crazy, even recording a single first track. often when i speak into the microphone, what comes out of the speakers sounds like it has a crazy chorus effect on it. i also wonder what the correct settings to use in the alsamixer gui are. below is some system info gleaned from a modified shell script somebody posted on another list. if anyone has any ideas please let me know! and if there's any other system info i could provide. thanks! ========================= ALSA Audio Debug v0.0.2 - Wed Jun 2 11:06:32 EDT 2004 Kernel ---------------------------------------------------- Linux 66.149.213.131 2.4.22-21mm.2mdksmp #1 SMP Sun Nov 2 15:20:57 CET 2003 i686 unknown unknown GNU/Linux Loaded Modules -------------------------------------------- snd-seq-midi 5632 0 (autoclean) (unused) snd-seq-oss 37824 0 (unused) snd-seq-midi-event 6784 0 [snd-seq-midi snd-seq-oss] snd-seq 51920 2 [snd-seq-midi snd-seq-oss snd-seq-midi-event] snd-pcm-oss 46852 0 (unused) snd-mixer-oss 16016 1 [snd-pcm-oss] snd-cs46xx 84232 1 snd-pcm 92128 0 [snd-pcm-oss snd-cs46xx] snd-timer 22404 0 [snd-seq snd-pcm] snd-ac97-codec 51352 0 [snd-cs46xx] snd-rawmidi 20128 0 [snd-seq-midi snd-cs46xx] snd-seq-device 6268 0 [snd-seq-midi snd-seq-oss snd-seq snd-rawmidi] snd-page-alloc 10228 0 [snd-cs46xx snd-pcm] snd 50692 0 [snd-seq-midi snd-seq-oss snd-seq-midi-event snd-seq snd-pcm-oss snd-mixer-oss snd-cs46xx snd-pcm snd-timer snd-ac97-codec snd-rawmidi snd-seq-device] Modules Conf ---------------------------------------------- alias sound-slot-0 snd-cs46xx above snd-cs46xx snd-pcm-oss Proc Asound ----------------------------------------------- Advanced Linux Sound Architecture Driver Version 0.9.8. Compiled on Nov 2 2003 for kernel 2.4.22-21mm.2mdksmp (SMP) with versioned symbols. 0 [CS46xx ]: CS46xx - Sound Fusion CS46xx Sound Fusion CS46xx at 0xf8ffe000/0xf8e00000, irq 18 0: [0- 0]: ctl 8: [0- 0]: raw midi 18: [0- 2]: digital audio playback 17: [0- 1]: digital audio playback 16: [0- 0]: digital audio playback 24: [0- 0]: digital audio capture 1: : sequencer 33: : timer 00-00: CS46xx : CS46xx : playback 31 : capture 1 00-01: CS46xx - Rear : CS46xx - Rear : playback 31 00-02: CS46xx - IEC958 : CS46xx - IEC958 : playback 1 CPU ------------------------------------------------------- model name : Pentium III (Coppermine) model name : Pentium III (Coppermine) cpu MHz : 864.478 cpu MHz : 864.478 RAM ------------------------------------------------------- MemTotal: 513760 kB SwapTotal: 1028120 kB Hardware -------------------------------------------------- 00:00.0 Host bridge: Intel Corp. 82840 840 (Carmel) Chipset Host Bridge (Hub A) (rev 02) 02:06.0 Multimedia audio controller: Cirrus Logic CS 4614/22/24 [CrystalClear SoundFusion Audio Accelerator] (rev 01) HDPARM /dev/hda: Timing buffered disk reads: 122 MB in 3.03 seconds = 40.26 MB/sec /dev/sda: Timing buffered disk reads: 124 MB in 3.04 seconds = 40.79 MB/sec __________________________________ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/

21 years, 5 months

[linux-audio-user] syncing ardour with video

by Tim Blechmann

hi all, is there any possibility to sync ardour (probably using jack) with a video file player? i want to compose a soundtrack to for some visuals... cheer... -- Tim mailto:TimBlechmann@gmx.de ICQ: 96771783 -- The only people for me are the mad ones, the ones who are mad to live, mad to talk, mad to be saved, desirous of everything at the same time, the ones who never yawn or say a commonplace thing, but burn, burn, burn, like fabulous yellow roman candles exploding like spiders across the stars and in the middle you see the blue centerlight pop and everybody goes "Awww!" Jack Kerouac

21 years, 5 months

Re: [linux-audio-user] Lower Output Level after Mono to Stereo Conversion with SOX

by Martin McCormick

Malcolm Baldridge writes: >It's most likely a consequence of being run through a filter. Remember, >filters generally REMOVE power rather than add it. And even though you're >upsampling, some filtering is being done to eliminate synthetically created >aliasing noise. That turned out to be sort of what was going on. I noticed the reduced audio on both the up-sampling script and a script that did no sample conversion but was supposed to send one mono feed to two stereo channels. I tried the -e -stat function and it showed a gain factor of 1.0 in the mono to stereo script. I stuck a gain conversion of 2 in the output and my level came right up to where it should be except for lots of clipping. After another look at the manual, I found that I had overlooked the avg settings. These settings control how much audio is sent to each channel. sox invokes this function but with defaults if you don't set it and I had not. Apparently, the default is to send half the audio to each channel. avg 1,1 did the trick and my stereo from mono recording is now as loud as the original mono .wav without clipping. That script I included that produces a stereo 44100 .wav out of 8000-HZ PCM probably also needs the avg settings adjusted. I just didn't realize that I had lost some level at the time. My thanks for answers that got me pointed in the right direction. Martin McCormick WB5AGZ Stillwater, OK OSU Information Technology Division Network Operations Group

21 years, 5 months

[linux-audio-user] digial audio synch between two machines

by derek holzer

Hi all, I've been sending two channels of audio via SPDIF from an RME Hammerfall DSP multiface to to a Delta 66. Listening to the Delta, I notice some little periodic glitches which seem to indicate a synch problem [these glitches are not present when receiving from the same card via analog]. Can anybody suggest how I can troubleshoot this? Is it necessary to make one a master and one a slave? If so, how is this done? thx, d. -- derek holzer ::: http://www.umatic.nl ---Oblique Strategy # 72: "Find a safe part and use it as an anchor"

21 years, 5 months

[linux-audio-user] Lower Output Level after Mono to Stereo Conversion with SOX

by Martin McCormick

I recorded a mono .wav file at 14400 samples per second and it sounds loud and clear. Then, I got fancy and decided to convert it in to a stereo file suitable for putting on a CD. It still sounds clear, but about 3DB down from the original mono recording. Just guessing at this, but it is noticeably lower in volume. Here's how I did all that: #! /bin/sh arecord -d 454 -t wav -r 44100 -c 1 -f S16_LE \ | sox -t .wav -c 1 - -c 2 -w -r44100 outputcd.wav The top half of that shell script originally outputted to a file which was mono and did play at 44100 samples/sec. If I take that file and use it as input to the lower half of the script, I get exactly the same lower audio level so I know it isn't the way I am piping the arecord output in to sox. Besides, that is normal UNIX practice and there is no modification to the data. I expected sox to simply duplicate the mono data in to both left and right channels. Is this normal or do I need to tell sox to increase the level. I may accidentally be causing sox to expect another channel of audio which would be right if one was mixing two channels. In this case, 100% of the mono signal should appear on both channels as if I had a Y connector. Sox is amazing in that I was able to produce a CD-ready audio track from some unsigned 8-bit PCM audio and it sounded as good as it did originally. This isn't all that good for anything but voice, but the conversion didn't make it any worse. For those who are curious, that script looks like: #! /bin/sh sox -t ub voxaudio.ub -t wav -c 2 -w -r44100 output.wav resample .95 I don't have any particular question about this script, but I do wonder why my mono/stereo conversion is a few DB down? Thank you. Martin McCormick WB5AGZ Stillwater, OK OSU Information Technology Division Network Operations Group

21 years, 5 months

Re: [linux-audio-user] Make Stereo WAV out of Two Separate Files and Other Weird Activities

by Martin McCormick

My thanks to every one who had a suggestion to try. I will install ecasound and learn all about it. Martin McCormick

21 years, 5 months

[linux-audio-user] SBLive Capture problem

by Doug Asherman

Hi: I'm fairly new to ALSA, and I'm having a problem with capturing sound on the SBLive cards line in jack. Long story short: using alsa 1.0.5 with kernel 2.4.26. Recompiled all the drivers including SBLive when I rebuilt the kernel. Ran alsaconf to do automatic set up. Altered the defaults like so: amixer sset Capture 33%,33% amixer sset Line cap 100%,100% ...and what I'm able to record is one channel of what's coming into the Line In. I know both channels are coming in -- if I set the playback of Line to 100% I can hear both channels fine. Has anyone had this problem? I'd appreciate any pointers to a solution. Thanks, Doug

21 years, 5 months

[linux-audio-user] AGNULA/DEMUDI 1.2.0-beta0 IS OUT

by The AGNULA Team

+-----------------------------------------------------------------+ | ______ ______ _ _ _ | | /\ / _____) ___ \| | | | | /\ | | / \ | / ___| | | | | | | | / \ | | / /\ \| | (___) | | | | | | | / /\ \ | | | |__| | \____/| | | | |___| | |_____| |__| | | | |______|\_____/|_| |_|\______|_______)______| | | | +-----------------------------------------------------------------+ [Sorry for cross-posting. Feel free to forward around] Florence, 08 June 2004 +++ AGNULA/DEMUDI 1.2.0-beta0 IS OUT AGNULA/DeMuDi 1.2.0-beta0, the Debian-based GNU/Linux distribution for audio/video, has been released. +++ AGNULA/DeMuDi 1.2.0-beta0, the Debian-based GNU/Linux distribution for audio/video, has been released. This version is the first beta of the 1.2.0 series, which sport tighter integration with Debian, using the Sarge Debian Installer and the CDD (Custom Debian Distributions) framework. You can download AGNULA/DeMuDi 1.2.0-beta0 here: http://download.agnula.org/1.2/1.2.0/demudi_1.2.0-beta0_i386.iso MD5SUM files are available here: http://download.agnula.org/1.2/1.2.0/MD5SUMS But PLEASE, PLEASE, PLEASE use the relevant mirrors: http://freesoftware.ircam.fr/mirrors/agnula/ (IRCAM, Paris) http://ccrma.stanford.edu/mirrors/agnula/agnula-iso/ (CCRMA, Stanford) Please report all bugs, requests, criticisms using our development portal [0]. Instructions on how to report bugs and requests are available here: http://www.agnula.org/development/agnula_bugs_requests/ We hope you enjoy AGNULA/DeMuDi! For any information, do not hesitate to contact us writing to: <info(a)agnula.org> And/or visiting our web site, http://www.agnula.org/. +++ About AGNULA: Agnula (acronym for A GNU/Linux Audio distribution, pronounced with a strong g) is the name of a project funded until April 2004 by the European Commission (number of contract: IST-2001-34879; key action IV.3.3, Free Software: towards the critical mass). After the end of the funded period, AGNULA is continuing as a volunteer based project, aiming to spread Libre Software in the professional audio/video arena. Best regards, -- The AGNULA Team info(a)agnula.org Our mailing lists: http://lists.agnula.org/ Our web site: http://www.agnula.org/ "There's no free expression without control on the tools you use" [0] http://devel.agnula.org/

21 years, 5 months

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Linux-audio-user